diff options
Diffstat (limited to 'doc')
101 files changed, 35370 insertions, 6862 deletions
diff --git a/doc/APIchanges b/doc/APIchanges index 6f7a14189f..d253a66280 100644 --- a/doc/APIchanges +++ b/doc/APIchanges @@ -1,5 +1,5 @@ Never assume the API of libav* to be stable unless at least 1 month has passed -since the last major version increase. +since the last major version increase or the API was added. The last version increases were: libavcodec: 2015-08-28 @@ -7,47 +7,63 @@ libavdevice: 2015-08-28 libavfilter: 2015-08-28 libavformat: 2015-08-28 libavresample: 2015-08-28 +libpostproc: 2015-08-28 +libswresample: 2015-08-28 libswscale: 2015-08-28 libavutil: 2015-08-28 API changes, most recent first: -2016-xx-xx - xxxxxxx - lavf 57.3.0 - avformat.h +2016-xx-xx - xxxxxxx - lavf 57.25.0 - avformat.h Add AVFormatContext.opaque, io_open and io_close, allowing custom IO - for muxers and demuxers that open additional files. -2015-xx-xx - xxxxxxx - lavc 57.12.0 - avcodec.h +2016-02-01 - xxxxxxx - lavf 57.24.100 + Add protocol_whitelist to AVFormatContext, AVIOContext + +2016-01-31 - xxxxxxx - lavu 55.17.100 + Add AV_FRAME_DATA_GOP_TIMECODE for exporting MPEG1/2 GOP timecodes. + +2016-01-01 - xxxxxxx - lavc 57.21.100 / 57.12.0 - avcodec.h Add AVCodecDescriptor.profiles and avcodec_profile_name(). -2015-xx-xx - xxxxxxx - lavc 57.11.0 - avcodec.h dirac.h +2015-12-28 - xxxxxxx - lavf 57.21.100 - avformat.h + Add automatic bitstream filtering; add av_apply_bitstream_filters() + +2015-12-22 - xxxxxxx - lavfi 6.21.101 - avfilter.h + Deprecate avfilter_link_set_closed(). + Applications are not supposed to mess with links, + they should close the sinks. + +2015-12-17 - lavc 57.18.100 / 57.11.0 - avcodec.h dirac.h xxxxxxx - Add av_packet_add_side_data(). xxxxxxx - Add AVCodecContext.coded_side_data. xxxxxxx - Add AVCPBProperties API. xxxxxxx - Add a new public header dirac.h containing av_dirac_parse_sequence_header() -2015-xx-xx - xxxxxxx - lavc 57.9.1 - avcodec.h +2015-12-11 - xxxxxxx - lavf 57.20.100 - avformat.h + Add av_program_add_stream_index() + +2015-11-29 - xxxxxxx - lavc 57.16.101 - avcodec.h Deprecate rtp_callback without replacement, i.e. it won't be possible to get image slices before the full frame is encoded any more. The libavformat rtpenc muxer can still be used for RFC-2190 packetization. -2015-11-xx - xxxxxxx - lavc 57.9.0 - avcodec.h +2015-11-xx - xxxxxxx - lavc 57.16.0 - avcodec.h Add AV_PKT_DATA_FALLBACK_TRACK for making fallback associations between streams. -2015-11-xx - xxxxxxx - lavf 57.1.0 - avformat.h +2015-11-xx - xxxxxxx - lavf 57.19.100 - avformat.h Add av_stream_new_side_data(). -2015-11-xx - xxxxxxx - lavu 55.3.0 - xtea.h +2015-11-xx - xxxxxxx - lavu 55.8.100 - xtea.h Add av_xtea_le_init and av_xtea_le_crypt -2015-11-xx - xxxxxxx - lavfi 6.1.0 - avfilter.h - Add a frame_rate field to AVFilterLink +2015-11-18 - lavu 55.7.100 - mem.h + Add av_fast_mallocz() -2015-xx-xx - xxxxxxx - lavc 57.6.0 - avcodec.h - -2015-xx-xx - lavc 57.7.0 - avcodec.h +2015-10-29 - lavc 57.12.100 / 57.8.0 - avcodec.h xxxxxx - Deprecate av_free_packet(). Use av_packet_unref() as replacement, it resets the packet in a more consistent way. xxxxxx - Deprecate av_dup_packet(), it is a no-op for most cases. @@ -55,142 +71,263 @@ API changes, most recent first: xxxxxx - Add av_packet_alloc(), av_packet_clone(), av_packet_free(). They match the AVFrame functions with the same name. -2015-xx-xx - xxxxxxx - lavc 57.5.0 - avcodec.h +2015-10-27 - xxxxxxx - lavu 55.5.100 - cpu.h + Add AV_CPU_FLAG_AESNI. + +2015-10-22 - xxxxxxx - lavc 57.9.100 / lavc 57.5.0 - avcodec.h Add data and linesize array to AVSubtitleRect, to be used instead of the ones from the embedded AVPicture. -2015-xx-xx - xxxxxxx - lavc 57.0.0 - qsv.h +2015-10-22 - xxxxxxx - lavc 57.8.100 / lavc 57.0.0 - qsv.h Add an API for allocating opaque surfaces. -2015-xx-xx - xxxxxxx - lavu 55.2.0 - dict.h +2015-10-15 - xxxxxxx - lavf 57.4.100 + Remove the latm demuxer that was a duplicate of the loas demuxer. + +2015-10-14 - xxxxxxx - lavu 55.4.100 / lavu 55.2.0 - dict.h Change return type of av_dict_copy() from void to int, so that a proper error code can be reported. -2015-09-29 - 948f3c1 - lavc 57.0.0 - avcodec.h +2015-09-29 - b01891a / 948f3c1 - lavc 57.3.100 / lavc 57.2.0 - avcodec.h Change type of AVPacket.duration from int to int64_t. -2015-09-18 - e3d4784 - lavc 57.2.0 - d3d11va.h +2015-09-17 - 7c46f24 / e3d4784 - lavc 57.3.100 / lavc 57.2.0 - d3d11va.h Add av_d3d11va_alloc_context(). This function must from now on be used for allocating AVD3D11VAContext. -2015-09-07 - lavu 55.0.0 - b8b5d82 - Change type of AVPixFmtDescriptor.flags from uint8_t to uint64_t. - 6b3ef7f - Change type of AVComponentDescriptor fields from uint16_t to int +2015-09-15 - lavf 57.2.100 - avformat.h + probesize and max_analyze_duration switched to 64bit, both + are only accessible through AVOptions + +2015-09-15 - lavf 57.1.100 - avformat.h + bit_rate was changed to 64bit, make sure you update any + printf() or other type sensitive code + +2015-09-15 - lavc 57.2.100 - avcodec.h + bit_rate/rc_max_rate/rc_min_rate were changed to 64bit, make sure you update + any printf() or other type sensitive code + +2015-09-07 - lavu 55.0.100 / lavu 55.0.0 + c734b34 / b8b5d82 - Change type of AVPixFmtDescriptor.flags from uint8_t to uint64_t. + f53569a / 6b3ef7f - Change type of AVComponentDescriptor fields from uint16_t to int and drop bit packing. - 2268db2 - Add step, offset, and depth to AVComponentDescriptor to replace + 151aa2e / 2268db2 - Add step, offset, and depth to AVComponentDescriptor to replace the deprecated step_minus1, offset_plus1, and depth_minus1. -2015-07-31 - lavu 54.17.0 - 7a7df34 - Add av_blowfish_alloc(). - ae36545 - Add av_rc4_alloc(). - 5d8bea3 - Add av_xtea_alloc(). - d9e8b47 - Add av_des_alloc(). +-------- 8< --------- FFmpeg 2.8 was cut here -------- 8< --------- + +2015-08-27 - 1dd854e1 - lavc 56.58.100 - vaapi.h + Deprecate old VA-API context (vaapi_context) fields that were only + set and used by libavcodec. They are all managed internally now. + +2015-08-19 - 9f8e57ef - lavu 54.31.100 - pixfmt.h + Add a unique pixel format for VA-API (AV_PIX_FMT_VAAPI) that + indicates the nature of the underlying storage: a VA surface. This + yields the same value as AV_PIX_FMT_VAAPI_VLD. + Deprecate old VA-API related pixel formats: AV_PIX_FMT_VAAPI_MOCO, + AV_PIX_FMT_VAAPI_IDCT, AV_PIX_FMT_VAAPI_VLD. -2015-07-29 - 7e38340 - lavu 54.16.0 - hmac.h - Add AV_HMAC_SHA224 and AV_HMAC_SHA256. +2015-08-02 - lavu 54.30.100 / 54.17.0 + 9ed59f1 / 7a7df34c - Add av_blowfish_alloc(). + a130ec9 / ae365453 - Add av_rc4_alloc(). + 9ca1997 / 5d8bea3b - Add av_xtea_alloc(). + 3cf08e9 / d9e8b47e - Add av_des_alloc(). -2015-07-27 - lavc 56.35.0 - avcodec.h - 7c6eb0a - Rename CODEC_FLAG* defines to AV_CODEC_FLAG*. - def9785 - Rename CODEC_CAP_* defines to AV_CODEC_CAP_*. - 059a934 - Rename FF_INPUT_BUFFER_PADDING_SIZE and FF_MIN_BUFFER_SIZE - to AV_INPUT_BUFFER_PADDING_SIZE and AV_INPUT_BUFFER_MIN_SIZE. +2015-07-27 - lavc 56.56.100 / 56.35.0 - avcodec.h + 94d68a4 / 7c6eb0a1 - Rename CODEC_FLAG* defines to AV_CODEC_FLAG*. + 444e987 / def97856 - Rename CODEC_CAP_* defines to AV_CODEC_CAP_*. + 29d147c / 059a9348 - Rename FF_INPUT_BUFFER_PADDING_SIZE and FF_MIN_BUFFER_SIZE + to AV_INPUT_BUFFER_PADDING_SIZE and AV_INPUT_BUFFER_MIN_SIZE. -2015-07-20 - 5d3addb - lavc 56.33.0 - avcodec.h - Add AV_PKT_DATA_QUALITY_FACTOR to export the quality value of an AVPacket. +2015-07-22 - c40ecff - lavc 56.51.100 - avcodec.h + Add AV_PKT_DATA_QUALITY_STATS to export the quality value, PSNR, and pict_type + of an AVPacket. -2015-07-02 - 1316df7 - lavu 56.15.0 +2015-07-16 - 8dad213 - lavc 56.49.100 + Add av_codec_get_codec_properties(), FF_CODEC_PROPERTY_LOSSLESS + and FF_CODEC_PROPERTY_CLOSED_CAPTIONS + +2015-07-03 - d563e13 / 83212943 - lavu 54.28.100 / 56.15.0 Add av_version_info(). -2015-07-01 - 0d449c8 - lavfi 5.1.0 - version.h - 0f87f9b - lavd 55.2.0 - version.h - Add library identification symbols, LIBAVFILTER_IDENT and LIBAVDEVICE_IDENT. +-------- 8< --------- FFmpeg 2.7 was cut here -------- 8< --------- -2015-06-07 - 252d620 - lavf 56.20.0 - avio.h - Add avio_put_str16be. +2015-06-04 - cc17b43 - lswr 1.2.100 + Add swr_get_out_samples() -2015-05-13 - f7cafb5 - lavu 54.14.0 - cpu.h +2015-05-27 - c312bfa - lavu 54.26.100 - cpu.h Add AV_CPU_FLAG_AVXSLOW. -2015-05-13 - e7c5e17 - lavc 56.23.0 +2015-05-26 - 1fb9b2a - lavu 54.25.100 - rational.h + Add av_q2intfloat(). + +2015-05-13 - cc48409 / e7c5e17 - lavc 56.39.100 / 56.23.0 Add av_vda_default_init2. -2015-04-19 - c253340 - lavu 54.12.0 +2015-05-11 - 541d75f - lavf 56.33.100 - avformat.h + Add AVOpenCallback AVFormatContext.open_cb + +2015-05-07 - a7dd933 - 56.38.100 - avcodec.h + Add av_packet_side_data_name(). + +2015-05-07 - 01e59d4 - 56.37.102 - avcodec.h + Add FF_PROFILE_VP9_2 and FF_PROFILE_VP9_3. + +2015-05-04 - 079b7f6 - 56.37.100 - avcodec.h + Add FF_PROFILE_VP9_0 and FF_PROFILE_VP9_1. + +2015-04-22 - 748d481 - lavf 56.31.100 - avformat.h + Add AVFMT_FLAG_FAST_SEEK flag. Some formats (initially mp3) use it to enable + fast, but inaccurate seeking. + +2015-04-20 - 8e8219e / c253340 - lavu 54.23.100 / 54.12.0 - log.h Add AV_LOG_TRACE for extremely verbose debugging. -2015-04-07 - 27f2746 - lavu 54.11.0 - Add av_small_strptime(). +2015-04-02 - 26e0e393 - lavf 56.29.100 - avio.h + Add AVIODirEntryType.AVIO_ENTRY_SERVER. + Add AVIODirEntryType.AVIO_ENTRY_SHARE. + Add AVIODirEntryType.AVIO_ENTRY_WORKGROUP. -2015-03-29 - 6fe2641 - lavc 56.22.0 - Add FF_PROFILE_DTS_EXPRESS. +2015-03-31 - 3188696 - lavu 54.22.100 - avstring.h + Add av_append_path_component() -2015-03-29 - c484561 - lavu 54.10.0 +2015-03-27 - 184084c - lavf 56.27.100 - avio.h url.h + New directory listing API. + + Add AVIODirEntryType enum. + Add AVIODirEntry, AVIODirContext structures. + Add avio_open_dir(), avio_read_dir(), avio_close_dir(), avio_free_directory_entry(). + Add ff_alloc_dir_entry(). + Extend URLProtocol with url_open_dir(), url_read_dir(), url_close_dir(). + +2015-03-29 - 268ff17 / c484561 - lavu 54.21.100 / 54.10.0 - pixfmt.h Add AV_PIX_FMT_MMAL for MMAL hardware acceleration. -2015-02-19 - 31d2039 - lavc 56.13 +2015-03-19 - 11fe56c - 56.29.100 / lavc 56.22.0 + Add FF_PROFILE_DTS_EXPRESS. + +-------- 8< --------- FFmpeg 2.6 was cut here -------- 8< --------- + +2015-03-04 - cca4476 - lavf 56.25.100 + Add avformat_flush() + +2015-03-03 - 81a9126 - lavf 56.24.100 + Add avio_put_str16be() + +2015-02-19 - 560eb71 / 31d2039 - lavc 56.23.100 / 56.13.0 Add width, height, coded_width, coded_height and format to AVCodecParserContext. -2015-02-19 - 5b1d9ce - lavu 54.9.0 +2015-02-19 - e375511 / 5b1d9ce - lavu 54.19.100 / 54.9.0 Add AV_PIX_FMT_QSV for QSV hardware acceleration. -2015-01-27 - 728685f - lavc 56.12.0, lavu 54.8.0 - avcodec.h, frame.h +2015-02-14 - ba22295 - lavc 56.21.102 + Deprecate VIMA decoder. + +2015-01-27 - 62a82c6 / 728685f - lavc 56.21.100 / 56.12.0, lavu 54.18.100 / 54.8.0 - avcodec.h, frame.h Add AV_PKT_DATA_AUDIO_SERVICE_TYPE and AV_FRAME_DATA_AUDIO_SERVICE_TYPE for storing the audio service type as side data. -2015-01-14 - e2ad0b6 - lavu 54.6.0 - imgutils.h - Add utility functions for image manipulation: av_image_get_buffer_size() - av_image_fill_arrays() and av_image_copy_to_buffer(). +2015-01-16 - a47c933 - lavf 56.19.100 - avformat.h + Add data_codec and data_codec_id for storing codec of data stream -2014-12-25 - c220a60 - lavc 56.10.0 - vdpau.h +2015-01-11 - 007c33d - lavd 56.4.100 - avdevice.h + Add avdevice_list_input_sources(). + Add avdevice_list_output_sinks(). + +2014-12-25 - d7aaeea / c220a60 - lavc 56.19.100 / 56.10.0 - vdpau.h Add av_vdpau_get_surface_parameters(). -2014-12-25 - 6c99c92 - lavc 56.9.0 - avcodec.h +2014-12-25 - ddb9a24 / 6c99c92 - lavc 56.18.100 / 56.9.0 - avcodec.h Add AV_HWACCEL_FLAG_ALLOW_HIGH_DEPTH flag to av_vdpau_bind_context(). -2014-12-25 - 57b6704 - lavc 56.8.0 - avcodec.h +2014-12-25 - d16079a / 57b6704 - lavc 56.17.100 / 56.8.0 - avcodec.h Add AVCodecContext.sw_pix_fmt. -2014-11-07 - 1384df6 - lavf 56.06.3 - avformat.h - Add AVFormatContext.avoid_negative_ts. +2014-12-04 - 6e9ac02 - lavc 56.14.100 - dv_profile.h + Add av_dv_codec_profile2(). -2014-11-06 - 5e80fb7 - lavc 56.6.0 - vorbis_parser.h - Add a public API for parsing vorbis packets. +-------- 8< --------- FFmpeg 2.5 was cut here -------- 8< --------- + +2014-11-21 - ab922f9 - lavu 54.15.100 - dict.h + Add av_dict_get_string(). -2014-10-24 - 1bd0bdc - lavu 54.5.0 - time.h - Add av_gettime_relative(). +2014-11-18 - a54a51c - lavu 54.14.100 - float_dsp.h + Add avpriv_float_dsp_alloc(). -2014-10-15 - 7ea1b34 - lavc 56.5.0 - avcodec.h +2014-11-16 - 6690d4c3 - lavf 56.13.100 - avformat.h + Add AVStream.recommended_encoder_configuration with accessors. + +2014-11-16 - bee5844d - lavu 54.13.100 - opt.h + Add av_opt_serialize(). + +2014-11-16 - eec69332 - lavu 54.12.100 - opt.h + Add av_opt_is_set_to_default(). + +2014-11-06 - 44fa267 / 5e80fb7 - lavc 56.11.100 / 56.6.0 - vorbis_parser.h + Add a public API for parsing vorbis packets. + +2014-10-15 - 17085a0 / 7ea1b34 - lavc 56.7.100 / 56.5.0 - avcodec.h Replace AVCodecContext.time_base used for decoding with AVCodecContext.framerate. -2014-10-15 - d565fef1- lavc 56.4.0 - avcodec.h +2014-10-15 - 51c810e / d565fef1 - lavc 56.6.100 / 56.4.0 - avcodec.h Add AV_HWACCEL_FLAG_IGNORE_LEVEL flag to av_vdpau_bind_context(). -2014-10-13 - 2df0c32 - lavc 56.03.0 - avcodec.h +2014-10-13 - da21895 / 2df0c32e - lavc 56.5.100 / 56.3.0 - avcodec.h Add AVCodecContext.initial_padding. Deprecate the use of AVCodecContext.delay for audio encoding. -2014-10-08 - 5a419b2 - lavu 54.04.0 - pixdesc.h +2014-10-08 - bb44f7d / 5a419b2 - lavu 54.10.100 / 54.4.0 - pixdesc.h Add API to return the name of frame and context color properties. -2014-10-06 - e3e158e - lavc 56.2.0 - vdpau.h +2014-10-06 - a61899a / e3e158e - lavc 56.3.100 / 56.2.0 - vdpau.h Add av_vdpau_bind_context(). This function should now be used for creating (or resetting) a AVVDPAUContext instead of av_vdpau_alloc_context(). -2014-08-25 - b263f8f - lavf 56.03.0 - avformat.h - Add AVFormatContext.max_ts_probe. +2014-10-02 - cdd6f05 - lavc 56.2.100 - avcodec.h +2014-10-02 - cdd6f05 - lavu 54.9.100 - frame.h + Add AV_FRAME_DATA_SKIP_SAMPLES. Add lavc CODEC_FLAG2_SKIP_MANUAL and + AVOption "skip_manual", which makes lavc export skip information via + AV_FRAME_DATA_SKIP_SAMPLES AVFrame side data, instead of skipping and + discarding samples automatically. + +2014-10-02 - 0d92b0d - lavu 54.8.100 - avstring.h + Add av_match_list() + +2014-09-24 - ac68295 - libpostproc 53.1.100 + Add visualization support + +2014-09-19 - 6edd6a4 - lavc 56.1.101 - dv_profile.h + deprecate avpriv_dv_frame_profile2(), which was made public by accident. -------------------------------8<------------------------------------- - 11 branch was cut here ------------------------------>8-------------------------------------- -2014-08-28 - 9301486 - lavc 56.1.0 - avcodec.h +-------- 8< --------- FFmpeg 2.4 was cut here -------- 8< --------- + +2014-08-25 - 215db29 / b263f8f - lavf 56.3.100 / 56.3.0 - avformat.h + Add AVFormatContext.max_ts_probe. + +2014-08-28 - f30a815 / 9301486 - lavc 56.1.100 / 56.1.0 - avcodec.h Add AV_PKT_DATA_STEREO3D to export container-level stereo3d information. -2014-08-13 - 8ddc326 - lavu 54.03.0 - mem.h +2014-08-23 - 8fc9bd0 - lavu 54.7.100 - dict.h + AV_DICT_DONT_STRDUP_KEY and AV_DICT_DONT_STRDUP_VAL arguments are now + freed even on error. This is consistent with the behaviour all users + of it we could find expect. + +2014-08-21 - 980a5b0 - lavu 54.6.100 - frame.h motion_vector.h + Add AV_FRAME_DATA_MOTION_VECTORS side data and AVMotionVector structure + +2014-08-16 - b7d5e01 - lswr 1.1.100 - swresample.h + Add AVFrame based API + +2014-08-16 - c2829dc - lavu 54.4.100 - dict.h + Add av_dict_set_int helper function. + +2014-08-13 - c8571c6 / 8ddc326 - lavu 54.3.100 / 54.3.0 - mem.h Add av_strndup(). -2014-08-13 - a8c104a - lavu 54.02.0 - opt.h +2014-08-13 - 2ba4577 / a8c104a - lavu 54.2.100 / 54.2.0 - opt.h Add av_opt_get_dict_val/set_dict_val with AV_OPT_TYPE_DICT to support dictionary types being set as options. @@ -198,334 +335,740 @@ API changes, most recent first: Add AVFormatContext.event_flags and AVStream.event_flags for signaling to the user when events happen in the file/stream. -2014-08-10 - fb1ddcd - lavr 2.1.0 - avresample.h +2014-08-10 - 78eaaa8 / fb1ddcd - lavr 2.1.0 - avresample.h Add avresample_convert_frame() and avresample_config(). -2014-08-10 - fb1ddcd - lavu 54.1.0 - error.h +2014-08-10 - 78eaaa8 / fb1ddcd - lavu 54.1.100 / 54.1.0 - error.h Add AVERROR_INPUT_CHANGED and AVERROR_OUTPUT_CHANGED. -2014-08-08 - d35b94f - lavc 55.57.4 - avcodec.h +2014-08-08 - 3841f2a / d35b94f - lavc 55.73.102 / 55.57.4 - avcodec.h Deprecate FF_IDCT_XVIDMMX define and xvidmmx idct option. Replaced by FF_IDCT_XVID and xvid respectively. +2014-08-08 - 5c3c671 - lavf 55.53.100 - avio.h + Add avio_feof() and deprecate url_feof(). + 2014-08-07 - bb78903 - lsws 2.1.3 - swscale.h - sws_getCachedContext is not going to be removed in the future. + sws_getContext is not going to be removed in the future. -2014-08-07 - ad1ee5f - lavc 55.57.3 - avcodec.h +2014-08-07 - a561662 / ad1ee5f - lavc 55.73.101 / 55.57.3 - avcodec.h reordered_opaque is not going to be removed in the future. -2014-08-04 - e9abafc - lavu 53.22.0 - pixfmt.h +2014-08-02 - 28a2107 - lavu 52.98.100 - pixelutils.h + Add pixelutils API with SAD functions + +2014-08-04 - 6017c98 / e9abafc - lavu 52.97.100 / 53.22.0 - pixfmt.h Add AV_PIX_FMT_YA16 pixel format for 16 bit packed gray with alpha. -2014-08-04 - e96c3b8 - lavu 53.21.1 - avstring.h +2014-08-04 - 4c8bc6f / e96c3b8 - lavu 52.96.101 / 53.21.1 - avstring.h Rename AV_PIX_FMT_Y400A to AV_PIX_FMT_YA8 to better identify the format. An alias pixel format and color space name are provided for compatibility. -2014-08-04 - d2962e9 - lavu 53.21.0 - pixdesc.h +2014-08-04 - 073c074 / d2962e9 - lavu 52.96.100 / 53.21.0 - pixdesc.h Support name aliases for pixel formats. -2014-08-03 - 1ef9e83 - lavc 55.57.2 - avcodec.h -2014-08-03 - 1ef9e83 - lavu 53.20.0 - frame.h +2014-08-03 - 71d008e / 1ef9e83 - lavc 55.72.101 / 55.57.2 - avcodec.h +2014-08-03 - 71d008e / 1ef9e83 - lavu 52.95.100 / 53.20.0 - frame.h Deprecate AVCodecContext.dtg_active_format and use side-data instead. -2014-08-03 - 9f17685 - lavc 55.57.1 - avcodec.h +2014-08-03 - e680c73 - lavc 55.72.100 - avcodec.h + Add get_pixels() to AVDCT + +2014-08-03 - 9400603 / 9f17685 - lavc 55.71.101 / 55.57.1 - avcodec.h Deprecate unused FF_IDCT_IPP define and ipp avcodec option. Deprecate unused FF_DEBUG_PTS define and pts avcodec option. Deprecate unused FF_CODER_TYPE_DEFLATE define and deflate avcodec option. Deprecate unused FF_DCT_INT define and int avcodec option. Deprecate unused avcodec option scenechange_factor. -2014-07-29 - 69e7336 - lavu 53.19.0 - avstring.h +2014-07-30 - ba3e331 - lavu 52.94.100 - frame.h + Add av_frame_side_data_name() + +2014-07-29 - 80a3a66 / 3a19405 - lavf 56.01.100 / 56.01.0 - avformat.h + Add mime_type field to AVProbeData, which now MUST be initialized in + order to avoid uninitialized reads of the mime_type pointer, likely + leading to crashes. + Typically, this means you will do 'AVProbeData pd = { 0 };' instead of + 'AVProbeData pd;'. + +2014-07-29 - 31e0b5d / 69e7336 - lavu 52.92.100 / 53.19.0 - avstring.h Make name matching function from lavf public as av_match_name(). -2014-07-28 - c5fca01 - lavc 55.57.0 - avcodec.h +2014-07-28 - 2e5c8b0 / c5fca01 - lavc 55.71.100 / 55.57.0 - avcodec.h Add AV_CODEC_PROP_REORDER to mark codecs supporting frame reordering. -2014-07-09 - a54f03b - lavu 53.18.0 - display.h +2014-07-27 - ff9a154 - lavf 55.50.100 - avformat.h + New field int64_t probesize2 instead of deprecated + field int probesize. + +2014-07-27 - 932ff70 - lavc 55.70.100 - avdct.h + Add AVDCT / avcodec_dct_alloc() / avcodec_dct_init(). + +2014-07-23 - 8a4c086 - lavf 55.49.100 - avio.h + Add avio_read_to_bprint() + + +-------- 8< --------- FFmpeg 2.3 was cut here -------- 8< --------- + +2014-07-14 - 62227a7 - lavf 55.47.100 - avformat.h + Add av_stream_get_parser() + +2014-07-09 - c67690f / a54f03b - lavu 52.92.100 / 53.18.0 - display.h Add av_display_matrix_flip() to flip the transformation matrix. -2014-07-09 - f6ee61f - lavc 55.56.0 - dv_profile.h +2014-07-09 - 1b58f13 / f6ee61f - lavc 55.69.100 / 55.56.0 - dv_profile.h Add a public API for DV profile handling. -2014-06-20 - 9e500ef - lavu 53.17.0 - imgutils.h +2014-06-20 - 0dceefc / 9e500ef - lavu 52.90.100 / 53.17.0 - imgutils.h Add av_image_check_sar(). -2014-06-20 - 874390e - lavc 55.55.0 - avcodec.h +2014-06-20 - 4a99333 / 874390e - lavc 55.68.100 / 55.55.0 - avcodec.h Add av_packet_rescale_ts() to simplify timestamp conversion. -2014-06-18 - 194be1f - lavf 55.20.0 - avformat.h +2014-06-18 - ac293b6 / 194be1f - lavf 55.44.100 / 55.20.0 - avformat.h The proper way for providing a hint about the desired timebase to the muxers is now setting AVStream.time_base, instead of AVStream.codec.time_base as was done previously. The old method is now deprecated. -2014-06-01 - 0957b27 - lavc 55.54.0 - avcodec.h +2014-06-11 - 67d29da - lavc 55.66.101 - avcodec.h + Increase FF_INPUT_BUFFER_PADDING_SIZE to 32 due to some corner cases needing + it + +2014-06-10 - 5482780 - lavf 55.43.100 - avformat.h + New field int64_t max_analyze_duration2 instead of deprecated + int max_analyze_duration. + +2014-05-30 - 00759d7 - lavu 52.89.100 - opt.h + Add av_opt_copy() + +2014-06-01 - 03bb99a / 0957b27 - lavc 55.66.100 / 55.54.0 - avcodec.h Add AVCodecContext.side_data_only_packets to allow encoders to output packets with only side data. This option may become mandatory in the future, so all users are recommended to update their code and enable this option. -2014-06-01 - 8c02adc - lavu 53.16.0 - frame.h, pixfmt.h +2014-06-01 - 6e8e9f1 / 8c02adc - lavu 52.88.100 / 53.16.0 - frame.h, pixfmt.h Move all color-related enums (AVColorPrimaries, AVColorSpace, AVColorRange, AVColorTransferCharacteristic, and AVChromaLocation) inside lavu. - Add AVFrame fields for them on the next lavu major bump. + And add AVFrame fields for them. -2014-05-28 - b2d4565 - lavr 1.3.0 - avresample.h +2014-05-29 - bdb2e80 / b2d4565 - lavr 1.3.0 - avresample.h Add avresample_max_output_samples -2014-05-28 - 6d21259 - lavf 55.19.0 - avformat.h +2014-05-28 - d858ee7 / 6d21259 - lavf 55.42.100 / 55.19.0 - avformat.h Add strict_std_compliance and related AVOptions to support experimental muxing. -2014-05-20 - c23c96b - lavf 55.18.0 - avformat.h +2014-05-26 - 55cc60c - lavu 52.87.100 - threadmessage.h + Add thread message queue API. + +2014-05-26 - c37d179 - lavf 55.41.100 - avformat.h + Add format_probesize to AVFormatContext. + +2014-05-20 - 7d25af1 / c23c96b - lavf 55.39.100 / 55.18.0 - avformat.h Add av_stream_get_side_data() to access stream-level side data in the same way as av_packet_get_side_data(). -2014-05-19 - bddd8cb - lavu 53.15.0 - frame.h, display.h +2014-05-20 - 7336e39 - lavu 52.86.100 - fifo.h + Add av_fifo_alloc_array() function. + +2014-05-19 - ef1d4ee / bddd8cb - lavu 52.85.100 / 53.15.0 - frame.h, display.h Add AV_FRAME_DATA_DISPLAYMATRIX for exporting frame-level spatial rendering on video frames for proper display. -2014-05-19 - bddd8cb - lavc 55.53.0 - avcodec.h +2014-05-19 - ef1d4ee / bddd8cb - lavc 55.64.100 / 55.53.0 - avcodec.h Add AV_PKT_DATA_DISPLAYMATRIX for exporting packet-level spatial rendering on video frames for proper display. -2014-05-19 - a312f71 - lavf 55.17.1 - avformat.h +2014-05-19 - 999a99c / a312f71 - lavf 55.38.101 / 55.17.1 - avformat.h Deprecate AVStream.pts and the AVFrac struct, which was its only use case. - Those fields were poorly defined and not meant to be public, so there is - no replacement for them. + See use av_stream_get_end_pts() -2014-05-18 - fd05602 - lavc 55.52.0 - avcodec.h +2014-05-18 - 68c0518 / fd05602 - lavc 55.63.100 / 55.52.0 - avcodec.h Add avcodec_free_context(). From now on it should be used for freeing AVCodecContext. -2014-05-15 - 0c1959b - lavf 55.17.0 - avformat.h +2014-05-17 - 0eec06e / 1bd0bdc - lavu 52.84.100 / 54.5.0 - time.h + Add av_gettime_relative() av_gettime_relative_is_monotonic() + +2014-05-15 - eacf7d6 / 0c1959b - lavf 55.38.100 / 55.17.0 - avformat.h Add AVFMT_FLAG_BITEXACT flag. Muxers now use it instead of checking CODEC_FLAG_BITEXACT on the first stream. -2014-05-11 - 66e6c8a - lavu 53.14.0 - pixfmt.h +2014-05-15 - 96cb4c8 - lswr 0.19.100 - swresample.h + Add swr_close() + +2014-05-11 - 14aef38 / 66e6c8a - lavu 52.83.100 / 53.14.0 - pixfmt.h Add AV_PIX_FMT_VDA for new-style VDA acceleration. -2014-05-01 - a2941c8 - lavc 55.50.3 - avcodec.h +2014-05-07 - 351f611 - lavu 52.82.100 - fifo.h + Add av_fifo_freep() function. + +2014-05-02 - ba52fb11 - lavu 52.81.100 - opt.h + Add av_opt_set_dict2() function. + +2014-05-01 - e77b985 / a2941c8 - lavc 55.60.103 / 55.50.3 - avcodec.h Deprecate CODEC_FLAG_MV0. It is replaced by the flag "mv0" in the "mpv_flags" private option of the mpegvideo encoders. -2014-05-01 - 6484149 - lavc 55.50.2 - avcodec.h +2014-05-01 - e40ae8c / 6484149 - lavc 55.60.102 / 55.50.2 - avcodec.h Deprecate CODEC_FLAG_GMC. It is replaced by the "gmc" private option of the libxvid encoder. -2014-05-01 - b2c3171 - lavc 55.50.1 - avcodec.h +2014-05-01 - 1851643 / b2c3171 - lavc 55.60.101 / 55.50.1 - avcodec.h Deprecate CODEC_FLAG_NORMALIZE_AQP. It is replaced by the flag "naq" in the "mpv_flags" private option of the mpegvideo encoders. -2014-05-01 - 5fcceda - avcodec.h +2014-05-01 - cac07d0 / 5fcceda - avcodec.h Deprecate CODEC_FLAG_INPUT_PRESERVED. Its functionality is replaced by passing reference-counted frames to encoders. -2014-04-28 - ed4b757 - lavc 55.50.0 - dxva2.h +2014-04-30 - 617e866 - lavu 52.81.100 - pixdesc.h + Add av_find_best_pix_fmt_of_2(), av_get_pix_fmt_loss() + Deprecate avcodec_get_pix_fmt_loss(), avcodec_find_best_pix_fmt_of_2() + +2014-04-29 - 1bf6396 - lavc 55.60.100 - avcodec.h + Add AVCodecDescriptor.mime_types field. + +2014-04-29 - b804eb4 - lavu 52.80.100 - hash.h + Add av_hash_final_bin(), av_hash_final_hex() and av_hash_final_b64(). + +2014-03-07 - 8b2a130 - lavc 55.50.0 / 55.53.100 - dxva2.h Add FF_DXVA2_WORKAROUND_INTEL_CLEARVIDEO for old Intel GPUs. -2014-04-22 - 502512e - lavu 53.13.0 - avutil.h +2014-04-22 - 502512e /dac7e8a - lavu 53.13.0 / 52.78.100 - avutil.h Add av_get_time_base_q(). -2014-04-17 - 0983d48 - lavu 53.12.0 - crc.h +2014-04-17 - a8d01a7 / 0983d48 - lavu 53.12.0 / 52.77.100 - crc.h Add AV_CRC_16_ANSI_LE crc variant. -2014-04-07 - 8b17243 - lavu 53.11.0 - pixfmt.h +2014-04-15 - ef818d8 - lavf 55.37.101 - avformat.h + Add av_format_inject_global_side_data() + +2014-04-12 - 4f698be - lavu 52.76.100 - log.h + Add av_log_get_flags() + +2014-04-11 - 6db42a2b - lavd 55.12.100 - avdevice.h + Add avdevice_capabilities_create() function. + Add avdevice_capabilities_free() function. + +2014-04-07 - 0a1cc04 / 8b17243 - lavu 52.75.100 / 53.11.0 - pixfmt.h Add AV_PIX_FMT_YVYU422 pixel format. -2014-04-04 - 8542f9c - lavu 53.10.0 - replaygain.h +2014-04-04 - c1d0536 / 8542f9c - lavu 52.74.100 / 53.10.0 - replaygain.h Full scale for peak values is now 100000 (instead of UINT32_MAX) and values may overflow. -2014-04-03 - 7763118 - lavu 53.09.0 - log.h +2014-04-03 - c16e006 / 7763118 - lavu 52.73.100 / 53.9.0 - log.h Add AV_LOG(c) macro to have 256 color debug messages. -2014-03-24 - d161ae0 - lavu 53.08.0 - frame.h +2014-04-03 - eaed4da9 - lavu 52.72.100 - opt.h + Add AV_OPT_MULTI_COMPONENT_RANGE define to allow return + multi-component option ranges. + +2014-03-29 - cd50a44b - lavu 52.70.100 - mem.h + Add av_dynarray_add_nofree() function. + +2014-02-24 - 3e1f241 / d161ae0 - lavu 52.69.100 / 53.8.0 - frame.h Add av_frame_remove_side_data() for removing a single side data instance from a frame. -2014-03-24 - 5a7e35d - lavu 53.07.0 - frame.h, replaygain.h +2014-03-24 - 83e8978 / 5a7e35d - lavu 52.68.100 / 53.7.0 - frame.h, replaygain.h Add AV_FRAME_DATA_REPLAYGAIN for exporting replaygain tags. Add a new header replaygain.h with the AVReplayGain struct. -2014-03-24 - 5a7e35d - lavc 55.36.0 - avcodec.h +2014-03-24 - 83e8978 / 5a7e35d - lavc 55.54.100 / 55.36.0 - avcodec.h Add AV_PKT_DATA_REPLAYGAIN for exporting replaygain tags. -2014-03-24 - 25b3258 - lavf 55.13.0 - avformat.h +2014-03-24 - 595ba3b / 25b3258 - lavf 55.35.100 / 55.13.0 - avformat.h Add AVStream.side_data and AVStream.nb_side_data for exporting stream-global side data (e.g. replaygain tags, video rotation) -2014-03-24 - 0e2c3ee - lavc 55.35.0 - avcodec.h +2014-03-24 - bd34e26 / 0e2c3ee - lavc 55.53.100 / 55.35.0 - avcodec.h Give the name AVPacketSideData to the previously anonymous struct used for AVPacket.side_data. -2014-03-16 - 1481d24 - lavu 53.06.0 - pixfmt.h - Add RGBA64 pixel format and variants. -2014-02-24 - 1155fd0 - lavu 53.05.0 - frame.h +-------- 8< --------- FFmpeg 2.2 was cut here -------- 8< --------- + +2014-03-18 - 37c07d4 - lsws 2.5.102 + Make gray16 full-scale. + +2014-03-16 - 6b1ca17 / 1481d24 - lavu 52.67.100 / 53.6.0 - pixfmt.h + Add RGBA64_LIBAV pixel format and variants for compatibility + +2014-03-11 - 3f3229c - lavf 55.34.101 - avformat.h + Set AVFormatContext.start_time_realtime when demuxing. + +2014-03-03 - 06fed440 - lavd 55.11.100 - avdevice.h + Add av_input_audio_device_next(). + Add av_input_video_device_next(). + Add av_output_audio_device_next(). + Add av_output_video_device_next(). + +2014-02-24 - fff5262 / 1155fd0 - lavu 52.66.100 / 53.5.0 - frame.h Add av_frame_copy() for copying the frame data. -2014-02-22 - 7e86c27 - lavr 1.2.0 - avresample.h +2014-02-24 - a66be60 - lswr 0.18.100 - swresample.h + Add swr_is_initialized() for checking whether a resample context is initialized. + +2014-02-22 - 5367c0b / 7e86c27 - lavr 1.2.0 - avresample.h Add avresample_is_open() for checking whether a resample context is open. -2014-02-19 - c3ecd96 - lavu 53.04.0 - opt.h +2014-02-19 - 6a24d77 / c3ecd96 - lavu 52.65.100 / 53.4.0 - opt.h Add AV_OPT_FLAG_EXPORT and AV_OPT_FLAG_READONLY to mark options meant (only) for reading. -2014-02-19 - 6bb8720 - lavu 53.03.01 - opt.h +2014-02-19 - f4c8d00 / 6bb8720 - lavu 52.64.101 / 53.3.1 - opt.h Deprecate unused AV_OPT_FLAG_METADATA. -------------------------------8<------------------------------------- - 10 branch was cut here ------------------------------>8-------------------------------------- +2014-02-16 - 81c3f81 - lavd 55.10.100 - avdevice.h + Add avdevice_list_devices() and avdevice_free_list_devices() + +2014-02-16 - db3c970 - lavf 55.33.100 - avio.h + Add avio_find_protocol_name() to find out the name of the protocol that would + be selected for a given URL. -2014-02-15 - c98f316 - lavu 53.3.0 - frame.h +2014-02-15 - a2bc6c1 / c98f316 - lavu 52.64.100 / 53.3.0 - frame.h Add AV_FRAME_DATA_DOWNMIX_INFO value to the AVFrameSideDataType enum and downmix_info.h API, which identify downmix-related metadata. -2014-02-04 - d9ae103 - lavf 55.11.0 - avformat.h +2014-02-11 - 1b05ac2 - lavf 55.32.100 - avformat.h + Add av_write_uncoded_frame() and av_interleaved_write_uncoded_frame(). + +2014-02-04 - 3adb5f8 / d9ae103 - lavf 55.30.100 / 55.11.0 - avformat.h Add AVFormatContext.max_interleave_delta for controlling amount of buffering when interleaving. -2014-01-20 - 93c553c - lavc 55.32.1 - avcodec.h +2014-02-02 - 5871ee5 - lavf 55.29.100 - avformat.h + Add output_ts_offset muxing option to AVFormatContext. + +2014-01-27 - 102bd64 - lavd 55.7.100 - avdevice.h + lavf 55.28.100 - avformat.h + Add avdevice_dev_to_app_control_message() function. + +2014-01-27 - 7151411 - lavd 55.6.100 - avdevice.h + lavf 55.27.100 - avformat.h + Add avdevice_app_to_dev_control_message() function. + +2014-01-24 - 86bee79 - lavf 55.26.100 - avformat.h + Add AVFormatContext option metadata_header_padding to allow control over the + amount of padding added. + +2014-01-20 - eef74b2 / 93c553c - lavc 55.48.102 / 55.32.1 - avcodec.h Edges are not required anymore on video buffers allocated by get_buffer2() (i.e. as if the CODEC_FLAG_EMU_EDGE flag was always on). Deprecate CODEC_FLAG_EMU_EDGE and avcodec_get_edge_width(). -2014-01-05 - 5b4797a - lavu 53.2.0 - frame.h +2014-01-19 - 1a193c4 - lavf 55.25.100 - avformat.h + Add avformat_get_mov_video_tags() and avformat_get_mov_audio_tags(). + +2014-01-19 - 3532dd5 - lavu 52.63.100 - rational.h + Add av_make_q() function. + +2014-01-05 - 4cf4da9 / 5b4797a - lavu 52.62.100 / 53.2.0 - frame.h Add AV_FRAME_DATA_MATRIXENCODING value to the AVFrameSideDataType enum, which identifies AVMatrixEncoding data. -2014-01-05 - 5c437fb - lavu 53.1.0 - channel_layout.h +2014-01-05 - 751385f / 5c437fb - lavu 52.61.100 / 53.1.0 - channel_layout.h Add values for various Dolby flags to the AVMatrixEncoding enum. -2013-12-20 - 2a41826 - lavc 55.30.0 - avcodec.h - Add HEVC profiles +2014-01-04 - b317f94 - lavu 52.60.100 - mathematics.h + Add av_add_stable() function. + +2013-12-22 - 911676c - lavu 52.59.100 - avstring.h + Add av_strnlen() function. + +2013-12-09 - 64f73ac - lavu 52.57.100 - opencl.h + Add av_opencl_benchmark() function. -2013-12-11 - b9fb59d,9431356,d7b3ee9 - lavc 55.28.1 - avcodec.h +2013-11-30 - 82b2e9c - lavu 52.56.100 - ffversion.h + Moves version.h to libavutil/ffversion.h. + Install ffversion.h and make it public. + +2013-12-11 - 29c83d2 / b9fb59d,409a143 / 9431356,44967ab / d7b3ee9 - lavc 55.45.101 / 55.28.1 - avcodec.h av_frame_alloc(), av_frame_unref() and av_frame_free() now can and should be used instead of avcodec_alloc_frame(), avcodec_get_frame_defaults() and avcodec_free_frame() respectively. The latter three functions are deprecated. -2013-12-09 - 7e244c6- - lavu 52.20.0 - frame.h +2013-12-09 - 7a60348 / 7e244c6- - lavu 52.58.100 / 52.20.0 - frame.h Add AV_FRAME_DATA_STEREO3D value to the AVFrameSideDataType enum and stereo3d.h API, that identify codec-independent stereo3d information. -2013-11-26 - 1eaac1d- - lavu 52.19.0 - frame.h +2013-11-26 - 625b290 / 1eaac1d- - lavu 52.55.100 / 52.19.0 - frame.h Add AV_FRAME_DATA_A53_CC value to the AVFrameSideDataType enum, which identifies ATSC A53 Part 4 Closed Captions data. -2013-11-14 - cce3e0a - lavu 52.18.0 - mem.h +2013-11-22 - 6859065 - lavu 52.54.100 - avstring.h + Add av_utf8_decode() function. + +2013-11-22 - fb7d70c - lavc 55.44.100 - avcodec.h + Add HEVC profiles + +2013-11-20 - c28b61c - lavc 55.44.100 - avcodec.h + Add av_packet_{un,}pack_dictionary() + Add AV_PKT_METADATA_UPDATE side data type, used to transmit key/value + strings between a stream and the application. + +2013-11-14 - 7c888ae / cce3e0a - lavu 52.53.100 / 52.18.0 - mem.h Move av_fast_malloc() and av_fast_realloc() for libavcodec to libavutil. -2013-11-14 - 8941971 - lavc 55.27.0 - avcodec.h +2013-11-14 - b71e4d8 / 8941971 - lavc 55.43.100 / 55.27.0 - avcodec.h Deprecate AVCodecContext.error_rate, it is replaced by the 'error_rate' private option of the mpegvideo encoder family. -2013-11-14 - 728c465 - lavc 55.26.0 - vdpau.h +2013-11-14 - 31c09b7 / 728c465 - lavc 55.42.100 / 55.26.0 - vdpau.h Add av_vdpau_get_profile(). Add av_vdpau_alloc_context(). This function must from now on be used for allocating AVVDPAUContext. -2013-11-04 - cd8f772 - lavc 55.25.0 - avcodec.h +2013-11-04 - be41f21 / cd8f772 - lavc 55.41.100 / 55.25.0 - avcodec.h + lavu 52.51.100 - frame.h Add ITU-R BT.2020 and other not yet included values to color primaries, transfer characteristics and colorspaces. -2013-10-31 - 28096e0 - lavu 52.17.0 - frame.h +2013-11-04 - 85cabf1 - lavu 52.50.100 - avutil.h + Add av_fopen_utf8() + +2013-10-31 - 78265fc / 28096e0 - lavu 52.49.100 / 52.17.0 - frame.h Add AVFrame.flags and AV_FRAME_FLAG_CORRUPT. -2013-09-28 - 0767bfd - lavfi 3.11.0 - avfilter.h + +-------- 8< --------- FFmpeg 2.1 was cut here -------- 8< --------- + +2013-10-27 - dbe6f9f - lavc 55.39.100 - avcodec.h + Add CODEC_CAP_DELAY support to avcodec_decode_subtitle2. + +2013-10-27 - d61617a - lavu 52.48.100 - parseutils.h + Add av_get_known_color_name(). + +2013-10-17 - 8696e51 - lavu 52.47.100 - opt.h + Add AV_OPT_TYPE_CHANNEL_LAYOUT and channel layout option handlers + av_opt_get_channel_layout() and av_opt_set_channel_layout(). + +2013-10-06 - ccf96f8 -libswscale 2.5.101 - options.c + Change default scaler to bicubic + +2013-10-03 - e57dba0 - lavc 55.34.100 - avcodec.h + Add av_codec_get_max_lowres() + +2013-10-02 - 5082fcc - lavf 55.19.100 - avformat.h + Add audio/video/subtitle AVCodec fields to AVFormatContext to force specific + decoders + +2013-09-28 - 7381d31 / 0767bfd - lavfi 3.88.100 / 3.11.0 - avfilter.h Add AVFilterGraph.execute and AVFilterGraph.opaque for custom slice threading implementations. -2013-09-21 - e208e6d - lavu 52.16.0 - pixfmt.h +2013-09-21 - 85f8a3c / e208e6d - lavu 52.46.100 / 52.16.0 - pixfmt.h Add interleaved 4:2:2 8/10-bit formats AV_PIX_FMT_NV16 and AV_PIX_FMT_NV20. -2013-09-16 - 3feb3d6 - lavu 52.15.0 - mem.h +2013-09-16 - c74c3fb / 3feb3d6 - lavu 52.44.100 / 52.15.0 - mem.h Add av_reallocp. -2013-08-10 - 5a9a9d4 - lavc 55.16.0 - avcodec.h +2013-09-04 - 3e1f507 - lavc 55.31.101 - avcodec.h + avcodec_close() argument can be NULL. + +2013-09-04 - 36cd017a - lavf 55.16.101 - avformat.h + avformat_close_input() argument can be NULL and point on NULL. + +2013-08-29 - e31db62 - lavf 55.15.100 - avformat.h + Add av_format_get_probe_score(). + +2013-08-15 - 1e0e193 - lsws 2.5.100 - + Add a sws_dither AVOption, allowing to set the dither algorithm used + +2013-08-11 - d404fe35 - lavc 55.27.100 - vdpau.h + Add a render2 alternative to the render callback function. + +2013-08-11 - af05edc - lavc 55.26.100 - vdpau.h + Add allocation function for AVVDPAUContext, allowing + to extend it in the future without breaking ABI/API. + +2013-08-10 - 67a580f / 5a9a9d4 - lavc 55.25.100 / 55.16.0 - avcodec.h Extend AVPacket API with av_packet_unref, av_packet_ref, av_packet_move_ref, av_packet_copy_props, av_packet_free_side_data. -2013-08-05 - f824535 - lavc 55.13.0 - avcodec.h +2013-08-05 - 9547e3e / f824535 - lavc 55.22.100 / 55.13.0 - avcodec.h Deprecate the bitstream-related members from struct AVVDPAUContext. - The bistream buffers no longer need to be explicitly freed. + The bitstream buffers no longer need to be explicitly freed. -2013-08-05 - 549294f - lavc 55.12.0 - avcodec.h +2013-08-05 - 3b805dc / 549294f - lavc 55.21.100 / 55.12.0 - avcodec.h Deprecate the CODEC_CAP_HWACCEL_VDPAU codec capability. Use CODEC_CAP_HWACCEL and select the AV_PIX_FMT_VDPAU format with get_format() instead. -2013-08-05 - a0ad5d0 - lavu 52.14.0 - pixfmt.h +2013-08-05 - 4ee0984 / a0ad5d0 - lavu 52.41.100 / 52.14.0 - pixfmt.h Deprecate AV_PIX_FMT_VDPAU_*. Use AV_PIX_FMT_VDPAU instead. -2013-08-02 - a8b1927 - lavc 55.11.0 - avcodec.h +2013-08-02 - 82fdfe8 / a8b1927 - lavc 55.20.100 / 55.11.0 - avcodec.h Add output_picture_number to AVCodecParserContext. -2013-06-24 - 95d5246 - lavc 55.10.0 - avcodec.h +2013-07-23 - abc8110 - lavc 55.19.100 - avcodec.h + Add avcodec_chroma_pos_to_enum() + Add avcodec_enum_to_chroma_pos() + + +-------- 8< --------- FFmpeg 2.0 was cut here -------- 8< --------- + +2013-07-03 - 838bd73 - lavfi 3.78.100 - avfilter.h + Deprecate avfilter_graph_parse() in favor of the equivalent + avfilter_graph_parse_ptr(). + +2013-06-24 - af5f9c0 / 95d5246 - lavc 55.17.100 / 55.10.0 - avcodec.h Add MPEG-2 AAC profiles -2013-06-04 - fc962d4 - lavu 52.13.0 - mem.h +2013-06-25 - af5f9c0 / 95d5246 - lavf 55.10.100 - avformat.h + Add AV_DISPOSITION_* flags to indicate text track kind. + +2013-06-15 - 99b8cd0 - lavu 52.36.100 + Add AVRIPEMD: + av_ripemd_alloc() + av_ripemd_init() + av_ripemd_update() + av_ripemd_final() + +2013-06-10 - 82ef670 - lavu 52.35.101 - hmac.h + Add AV_HMAC_SHA224, AV_HMAC_SHA256, AV_HMAC_SHA384, AV_HMAC_SHA512 + +2013-06-04 - 30b491f / fc962d4 - lavu 52.35.100 / 52.13.0 - mem.h Add av_realloc_array and av_reallocp_array -2013-05-24 - 129bb23 - lavfi 3.10.0 - avfilter.h +2013-05-30 - 682b227 - lavu 52.35.100 + Add AVSHA512: + av_sha512_alloc() + av_sha512_init() + av_sha512_update() + av_sha512_final() + +2013-05-24 - 8d4e969 / 129bb23 - lavfi 3.10.0 / 3.70.100 - avfilter.h Add support for slice multithreading to lavfi. Filters supporting threading are marked with AVFILTER_FLAG_SLICE_THREADS. New fields AVFilterContext.thread_type, AVFilterGraph.thread_type and AVFilterGraph.nb_threads (accessible directly or through AVOptions) may be used to configure multithreading. -2013-05-24 - 2a6eaea - lavu 52.12.0 - cpu.h +2013-05-24 - fe40a9f / 2a6eaea - lavu 52.12.0 / 52.34.100 - cpu.h Add av_cpu_count() function for getting the number of logical CPUs. -2013-05-24 - b493847 - lavc 55.7.0 - avcodec.h +2013-05-24 - 0c25c39 / b493847 - lavc 55.7.0 / 55.12.100 - avcodec.h Add picture_structure to AVCodecParserContext. -2013-05-15 - e6c4ac7 - lavu 52.11.0 - pixdesc.h +2013-05-17 - 3a751ea - lavu 52.33.100 - opt.h + Add AV_OPT_TYPE_COLOR value to AVOptionType enum. + +2013-05-13 - e398416 - lavu 52.31.100 - mem.h + Add av_dynarray2_add(). + +2013-05-12 - 1776177 - lavfi 3.65.100 + Add AVFILTER_FLAG_SUPPORT_TIMELINE* filter flags. + +2013-04-19 - 380cfce - lavc 55.4.100 + Add AV_CODEC_PROP_TEXT_SUB property for text based subtitles codec. + +2013-04-18 - 7c1a002 - lavf 55.3.100 + The matroska demuxer can now output proper verbatim ASS packets. It will + become the default starting lavf 56.0.100. + +2013-04-10 - af0d270 - lavu 25.26.100 - avutil.h,opt.h + Add av_int_list_length() + and av_opt_set_int_list(). + +2013-03-30 - 5c73645 - lavu 52.24.100 - samplefmt.h + Add av_samples_alloc_array_and_samples(). + +2013-03-29 - ef7b6b4 - lavf 55.1.100 - avformat.h + Add av_guess_frame_rate() + +2013-03-20 - 8d928a9 - lavu 52.22.100 - opt.h + Add AV_OPT_TYPE_DURATION value to AVOptionType enum. + +2013-03-17 - 7aa9af5 - lavu 52.20.100 - opt.h + Add AV_OPT_TYPE_VIDEO_RATE value to AVOptionType enum. + + +-------- 8< --------- FFmpeg 1.2 was cut here -------- 8< --------- + +2013-03-07 - 9767ec6 - lavu 52.18.100 - avstring.h,bprint.h + Add av_escape() and av_bprint_escape() API. + +2013-02-24 - b59cd08 - lavfi 3.41.100 - buffersink.h + Add sample_rates field to AVABufferSinkParams. + +2013-01-17 - a1a707f - lavf 54.61.100 + Add av_codec_get_tag2(). + +2013-01-01 - 2eb2e17 - lavfi 3.34.100 + Add avfilter_get_audio_buffer_ref_from_arrays_channels. + + +-------- 8< --------- FFmpeg 1.1 was cut here -------- 8< --------- + +2012-12-20 - 34de47aa - lavfi 3.29.100 - avfilter.h + Add AVFilterLink.channels, avfilter_link_get_channels() + and avfilter_ref_get_channels(). + +2012-12-15 - 96d815fc - lavc 54.80.100 - avcodec.h + Add pkt_size field to AVFrame. + +2012-11-25 - c70ec631 - lavu 52.9.100 - opt.h + Add the following convenience functions to opt.h: + av_opt_get_image_size + av_opt_get_pixel_fmt + av_opt_get_sample_fmt + av_opt_set_image_size + av_opt_set_pixel_fmt + av_opt_set_sample_fmt + +2012-11-17 - 4cd74c81 - lavu 52.8.100 - bprint.h + Add av_bprint_strftime(). + +2012-11-15 - 92648107 - lavu 52.7.100 - opt.h + Add av_opt_get_key_value(). + +2012-11-13 - 79456652 - lavfi 3.23.100 - avfilter.h + Add channels field to AVFilterBufferRefAudioProps. + +2012-11-03 - 481fdeee - lavu 52.3.100 - opt.h + Add AV_OPT_TYPE_SAMPLE_FMT value to AVOptionType enum. + +2012-10-21 - 6fb2fd8 - lavc 54.68.100 - avcodec.h + lavfi 3.20.100 - avfilter.h + Add AV_PKT_DATA_STRINGS_METADATA side data type, used to transmit key/value + strings between AVPacket and AVFrame, and add metadata field to + AVCodecContext (which shall not be accessed by users; see AVFrame metadata + instead). + +2012-09-27 - a70b493 - lavd 54.3.100 - version.h + Add LIBAVDEVICE_IDENT symbol. + +2012-09-27 - a70b493 - lavfi 3.18.100 - version.h + Add LIBAVFILTER_IDENT symbol. + +2012-09-27 - a70b493 - libswr 0.16.100 - version.h + Add LIBSWRESAMPLE_VERSION, LIBSWRESAMPLE_BUILD + and LIBSWRESAMPLE_IDENT symbols. + + +-------- 8< --------- FFmpeg 1.0 was cut here -------- 8< --------- + +2012-09-06 - 29e972f - lavu 51.72.100 - parseutils.h + Add av_small_strptime() time parsing function. + + Can be used as a stripped-down replacement for strptime(), on + systems which do not support it. + +2012-08-25 - 2626cc4 - lavf 54.28.100 + Matroska demuxer now identifies SRT subtitles as AV_CODEC_ID_SUBRIP instead + of AV_CODEC_ID_TEXT. + +2012-08-13 - 5c0d8bc - lavfi 3.8.100 - avfilter.h + Add avfilter_get_class() function, and priv_class field to AVFilter + struct. + +2012-08-12 - a25346e - lavu 51.69.100 - opt.h + Add AV_OPT_FLAG_FILTERING_PARAM symbol in opt.h. + +2012-07-31 - 23fc4dd - lavc 54.46.100 + Add channels field to AVFrame. + +2012-07-30 - f893904 - lavu 51.66.100 + Add av_get_channel_description() + and av_get_standard_channel_layout() functions. + +2012-07-21 - 016a472 - lavc 54.43.100 + Add decode_error_flags field to AVFrame. + +2012-07-20 - b062936 - lavf 54.18.100 + Add avformat_match_stream_specifier() function. + +2012-07-14 - f49ec1b - lavc 54.38.100 - avcodec.h + Add metadata to AVFrame, and the accessor functions + av_frame_get_metadata() and av_frame_set_metadata(). + +2012-07-10 - 0e003d8 - lavc 54.33.100 + Add av_fast_padded_mallocz(). + +2012-07-10 - 21d5609 - lavfi 3.2.0 - avfilter.h + Add init_opaque() callback to AVFilter struct. + +2012-06-26 - e6674e4 - lavu 51.63.100 - imgutils.h + Add functions to libavutil/imgutils.h: + av_image_get_buffer_size() + av_image_fill_arrays() + av_image_copy_to_buffer() + +2012-06-24 - c41899a - lavu 51.62.100 - version.h + version moved from avutil.h to version.h + +2012-04-11 - 359abb1 - lavu 51.58.100 - error.h + Add av_make_error_string() and av_err2str() utilities to + libavutil/error.h. + +2012-06-05 - 62b39d4 - lavc 54.24.100 + Add pkt_duration field to AVFrame. + +2012-05-24 - f2ee065 - lavu 51.54.100 + Move AVPALETTE_SIZE and AVPALETTE_COUNT macros from + libavcodec/avcodec.h to libavutil/pixfmt.h. + +2012-05-14 - 94a9ac1 - lavf 54.5.100 + Add av_guess_sample_aspect_ratio() function. + +2012-04-20 - 65fa7bc - lavfi 2.70.100 + Add avfilter_unref_bufferp() to avfilter.h. + +2012-04-13 - 162e400 - lavfi 2.68.100 + Install libavfilter/asrc_abuffer.h public header. + +2012-03-26 - a67d9cf - lavfi 2.66.100 + Add avfilter_fill_frame_from_{audio_,}buffer_ref() functions. + +2013-05-15 - ff46809 / e6c4ac7 - lavu 52.32.100 / 52.11.0 - pixdesc.h Replace PIX_FMT_* flags with AV_PIX_FMT_FLAG_*. -2013-04-03 - 507b1e4 - lavc 55.4.0 - avcodec.h +2013-04-03 - 6fc58a8 / 507b1e4 - lavc 55.7.100 / 55.4.0 - avcodec.h Add field_order to AVCodecParserContext. -2013-04-19 - 5e83d9a - lavc 55.2.0 - avcodec.h +2013-04-19 - f4b05cd / 5e83d9a - lavc 55.5.100 / 55.2.0 - avcodec.h Add CODEC_FLAG_UNALIGNED to allow decoders to produce unaligned output. -2013-04-11 - lavfi 3.8.0 - 38f0c07 - Move all content from avfiltergraph.h to avfilter.h. Deprecate +2013-04-11 - lavfi 3.53.100 / 3.8.0 + 231fd44 / 38f0c07 - Move all content from avfiltergraph.h to avfilter.h. Deprecate avfilterhraph.h, user applications should include just avfilter.h - bc1a985 - Add avfilter_graph_alloc_filter(), deprecate avfilter_open() and + 86070b8 / bc1a985 - Add avfilter_graph_alloc_filter(), deprecate avfilter_open() and avfilter_graph_add_filter(). - 1113672 - Add AVFilterContext.graph pointing to the AVFilterGraph that contains the + 4fde705 / 1113672 - Add AVFilterContext.graph pointing to the AVFilterGraph that contains the filter. - 48a5ada - Add avfilter_init_str(), deprecate avfilter_init_filter(). - 1ba95a9 - Add avfilter_init_dict(). - 7cdd737 - Add AVFilter.flags field and AVFILTER_FLAG_DYNAMIC_{INPUTS,OUTPUTS} flags. - 7e8fe4b - Add avfilter_pad_count() for counting filter inputs/outputs. - fa2a34c - Add avfilter_next(), deprecate av_filter_next(). + 710b0aa / 48a5ada - Add avfilter_init_str(), deprecate avfilter_init_filter(). + 46de9ba / 1ba95a9 - Add avfilter_init_dict(). + 16fc24b / 7cdd737 - Add AVFilter.flags field and AVFILTER_FLAG_DYNAMIC_{INPUTS,OUTPUTS} flags. + f4db6bf / 7e8fe4b - Add avfilter_pad_count() for counting filter inputs/outputs. + 835cc0f / fa2a34c - Add avfilter_next(), deprecate av_filter_next(). Deprecate avfilter_uninit(). -2013-04-09 - lavfi 3.7.0 - avfilter.h - b439c99 - Add AVFilter.priv_class for exporting filter options through the +2013-04-09 - lavfi 3.51.100 / 3.7.0 - avfilter.h + 0594ef0 / b439c99 - Add AVFilter.priv_class for exporting filter options through the AVOptions API in the similar way private options work in lavc and lavf. - 8114c10 - Add avfilter_get_class(). + 44d4488 / 8114c10 - Add avfilter_get_class(). Switch all filters to use AVOptions. -2013-03-19 - 2c328a9 - lavu 52.9.0 - pixdesc.h +2013-03-19 - 17ebef2 / 2c328a9 - lavu 52.20.100 / 52.9.0 - pixdesc.h Add av_pix_fmt_count_planes() function for counting planes in a pixel format. -2013-03-16 - 42c7c61 - lavfi 3.6.0 +2013-03-16 - ecade98 / 42c7c61 - lavfi 3.47.100 / 3.6.0 Add AVFilterGraph.nb_filters, deprecate AVFilterGraph.filter_count. -2013-03-08 - Reference counted buffers - lavu 52.8.0, lavc 55.0.0, lavf 55.0.0, -lavd 54.0.0, lavfi 3.5.0 - 8e401db, 1cec062 - add a new API for reference counted buffers and buffer +2013-03-08 - Reference counted buffers - lavu 52.8.0, lavc 55.0.100 / 55.0.0, lavf 55.0.100 / 55.0.0, +lavd 54.4.100 / 54.0.0, lavfi 3.5.0 + 36099df / 8e401db, 532f31a / 1cec062 - add a new API for reference counted buffers and buffer pools (new header libavutil/buffer.h). - 1afddbe - add AVPacket.buf to allow reference counting for the AVPacket data. + 2653e12 / 1afddbe - add AVPacket.buf to allow reference counting for the AVPacket data. Add av_packet_from_data() function for constructing packets from av_malloc()ed data. - 7ecc2d4 - move AVFrame from lavc to lavu (new header libavutil/frame.h), add + c4e8821 / 7ecc2d4 - move AVFrame from lavc to lavu (new header libavutil/frame.h), add AVFrame.buf/extended_buf to allow reference counting for the AVFrame data. Add new API for working with reference-counted AVFrames. - 759001c - add the refcounted_frames field to AVCodecContext to make audio and + 80e9e63 / 759001c - add the refcounted_frames field to AVCodecContext to make audio and video decoders return reference-counted frames. Add get_buffer2() callback to AVCodecContext which allocates reference-counted frames. Add avcodec_default_get_buffer2() as the default get_buffer2() @@ -543,70 +1086,62 @@ lavd 54.0.0, lavfi 3.5.0 * qscale_table, qstride, qscale_type, mbskip_table, motion_val, mb_type, dct_coeff, ref_index -- mpegvideo-specific tables, which are not exported anymore. - 7e35037 - switch libavfilter to use AVFrame instead of AVFilterBufferRef. Add + a05a44e / 7e35037 - switch libavfilter to use AVFrame instead of AVFilterBufferRef. Add av_buffersrc_add_frame(), deprecate av_buffersrc_buffer(). Add av_buffersink_get_frame() and av_buffersink_get_samples(), deprecate av_buffersink_read() and av_buffersink_read_samples(). Deprecate AVFilterBufferRef and all functions for working with it. -2013-03-17 - 12c5c1d - lavu 52.8.0 - avstring.h +2013-03-17 - 6c17ff8 / 12c5c1d - lavu 52.19.100 / 52.8.0 - avstring.h Add av_isdigit, av_isgraph, av_isspace, av_isxdigit. -2013-02-23 - 9f12235 - lavfi 3.4.0 - avfiltergraph.h +2013-02-23 - 71cf094 / 9f12235 - lavfi 3.40.100 / 3.4.0 - avfiltergraph.h Add resample_lavr_opts to AVFilterGraph for setting libavresample options for auto-inserted resample filters. -2013-01-25 - 38c1466 - lavu 52.7.0 - dict.h +2013-01-25 - e7e14bc / 38c1466 - lavu 52.17.100 / 52.7.0 - dict.h Add av_dict_parse_string() to set multiple key/value pairs at once from a string. -2013-01-25 - b85a5e8 - lavu 52.6.0 - avstring.h +2013-01-25 - 25be630 / b85a5e8 - lavu 52.16.100 / 52.6.0 - avstring.h Add av_strnstr() -2013-01-15 - 8ee288d - lavu 52.5.0 - hmac.h +2013-01-15 - e7e0186 / 8ee288d - lavu 52.15.100 / 52.5.0 - hmac.h Add AVHMAC. -2013-01-13 - 44e065d - lavc 54.36.0 - vdpau.h +2013-01-13 - 8ee7b38 / 44e065d - lavc 54.87.100 / 54.36.0 - vdpau.h Add AVVDPAUContext struct for VDPAU hardware-accelerated decoding. -2013-01-12 - 169fb94 - lavu 52.4.0 - pixdesc.h +2013-01-12 - dae382b / 169fb94 - lavu 52.14.100 / 52.4.0 - pixdesc.h Add AV_PIX_FMT_VDPAU flag. -2013-01-07 - 074a00d - lavr 1.1.0 +2013-01-07 - 249fca3 / 074a00d - lavr 1.1.0 Add avresample_set_channel_mapping() for input channel reordering, duplication, and silencing. -------------------------------8<------------------------------------- - 9 branch was cut here ------------------------------>8-------------------------------------- - -2012-12-29 - lavu 52.3.0 - d8fd06c - Add av_basename() and av_dirname(). - c1a02e8 - Add av_pix_fmt_get_chroma_sub_sample and deprecate - avcodec_get_chroma_sub_sample. +2012-12-29 - lavu 52.13.100 / 52.3.0 - avstring.h + 2ce43b3 / d8fd06c - Add av_basename() and av_dirname(). + e13d5e9 / c1a02e8 - Add av_pix_fmt_get_chroma_sub_sample and deprecate + avcodec_get_chroma_sub_sample. -2012-11-11 - 5980f5d - lavu 52.2.0 - audioconvert.h +2012-11-11 - 03b0787 / 5980f5d - lavu 52.6.100 / 52.2.0 - audioconvert.h Rename audioconvert.h to channel_layout.h. audioconvert.h is now deprecated. -2012-11-05 - dfde8a3 - lavu 52.1.0 - intmath.h - Add av_ctz() for trailing zero bit count - -2012-10-21 - a893655 - lavu 51.45.0 - error.h +2012-10-21 - e3a91c5 / a893655 - lavu 51.77.100 / 51.45.0 - error.h Add AVERROR_EXPERIMENTAL -2012-10-12 - d2fcb35 - lavu 51.44.0 - pixdesc.h +2012-10-12 - a33ed6b / d2fcb35 - lavu 51.76.100 / 51.44.0 - pixdesc.h Add functions for accessing pixel format descriptors. Accessing the av_pix_fmt_descriptors array directly is now deprecated. -2012-10-11 - 9a92aea - lavu 51.43.0 - aes.h, md5.h, sha.h, tree.h +2012-10-11 - f391e40 / 9a92aea - lavu 51.75.100 / 51.43.0 - aes.h, md5.h, sha.h, tree.h Add functions for allocating the opaque contexts for the algorithms, - deprecate the context size variables. -2012-10-10 - b522000 - lavf 54.18.0 - avio.h +2012-10-10 - de31814 / b522000 - lavf 54.32.100 / 54.18.0 - avio.h Add avio_closep to complement avio_close. -2012-10-08 - 78071a1 - lavu 51.42.0 - pixfmt.h +2012-10-08 - ae77266 / 78071a1 - lavu 51.74.100 / 51.42.0 - pixfmt.h Rename PixelFormat to AVPixelFormat and all PIX_FMT_* to AV_PIX_FMT_*. To provide backwards compatibility, PixelFormat is now #defined as AVPixelFormat. @@ -614,23 +1149,26 @@ lavd 54.0.0, lavfi 3.5.0 'PixelFormat' identifier. Such code should either #undef PixelFormat or stop using the PixelFormat name. -2012-10-05 - e7ba5b1 - lavr 1.0.0 - avresample.h +2012-10-05 - 55c49af / e7ba5b1 - lavr 1.0.0 - avresample.h Data planes parameters to avresample_convert() and avresample_read() are now uint8_t** instead of void**. Libavresample is now stable. -2012-09-24 - a42aada - lavc 54.28.0 - avcodec.h +2012-09-26 - 3ba0dab7 / 1384df64 - lavf 54.29.101 / 56.06.3 - avformat.h + Add AVFormatContext.avoid_negative_ts. + +2012-09-24 - 46a3595 / a42aada - lavc 54.59.100 / 54.28.0 - avcodec.h Add avcodec_free_frame(). This function must now be used for freeing an AVFrame. -2012-09-12 - 8919fee - lavu 51.41.0 - audioconvert.h +2012-09-12 - e3e09f2 / 8919fee - lavu 51.73.100 / 51.41.0 - audioconvert.h Added AV_CH_LOW_FREQUENCY_2 channel mask value. -2012-09-04 - 686a329 - lavu 51.40.0 - opt.h +2012-09-04 - b21b5b0 / 686a329 - lavu 51.71.100 / 51.40.0 - opt.h Reordered the fields in default_val in AVOption, changed which default_val field is used for which AVOptionType. -2012-08-30 - a231832 - lavc 54.26.1 - avcodec.h +2012-08-30 - 98298eb / a231832 - lavc 54.54.101 / 54.26.1 - avcodec.h Add codec descriptor properties AV_CODEC_PROP_LOSSY and AV_CODEC_PROP_LOSSLESS. @@ -638,101 +1176,90 @@ lavd 54.0.0, lavfi 3.5.0 Add codec descriptors for accessing codec properties without having to refer to a specific decoder or encoder. - c223d79 - Add an AVCodecDescriptor struct and functions + f5f3684 / c223d79 - Add an AVCodecDescriptor struct and functions avcodec_descriptor_get() and avcodec_descriptor_next(). - 51efed1 - Add AVCodecDescriptor.props and AV_CODEC_PROP_INTRA_ONLY. - 91e59fe - Add avcodec_descriptor_get_by_name(). - + f5f3684 / 51efed1 - Add AVCodecDescriptor.props and AV_CODEC_PROP_INTRA_ONLY. + 6c180b3 / 91e59fe - Add avcodec_descriptor_get_by_name(). -2012-08-08 - 1d9c2dc - lavu 51.39 - avutil.h - Don't implicitly include libavutil/common.h in avutil.h - -2012-08-08 - 987170c - lavu 51.38 - dict.h +2012-08-08 - f5f3684 / 987170c - lavu 51.68.100 / 51.38.0 - dict.h Add av_dict_count(). -2012-08-07 - 104e10f - lavc 54.25 - avcodec.h +2012-08-07 - 7a72695 / 104e10f - lavc 54.51.100 / 54.25.0 - avcodec.h Rename CodecID to AVCodecID and all CODEC_ID_* to AV_CODEC_ID_*. To provide backwards compatibility, CodecID is now #defined as AVCodecID. Note that this can break user code that includes avcodec.h and uses the 'CodecID' identifier. Such code should either #undef CodecID or stop using the CodecID name. -2012-08-03 - 239fdf1 - lavu 51.37.1 - cpu.h +2012-08-03 - e776ee8 / 239fdf1 - lavu 51.66.101 / 51.37.1 - cpu.h lsws 2.1.1 - swscale.h Rename AV_CPU_FLAG_MMX2 ---> AV_CPU_FLAG_MMXEXT. Rename SWS_CPU_CAPS_MMX2 ---> SWS_CPU_CAPS_MMXEXT. -2012-07-29 - 681ed00 - lavf 54.13.0 - avformat.h +2012-07-29 - 7c26761 / 681ed00 - lavf 54.22.100 / 54.13.0 - avformat.h Add AVFMT_FLAG_NOBUFFER for low latency use cases. -2012-07-20 - b70d89a - lavfi 3.0.0 - avfilter.h - Add avfilter_unref_bufferp(). - -2012-07-10 - 5fade8a - lavu 51.37.0 +2012-07-10 - fbe0245 / f3e5e6f - lavu 51.65.100 / 51.37.0 Add av_malloc_array() and av_mallocz_array() -2012-06-22 - d3d3a32 - lavu 51.34.0 +2012-06-22 - e847f41 / d3d3a32 - lavu 51.61.100 / 51.34.0 Add av_usleep() -2012-06-20 - ae0a301 - lavu 51.33.0 +2012-06-20 - 4da42eb / ae0a301 - lavu 51.60.100 / 51.33.0 Move av_gettime() to libavutil, add libavutil/time.h -2012-06-09 - 3971be0 - lavr 0.0.3 +2012-06-09 - 82edf67 / 3971be0 - lavr 0.0.3 Add a parameter to avresample_build_matrix() for Dolby/DPLII downmixing. -2012-06-12 - 9baeff9 - lavfi 2.23.0 - avfilter.h +2012-06-12 - c7b9eab / 9baeff9 - lavfi 2.79.100 / 2.23.0 - avfilter.h Add AVFilterContext.nb_inputs/outputs. Deprecate AVFilterContext.input/output_count. -2012-06-12 - 84b9fbe - lavfi 2.22.0 - avfilter.h +2012-06-12 - c7b9eab / 84b9fbe - lavfi 2.79.100 / 2.22.0 - avfilter.h Add avfilter_pad_get_type() and avfilter_pad_get_name(). Those should now be used instead of accessing AVFilterPad members directly. -2012-06-12 - b0f0dfc - lavu 51.32.0 - audioconvert.h +2012-06-12 - 3630a07 / b0f0dfc - lavu 51.57.100 / 51.32.0 - audioconvert.h Add av_get_channel_layout_channel_index(), av_get_channel_name() and av_channel_layout_extract_channel(). -2012-05-25 - 154486f - lavu 51.31.0 - opt.h +2012-05-25 - 53ce990 / 154486f - lavu 51.55.100 / 51.31.0 - opt.h Add av_opt_set_bin() -2012-05-26 - e9cef89 - lavf 54.3.0 - Add AVFMT_TS_NONSTRICT format flag to indicate that a muxer supports - non-increasing monotone timestamps. - -2012-05-15 - lavfi 2.17.0 +2012-05-15 - lavfi 2.74.100 / 2.17.0 Add support for audio filters - ac71230/a2cd9be - add video/audio buffer sink in a new installed + 61930bd / ac71230, 1cbf7fb / a2cd9be - add video/audio buffer sink in a new installed header buffersink.h - 720c6b7 - add av_buffersrc_write_frame(), deprecate + 1cbf7fb / 720c6b7 - add av_buffersrc_write_frame(), deprecate av_vsrc_buffer_add_frame() - ab16504 - add avfilter_copy_buf_props() - 9453c9e - add extended_data to AVFilterBuffer - 1b8c927 - add avfilter_get_audio_buffer_ref_from_arrays() + 61930bd / ab16504 - add avfilter_copy_buf_props() + 61930bd / 9453c9e - add extended_data to AVFilterBuffer + 61930bd / 1b8c927 - add avfilter_get_audio_buffer_ref_from_arrays() -2012-05-09 - lavu 51.30.0 - samplefmt.h - 142e740 - add av_samples_copy() - 6d7f617 - add av_samples_set_silence() +2012-05-09 - lavu 51.53.100 / 51.30.0 - samplefmt.h + 61930bd / 142e740 - add av_samples_copy() + 61930bd / 6d7f617 - add av_samples_set_silence() -2012-05-09 - a5117a2 - lavc 54.13.1 +2012-05-09 - 61930bd / a5117a2 - lavc 54.21.101 / 54.13.1 For audio formats with fixed frame size, the last frame no longer needs to be padded with silence, libavcodec will handle this internally (effectively all encoders behave as if they had CODEC_CAP_SMALL_LAST_FRAME set). -2012-05-07 - 828bd08 - lavc 54.13.0 - avcodec.h +2012-05-07 - 653d117 / 828bd08 - lavc 54.20.100 / 54.13.0 - avcodec.h Add sample_rate and channel_layout fields to AVFrame. -2012-05-01 - 4010d72 - lavr 0.0.1 +2012-05-01 - 2330eb1 / 4010d72 - lavr 0.0.1 Change AV_MIX_COEFF_TYPE_Q6 to AV_MIX_COEFF_TYPE_Q8. -2012-04-25 - 3527a73 - lavu 51.29.0 - cpu.h +2012-04-25 - e890b68 / 3527a73 - lavu 51.48.100 / 51.29.0 - cpu.h Add av_parse_cpu_flags() -2012-04-24 - c8af852 - lavr 0.0.0 +2012-04-24 - 3ead79e / c8af852 - lavr 0.0.0 Add libavresample audio conversion library -2012-04-20 - 0c0d1bc - lavu 51.28.0 - audio_fifo.h +2012-04-20 - 3194ab7 / 0c0d1bc - lavu 51.47.100 / 51.28.0 - audio_fifo.h Add audio FIFO functions: av_audio_fifo_free() av_audio_fifo_alloc() @@ -744,144 +1271,165 @@ lavd 54.0.0, lavfi 3.5.0 av_audio_fifo_size() av_audio_fifo_space() -2012-04-14 - lavfi 2.16.0 - avfiltergraph.h - d7bcc71 Add avfilter_graph_parse2(). - 91d3cbe Add avfilter_inout_alloc() and avfilter_inout_free(). +2012-04-14 - lavfi 2.70.100 / 2.16.0 - avfiltergraph.h + 7432bcf / d7bcc71 Add avfilter_graph_parse2(). -2012-04-08 - 4d693b0 - lavu 51.27.0 - samplefmt.h +2012-04-08 - 6bfb304 / 4d693b0 - lavu 51.46.100 / 51.27.0 - samplefmt.h Add av_get_packed_sample_fmt() and av_get_planar_sample_fmt() -2012-04-05 - 5cc51a5 - lavu 51.26.0 - audioconvert.h - Add av_get_default_channel_layout() +2012-03-21 - b75c67d - lavu 51.43.100 + Add bprint.h for bprint API. + +2012-02-21 - 9cbf17e - lavc 54.4.100 + Add av_get_pcm_codec() function. + +2012-02-16 - 560b224 - libswr 0.7.100 + Add swr_set_matrix() function. + +2012-02-09 - c28e7af - lavu 51.39.100 + Add a new installed header libavutil/timestamp.h with timestamp + utilities. + +2012-02-06 - 70ffda3 - lavu 51.38.100 + Add av_parse_ratio() function to parseutils.h. + +2012-02-06 - 70ffda3 - lavu 51.38.100 + Add AV_LOG_MAX_OFFSET macro to log.h. -2012-03-20 - 3c90cc2 - lavfo 54.2.0 +2012-02-02 - 0eaa123 - lavu 51.37.100 + Add public timecode helpers. + +2012-01-24 - 0c3577b - lavfi 2.60.100 + Add avfilter_graph_dump. + +2012-03-20 - 0ebd836 / 3c90cc2 - lavfo 54.2.0 Deprecate av_read_packet(), use av_read_frame() with AVFMT_FLAG_NOPARSE | AVFMT_FLAG_NOFILLIN in AVFormatContext.flags -2012-03-06 - 4d851f8 - lavu 51.25.0 - cpu.h - Add av_set_cpu_flags_mask(). +2012-03-05 - lavc 54.10.100 / 54.8.0 + f095391 / 6699d07 Add av_get_exact_bits_per_sample() + f095391 / 9524cf7 Add av_get_audio_frame_duration() -2012-03-05 - lavc 54.8.0 - 6699d07 Add av_get_exact_bits_per_sample() - 9524cf7 Add av_get_audio_frame_duration() - -2012-03-04 - 44fe77b - lavc 54.7.0 - avcodec.h +2012-03-04 - 2af8f2c / 44fe77b - lavc 54.8.100 / 54.7.0 - avcodec.h Add av_codec_is_encoder/decoder(). -2012-03-01 - 442c132 - lavc 54.3.0 - avcodec.h +2012-03-01 - 1eb7f39 / 442c132 - lavc 54.5.100 / 54.3.0 - avcodec.h Add av_packet_shrink_side_data. -2012-02-29 - dd2a4bc - lavf 54.2.0 - avformat.h +2012-02-29 - 79ae084 / dd2a4bc - lavf 54.2.100 / 54.2.0 - avformat.h Add AVStream.attached_pic and AV_DISPOSITION_ATTACHED_PIC, used for dealing with attached pictures/cover art. -2012-02-25 - c9bca80 - lavu 51.24.0 - error.h +2012-02-25 - 305e4b3 / c9bca80 - lavu 51.41.100 / 51.24.0 - error.h Add AVERROR_UNKNOWN NOTE: this was backported to 0.8 -2012-02-20 - e9cda85 - lavc 54.2.0 +2012-02-20 - eadd426 / e9cda85 - lavc 54.2.100 / 54.2.0 Add duration field to AVCodecParserContext -2012-02-20 - 0b42a93 - lavu 51.23.1 - mathematics.h +2012-02-20 - eadd426 / 0b42a93 - lavu 51.40.100 / 51.23.1 - mathematics.h Add av_rescale_q_rnd() -2012-02-08 - 38d5533 - lavu 51.22.1 - pixdesc.h +2012-02-08 - f2b20b7 / 38d5533 - lavu 51.38.101 / 51.22.1 - pixdesc.h Add PIX_FMT_PSEUDOPAL flag. -2012-02-08 - 52f82a1 - lavc 54.01.0 +2012-02-08 - f2b20b7 / 52f82a1 - lavc 54.2.100 / 54.1.0 Add avcodec_encode_video2() and deprecate avcodec_encode_video(). -2012-02-01 - 316fc74 - lavc 54.01.0 +2012-02-01 - 4c677df / 316fc74 - lavc 54.1.0 Add av_fast_padded_malloc() as alternative for av_realloc() when aligned memory is required. The buffer will always have FF_INPUT_BUFFER_PADDING_SIZE zero-padded bytes at the end. -2012-01-31 - dd6d3b0 - lavf 54.01.0 +2012-01-31 - a369a6b / dd6d3b0 - lavf 54.1.0 Add avformat_get_riff_video_tags() and avformat_get_riff_audio_tags(). NOTE: this was backported to 0.8 -2012-01-31 - af08d9a - lavc 54.01.0 +2012-01-31 - a369a6b / af08d9a - lavc 54.1.0 Add avcodec_is_open() function. NOTE: this was backported to 0.8 -2012-01-30 - 8b93312 - lavu 51.22.0 - intfloat.h +2012-01-30 - 151ecc2 / 8b93312 - lavu 51.36.100 / 51.22.0 - intfloat.h Add a new installed header libavutil/intfloat.h with int/float punning functions. NOTE: this was backported to 0.8 -2012-01-25 - lavf 53.22.0 - f1caf01 Allow doing av_write_frame(ctx, NULL) for flushing possible +2012-01-25 - lavf 53.31.100 / 53.22.0 + 3c5fe5b / f1caf01 Allow doing av_write_frame(ctx, NULL) for flushing possible buffered data within a muxer. Added AVFMT_ALLOW_FLUSH for muxers supporting it (av_write_frame makes sure it is called only for muxers with this flag). -------------------------------8<------------------------------------- - 0.8 branch was cut here ------------------------------>8-------------------------------------- - -2012-01-15 - lavc 53.34.0 +2012-01-15 - lavc 53.56.105 / 53.34.0 New audio encoding API: - b2c75b6 Add CODEC_CAP_VARIABLE_FRAME_SIZE capability for use by audio + 67f5650 / b2c75b6 Add CODEC_CAP_VARIABLE_FRAME_SIZE capability for use by audio encoders. - 5ee5fa0 Add avcodec_fill_audio_frame() as a convenience function. - b2c75b6 Add avcodec_encode_audio2() and deprecate avcodec_encode_audio(). + 67f5650 / 5ee5fa0 Add avcodec_fill_audio_frame() as a convenience function. + 67f5650 / b2c75b6 Add avcodec_encode_audio2() and deprecate avcodec_encode_audio(). Add AVCodec.encode2(). -2012-01-12 - 3167dc9 - lavfi 2.15.0 +2012-01-12 - b18e17e / 3167dc9 - lavfi 2.59.100 / 2.15.0 Add a new installed header -- libavfilter/version.h -- with version macros. -2011-01-03 - b73ec05 - lavu 51.21.0 + +-------- 8< --------- FFmpeg 0.9 was cut here -------- 8< --------- + +2011-12-08 - a502939 - lavfi 2.52.0 + Add av_buffersink_poll_frame() to buffersink.h. + +2011-12-08 - 26c6fec - lavu 51.31.0 + Add av_log_format_line. + +2011-12-03 - 976b095 - lavu 51.30.0 + Add AVERROR_BUG. + +2011-11-24 - 573ffbb - lavu 51.28.1 + Add av_get_alt_sample_fmt() to samplefmt.h. + +2011-11-03 - 96949da - lavu 51.23.0 + Add av_strcasecmp() and av_strncasecmp() to avstring.h. + +2011-10-20 - b35e9e1 - lavu 51.22.0 + Add av_strtok() to avstring.h. + +2012-01-03 - ad1c8dd / b73ec05 - lavu 51.34.100 / 51.21.0 Add av_popcount64 -2011-12-25 - lavfi 2.14.0 - e1d9dbf Add a new installed header - buffersrc.h - It contains a new function av_buffersrc_buffer() that allows passing - frames to the 'buffer' filter, but unlike av_vsrc_buffer_add_frame() - it allows for direct rendering. - 1c9e340 Add avfilter_copy_frame_props() for copying properties from - AVFrame to AVFilterBufferRef. - -2011-12-25 - lavc 53.31.0 - Add the following new fields to AVFrame: - b58dbb5 sample_aspect_ratio - 3a2ddf7 width, height - 8a4a5f6 format - -2011-12-18 - 8400b12 - lavc 53.28.1 +2011-12-18 - 7c29313 / 8400b12 - lavc 53.46.1 / 53.28.1 Deprecate AVFrame.age. The field is unused. -2011-12-12 - 5266045 - lavf 53.17.0 +2011-12-12 - 8bc7fe4 / 5266045 - lavf 53.25.0 / 53.17.0 Add avformat_close_input(). Deprecate av_close_input_file() and av_close_input_stream(). -2011-12-09 - b2890f5 - lavu 51.20.0 - audioconvert.h +2011-12-09 - c59b80c / b2890f5 - lavu 51.32.0 / 51.20.0 - audioconvert.h Expand the channel layout list. -2011-12-02 - 0eea212 - lavc 53.25.0 +2011-12-02 - e4de716 / 0eea212 - lavc 53.40.0 / 53.25.0 Add nb_samples and extended_data fields to AVFrame. Deprecate AVCODEC_MAX_AUDIO_FRAME_SIZE. Deprecate avcodec_decode_audio3() in favor of avcodec_decode_audio4(). avcodec_decode_audio4() writes output samples to an AVFrame, which allows audio decoders to use get_buffer(). -2011-12-04 - 560f773 - lavc 53.24.0 +2011-12-04 - e4de716 / 560f773 - lavc 53.40.0 / 53.24.0 Change AVFrame.data[4]/base[4]/linesize[4]/error[4] to [8] at next major bump. Change AVPicture.data[4]/linesize[4] to [8] at next major bump. Change AVCodecContext.error[4] to [8] at next major bump. Add AV_NUM_DATA_POINTERS to simplify the bump transition. -2011-11-24 - lavu 51.19.0 - bd97b2e - add planar RGB pixel formats - 6b0768e - add PIX_FMT_PLANAR and PIX_FMT_RGB pixel descriptions +2011-11-24 - lavu 51.29.0 / 51.19.0 + 92afb43 / bd97b2e - add planar RGB pixel formats + 92afb43 / 6b0768e - add PIX_FMT_PLANAR and PIX_FMT_RGB pixel descriptions -2011-11-23 - bbb46f3 - lavu 51.18.0 +2011-11-23 - 8e576d5 / bbb46f3 - lavu 51.27.0 / 51.18.0 Add av_samples_get_buffer_size(), av_samples_fill_arrays(), and av_samples_alloc(), to samplefmt.h. -2011-11-23 - 8889cc4 - lavu 51.17.0 +2011-11-23 - 8e576d5 / 8889cc4 - lavu 51.27.0 / 51.17.0 Add planar sample formats and av_sample_fmt_is_planar() to samplefmt.h. -2011-11-19 - f3a29b7 - lavc 53.21.0 +2011-11-19 - dbb38bc / f3a29b7 - lavc 53.36.0 / 53.21.0 Move some AVCodecContext fields to a new private struct, AVCodecInternal, which is accessed from a new field, AVCodecContext.internal. - fields moved: @@ -889,220 +1437,371 @@ lavd 54.0.0, lavfi 3.5.0 AVCodecContext.internal_buffer_count --> AVCodecInternal.buffer_count AVCodecContext.is_copy --> AVCodecInternal.is_copy -2011-11-16 - 6270671 - lavu 51.16.0 +2011-11-16 - 8709ba9 / 6270671 - lavu 51.26.0 / 51.16.0 Add av_timegm() -2011-11-13 - lavf 53.15.0 +2011-11-13 - lavf 53.21.0 / 53.15.0 New interrupt callback API, allowing per-AVFormatContext/AVIOContext interrupt callbacks. - 6aa0b98 Add AVIOInterruptCB struct and the interrupt_callback field to + 5f268ca / 6aa0b98 Add AVIOInterruptCB struct and the interrupt_callback field to AVFormatContext. - 1dee0ac Add avio_open2() with additional parameters. Those are + 5f268ca / 1dee0ac Add avio_open2() with additional parameters. Those are an interrupt callback and an options AVDictionary. This will allow passing AVOptions to protocols after lavf 54.0. -2011-11-06 - ba04ecf - lavu 51.14.0 +2011-11-06 - 13b7781 / ba04ecf - lavu 51.24.0 / 51.14.0 Add av_strcasecmp() and av_strncasecmp() to avstring.h. -2011-11-06 - 07b172f - lavu 51.13.0 +2011-11-06 - 13b7781 / 07b172f - lavu 51.24.0 / 51.13.0 Add av_toupper()/av_tolower() -2011-11-05 - b6d08f4 - lavf 53.13.0 +2011-11-05 - d8cab5c / b6d08f4 - lavf 53.19.0 / 53.13.0 Add avformat_network_init()/avformat_network_deinit() -2011-10-27 - 512557b - lavc 53.15.0 +2011-10-27 - 6faf0a2 / 512557b - lavc 53.24.0 / 53.15.0 Remove avcodec_parse_frame. Deprecate AVCodecContext.parse_only and CODEC_CAP_PARSE_ONLY. -2011-10-19 - 569129a - lavf 53.10.0 +2011-10-19 - d049257 / 569129a - lavf 53.17.0 / 53.10.0 Add avformat_new_stream(). Deprecate av_new_stream(). -2011-10-13 - b631fba - lavf 53.9.0 +2011-10-13 - 91eb1b1 / b631fba - lavf 53.16.0 / 53.9.0 Add AVFMT_NO_BYTE_SEEK AVInputFormat flag. -2011-10-12 - lavu 51.12.0 +2011-10-12 - lavu 51.21.0 / 51.12.0 AVOptions API rewrite. - - 145f741 FF_OPT_TYPE* renamed to AV_OPT_TYPE_* + - f884ef0 / 145f741 FF_OPT_TYPE* renamed to AV_OPT_TYPE_* - new setting/getting functions with slightly different semantics: - dac66da av_set_string3 -> av_opt_set + f884ef0 / dac66da av_set_string3 -> av_opt_set av_set_double -> av_opt_set_double av_set_q -> av_opt_set_q av_set_int -> av_opt_set_int - 41d9d51 av_get_string -> av_opt_get + f884ef0 / 41d9d51 av_get_string -> av_opt_get av_get_double -> av_opt_get_double av_get_q -> av_opt_get_q av_get_int -> av_opt_get_int - - 8c5dcaa trivial rename av_next_option -> av_opt_next - - 641c7af new functions - av_opt_child_next, av_opt_child_class_next + - f884ef0 / 8c5dcaa trivial rename av_next_option -> av_opt_next + - f884ef0 / 641c7af new functions - av_opt_child_next, av_opt_child_class_next and av_opt_find2() -2011-09-03 - fb4ca26 - lavc 53.10.0 - lavf 53.6.0 +2011-09-22 - a70e787 - lavu 51.17.0 + Add av_x_if_null(). + +2011-09-18 - 645cebb - lavc 53.16.0 + Add showall flag2 + +2011-09-16 - ea8de10 - lavfi 2.42.0 + Add avfilter_all_channel_layouts. + +2011-09-16 - 9899037 - lavfi 2.41.0 + Rename avfilter_all_* function names to avfilter_make_all_*. + + In particular, apply the renames: + avfilter_all_formats -> avfilter_make_all_formats + avfilter_all_channel_layouts -> avfilter_make_all_channel_layouts + avfilter_all_packing_formats -> avfilter_make_all_packing_formats + +2011-09-12 - 4381bdd - lavfi 2.40.0 + Change AVFilterBufferRefAudioProps.sample_rate type from uint32_t to int. + +2011-09-12 - 2c03174 - lavfi 2.40.0 + Simplify signature for avfilter_get_audio_buffer(), make it + consistent with avfilter_get_video_buffer(). + +2011-09-06 - 4f7dfe1 - lavfi 2.39.0 + Rename libavfilter/vsink_buffer.h to libavfilter/buffersink.h. + +2011-09-06 - c4415f6 - lavfi 2.38.0 + Unify video and audio sink API. + + In particular, add av_buffersink_get_buffer_ref(), deprecate + av_vsink_buffer_get_video_buffer_ref() and change the value for the + opaque field passed to the abuffersink init function. + +2011-09-04 - 61e2e29 - lavu 51.16.0 + Add av_asprintf(). + +2011-08-22 - dacd827 - lavf 53.10.0 + Add av_find_program_from_stream(). + +2011-08-20 - 69e2c1a - lavu 51.13.0 + Add av_get_media_type_string(). + +2011-09-03 - 1889c67 / fb4ca26 - lavc 53.13.0 + lavf 53.11.0 lsws 2.1.0 Add {avcodec,avformat,sws}_get_class(). -2011-09-03 - c11fb82 - lavu 51.10.0 +2011-08-03 - 1889c67 / c11fb82 - lavu 51.15.0 Add AV_OPT_SEARCH_FAKE_OBJ flag for av_opt_find() function. -2011-08-26 - lavu 51.9.0 - - f2011ed Add av_fifo_peek2(), deprecate av_fifo_peek(). - - add41de..abc78a5 Do not include intfloat_readwrite.h, +2011-08-14 - 323b930 - lavu 51.12.0 + Add av_fifo_peek2(), deprecate av_fifo_peek(). + +2011-08-26 - lavu 51.14.0 / 51.9.0 + - 976a8b2 / add41de..976a8b2 / abc78a5 Do not include intfloat_readwrite.h, mathematics.h, rational.h, pixfmt.h, or log.h from avutil.h. -2011-08-16 - 48f9e45 - lavf 53.4.0 +2011-08-16 - 27fbe31 / 48f9e45 - lavf 53.11.0 / 53.8.0 Add avformat_query_codec(). -2011-08-16 - bca06e7 - lavc 53.8.0 +2011-08-16 - 27fbe31 / bca06e7 - lavc 53.11.0 Add avcodec_get_type(). -2011-08-06 - 2f63440 - lavf 53.4.0 +2011-08-06 - 0cb233c / 2f63440 - lavf 53.7.0 Add error_recognition to AVFormatContext. -2011-08-02 - 9d39cbf - lavc 53.7.1 +2011-08-02 - 1d186e9 / 9d39cbf - lavc 53.9.1 Add AV_PKT_FLAG_CORRUPT AVPacket flag. -2011-07-10 - a67c061 - lavf 53.3.0 +2011-07-16 - b57df29 - lavfi 2.27.0 + Add audio packing negotiation fields and helper functions. + + In particular, add AVFilterPacking enum, planar, in_packings and + out_packings fields to AVFilterLink, and the functions: + avfilter_set_common_packing_formats() + avfilter_all_packing_formats() + +2011-07-10 - 3602ad7 / a67c061 - lavf 53.6.0 Add avformat_find_stream_info(), deprecate av_find_stream_info(). NOTE: this was backported to 0.7 -2011-07-10 - 0b950fe - lavc 53.6.0 +2011-07-10 - 3602ad7 / 0b950fe - lavc 53.8.0 Add avcodec_open2(), deprecate avcodec_open(). NOTE: this was backported to 0.7 Add avcodec_alloc_context3. Deprecate avcodec_alloc_context() and avcodec_alloc_context2(). -2011-06-23 - 67e9ae1 - lavu 51.8.0 - attributes.h - Add av_printf_format(). +2011-07-01 - b442ca6 - lavf 53.5.0 - avformat.h + Add function av_get_output_timestamp(). + +2011-06-28 - 5129336 - lavu 51.11.0 - avutil.h + Define the AV_PICTURE_TYPE_NONE value in AVPictureType enum. + + +-------- 8< --------- FFmpeg 0.7 was cut here -------- 8< --------- + + + +-------- 8< --------- FFmpeg 0.8 was cut here -------- 8< --------- + +2011-06-19 - fd2c0a5 - lavfi 2.23.0 - avfilter.h + Add layout negotiation fields and helper functions. + + In particular, add in_chlayouts and out_chlayouts to AVFilterLink, + and the functions: + avfilter_set_common_sample_formats() + avfilter_set_common_channel_layouts() + avfilter_all_channel_layouts() + +2011-06-19 - 527ca39 - lavfi 2.22.0 - AVFilterFormats + Change type of AVFilterFormats.formats from int * to int64_t *, + and update formats handling API accordingly. + + avfilter_make_format_list() still takes a int32_t array and converts + it to int64_t. A new function, avfilter_make_format64_list(), that + takes int64_t arrays has been added. + +2011-06-19 - 44f669e - lavfi 2.21.0 - vsink_buffer.h + Add video sink buffer and vsink_buffer.h public header. -------------------------------8<------------------------------------- - 0.7 branch was cut here ------------------------------>8-------------------------------------- +2011-06-12 - 9fdf772 - lavfi 2.18.0 - avcodec.h + Add avfilter_get_video_buffer_ref_from_frame() function in + libavfilter/avcodec.h. -2011-06-16 - 05e84c9, 25de595 - lavf 53.2.0 - avformat.h +2011-06-12 - c535494 - lavfi 2.17.0 - avfiltergraph.h + Add avfilter_inout_alloc() and avfilter_inout_free() functions. + +2011-06-12 - 6119b23 - lavfi 2.16.0 - avfilter_graph_parse() + Change avfilter_graph_parse() signature. + +2011-06-23 - 686959e / 67e9ae1 - lavu 51.10.0 / 51.8.0 - attributes.h + Add av_printf_format(). + +2011-06-16 - 2905e3f / 05e84c9, 2905e3f / 25de595 - lavf 53.4.0 / 53.2.0 - avformat.h Add avformat_open_input and avformat_write_header(). Deprecate av_open_input_stream, av_open_input_file, AVFormatParameters and av_write_header. -2011-06-16 - 7e83e1c, dc59ec5 - lavu 51.7.0 - opt.h +2011-06-16 - 2905e3f / 7e83e1c, 2905e3f / dc59ec5 - lavu 51.9.0 / 51.7.0 - opt.h Add av_opt_set_dict() and av_opt_find(). Deprecate av_find_opt(). Add AV_DICT_APPEND flag. -2011-06-10 - cb7c11c - lavu 51.6.0 - opt.h +2011-06-10 - 45fb647 / cb7c11c - lavu 51.6.0 - opt.h Add av_opt_flag_is_set(). -2011-06-08 - d9f80ea - lavu 51.5.0 - AVMetadata +2011-06-10 - c381960 - lavfi 2.15.0 - avfilter_get_audio_buffer_ref_from_arrays + Add avfilter_get_audio_buffer_ref_from_arrays() to avfilter.h. + +2011-06-09 - f9ecb84 / d9f80ea - lavu 51.8.0 - AVMetadata Move AVMetadata from lavf to lavu and rename it to AVDictionary -- new installed header dict.h. All av_metadata_* functions renamed to av_dict_*. -2011-06-07 - a6703fa - lavu 51.4.0 - av_get_bytes_per_sample() +2011-06-07 - d552f61 / a6703fa - lavu 51.8.0 - av_get_bytes_per_sample() Add av_get_bytes_per_sample() in libavutil/samplefmt.h. Deprecate av_get_bits_per_sample_fmt(). -2011-06-05 - b39b062 - lavu 51.3.0 - opt.h +2011-06-05 - f956924 / b39b062 - lavu 51.8.0 - opt.h Add av_opt_free convenience function. -2011-05-28 - 0420bd7 - lavu 51.2.0 - pixdesc.h +2011-06-06 - 95a0242 - lavfi 2.14.0 - AVFilterBufferRefAudioProps + Remove AVFilterBufferRefAudioProps.size, and use nb_samples in + avfilter_get_audio_buffer() and avfilter_default_get_audio_buffer() in + place of size. + +2011-06-06 - 0bc2cca - lavu 51.6.0 - av_samples_alloc() + Switch nb_channels and nb_samples parameters order in + av_samples_alloc(). + +2011-06-06 - e1c7414 - lavu 51.5.0 - av_samples_* + Change the data layout created by av_samples_fill_arrays() and + av_samples_alloc(). + +2011-06-06 - 27bcf55 - lavfi 2.13.0 - vsrc_buffer.h + Make av_vsrc_buffer_add_video_buffer_ref() accepts an additional + flags parameter in input. + +2011-06-03 - e977ca2 - lavfi 2.12.0 - avfilter_link_free() + Add avfilter_link_free() function. + +2011-06-02 - 5ad38d9 - lavu 51.4.0 - av_force_cpu_flags() + Add av_cpu_flags() in libavutil/cpu.h. + +2011-05-28 - e71f260 - lavu 51.3.0 - pixdesc.h Add av_get_pix_fmt_name() in libavutil/pixdesc.h, and deprecate avcodec_get_pix_fmt_name() in libavcodec/avcodec.h in its favor. -2011-05-25 - 30315a8 - lavf 53.1.0 - avformat.h +2011-05-25 - 39e4206 / 30315a8 - lavf 53.3.0 - avformat.h Add fps_probe_size to AVFormatContext. -2011-05-18 - 64150ff - lavc 53.4.0 - AVCodecContext.request_sample_fmt +2011-05-22 - 5ecdfd0 - lavf 53.2.0 - avformat.h + Introduce avformat_alloc_output_context2() and deprecate + avformat_alloc_output_context(). + +2011-05-22 - 83db719 - lavfi 2.10.0 - vsrc_buffer.h + Make libavfilter/vsrc_buffer.h public. + +2011-05-19 - c000a9f - lavfi 2.8.0 - avcodec.h + Add av_vsrc_buffer_add_frame() to libavfilter/avcodec.h. + +2011-05-14 - 9fdf772 - lavfi 2.6.0 - avcodec.h + Add avfilter_get_video_buffer_ref_from_frame() to libavfilter/avcodec.h. + +2011-05-18 - 75a37b5 / 64150ff - lavc 53.7.0 - AVCodecContext.request_sample_fmt Add request_sample_fmt field to AVCodecContext. -2011-05-10 - 188dea1 - lavc 53.3.0 - avcodec.h +2011-05-10 - 59eb12f / 188dea1 - lavc 53.6.0 - avcodec.h Deprecate AVLPCType and the following fields in AVCodecContext: lpc_coeff_precision, prediction_order_method, min_partition_order, max_partition_order, lpc_type, lpc_passes. Corresponding FLAC encoder options should be used instead. -2011-04-26 - bebe72f - lavu 51.1.0 - avutil.h +2011-05-07 - 9fdf772 - lavfi 2.5.0 - avcodec.h + Add libavfilter/avcodec.h header and avfilter_copy_frame_props() + function. + +2011-05-07 - 18ded93 - lavc 53.5.0 - AVFrame + Add format field to AVFrame. + +2011-05-07 - 22333a6 - lavc 53.4.0 - AVFrame + Add width and height fields to AVFrame. + +2011-05-01 - 35fe66a - lavfi 2.4.0 - avfilter.h + Rename AVFilterBufferRefVideoProps.pixel_aspect to + sample_aspect_ratio. + +2011-05-01 - 77e9dee - lavc 53.3.0 - AVFrame + Add a sample_aspect_ratio field to AVFrame. + +2011-05-01 - 1ba5727 - lavc 53.2.0 - AVFrame + Add a pkt_pos field to AVFrame. + +2011-04-29 - 35ceaa7 - lavu 51.2.0 - mem.h + Add av_dynarray_add function for adding + an element to a dynamic array. + +2011-04-26 - d7e5aeb / bebe72f - lavu 51.1.0 - avutil.h Add AVPictureType enum and av_get_picture_type_char(), deprecate FF_*_TYPE defines and av_get_pict_type_char() defined in libavcodec/avcodec.h. -2011-04-26 - 10d3940 - lavfi 2.3.0 - avfilter.h +2011-04-26 - d7e5aeb / 10d3940 - lavfi 2.3.0 - avfilter.h Add pict_type and key_frame fields to AVFilterBufferRefVideo. -2011-04-26 - 7a11c82 - lavfi 2.2.0 - vsrc_buffer +2011-04-26 - d7e5aeb / 7a11c82 - lavfi 2.2.0 - vsrc_buffer Add sample_aspect_ratio fields to vsrc_buffer arguments -2011-04-21 - 94f7451 - lavc 53.1.0 - avcodec.h +2011-04-21 - 8772156 / 94f7451 - lavc 53.1.0 - avcodec.h Add CODEC_CAP_SLICE_THREADS for codecs supporting sliced threading. 2011-04-15 - lavc 52.120.0 - avcodec.h AVPacket structure got additional members for passing side information: - 4de339e introduce side information for AVPacket - 2d8591c make containers pass palette change in AVPacket + c407984 / 4de339e introduce side information for AVPacket + c407984 / 2d8591c make containers pass palette change in AVPacket 2011-04-12 - lavf 52.107.0 - avio.h Avio cleanup, part II - deprecate the entire URLContext API: - 175389c add avio_check as a replacement for url_exist - ff1ec0c add avio_pause and avio_seek_time as replacements + c55780d / 175389c add avio_check as a replacement for url_exist + 9891004 / ff1ec0c add avio_pause and avio_seek_time as replacements for _av_url_read_fseek/fpause - cdc6a87 deprecate av_protocol_next(), avio_enum_protocols + d4d0932 / cdc6a87 deprecate av_protocol_next(), avio_enum_protocols should be used instead. - 80c6e23 rename url_set_interrupt_cb->avio_set_interrupt_cb. - f87b1b3 rename open flags: URL_* -> AVIO_* - f8270bb add avio_enum_protocols. - 5593f03 deprecate URLProtocol. - c486dad deprecate URLContext. - 026e175 deprecate the typedef for URLInterruptCB - 8e76a19 deprecate av_register_protocol2. - b840484 deprecate URL_PROTOCOL_FLAG_NESTED_SCHEME - 1305d93 deprecate av_url_read_seek - fa104e1 deprecate av_url_read_pause - 727c7aa deprecate url_get_filename(). - 5958df3 deprecate url_max_packet_size(). - 1869ea0 deprecate url_get_file_handle(). - 32a97d4 deprecate url_filesize(). - e52a914 deprecate url_close(). - 58a48c6 deprecate url_seek(). - 925e908 deprecate url_write(). - dce3756 deprecate url_read_complete(). - bc371ac deprecate url_read(). - 0589da0 deprecate url_open(). - 62eaaea deprecate url_connect. - 5652bb9 deprecate url_alloc. - 333e894 deprecate url_open_protocol - e230705 deprecate url_poll and URLPollEntry - -2011-04-10 - lavu 50.40.0 - pixfmt.h - Add PIX_FMT_BGR48LE and PIX_FMT_BGR48BE pixel formats + c88caa5 / 80c6e23 rename url_set_interrupt_cb->avio_set_interrupt_cb. + c88caa5 / f87b1b3 rename open flags: URL_* -> AVIO_* + d4d0932 / f8270bb add avio_enum_protocols. + d4d0932 / 5593f03 deprecate URLProtocol. + d4d0932 / c486dad deprecate URLContext. + d4d0932 / 026e175 deprecate the typedef for URLInterruptCB + c88caa5 / 8e76a19 deprecate av_register_protocol2. + 11d7841 / b840484 deprecate URL_PROTOCOL_FLAG_NESTED_SCHEME + 11d7841 / 1305d93 deprecate av_url_read_seek + 11d7841 / fa104e1 deprecate av_url_read_pause + 434f248 / 727c7aa deprecate url_get_filename(). + 434f248 / 5958df3 deprecate url_max_packet_size(). + 434f248 / 1869ea0 deprecate url_get_file_handle(). + 434f248 / 32a97d4 deprecate url_filesize(). + 434f248 / e52a914 deprecate url_close(). + 434f248 / 58a48c6 deprecate url_seek(). + 434f248 / 925e908 deprecate url_write(). + 434f248 / dce3756 deprecate url_read_complete(). + 434f248 / bc371ac deprecate url_read(). + 434f248 / 0589da0 deprecate url_open(). + 434f248 / 62eaaea deprecate url_connect. + 434f248 / 5652bb9 deprecate url_alloc. + 434f248 / 333e894 deprecate url_open_protocol + 434f248 / e230705 deprecate url_poll and URLPollEntry 2011-04-08 - lavf 52.106.0 - avformat.h Minor avformat.h cleanup: - a9bf9d8 deprecate av_guess_image2_codec - c3675df rename avf_sdp_create->av_sdp_create + d4d0932 / a9bf9d8 deprecate av_guess_image2_codec + d4d0932 / c3675df rename avf_sdp_create->av_sdp_create 2011-04-03 - lavf 52.105.0 - avio.h Large-scale renaming/deprecating of AVIOContext-related functions: - 724f6a0 deprecate url_fdopen - 403ee83 deprecate url_open_dyn_packet_buf - 6dc7d80 rename url_close_dyn_buf -> avio_close_dyn_buf - b92c545 rename url_open_dyn_buf -> avio_open_dyn_buf - 8978fed introduce an AVIOContext.seekable field as a replacement for + 2cae980 / 724f6a0 deprecate url_fdopen + 2cae980 / 403ee83 deprecate url_open_dyn_packet_buf + 2cae980 / 6dc7d80 rename url_close_dyn_buf -> avio_close_dyn_buf + 2cae980 / b92c545 rename url_open_dyn_buf -> avio_open_dyn_buf + 2cae980 / 8978fed introduce an AVIOContext.seekable field as a replacement for AVIOContext.is_streamed and url_is_streamed() - b64030f deprecate get_checksum() - 4c4427a deprecate init_checksum() - 4ec153b deprecate udp_set_remote_url/get_local_port - 933e90a deprecate av_url_read_fseek/fpause - 8d9769a deprecate url_fileno - b7f2fdd rename put_flush_packet -> avio_flush - 35f1023 deprecate url_close_buf - 83fddae deprecate url_open_buf - d9d86e0 rename url_fprintf -> avio_printf - 59f65d9 deprecate url_setbufsize - 3e68b3b deprecate url_ferror - 66e5b1d deprecate url_feof + 1caa412 / b64030f deprecate get_checksum() + 1caa412 / 4c4427a deprecate init_checksum() + 2fd41c9 / 4ec153b deprecate udp_set_remote_url/get_local_port + 4fa0e24 / 933e90a deprecate av_url_read_fseek/fpause + 4fa0e24 / 8d9769a deprecate url_fileno + 0fecf26 / b7f2fdd rename put_flush_packet -> avio_flush + 0fecf26 / 35f1023 deprecate url_close_buf + 0fecf26 / 83fddae deprecate url_open_buf + 0fecf26 / d9d86e0 rename url_fprintf -> avio_printf + 0fecf26 / 59f65d9 deprecate url_setbufsize + 6947b0c / 3e68b3b deprecate url_ferror e8bb2e2 deprecate url_fget_max_packet_size 76aa876 rename url_fsize -> avio_size e519753 deprecate url_fgetc @@ -1123,9 +1822,12 @@ lavd 54.0.0, lavfi 3.5.0 b3db9ce deprecate get_partial_buffer 8d9ac96 rename av_alloc_put_byte -> avio_alloc_context -2011-03-25 - 34b47d7 - lavc 52.115.0 - AVCodecContext.audio_service_type +2011-03-25 - 27ef7b1 / 34b47d7 - lavc 52.115.0 - AVCodecContext.audio_service_type Add audio_service_type field to AVCodecContext. +2011-03-17 - e309fdc - lavu 50.40.0 - pixfmt.h + Add PIX_FMT_BGR48LE and PIX_FMT_BGR48BE pixel formats + 2011-03-02 - 863c471 - lavf 52.103.0 - av_pkt_dump2, av_pkt_dump_log2 Add new functions av_pkt_dump2, av_pkt_dump_log2 that uses the source stream timebase for outputting timestamps. Deprecate @@ -1158,11 +1860,11 @@ lavd 54.0.0, lavfi 3.5.0 2011-02-10 - 12c14cd - lavf 52.99.0 - AVStream.disposition Add AV_DISPOSITION_HEARING_IMPAIRED and AV_DISPOSITION_VISUAL_IMPAIRED. -2011-02-09 - 5592734 - lavc 52.112.0 - avcodec_thread_init() +2011-02-09 - c0b102c - lavc 52.112.0 - avcodec_thread_init() Deprecate avcodec_thread_init()/avcodec_thread_free() use; instead set thread_count before calling avcodec_open. -2011-02-09 - 778b08a - lavc 52.111.0 - threading API +2011-02-09 - 37b00b4 - lavc 52.111.0 - threading API Add CODEC_CAP_FRAME_THREADS with new restrictions on get_buffer()/ release_buffer()/draw_horiz_band() callbacks for appropriate codecs. Add thread_type and active_thread_type fields to AVCodecContext. @@ -1192,6 +1894,12 @@ lavd 54.0.0, lavfi 3.5.0 2011-02-02 - dfd2a00 - lavu 50.37.0 - log.h Make av_dlog public. +2011-01-31 - 7b3ea55 - lavfi 1.76.0 - vsrc_buffer + Add sample_aspect_ratio fields to vsrc_buffer arguments + +2011-01-31 - 910b5b8 - lavfi 1.75.0 - AVFilterLink sample_aspect_ratio + Add sample_aspect_ratio field to AVFilterLink. + 2011-01-15 - a242ac3 - lavfi 1.74.0 - AVFilterBufferRefAudioProps Rename AVFilterBufferRefAudioProps.samples_nb to nb_samples. @@ -1559,6 +2267,9 @@ lavd 54.0.0, lavfi 3.5.0 2010-06-02 - 7e566bb - lavc 52.73.0 - av_get_codec_tag_string() Add av_get_codec_tag_string(). + +-------- 8< --------- FFmpeg 0.6 was cut here -------- 8< --------- + 2010-06-01 - 2b99142 - lsws 0.11.0 - convertPalette API Add sws_convertPalette8ToPacked32() and sws_convertPalette8ToPacked24(). @@ -1576,10 +2287,6 @@ lavd 54.0.0, lavfi 3.5.0 2010-05-09 - b6bc205 - lavfi 1.20.0 - AVFilterPicRef Add interlaced and top_field_first fields to AVFilterPicRef. -------------------------------8<------------------------------------- - 0.6 branch was cut here ------------------------------>8-------------------------------------- - 2010-05-01 - 8e2ee18 - lavf 52.62.0 - probe function Add av_probe_input_format2 to API, it allows ignoring probe results below given score and returns the actual probe score. diff --git a/doc/Doxyfile b/doc/Doxyfile index 58f7dfbece..1ad9f3041f 100644 --- a/doc/Doxyfile +++ b/doc/Doxyfile @@ -25,7 +25,7 @@ DOXYFILE_ENCODING = UTF-8 # The PROJECT_NAME tag is a single word (or a sequence of words surrounded # by quotes) that should identify the project. -PROJECT_NAME = Libav +PROJECT_NAME = FFmpeg # The PROJECT_NUMBER tag can be used to enter a project or revision number. # This could be handy for archiving the generated documentation or @@ -33,7 +33,7 @@ PROJECT_NAME = Libav PROJECT_NUMBER = -# With the PROJECT_LOGO tag one can specify an logo or icon that is included +# With the PROJECT_LOGO tag one can specify a logo or icon that is included # in the documentation. The maximum height of the logo should not exceed 55 # pixels and the maximum width should not exceed 200 pixels. Doxygen will # copy the logo to the output directory. @@ -639,14 +639,14 @@ EXCLUDE_SYMBOLS = # directories that contain example code fragments that are included (see # the \include command). -EXAMPLE_PATH = +EXAMPLE_PATH = doc/examples/ # If the value of the EXAMPLE_PATH tag contains directories, you can use the # EXAMPLE_PATTERNS tag to specify one or more wildcard pattern (like *.cpp # and *.h) to filter out the source-files in the directories. If left # blank all files are included. -EXAMPLE_PATTERNS = *-example.c +EXAMPLE_PATTERNS = *.c # If the EXAMPLE_RECURSIVE tag is set to YES then subdirectories will be # searched for input files to be used with the \include or \dontinclude @@ -709,7 +709,7 @@ INLINE_SOURCES = NO # doxygen to hide any special comment blocks from generated source code # fragments. Normal C and C++ comments will always remain visible. -STRIP_CODE_COMMENTS = YES +STRIP_CODE_COMMENTS = NO # If the REFERENCED_BY_RELATION tag is set to YES # then for each documented function all documented @@ -759,7 +759,7 @@ ALPHABETICAL_INDEX = YES # the COLS_IN_ALPHA_INDEX tag can be used to specify the number of columns # in which this list will be split (can be a number in the range [1..20]) -COLS_IN_ALPHA_INDEX = 2 +COLS_IN_ALPHA_INDEX = 5 # In case all classes in a project start with a common prefix, all # classes will be put under the same header in the alphabetical index. @@ -818,7 +818,7 @@ HTML_STYLESHEET = # 180 is cyan, 240 is blue, 300 purple, and 360 is red again. # The allowed range is 0 to 359. -HTML_COLORSTYLE_HUE = 120 +#HTML_COLORSTYLE_HUE = 120 # The HTML_COLORSTYLE_SAT tag controls the purity (or saturation) of # the colors in the HTML output. For a value of 0 the output will use @@ -1056,7 +1056,7 @@ FORMULA_TRANSPARENT = YES # typically be disabled. For large projects the javascript based search engine # can be slow, then enabling SERVER_BASED_SEARCH may provide a better solution. -SEARCHENGINE = NO +SEARCHENGINE = YES # When the SERVER_BASED_SEARCH tag is enabled the search engine will be # implemented using a PHP enabled web server instead of at the web client @@ -1359,6 +1359,9 @@ PREDEFINED = "__attribute__(x)=" \ "DECLARE_ALIGNED(a,t,n)=t n" \ "offsetof(x,y)=0x42" \ av_alloc_size \ + AV_GCC_VERSION_AT_LEAST(x,y)=1 \ + AV_GCC_VERSION_AT_MOST(x,y)=0 \ + __GNUC__=1 \ # If the MACRO_EXPANSION and EXPAND_ONLY_PREDEF tags are set to YES then # this tag can be used to specify a list of macro names that should be expanded. diff --git a/doc/Makefile b/doc/Makefile index 2f6a5fb0c1..4a77aac10d 100644 --- a/doc/Makefile +++ b/doc/Makefile @@ -1,40 +1,80 @@ -ALLMANPAGES = $(AVBASENAMES:%=%.1) -MANPAGES = $(AVPROGS-yes:%=doc/%.1) -PODPAGES = $(AVPROGS-yes:%=doc/%.pod) -HTMLPAGES = $(AVPROGS-yes:%=doc/%.html) \ +LIBRARIES-$(CONFIG_AVUTIL) += libavutil +LIBRARIES-$(CONFIG_SWSCALE) += libswscale +LIBRARIES-$(CONFIG_SWRESAMPLE) += libswresample +LIBRARIES-$(CONFIG_AVCODEC) += libavcodec +LIBRARIES-$(CONFIG_AVFORMAT) += libavformat +LIBRARIES-$(CONFIG_AVDEVICE) += libavdevice +LIBRARIES-$(CONFIG_AVFILTER) += libavfilter + +COMPONENTS-$(CONFIG_AVUTIL) += ffmpeg-utils +COMPONENTS-$(CONFIG_SWSCALE) += ffmpeg-scaler +COMPONENTS-$(CONFIG_SWRESAMPLE) += ffmpeg-resampler +COMPONENTS-$(CONFIG_AVCODEC) += ffmpeg-codecs ffmpeg-bitstream-filters +COMPONENTS-$(CONFIG_AVFORMAT) += ffmpeg-formats ffmpeg-protocols +COMPONENTS-$(CONFIG_AVDEVICE) += ffmpeg-devices +COMPONENTS-$(CONFIG_AVFILTER) += ffmpeg-filters + +MANPAGES1 = $(AVPROGS-yes:%=doc/%.1) $(AVPROGS-yes:%=doc/%-all.1) $(COMPONENTS-yes:%=doc/%.1) +MANPAGES3 = $(LIBRARIES-yes:%=doc/%.3) +MANPAGES = $(MANPAGES1) $(MANPAGES3) +PODPAGES = $(AVPROGS-yes:%=doc/%.pod) $(AVPROGS-yes:%=doc/%-all.pod) $(COMPONENTS-yes:%=doc/%.pod) $(LIBRARIES-yes:%=doc/%.pod) +HTMLPAGES = $(AVPROGS-yes:%=doc/%.html) $(AVPROGS-yes:%=doc/%-all.html) $(COMPONENTS-yes:%=doc/%.html) $(LIBRARIES-yes:%=doc/%.html) \ doc/developer.html \ doc/faq.html \ doc/fate.html \ doc/general.html \ doc/git-howto.html \ - doc/libavfilter.html \ doc/nut.html \ doc/platform.html \ -DOCS-$(CONFIG_POD2MAN) += $(MANPAGES) $(PODPAGES) -DOCS-$(CONFIG_TEXI2HTML) += $(HTMLPAGES) -DOCS = $(DOCS-yes) +TXTPAGES = doc/fate.txt \ -DOC_EXAMPLES-$(CONFIG_AVCODEC_EXAMPLE) += avcodec -DOC_EXAMPLES-$(CONFIG_FILTER_AUDIO_EXAMPLE) += filter_audio -DOC_EXAMPLES-$(CONFIG_METADATA_EXAMPLE) += metadata -DOC_EXAMPLES-$(CONFIG_OUTPUT_EXAMPLE) += output -DOC_EXAMPLES-$(CONFIG_QSVDEC_EXAMPLE) += qsvdec -DOC_EXAMPLES-$(CONFIG_TRANSCODE_AAC_EXAMPLE) += transcode_aac -ALL_DOC_EXAMPLES = avcodec filter_audio metadata output transcode_aac -DOC_EXAMPLES := $(DOC_EXAMPLES-yes:%=doc/examples/%$(EXESUF)) -ALL_DOC_EXAMPLES := $(ALL_DOC_EXAMPLES:%=doc/examples/%$(EXESUF)) -PROGS += $(DOC_EXAMPLES) +DOCS-$(CONFIG_HTMLPAGES) += $(HTMLPAGES) +DOCS-$(CONFIG_PODPAGES) += $(PODPAGES) +DOCS-$(CONFIG_MANPAGES) += $(MANPAGES) +DOCS-$(CONFIG_TXTPAGES) += $(TXTPAGES) +DOCS = $(DOCS-yes) -all: $(DOCS) +DOC_EXAMPLES-$(CONFIG_AVIO_DIR_CMD_EXAMPLE) += avio_dir_cmd +DOC_EXAMPLES-$(CONFIG_AVIO_READING_EXAMPLE) += avio_reading +DOC_EXAMPLES-$(CONFIG_AVCODEC_EXAMPLE) += avcodec +DOC_EXAMPLES-$(CONFIG_DECODING_ENCODING_EXAMPLE) += decoding_encoding +DOC_EXAMPLES-$(CONFIG_DEMUXING_DECODING_EXAMPLE) += demuxing_decoding +DOC_EXAMPLES-$(CONFIG_EXTRACT_MVS_EXAMPLE) += extract_mvs +DOC_EXAMPLES-$(CONFIG_FILTER_AUDIO_EXAMPLE) += filter_audio +DOC_EXAMPLES-$(CONFIG_FILTERING_AUDIO_EXAMPLE) += filtering_audio +DOC_EXAMPLES-$(CONFIG_FILTERING_VIDEO_EXAMPLE) += filtering_video +DOC_EXAMPLES-$(CONFIG_METADATA_EXAMPLE) += metadata +DOC_EXAMPLES-$(CONFIG_MUXING_EXAMPLE) += muxing +DOC_EXAMPLES-$(CONFIG_QSVDEC_EXAMPLE) += qsvdec +DOC_EXAMPLES-$(CONFIG_REMUXING_EXAMPLE) += remuxing +DOC_EXAMPLES-$(CONFIG_RESAMPLING_AUDIO_EXAMPLE) += resampling_audio +DOC_EXAMPLES-$(CONFIG_SCALING_VIDEO_EXAMPLE) += scaling_video +DOC_EXAMPLES-$(CONFIG_TRANSCODE_AAC_EXAMPLE) += transcode_aac +DOC_EXAMPLES-$(CONFIG_TRANSCODING_EXAMPLE) += transcoding +ALL_DOC_EXAMPLES_LIST = $(DOC_EXAMPLES-) $(DOC_EXAMPLES-yes) + +DOC_EXAMPLES := $(DOC_EXAMPLES-yes:%=doc/examples/%$(PROGSSUF)$(EXESUF)) +ALL_DOC_EXAMPLES := $(ALL_DOC_EXAMPLES_LIST:%=doc/examples/%$(PROGSSUF)$(EXESUF)) +ALL_DOC_EXAMPLES_G := $(ALL_DOC_EXAMPLES_LIST:%=doc/examples/%$(PROGSSUF)_g$(EXESUF)) +PROGS += $(DOC_EXAMPLES) + +all-$(CONFIG_DOC): doc + +doc: documentation apidoc: doc/doxy/html documentation: $(DOCS) examples: $(DOC_EXAMPLES) -TEXIDEP = awk '/^@include/ { printf "$@: $(@D)/%s\n", $$2 }' <$< >$(@:%=%.d) +TEXIDEP = perl $(SRC_PATH)/doc/texidep.pl $(SRC_PATH) $< $@ >$(@:%=%.d) + +doc/%.txt: TAG = TXT +doc/%.txt: doc/%.texi + $(Q)$(TEXIDEP) + $(M)makeinfo --force --no-headers -o $@ $< 2>/dev/null GENTEXI = format codec GENTEXI := $(GENTEXI:%=doc/avoptions_%.texi) @@ -44,55 +84,102 @@ $(GENTEXI): doc/avoptions_%.texi: doc/print_options$(HOSTEXESUF) $(M)doc/print_options $* > $@ doc/%.html: TAG = HTML +doc/%-all.html: TAG = HTML + +ifdef HAVE_MAKEINFO_HTML +doc/%.html: doc/%.texi $(SRC_PATH)/doc/t2h.pm $(GENTEXI) + $(Q)$(TEXIDEP) + $(M)makeinfo --html -I doc --no-split -D config-not-all --init-file=$(SRC_PATH)/doc/t2h.pm --output $@ $< + +doc/%-all.html: doc/%.texi $(SRC_PATH)/doc/t2h.pm $(GENTEXI) + $(Q)$(TEXIDEP) + $(M)makeinfo --html -I doc --no-split -D config-all --init-file=$(SRC_PATH)/doc/t2h.pm --output $@ $< +else doc/%.html: doc/%.texi $(SRC_PATH)/doc/t2h.init $(GENTEXI) $(Q)$(TEXIDEP) - $(M)texi2html -I doc -monolithic --init-file $(SRC_PATH)/doc/t2h.init --output $@ $< + $(M)texi2html -I doc -monolithic --D=config-not-all --init-file $(SRC_PATH)/doc/t2h.init --output $@ $< + +doc/%-all.html: doc/%.texi $(SRC_PATH)/doc/t2h.init $(GENTEXI) + $(Q)$(TEXIDEP) + $(M)texi2html -I doc -monolithic --D=config-all --init-file $(SRC_PATH)/doc/t2h.init --output $@ $< +endif doc/%.pod: TAG = POD doc/%.pod: doc/%.texi $(SRC_PATH)/doc/texi2pod.pl $(GENTEXI) $(Q)$(TEXIDEP) - $(M)$(SRC_PATH)/doc/texi2pod.pl -Idoc $< $@ + $(M)perl $(SRC_PATH)/doc/texi2pod.pl -Dconfig-not-all=yes -Idoc $< $@ -doc/%.1: TAG = MAN +doc/%-all.pod: TAG = POD +doc/%-all.pod: doc/%.texi $(SRC_PATH)/doc/texi2pod.pl $(GENTEXI) + $(Q)$(TEXIDEP) + $(M)perl $(SRC_PATH)/doc/texi2pod.pl -Dconfig-all=yes -Idoc $< $@ + +doc/%.1 doc/%.3: TAG = MAN doc/%.1: doc/%.pod $(GENTEXI) - $(M)pod2man --section=1 --center=" " --release=" " $< > $@ + $(M)pod2man --section=1 --center=" " --release=" " --date=" " $< > $@ +doc/%.3: doc/%.pod $(GENTEXI) + $(M)pod2man --section=3 --center=" " --release=" " --date=" " $< > $@ $(DOCS) doc/doxy/html: | doc/ $(DOC_EXAMPLES:%$(EXESUF)=%.o): | doc/examples OBJDIRS += doc/examples -DOXY_INPUT = $(addprefix $(SRC_PATH)/, $(INSTHEADERS) $(DOC_EXAMPLES:%$(EXESUF)=%.c) $(LIB_EXAMPLES:%$(EXESUF)=%.c)) -DOXY_TEMPLATES = doxy_stylesheet.css footer.html header.html -DOXY_TEMPLATES := $(addprefix $(SRC_PATH)/doc/doxy/, $(DOXY_TEMPLATES)) +DOXY_INPUT = $(INSTHEADERS) $(DOC_EXAMPLES:%$(EXESUF)=%.c) $(LIB_EXAMPLES:%$(EXESUF)=%.c) +DOXY_INPUT_DEPS = $(addprefix $(SRC_PATH)/, $(DOXY_INPUT)) + +doc/doxy/html: TAG = DOXY +doc/doxy/html: $(SRC_PATH)/doc/Doxyfile $(SRC_PATH)/doc/doxy-wrapper.sh $(DOXY_INPUT_DEPS) + $(M)OUT_DIR=$$PWD/doc/doxy; cd $(SRC_PATH); ./doc/doxy-wrapper.sh $$OUT_DIR $< $(DOXYGEN) $(DOXY_INPUT); + +install-doc: install-html install-man + +install-html: -doc/doxy/html: $(SRC_PATH)/doc/Doxyfile $(DOXY_INPUT) $(DOXY_TEMPLATES) - $(M)$(SRC_PATH)/doc/doxy-wrapper.sh $(SRC_PATH) $< $(DOXY_INPUT) +install-man: -install-progs-$(CONFIG_POD2MAN): install-man -install-progs-$(CONFIG_TEXI2HTML): install-doc +ifdef CONFIG_HTMLPAGES +install-progs-$(CONFIG_DOC): install-html -install-doc: $(HTMLPAGES) +install-html: $(HTMLPAGES) $(Q)mkdir -p "$(DOCDIR)" $(INSTALL) -m 644 $(HTMLPAGES) "$(DOCDIR)" +endif + +ifdef CONFIG_MANPAGES +install-progs-$(CONFIG_DOC): install-man install-man: $(MANPAGES) $(Q)mkdir -p "$(MANDIR)/man1" - $(INSTALL) -m 644 $(MANPAGES) "$(MANDIR)/man1" + $(INSTALL) -m 644 $(MANPAGES1) "$(MANDIR)/man1" + $(Q)mkdir -p "$(MANDIR)/man3" + $(INSTALL) -m 644 $(MANPAGES3) "$(MANDIR)/man3" +endif + +uninstall: uninstall-doc -uninstall: uninstall-doc uninstall-man +uninstall-doc: uninstall-html uninstall-man -uninstall-doc: +uninstall-html: $(RM) -r "$(DOCDIR)" uninstall-man: - $(RM) $(addprefix "$(MANDIR)/man1/",$(ALLMANPAGES)) + $(RM) $(addprefix "$(MANDIR)/man1/",$(AVPROGS-yes:%=%.1) $(AVPROGS-yes:%=%-all.1) $(COMPONENTS-yes:%=%.1)) + $(RM) $(addprefix "$(MANDIR)/man3/",$(LIBRARIES-yes:%=%.3)) + +clean:: docclean + +distclean:: docclean + $(RM) doc/config.texi + +examplesclean: + $(RM) $(ALL_DOC_EXAMPLES) $(ALL_DOC_EXAMPLES_G) + $(RM) $(CLEANSUFFIXES:%=doc/examples/%) -clean:: - $(RM) $(ALL_DOC_EXAMPLES) - $(RM) $(CLEANSUFFIXES:%=doc/%) $(CLEANSUFFIXES:%=doc/examples/%) - $(RM) doc/*.html doc/*.pod doc/*.1 doc/avoptions_*.texi +docclean: examplesclean + $(RM) $(CLEANSUFFIXES:%=doc/%) + $(RM) $(TXTPAGES) doc/*.html doc/*.pod doc/*.1 doc/*.3 doc/avoptions_*.texi $(RM) -r doc/doxy/html -include $(wildcard $(DOCS:%=%.d)) -.PHONY: apidoc documentation +.PHONY: apidoc doc documentation diff --git a/doc/RELEASE_NOTES b/doc/RELEASE_NOTES deleted file mode 100644 index bda4789aff..0000000000 --- a/doc/RELEASE_NOTES +++ /dev/null @@ -1,75 +0,0 @@ -Release Notes -============= - -* 11 "One Louder" - -General notes -------------- - -With this release we are trying to answer the numerous calls from our users for -shorter development cycles. From now on we will aim for approximately two major -releases per year. - -Libav 11 is API-, but not ABI-compatible with the previous major release. This -means that the code using our libraries needs to be rebuilt, but no source -changes should be required. Note however, that a number of old APIs remain -deprecated and will be dropped in the near future. All users are strongly -encouraged to update their code as soon as possible. The doc/APIchanges file in -the Libav source tree and the migration guide on the wiki should help with -migration to the new APIs. If those are not sufficient, do not hesitate to -contact us on IRC or through the user mailing list. - -One specific API issue in libavformat deserves mentioning here. When using -libavcodec for decoding or encoding and libavformat for demuxing or muxing, -the standard practice was to use the stream codec context (AVStream.codec) for -actual decoding or encoding. There are multiple problems with this pattern -(the main one is that the decoder/demuxer or encoder/muxer are not necessarily -synchronized and may overwrite each other's state), so it is now strongly -discouraged and will likely be deprecated in the future. Users should instead -allocate a separate decoding or encoding context and populate it from the -demuxing codec context (or the reverse for encoding) with the -avcodec_copy_context() function. - -The main highlights of this release include native Opus, VP7, OpenEXR, and On2 -AVC decoders, HEVC encoding through libx265, new APIs for exporting ReplayGain -and display transformation metadata and countless bug fixes. A large effort was -also expended on internal cleanups which are not very visible to our users, -but should make the codebase cleaner, safer and easier to maintain and extend. -One point worth mentioning is refactoring the large monolithic framework for -architecture-specific codec optimizations into small blocks, which reduces the -size of configurations that selectively enable or disable certain codecs. - -The avserver streaming tool, which has not been maintained for many years and -was mostly broken, was removed from the tree. It was decided that it is a -significant maintenance burden and that we do our users no service by pretending -to support it, while we in fact do not. - -See the Changelog file for a more extensive list of significant changes. - -API changes ------------ - -A number of additional APIs have been introduced and some existing functions -have been deprecated and are scheduled for removal in the next release. -Significant API changes include: - -[libavcodec] -+ Added the avcodec_copy_context() function that must from now on be used for - freeing codec contexts. -+- Added a new VDA hardware acceleration API, since the old one was broken and - not fixable in a compatible way. Deprecated the old VDA API. - -[libavformat] -+ Added support for exporting stream-global (as opposed to per-packet) side - data. This feature is now used by some demuxers to export ReplayGain or - display transformation matrix (aka rotation) or stereoscopic 3D mode. -+ Added an API for live metadata updates through event flags. -+- Changed the way to provide a hint about the desired timebase to muxers. - Previously it was done by setting AVStream.codec.time_base. Now callers - should set AVStream.time_base. - -[libavresample] -+ Added an API for working with AVFrames. - -Please see the file doc/APIchanges for details along with similar -programmer-centric information. diff --git a/doc/authors.texi b/doc/authors.texi new file mode 100644 index 0000000000..6c8c1d7efa --- /dev/null +++ b/doc/authors.texi @@ -0,0 +1,11 @@ +@chapter Authors + +The FFmpeg developers. + +For details about the authorship, see the Git history of the project +(git://source.ffmpeg.org/ffmpeg), e.g. by typing the command +@command{git log} in the FFmpeg source directory, or browsing the +online repository at @url{http://source.ffmpeg.org}. + +Maintainers for the specific components are listed in the file +@file{MAINTAINERS} in the source code tree. diff --git a/doc/avplay.texi b/doc/avplay.texi deleted file mode 100644 index d143f75201..0000000000 --- a/doc/avplay.texi +++ /dev/null @@ -1,186 +0,0 @@ -\input texinfo @c -*- texinfo -*- - -@settitle avplay Documentation -@titlepage -@center @titlefont{avplay Documentation} -@end titlepage - -@top - -@contents - -@chapter Synopsis - -@example -@c man begin SYNOPSIS -avplay [options] @file{input_file} -@c man end -@end example - -@chapter Description -@c man begin DESCRIPTION - -AVplay is a very simple and portable media player using the Libav -libraries and the SDL library. It is mostly used as a testbed for the -various Libav APIs. -@c man end - -@chapter Options -@c man begin OPTIONS - -@include avtools-common-opts.texi - -@section Main options - -@table @option -@item -x @var{width} -Force displayed width. -@item -y @var{height} -Force displayed height. -@item -s @var{size} -This option has been removed. Use private format options for specifying the -input video size. For example with the rawvideo demuxer you need to specify the -option @var{video_size}. -@item -an -Disable audio. -@item -vn -Disable video. -@item -ss @var{pos} -Seek to a given position in seconds. -@item -t @var{duration} -play <duration> seconds of audio/video -@item -bytes -Seek by bytes. -@item -nodisp -Disable graphical display. -@item -f @var{fmt} -Force format. -@item -window_title @var{title} -Set window title (default is the input filename). -@item -loop @var{number} -Loops movie playback <number> times. 0 means forever. -@item -vf @var{filter_graph} -@var{filter_graph} is a description of the filter graph to apply to -the input video. -Use the option "-filters" to show all the available filters (including -also sources and sinks). - -@end table - -@section Advanced options -@table @option -@item -pix_fmt @var{format} -This option has been removed. Use private options for specifying the -input pixel format. For example with the rawvideo demuxer you need to specify -the option @var{pixel_format}. -@item -stats -Show the stream duration, the codec parameters, the current position in -the stream and the audio/video synchronisation drift. -@item -bug -Work around bugs. -@item -fast -Non-spec-compliant optimizations. -@item -genpts -Generate pts. -@item -rtp_tcp -Force RTP/TCP protocol usage instead of RTP/UDP. It is only meaningful -if you are streaming with the RTSP protocol. -@item -sync @var{type} -Set the master clock to audio (@code{type=audio}), video -(@code{type=video}) or external (@code{type=ext}). Default is audio. The -master clock is used to control audio-video synchronization. Most media -players use audio as master clock, but in some cases (streaming or high -quality broadcast) it is necessary to change that. This option is mainly -used for debugging purposes. -@item -threads @var{count} -Set the thread count. -@item -ast @var{audio_stream_number} -Select the desired audio stream number, counting from 0. The number -refers to the list of all the input audio streams. If it is greater -than the number of audio streams minus one, then the last one is -selected, if it is negative the audio playback is disabled. -@item -vst @var{video_stream_number} -Select the desired video stream number, counting from 0. The number -refers to the list of all the input video streams. If it is greater -than the number of video streams minus one, then the last one is -selected, if it is negative the video playback is disabled. -@item -sst @var{subtitle_stream_number} -Select the desired subtitle stream number, counting from 0. The number -refers to the list of all the input subtitle streams. If it is greater -than the number of subtitle streams minus one, then the last one is -selected, if it is negative the subtitle rendering is disabled. -@item -noautoexit -Do not exit after playback is finished. -@item -exitonkeydown -Exit if any key is pressed. -@item -exitonmousedown -Exit if any mouse button is pressed. -@item -noautorotate -Disable automatically rotating video based on file metadata. -@end table - -@section While playing - -@table @key -@item q, ESC -Quit. - -@item f -Toggle full screen. - -@item p, SPC -Pause. - -@item a -Cycle audio channel. - -@item v -Cycle video channel. - -@item t -Cycle subtitle channel. - -@item w -Show audio waves. - -@item left/right -Seek backward/forward 10 seconds. - -@item down/up -Seek backward/forward 1 minute. - -@item PGDOWN/PGUP -Seek to the previous/next chapter. - -@item mouse click -Seek to percentage in file corresponding to fraction of width. - -@end table - -@c man end - -@include eval.texi -@include decoders.texi -@include demuxers.texi -@include muxers.texi -@include indevs.texi -@include outdevs.texi -@include protocols.texi -@include filters.texi - -@ignore - -@setfilename avplay -@settitle AVplay media player - -@c man begin SEEALSO -avconv(1), avprobe(1) and the Libav HTML documentation -@c man end - -@c man begin AUTHORS -The Libav developers -@c man end - -@end ignore - -@bye diff --git a/doc/avprobe.texi b/doc/avprobe.texi deleted file mode 100644 index 7e6fedf5c4..0000000000 --- a/doc/avprobe.texi +++ /dev/null @@ -1,141 +0,0 @@ -\input texinfo @c -*- texinfo -*- - -@settitle avprobe Documentation -@titlepage -@center @titlefont{avprobe Documentation} -@end titlepage - -@top - -@contents - -@chapter Synopsis - -The generic syntax is: - -@example -@c man begin SYNOPSIS -avprobe [options] [@file{input_file}] -@c man end -@end example - -@chapter Description -@c man begin DESCRIPTION - -avprobe gathers information from multimedia streams and prints it in -human- and machine-readable fashion. - -For example it can be used to check the format of the container used -by a multimedia stream and the format and type of each media stream -contained in it. - -If a filename is specified in input, avprobe will try to open and -probe the file content. If the file cannot be opened or recognized as -a multimedia file, a positive exit code is returned. - -avprobe may be employed both as a standalone application or in -combination with a textual filter, which may perform more -sophisticated processing, e.g. statistical processing or plotting. - -Options are used to list some of the formats supported by avprobe or -for specifying which information to display, and for setting how -avprobe will show it. - -avprobe output is designed to be easily parsable by any INI or JSON -parsers. - -@c man end - -@chapter Options -@c man begin OPTIONS - -@include avtools-common-opts.texi - -@section Main options - -@table @option - -@item -f @var{format} -Force format to use. - -@item -of @var{formatter} -Use a specific formatter to output the document. The following -formatters are available -@table @option -@item ini - -@item json - -@item old -Pseudo-INI format that used to be the only one available in old -avprobe versions. -@end table - -@item -unit -Show the unit of the displayed values. - -@item -prefix -Use SI prefixes for the displayed values. -Unless the "-byte_binary_prefix" option is used all the prefixes -are decimal. - -@item -byte_binary_prefix -Force the use of binary prefixes for byte values. - -@item -sexagesimal -Use sexagesimal format HH:MM:SS.MICROSECONDS for time values. - -@item -pretty -Prettify the format of the displayed values, it corresponds to the -options "-unit -prefix -byte_binary_prefix -sexagesimal". - -@item -show_format -Show information about the container format of the input multimedia -stream. - -All the container format information is printed within a section with -name "FORMAT". - -@item -show_format_entry @var{name} -Like @option{-show_format}, but only prints the specified entry of the -container format information, rather than all. This option may be given more -than once, then all specified entries will be shown. - -@item -show_packets -Show information about each packet contained in the input multimedia -stream. - -The information for each single packet is printed within a dedicated -section with name "PACKET". - -@item -show_streams -Show information about each media stream contained in the input -multimedia stream. - -Each media stream information is printed within a dedicated section -with name "STREAM". - -@end table -@c man end - -@include demuxers.texi -@include muxers.texi -@include protocols.texi -@include indevs.texi - -@ignore - -@setfilename avprobe -@settitle avprobe media prober - -@c man begin SEEALSO -avconv(1), avplay(1) and the Libav HTML documentation -@c man end - -@c man begin AUTHORS -The Libav developers -@c man end - -@end ignore - -@bye diff --git a/doc/avtools-common-opts.texi b/doc/avtools-common-opts.texi deleted file mode 100644 index 79f764b58c..0000000000 --- a/doc/avtools-common-opts.texi +++ /dev/null @@ -1,197 +0,0 @@ -All the numerical options, if not specified otherwise, accept in input -a string representing a number, which may contain one of the -SI unit prefixes, for example 'K', 'M', 'G'. -If 'i' is appended after the prefix, binary prefixes are used, -which are based on powers of 1024 instead of powers of 1000. -The 'B' postfix multiplies the value by 8, and can be -appended after a unit prefix or used alone. This allows using for -example 'KB', 'MiB', 'G' and 'B' as number postfix. - -Options which do not take arguments are boolean options, and set the -corresponding value to true. They can be set to false by prefixing -with "no" the option name, for example using "-nofoo" in the -command line will set to false the boolean option with name "foo". - -@anchor{Stream specifiers} -@section Stream specifiers -Some options are applied per-stream, e.g. bitrate or codec. Stream specifiers -are used to precisely specify which stream(s) does a given option belong to. - -A stream specifier is a string generally appended to the option name and -separated from it by a colon. E.g. @code{-codec:a:1 ac3} option contains -@code{a:1} stream specifer, which matches the second audio stream. Therefore it -would select the ac3 codec for the second audio stream. - -A stream specifier can match several stream, the option is then applied to all -of them. E.g. the stream specifier in @code{-b:a 128k} matches all audio -streams. - -An empty stream specifier matches all streams, for example @code{-codec copy} -or @code{-codec: copy} would copy all the streams without reencoding. - -Possible forms of stream specifiers are: -@table @option -@item @var{stream_index} -Matches the stream with this index. E.g. @code{-threads:1 4} would set the -thread count for the second stream to 4. -@item @var{stream_type}[:@var{stream_index}] -@var{stream_type} is one of: 'v' for video, 'a' for audio, 's' for subtitle, -'d' for data and 't' for attachments. If @var{stream_index} is given, then -matches stream number @var{stream_index} of this type. Otherwise matches all -streams of this type. -@item p:@var{program_id}[:@var{stream_index}] -If @var{stream_index} is given, then matches stream number @var{stream_index} in -program with id @var{program_id}. Otherwise matches all streams in this program. -@item i:@var{stream_id} -Match the stream by stream id (e.g. PID in MPEG-TS container). -@item m:@var{key}[:@var{value}] -Matches streams with the metadata tag @var{key} having the specified value. If -@var{value} is not given, matches streams that contain the given tag with any -value. -@item u -Matches streams with usable configuration, the codec must be defined and the -essential information such as video dimension or audio sample rate must be present. - -Note that in @command{avconv}, matching by metadata will only work properly for -input files. -@end table -@section Generic options - -These options are shared amongst the av* tools. - -@table @option - -@item -L -Show license. - -@item -h, -?, -help, --help [@var{arg}] -Show help. An optional parameter may be specified to print help about a specific -item. - -Possible values of @var{arg} are: -@table @option -@item decoder=@var{decoder_name} -Print detailed information about the decoder named @var{decoder_name}. Use the -@option{-decoders} option to get a list of all decoders. - -@item encoder=@var{encoder_name} -Print detailed information about the encoder named @var{encoder_name}. Use the -@option{-encoders} option to get a list of all encoders. - -@item demuxer=@var{demuxer_name} -Print detailed information about the demuxer named @var{demuxer_name}. Use the -@option{-formats} option to get a list of all demuxers and muxers. - -@item muxer=@var{muxer_name} -Print detailed information about the muxer named @var{muxer_name}. Use the -@option{-formats} option to get a list of all muxers and demuxers. - -@item filter=@var{filter_name} -Print detailed information about the filter name @var{filter_name}. Use the -@option{-filters} option to get a list of all filters. - -@end table - -@item -version -Show version. - -@item -formats -Show available formats. - -The fields preceding the format names have the following meanings: -@table @samp -@item D -Decoding available -@item E -Encoding available -@end table - -@item -codecs -Show all codecs known to libavcodec. - -Note that the term 'codec' is used throughout this documentation as a shortcut -for what is more correctly called a media bitstream format. - -@item -decoders -Show available decoders. - -@item -encoders -Show all available encoders. - -@item -bsfs -Show available bitstream filters. - -@item -protocols -Show available protocols. - -@item -filters -Show available libavfilter filters. - -@item -pix_fmts -Show available pixel formats. - -@item -sample_fmts -Show available sample formats. - -@item -loglevel @var{loglevel} | -v @var{loglevel} -Set the logging level used by the library. -@var{loglevel} is a number or a string containing one of the following values: -@table @samp -@item quiet -@item panic -@item fatal -@item error -@item warning -@item info -@item verbose -@item debug -@item trace -@end table - -By default the program logs to stderr, if coloring is supported by the -terminal, colors are used to mark errors and warnings. Log coloring -can be disabled setting the environment variable -@env{AV_LOG_FORCE_NOCOLOR} or @env{NO_COLOR}, or can be forced setting -the environment variable @env{AV_LOG_FORCE_COLOR}. -The use of the environment variable @env{NO_COLOR} is deprecated and -will be dropped in a following Libav version. - -@item -cpuflags mask (@emph{global}) -Set a mask that's applied to autodetected CPU flags. This option is intended -for testing. Do not use it unless you know what you're doing. - -@end table - -@section AVOptions - -These options are provided directly by the libavformat, libavdevice and -libavcodec libraries. To see the list of available AVOptions, use the -@option{-help} option. They are separated into two categories: -@table @option -@item generic -These options can be set for any container, codec or device. Generic options -are listed under AVFormatContext options for containers/devices and under -AVCodecContext options for codecs. -@item private -These options are specific to the given container, device or codec. Private -options are listed under their corresponding containers/devices/codecs. -@end table - -For example to write an ID3v2.3 header instead of a default ID3v2.4 to -an MP3 file, use the @option{id3v2_version} private option of the MP3 -muxer: -@example -avconv -i input.flac -id3v2_version 3 out.mp3 -@end example - -All codec AVOptions are obviously per-stream, so the chapter on stream -specifiers applies to them - -Note @option{-nooption} syntax cannot be used for boolean AVOptions, -use @option{-option 0}/@option{-option 1}. - -Note2 old undocumented way of specifying per-stream AVOptions by prepending -v/a/s to the options name is now obsolete and will be removed soon. - -@include avoptions_codec.texi -@include avoptions_format.texi diff --git a/doc/avutil.txt b/doc/avutil.txt deleted file mode 100644 index 0847683d1d..0000000000 --- a/doc/avutil.txt +++ /dev/null @@ -1,36 +0,0 @@ -AVUtil -====== -libavutil is a small lightweight library of generally useful functions. -It is not a library for code needed by both libavcodec and libavformat. - - -Overview: -========= -adler32.c adler32 checksum -aes.c AES encryption and decryption -fifo.c resizeable first in first out buffer -intfloat_readwrite.c portable reading and writing of floating point values -log.c "printf" with context and level -md5.c MD5 Message-Digest Algorithm -rational.c code to perform exact calculations with rational numbers -tree.c generic AVL tree -crc.c generic CRC checksumming code -integer.c 128bit integer math -lls.c -mathematics.c greatest common divisor, integer sqrt, integer log2, ... -mem.c memory allocation routines with guaranteed alignment - -Headers: -bswap.h big/little/native-endian conversion code -x86_cpu.h a few useful macros for unifying x86-64 and x86-32 code -avutil.h -common.h -intreadwrite.h reading and writing of unaligned big/little/native-endian integers - - -Goals: -====== -* Modular (few interdependencies and the possibility of disabling individual parts during ./configure) -* Small (source and object) -* Efficient (low CPU and memory usage) -* Useful (avoid useless features almost no one needs) diff --git a/doc/bitstream_filters.texi b/doc/bitstream_filters.texi index 6e7f8781ee..563049e281 100644 --- a/doc/bitstream_filters.texi +++ b/doc/bitstream_filters.texi @@ -1,7 +1,7 @@ @chapter Bitstream Filters @c man begin BITSTREAM FILTERS -When you configure your Libav build, all the supported bitstream +When you configure your FFmpeg build, all the supported bitstream filters are enabled by default. You can list all available ones using the configure option @code{--list-bsfs}. @@ -10,20 +10,91 @@ You can disable all the bitstream filters using the configure option the option @code{--enable-bsf=BSF}, or you can disable a particular bitstream filter using the option @code{--disable-bsf=BSF}. -The option @code{-bsfs} of the av* tools will display the list of +The option @code{-bsfs} of the ff* tools will display the list of all the supported bitstream filters included in your build. -Below is a description of the currently available bitstream filters. +The ff* tools have a -bsf option applied per stream, taking a +comma-separated list of filters, whose parameters follow the filter +name after a '='. + +@example +ffmpeg -i INPUT -c:v copy -bsf:v filter1[=opt1=str1/opt2=str2][,filter2] OUTPUT +@end example + +Below is a description of the currently available bitstream filters, +with their parameters, if any. @section aac_adtstoasc +Convert MPEG-2/4 AAC ADTS to MPEG-4 Audio Specific Configuration +bitstream filter. + +This filter creates an MPEG-4 AudioSpecificConfig from an MPEG-2/4 +ADTS header and removes the ADTS header. + +This is required for example when copying an AAC stream from a raw +ADTS AAC container to a FLV or a MOV/MP4 file. + @section chomp -@section dump_extradata +Remove zero padding at the end of a packet. + +@section dump_extra + +Add extradata to the beginning of the filtered packets. + +The additional argument specifies which packets should be filtered. +It accepts the values: +@table @samp +@item a +add extradata to all key packets, but only if @var{local_header} is +set in the @option{flags2} codec context field + +@item k +add extradata to all key packets + +@item e +add extradata to all packets +@end table + +If not specified it is assumed @samp{k}. + +For example the following @command{ffmpeg} command forces a global +header (thus disabling individual packet headers) in the H.264 packets +generated by the @code{libx264} encoder, but corrects them by adding +the header stored in extradata to the key packets: +@example +ffmpeg -i INPUT -map 0 -flags:v +global_header -c:v libx264 -bsf:v dump_extra out.ts +@end example @section h264_mp4toannexb -@section imx_dump_header +Convert an H.264 bitstream from length prefixed mode to start code +prefixed mode (as defined in the Annex B of the ITU-T H.264 +specification). + +This is required by some streaming formats, typically the MPEG-2 +transport stream format ("mpegts"). + +For example to remux an MP4 file containing an H.264 stream to mpegts +format with @command{ffmpeg}, you can use the command: + +@example +ffmpeg -i INPUT.mp4 -codec copy -bsf:v h264_mp4toannexb OUTPUT.ts +@end example + +@section imxdump + +Modifies the bitstream to fit in MOV and to be usable by the Final Cut +Pro decoder. This filter only applies to the mpeg2video codec, and is +likely not needed for Final Cut Pro 7 and newer with the appropriate +@option{-tag:v}. + +For example, to remux 30 MB/sec NTSC IMX to MOV: + +@example +ffmpeg -i input.mxf -c copy -bsf:v imxdump -tag:v mx3n output.mov +@end example @section mjpeg2jpeg @@ -34,7 +105,7 @@ JPEG image. The individual frames can be extracted without loss, e.g. by @example -avconv -i ../some_mjpeg.avi -c:v copy frames_%d.jpg +ffmpeg -i ../some_mjpeg.avi -c:v copy frames_%d.jpg @end example Unfortunately, these chunks are incomplete JPEG images, because @@ -57,21 +128,53 @@ stream (carrying the AVI1 header ID and lacking a DHT segment) to produce fully qualified JPEG images. @example -avconv -i mjpeg-movie.avi -c:v copy -bsf:v mjpeg2jpeg frame_%d.jpg +ffmpeg -i mjpeg-movie.avi -c:v copy -bsf:v mjpeg2jpeg frame_%d.jpg exiftran -i -9 frame*.jpg -avconv -i frame_%d.jpg -c:v copy rotated.avi +ffmpeg -i frame_%d.jpg -c:v copy rotated.avi @end example @section mjpega_dump_header @section movsub -@section mp3_header_compress - @section mp3_header_decompress +@section mpeg4_unpack_bframes + +Unpack DivX-style packed B-frames. + +DivX-style packed B-frames are not valid MPEG-4 and were only a +workaround for the broken Video for Windows subsystem. +They use more space, can cause minor AV sync issues, require more +CPU power to decode (unless the player has some decoded picture queue +to compensate the 2,0,2,0 frame per packet style) and cause +trouble if copied into a standard container like mp4 or mpeg-ps/ts, +because MPEG-4 decoders may not be able to decode them, since they are +not valid MPEG-4. + +For example to fix an AVI file containing an MPEG-4 stream with +DivX-style packed B-frames using @command{ffmpeg}, you can use the command: + +@example +ffmpeg -i INPUT.avi -codec copy -bsf:v mpeg4_unpack_bframes OUTPUT.avi +@end example + @section noise -@section remove_extradata +Damages the contents of packets without damaging the container. Can be +used for fuzzing or testing error resilience/concealment. + +Parameters: +A numeral string, whose value is related to how often output bytes will +be modified. Therefore, values below or equal to 0 are forbidden, and +the lower the more frequent bytes will be modified, with 1 meaning +every byte is modified. + +@example +ffmpeg -i INPUT -c copy -bsf noise[=1] output.mkv +@end example +applies the modification to every byte. + +@section remove_extra @c man end BITSTREAM FILTERS diff --git a/doc/bootstrap.min.css b/doc/bootstrap.min.css new file mode 100644 index 0000000000..6f68017d58 --- /dev/null +++ b/doc/bootstrap.min.css @@ -0,0 +1,5 @@ +/*! + * Bootstrap v3.2.0 (http://getbootstrap.com) + * Copyright 2011-2014 Twitter, Inc. + * Licensed under MIT (https://github.com/twbs/bootstrap/blob/master/LICENSE) + *//*! normalize.css v3.0.1 | MIT License | git.io/normalize 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print{.visible-print{display:block!important}table.visible-print{display:table}tr.visible-print{display:table-row!important}th.visible-print,td.visible-print{display:table-cell!important}}.visible-print-block{display:none!important}@media print{.visible-print-block{display:block!important}}.visible-print-inline{display:none!important}@media print{.visible-print-inline{display:inline!important}}.visible-print-inline-block{display:none!important}@media print{.visible-print-inline-block{display:inline-block!important}}@media print{.hidden-print{display:none!important}} diff --git a/doc/build_system.txt b/doc/build_system.txt index c3dede7cde..a9bd4eb6b1 100644 --- a/doc/build_system.txt +++ b/doc/build_system.txt @@ -1,4 +1,4 @@ -Libav currently uses a custom build system, this text attempts to document +FFmpeg currently uses a custom build system, this text attempts to document some of its obscure features and options. Makefile variables: @@ -7,18 +7,35 @@ V Disable the default terse mode, the full command issued by make and its output will be shown on the screen. +DBG + Preprocess x86 external assembler files to a .dbg.asm file in the object + directory, which then gets compiled. Helps in developing those assembler + files. + DESTDIR Destination directory for the install targets, useful to prepare packages - or install Libav in cross-environments. + or install FFmpeg in cross-environments. + +GEN + Set to ‘1’ to generate the missing or mismatched references. Makefile targets: all Default target, builds all the libraries and the executables. +fate + Run the fate test suite, note that you must have installed it. + +fate-list + List all fate/regression test targets. + install Install headers, libraries and programs. +examples + Build all examples located in doc/examples. + libavformat/output-example Build the libavformat basic example. @@ -26,4 +43,22 @@ libavcodec/api-example Build the libavcodec basic example. libswscale/swscale-test - Build the swscale self-test (useful also as example). + Build the swscale self-test (useful also as an example). + +config + Reconfigure the project with the current configuration. + + +Useful standard make commands: +make -t <target> + Touch all files that otherwise would be built, this is useful to reduce + unneeded rebuilding when changing headers, but note that you must force rebuilds + of files that actually need it by hand then. + +make -j<num> + Rebuild with multiple jobs at the same time. Faster on multi processor systems. + +make -k + Continue build in case of errors, this is useful for the regression tests + sometimes but note that it will still not run all reg tests. + diff --git a/doc/codecs.texi b/doc/codecs.texi new file mode 100644 index 0000000000..b481b4a053 --- /dev/null +++ b/doc/codecs.texi @@ -0,0 +1,1162 @@ +@anchor{codec-options} +@chapter Codec Options +@c man begin CODEC OPTIONS + +libavcodec provides some generic global options, which can be set on +all the encoders and decoders. In addition each codec may support +so-called private options, which are specific for a given codec. + +Sometimes, a global option may only affect a specific kind of codec, +and may be nonsensical or ignored by another, so you need to be aware +of the meaning of the specified options. Also some options are +meant only for decoding or encoding. + +Options may be set by specifying -@var{option} @var{value} in the +FFmpeg tools, or by setting the value explicitly in the +@code{AVCodecContext} options or using the @file{libavutil/opt.h} API +for programmatic use. + +The list of supported options follow: + +@table @option +@item b @var{integer} (@emph{encoding,audio,video}) +Set bitrate in bits/s. Default value is 200K. + +@item ab @var{integer} (@emph{encoding,audio}) +Set audio bitrate (in bits/s). Default value is 128K. + +@item bt @var{integer} (@emph{encoding,video}) +Set video bitrate tolerance (in bits/s). In 1-pass mode, bitrate +tolerance specifies how far ratecontrol is willing to deviate from the +target average bitrate value. This is not related to min/max +bitrate. Lowering tolerance too much has an adverse effect on quality. + +@item flags @var{flags} (@emph{decoding/encoding,audio,video,subtitles}) +Set generic flags. + +Possible values: +@table @samp +@item mv4 +Use four motion vector by macroblock (mpeg4). +@item qpel +Use 1/4 pel motion compensation. +@item loop +Use loop filter. +@item qscale +Use fixed qscale. +@item gmc +Use gmc. +@item mv0 +Always try a mb with mv=<0,0>. +@item input_preserved + +@item pass1 +Use internal 2pass ratecontrol in first pass mode. +@item pass2 +Use internal 2pass ratecontrol in second pass mode. +@item gray +Only decode/encode grayscale. +@item emu_edge +Do not draw edges. +@item psnr +Set error[?] variables during encoding. +@item truncated + +@item naq +Normalize adaptive quantization. +@item ildct +Use interlaced DCT. +@item low_delay +Force low delay. +@item global_header +Place global headers in extradata instead of every keyframe. +@item bitexact +Only write platform-, build- and time-independent data. (except (I)DCT). +This ensures that file and data checksums are reproducible and match between +platforms. Its primary use is for regression testing. +@item aic +Apply H263 advanced intra coding / mpeg4 ac prediction. +@item cbp +Deprecated, use mpegvideo private options instead. +@item qprd +Deprecated, use mpegvideo private options instead. +@item ilme +Apply interlaced motion estimation. +@item cgop +Use closed gop. +@end table + +@item me_method @var{integer} (@emph{encoding,video}) +Set motion estimation method. + +Possible values: +@table @samp +@item zero +zero motion estimation (fastest) +@item full +full motion estimation (slowest) +@item epzs +EPZS motion estimation (default) +@item esa +esa motion estimation (alias for full) +@item tesa +tesa motion estimation +@item dia +dia motion estimation (alias for epzs) +@item log +log motion estimation +@item phods +phods motion estimation +@item x1 +X1 motion estimation +@item hex +hex motion estimation +@item umh +umh motion estimation +@item iter +iter motion estimation +@end table + +@item extradata_size @var{integer} +Set extradata size. + +@item time_base @var{rational number} +Set codec time base. + +It is the fundamental unit of time (in seconds) in terms of which +frame timestamps are represented. For fixed-fps content, timebase +should be @code{1 / frame_rate} and timestamp increments should be +identically 1. + +@item g @var{integer} (@emph{encoding,video}) +Set the group of picture (GOP) size. Default value is 12. + +@item ar @var{integer} (@emph{decoding/encoding,audio}) +Set audio sampling rate (in Hz). + +@item ac @var{integer} (@emph{decoding/encoding,audio}) +Set number of audio channels. + +@item cutoff @var{integer} (@emph{encoding,audio}) +Set cutoff bandwidth. + +@item frame_size @var{integer} (@emph{encoding,audio}) +Set audio frame size. + +Each submitted frame except the last must contain exactly frame_size +samples per channel. May be 0 when the codec has +CODEC_CAP_VARIABLE_FRAME_SIZE set, in that case the frame size is not +restricted. It is set by some decoders to indicate constant frame +size. + +@item frame_number @var{integer} +Set the frame number. + +@item delay @var{integer} + +@item qcomp @var{float} (@emph{encoding,video}) +Set video quantizer scale compression (VBR). It is used as a constant +in the ratecontrol equation. Recommended range for default rc_eq: +0.0-1.0. + +@item qblur @var{float} (@emph{encoding,video}) +Set video quantizer scale blur (VBR). + +@item qmin @var{integer} (@emph{encoding,video}) +Set min video quantizer scale (VBR). Must be included between -1 and +69, default value is 2. + +@item qmax @var{integer} (@emph{encoding,video}) +Set max video quantizer scale (VBR). Must be included between -1 and +1024, default value is 31. + +@item qdiff @var{integer} (@emph{encoding,video}) +Set max difference between the quantizer scale (VBR). + +@item bf @var{integer} (@emph{encoding,video}) +Set max number of B frames between non-B-frames. + +Must be an integer between -1 and 16. 0 means that B-frames are +disabled. If a value of -1 is used, it will choose an automatic value +depending on the encoder. + +Default value is 0. + +@item b_qfactor @var{float} (@emph{encoding,video}) +Set qp factor between P and B frames. + +@item rc_strategy @var{integer} (@emph{encoding,video}) +Set ratecontrol method. + +@item b_strategy @var{integer} (@emph{encoding,video}) +Set strategy to choose between I/P/B-frames. + +@item ps @var{integer} (@emph{encoding,video}) +Set RTP payload size in bytes. + +@item mv_bits @var{integer} +@item header_bits @var{integer} +@item i_tex_bits @var{integer} +@item p_tex_bits @var{integer} +@item i_count @var{integer} +@item p_count @var{integer} +@item skip_count @var{integer} +@item misc_bits @var{integer} +@item frame_bits @var{integer} +@item codec_tag @var{integer} +@item bug @var{flags} (@emph{decoding,video}) +Workaround not auto detected encoder bugs. + +Possible values: +@table @samp +@item autodetect + +@item old_msmpeg4 +some old lavc generated msmpeg4v3 files (no autodetection) +@item xvid_ilace +Xvid interlacing bug (autodetected if fourcc==XVIX) +@item ump4 +(autodetected if fourcc==UMP4) +@item no_padding +padding bug (autodetected) +@item amv + +@item ac_vlc +illegal vlc bug (autodetected per fourcc) +@item qpel_chroma + +@item std_qpel +old standard qpel (autodetected per fourcc/version) +@item qpel_chroma2 + +@item direct_blocksize +direct-qpel-blocksize bug (autodetected per fourcc/version) +@item edge +edge padding bug (autodetected per fourcc/version) +@item hpel_chroma + +@item dc_clip + +@item ms +Workaround various bugs in microsoft broken decoders. +@item trunc +trancated frames +@end table + +@item lelim @var{integer} (@emph{encoding,video}) +Set single coefficient elimination threshold for luminance (negative +values also consider DC coefficient). + +@item celim @var{integer} (@emph{encoding,video}) +Set single coefficient elimination threshold for chrominance (negative +values also consider dc coefficient) + +@item strict @var{integer} (@emph{decoding/encoding,audio,video}) +Specify how strictly to follow the standards. + +Possible values: +@table @samp +@item very +strictly conform to a older more strict version of the spec or reference software +@item strict +strictly conform to all the things in the spec no matter what consequences +@item normal + +@item unofficial +allow unofficial extensions +@item experimental +allow non standardized experimental things, experimental +(unfinished/work in progress/not well tested) decoders and encoders. +Note: experimental decoders can pose a security risk, do not use this for +decoding untrusted input. +@end table + +@item b_qoffset @var{float} (@emph{encoding,video}) +Set QP offset between P and B frames. + +@item err_detect @var{flags} (@emph{decoding,audio,video}) +Set error detection flags. + +Possible values: +@table @samp +@item crccheck +verify embedded CRCs +@item bitstream +detect bitstream specification deviations +@item buffer +detect improper bitstream length +@item explode +abort decoding on minor error detection +@item ignore_err +ignore decoding errors, and continue decoding. +This is useful if you want to analyze the content of a video and thus want +everything to be decoded no matter what. This option will not result in a video +that is pleasing to watch in case of errors. +@item careful +consider things that violate the spec and have not been seen in the wild as errors +@item compliant +consider all spec non compliancies as errors +@item aggressive +consider things that a sane encoder should not do as an error +@end table + +@item has_b_frames @var{integer} + +@item block_align @var{integer} + +@item mpeg_quant @var{integer} (@emph{encoding,video}) +Use MPEG quantizers instead of H.263. + +@item qsquish @var{float} (@emph{encoding,video}) +How to keep quantizer between qmin and qmax (0 = clip, 1 = use +differentiable function). + +@item rc_qmod_amp @var{float} (@emph{encoding,video}) +Set experimental quantizer modulation. + +@item rc_qmod_freq @var{integer} (@emph{encoding,video}) +Set experimental quantizer modulation. + +@item rc_override_count @var{integer} + +@item rc_eq @var{string} (@emph{encoding,video}) +Set rate control equation. When computing the expression, besides the +standard functions defined in the section 'Expression Evaluation', the +following functions are available: bits2qp(bits), qp2bits(qp). Also +the following constants are available: iTex pTex tex mv fCode iCount +mcVar var isI isP isB avgQP qComp avgIITex avgPITex avgPPTex avgBPTex +avgTex. + +@item maxrate @var{integer} (@emph{encoding,audio,video}) +Set max bitrate tolerance (in bits/s). Requires bufsize to be set. + +@item minrate @var{integer} (@emph{encoding,audio,video}) +Set min bitrate tolerance (in bits/s). Most useful in setting up a CBR +encode. It is of little use elsewise. + +@item bufsize @var{integer} (@emph{encoding,audio,video}) +Set ratecontrol buffer size (in bits). + +@item rc_buf_aggressivity @var{float} (@emph{encoding,video}) +Currently useless. + +@item i_qfactor @var{float} (@emph{encoding,video}) +Set QP factor between P and I frames. + +@item i_qoffset @var{float} (@emph{encoding,video}) +Set QP offset between P and I frames. + +@item rc_init_cplx @var{float} (@emph{encoding,video}) +Set initial complexity for 1-pass encoding. + +@item dct @var{integer} (@emph{encoding,video}) +Set DCT algorithm. + +Possible values: +@table @samp +@item auto +autoselect a good one (default) +@item fastint +fast integer +@item int +accurate integer +@item mmx + +@item altivec + +@item faan +floating point AAN DCT +@end table + +@item lumi_mask @var{float} (@emph{encoding,video}) +Compress bright areas stronger than medium ones. + +@item tcplx_mask @var{float} (@emph{encoding,video}) +Set temporal complexity masking. + +@item scplx_mask @var{float} (@emph{encoding,video}) +Set spatial complexity masking. + +@item p_mask @var{float} (@emph{encoding,video}) +Set inter masking. + +@item dark_mask @var{float} (@emph{encoding,video}) +Compress dark areas stronger than medium ones. + +@item idct @var{integer} (@emph{decoding/encoding,video}) +Select IDCT implementation. + +Possible values: +@table @samp +@item auto + +@item int + +@item simple + +@item simplemmx + +@item simpleauto +Automatically pick a IDCT compatible with the simple one + +@item arm + +@item altivec + +@item sh4 + +@item simplearm + +@item simplearmv5te + +@item simplearmv6 + +@item simpleneon + +@item simplealpha + +@item ipp + +@item xvidmmx + +@item faani +floating point AAN IDCT +@end table + +@item slice_count @var{integer} + +@item ec @var{flags} (@emph{decoding,video}) +Set error concealment strategy. + +Possible values: +@table @samp +@item guess_mvs +iterative motion vector (MV) search (slow) +@item deblock +use strong deblock filter for damaged MBs +@item favor_inter +favor predicting from the previous frame instead of the current +@end table + +@item bits_per_coded_sample @var{integer} + +@item pred @var{integer} (@emph{encoding,video}) +Set prediction method. + +Possible values: +@table @samp +@item left + +@item plane + +@item median + +@end table + +@item aspect @var{rational number} (@emph{encoding,video}) +Set sample aspect ratio. + +@item debug @var{flags} (@emph{decoding/encoding,audio,video,subtitles}) +Print specific debug info. + +Possible values: +@table @samp +@item pict +picture info +@item rc +rate control +@item bitstream + +@item mb_type +macroblock (MB) type +@item qp +per-block quantization parameter (QP) +@item mv +motion vector +@item dct_coeff + +@item green_metadata +display complexity metadata for the upcoming frame, GoP or for a given duration. + +@item skip + +@item startcode + +@item pts + +@item er +error recognition +@item mmco +memory management control operations (H.264) +@item bugs + +@item vis_qp +visualize quantization parameter (QP), lower QP are tinted greener +@item vis_mb_type +visualize block types +@item buffers +picture buffer allocations +@item thread_ops +threading operations +@item nomc +skip motion compensation +@end table + +@item vismv @var{integer} (@emph{decoding,video}) +Visualize motion vectors (MVs). + +This option is deprecated, see the codecview filter instead. + +Possible values: +@table @samp +@item pf +forward predicted MVs of P-frames +@item bf +forward predicted MVs of B-frames +@item bb +backward predicted MVs of B-frames +@end table + +@item cmp @var{integer} (@emph{encoding,video}) +Set full pel me compare function. + +Possible values: +@table @samp +@item sad +sum of absolute differences, fast (default) +@item sse +sum of squared errors +@item satd +sum of absolute Hadamard transformed differences +@item dct +sum of absolute DCT transformed differences +@item psnr +sum of squared quantization errors (avoid, low quality) +@item bit +number of bits needed for the block +@item rd +rate distortion optimal, slow +@item zero +0 +@item vsad +sum of absolute vertical differences +@item vsse +sum of squared vertical differences +@item nsse +noise preserving sum of squared differences +@item w53 +5/3 wavelet, only used in snow +@item w97 +9/7 wavelet, only used in snow +@item dctmax + +@item chroma + +@end table + +@item subcmp @var{integer} (@emph{encoding,video}) +Set sub pel me compare function. + +Possible values: +@table @samp +@item sad +sum of absolute differences, fast (default) +@item sse +sum of squared errors +@item satd +sum of absolute Hadamard transformed differences +@item dct +sum of absolute DCT transformed differences +@item psnr +sum of squared quantization errors (avoid, low quality) +@item bit +number of bits needed for the block +@item rd +rate distortion optimal, slow +@item zero +0 +@item vsad +sum of absolute vertical differences +@item vsse +sum of squared vertical differences +@item nsse +noise preserving sum of squared differences +@item w53 +5/3 wavelet, only used in snow +@item w97 +9/7 wavelet, only used in snow +@item dctmax + +@item chroma + +@end table + +@item mbcmp @var{integer} (@emph{encoding,video}) +Set macroblock compare function. + +Possible values: +@table @samp +@item sad +sum of absolute differences, fast (default) +@item sse +sum of squared errors +@item satd +sum of absolute Hadamard transformed differences +@item dct +sum of absolute DCT transformed differences +@item psnr +sum of squared quantization errors (avoid, low quality) +@item bit +number of bits needed for the block +@item rd +rate distortion optimal, slow +@item zero +0 +@item vsad +sum of absolute vertical differences +@item vsse +sum of squared vertical differences +@item nsse +noise preserving sum of squared differences +@item w53 +5/3 wavelet, only used in snow +@item w97 +9/7 wavelet, only used in snow +@item dctmax + +@item chroma + +@end table + +@item ildctcmp @var{integer} (@emph{encoding,video}) +Set interlaced dct compare function. + +Possible values: +@table @samp +@item sad +sum of absolute differences, fast (default) +@item sse +sum of squared errors +@item satd +sum of absolute Hadamard transformed differences +@item dct +sum of absolute DCT transformed differences +@item psnr +sum of squared quantization errors (avoid, low quality) +@item bit +number of bits needed for the block +@item rd +rate distortion optimal, slow +@item zero +0 +@item vsad +sum of absolute vertical differences +@item vsse +sum of squared vertical differences +@item nsse +noise preserving sum of squared differences +@item w53 +5/3 wavelet, only used in snow +@item w97 +9/7 wavelet, only used in snow +@item dctmax + +@item chroma + +@end table + +@item dia_size @var{integer} (@emph{encoding,video}) +Set diamond type & size for motion estimation. + +@item last_pred @var{integer} (@emph{encoding,video}) +Set amount of motion predictors from the previous frame. + +@item preme @var{integer} (@emph{encoding,video}) +Set pre motion estimation. + +@item precmp @var{integer} (@emph{encoding,video}) +Set pre motion estimation compare function. + +Possible values: +@table @samp +@item sad +sum of absolute differences, fast (default) +@item sse +sum of squared errors +@item satd +sum of absolute Hadamard transformed differences +@item dct +sum of absolute DCT transformed differences +@item psnr +sum of squared quantization errors (avoid, low quality) +@item bit +number of bits needed for the block +@item rd +rate distortion optimal, slow +@item zero +0 +@item vsad +sum of absolute vertical differences +@item vsse +sum of squared vertical differences +@item nsse +noise preserving sum of squared differences +@item w53 +5/3 wavelet, only used in snow +@item w97 +9/7 wavelet, only used in snow +@item dctmax + +@item chroma + +@end table + +@item pre_dia_size @var{integer} (@emph{encoding,video}) +Set diamond type & size for motion estimation pre-pass. + +@item subq @var{integer} (@emph{encoding,video}) +Set sub pel motion estimation quality. + +@item dtg_active_format @var{integer} + +@item me_range @var{integer} (@emph{encoding,video}) +Set limit motion vectors range (1023 for DivX player). + +@item ibias @var{integer} (@emph{encoding,video}) +Set intra quant bias. + +@item pbias @var{integer} (@emph{encoding,video}) +Set inter quant bias. + +@item color_table_id @var{integer} + +@item global_quality @var{integer} (@emph{encoding,audio,video}) + +@item coder @var{integer} (@emph{encoding,video}) + +Possible values: +@table @samp +@item vlc +variable length coder / huffman coder +@item ac +arithmetic coder +@item raw +raw (no encoding) +@item rle +run-length coder +@item deflate +deflate-based coder +@end table + +@item context @var{integer} (@emph{encoding,video}) +Set context model. + +@item slice_flags @var{integer} + +@item xvmc_acceleration @var{integer} + +@item mbd @var{integer} (@emph{encoding,video}) +Set macroblock decision algorithm (high quality mode). + +Possible values: +@table @samp +@item simple +use mbcmp (default) +@item bits +use fewest bits +@item rd +use best rate distortion +@end table + +@item stream_codec_tag @var{integer} + +@item sc_threshold @var{integer} (@emph{encoding,video}) +Set scene change threshold. + +@item lmin @var{integer} (@emph{encoding,video}) +Set min lagrange factor (VBR). + +@item lmax @var{integer} (@emph{encoding,video}) +Set max lagrange factor (VBR). + +@item nr @var{integer} (@emph{encoding,video}) +Set noise reduction. + +@item rc_init_occupancy @var{integer} (@emph{encoding,video}) +Set number of bits which should be loaded into the rc buffer before +decoding starts. + +@item flags2 @var{flags} (@emph{decoding/encoding,audio,video}) + +Possible values: +@table @samp +@item fast +Allow non spec compliant speedup tricks. +@item sgop +Deprecated, use mpegvideo private options instead. +@item noout +Skip bitstream encoding. +@item ignorecrop +Ignore cropping information from sps. +@item local_header +Place global headers at every keyframe instead of in extradata. +@item chunks +Frame data might be split into multiple chunks. +@item showall +Show all frames before the first keyframe. +@item skiprd +Deprecated, use mpegvideo private options instead. +@item export_mvs +Export motion vectors into frame side-data (see @code{AV_FRAME_DATA_MOTION_VECTORS}) +for codecs that support it. See also @file{doc/examples/export_mvs.c}. +@end table + +@item error @var{integer} (@emph{encoding,video}) + +@item qns @var{integer} (@emph{encoding,video}) +Deprecated, use mpegvideo private options instead. + +@item threads @var{integer} (@emph{decoding/encoding,video}) +Set the number of threads to be used, in case the selected codec +implementation supports multi-threading. + +Possible values: +@table @samp +@item auto, 0 +automatically select the number of threads to set +@end table + +Default value is @samp{auto}. + +@item me_threshold @var{integer} (@emph{encoding,video}) +Set motion estimation threshold. + +@item mb_threshold @var{integer} (@emph{encoding,video}) +Set macroblock threshold. + +@item dc @var{integer} (@emph{encoding,video}) +Set intra_dc_precision. + +@item nssew @var{integer} (@emph{encoding,video}) +Set nsse weight. + +@item skip_top @var{integer} (@emph{decoding,video}) +Set number of macroblock rows at the top which are skipped. + +@item skip_bottom @var{integer} (@emph{decoding,video}) +Set number of macroblock rows at the bottom which are skipped. + +@item profile @var{integer} (@emph{encoding,audio,video}) + +Possible values: +@table @samp +@item unknown + +@item aac_main + +@item aac_low + +@item aac_ssr + +@item aac_ltp + +@item aac_he + +@item aac_he_v2 + +@item aac_ld + +@item aac_eld + +@item mpeg2_aac_low + +@item mpeg2_aac_he + +@item mpeg4_sp + +@item mpeg4_core + +@item mpeg4_main + +@item mpeg4_asp + +@item dts + +@item dts_es + +@item dts_96_24 + +@item dts_hd_hra + +@item dts_hd_ma + +@end table + +@item level @var{integer} (@emph{encoding,audio,video}) + +Possible values: +@table @samp +@item unknown + +@end table + +@item lowres @var{integer} (@emph{decoding,audio,video}) +Decode at 1= 1/2, 2=1/4, 3=1/8 resolutions. + +@item skip_threshold @var{integer} (@emph{encoding,video}) +Set frame skip threshold. + +@item skip_factor @var{integer} (@emph{encoding,video}) +Set frame skip factor. + +@item skip_exp @var{integer} (@emph{encoding,video}) +Set frame skip exponent. +Negative values behave identical to the corresponding positive ones, except +that the score is normalized. +Positive values exist primarily for compatibility reasons and are not so useful. + +@item skipcmp @var{integer} (@emph{encoding,video}) +Set frame skip compare function. + +Possible values: +@table @samp +@item sad +sum of absolute differences, fast (default) +@item sse +sum of squared errors +@item satd +sum of absolute Hadamard transformed differences +@item dct +sum of absolute DCT transformed differences +@item psnr +sum of squared quantization errors (avoid, low quality) +@item bit +number of bits needed for the block +@item rd +rate distortion optimal, slow +@item zero +0 +@item vsad +sum of absolute vertical differences +@item vsse +sum of squared vertical differences +@item nsse +noise preserving sum of squared differences +@item w53 +5/3 wavelet, only used in snow +@item w97 +9/7 wavelet, only used in snow +@item dctmax + +@item chroma + +@end table + +@item border_mask @var{float} (@emph{encoding,video}) +Increase the quantizer for macroblocks close to borders. + +@item mblmin @var{integer} (@emph{encoding,video}) +Set min macroblock lagrange factor (VBR). + +@item mblmax @var{integer} (@emph{encoding,video}) +Set max macroblock lagrange factor (VBR). + +@item mepc @var{integer} (@emph{encoding,video}) +Set motion estimation bitrate penalty compensation (1.0 = 256). + +@item skip_loop_filter @var{integer} (@emph{decoding,video}) +@item skip_idct @var{integer} (@emph{decoding,video}) +@item skip_frame @var{integer} (@emph{decoding,video}) + +Make decoder discard processing depending on the frame type selected +by the option value. + +@option{skip_loop_filter} skips frame loop filtering, @option{skip_idct} +skips frame IDCT/dequantization, @option{skip_frame} skips decoding. + +Possible values: +@table @samp +@item none +Discard no frame. + +@item default +Discard useless frames like 0-sized frames. + +@item noref +Discard all non-reference frames. + +@item bidir +Discard all bidirectional frames. + +@item nokey +Discard all frames excepts keyframes. + +@item all +Discard all frames. +@end table + +Default value is @samp{default}. + +@item bidir_refine @var{integer} (@emph{encoding,video}) +Refine the two motion vectors used in bidirectional macroblocks. + +@item brd_scale @var{integer} (@emph{encoding,video}) +Downscale frames for dynamic B-frame decision. + +@item keyint_min @var{integer} (@emph{encoding,video}) +Set minimum interval between IDR-frames. + +@item refs @var{integer} (@emph{encoding,video}) +Set reference frames to consider for motion compensation. + +@item chromaoffset @var{integer} (@emph{encoding,video}) +Set chroma qp offset from luma. + +@item trellis @var{integer} (@emph{encoding,audio,video}) +Set rate-distortion optimal quantization. + +@item sc_factor @var{integer} (@emph{encoding,video}) +Set value multiplied by qscale for each frame and added to +scene_change_score. + +@item mv0_threshold @var{integer} (@emph{encoding,video}) +@item b_sensitivity @var{integer} (@emph{encoding,video}) +Adjust sensitivity of b_frame_strategy 1. + +@item compression_level @var{integer} (@emph{encoding,audio,video}) +@item min_prediction_order @var{integer} (@emph{encoding,audio}) +@item max_prediction_order @var{integer} (@emph{encoding,audio}) +@item timecode_frame_start @var{integer} (@emph{encoding,video}) +Set GOP timecode frame start number, in non drop frame format. + +@item request_channels @var{integer} (@emph{decoding,audio}) +Set desired number of audio channels. + +@item bits_per_raw_sample @var{integer} +@item channel_layout @var{integer} (@emph{decoding/encoding,audio}) + +Possible values: +@table @samp +@end table +@item request_channel_layout @var{integer} (@emph{decoding,audio}) + +Possible values: +@table @samp +@end table +@item rc_max_vbv_use @var{float} (@emph{encoding,video}) +@item rc_min_vbv_use @var{float} (@emph{encoding,video}) +@item ticks_per_frame @var{integer} (@emph{decoding/encoding,audio,video}) +@item color_primaries @var{integer} (@emph{decoding/encoding,video}) +@item color_trc @var{integer} (@emph{decoding/encoding,video}) +@item colorspace @var{integer} (@emph{decoding/encoding,video}) + +@item color_range @var{integer} (@emph{decoding/encoding,video}) +If used as input parameter, it serves as a hint to the decoder, which +color_range the input has. + +@item chroma_sample_location @var{integer} (@emph{decoding/encoding,video}) + +@item log_level_offset @var{integer} +Set the log level offset. + +@item slices @var{integer} (@emph{encoding,video}) +Number of slices, used in parallelized encoding. + +@item thread_type @var{flags} (@emph{decoding/encoding,video}) +Select which multithreading methods to use. + +Use of @samp{frame} will increase decoding delay by one frame per +thread, so clients which cannot provide future frames should not use +it. + +Possible values: +@table @samp +@item slice +Decode more than one part of a single frame at once. + +Multithreading using slices works only when the video was encoded with +slices. + +@item frame +Decode more than one frame at once. +@end table + +Default value is @samp{slice+frame}. + +@item audio_service_type @var{integer} (@emph{encoding,audio}) +Set audio service type. + +Possible values: +@table @samp +@item ma +Main Audio Service +@item ef +Effects +@item vi +Visually Impaired +@item hi +Hearing Impaired +@item di +Dialogue +@item co +Commentary +@item em +Emergency +@item vo +Voice Over +@item ka +Karaoke +@end table + +@item request_sample_fmt @var{sample_fmt} (@emph{decoding,audio}) +Set sample format audio decoders should prefer. Default value is +@code{none}. + +@item pkt_timebase @var{rational number} + +@item sub_charenc @var{encoding} (@emph{decoding,subtitles}) +Set the input subtitles character encoding. + +@item field_order @var{field_order} (@emph{video}) +Set/override the field order of the video. +Possible values: +@table @samp +@item progressive +Progressive video +@item tt +Interlaced video, top field coded and displayed first +@item bb +Interlaced video, bottom field coded and displayed first +@item tb +Interlaced video, top coded first, bottom displayed first +@item bt +Interlaced video, bottom coded first, top displayed first +@end table + +@item skip_alpha @var{integer} (@emph{decoding,video}) +Set to 1 to disable processing alpha (transparency). This works like the +@samp{gray} flag in the @option{flags} option which skips chroma information +instead of alpha. Default is 0. + +@item codec_whitelist @var{list} (@emph{input}) +"," separated List of allowed decoders. By default all are allowed. + +@item dump_separator @var{string} (@emph{input}) +Separator used to separate the fields printed on the command line about the +Stream parameters. +For example to separate the fields with newlines and indention: +@example +ffprobe -dump_separator " + " -i ~/videos/matrixbench_mpeg2.mpg +@end example + +@end table + +@c man end CODEC OPTIONS + +@ifclear config-writeonly +@include decoders.texi +@end ifclear +@ifclear config-readonly +@include encoders.texi +@end ifclear diff --git a/doc/decoders.texi b/doc/decoders.texi index 99d2008101..35771140e1 100644 --- a/doc/decoders.texi +++ b/doc/decoders.texi @@ -1,10 +1,10 @@ @chapter Decoders @c man begin DECODERS -Decoders are configured elements in Libav which allow the decoding of +Decoders are configured elements in FFmpeg which allow the decoding of multimedia streams. -When you configure your Libav build, all the supported native decoders +When you configure your FFmpeg build, all the supported native decoders are enabled by default. Decoders requiring an external library must be enabled manually via the corresponding @code{--enable-lib} option. You can list all available decoders using the configure option @code{--list-decoders}. @@ -14,11 +14,48 @@ You can disable all the decoders with the configure option with the options @code{--enable-decoder=@var{DECODER}} / @code{--disable-decoder=@var{DECODER}}. -The option @code{-decoders} of the av* tools will display the list of +The option @code{-decoders} of the ff* tools will display the list of enabled decoders. @c man end DECODERS +@chapter Video Decoders +@c man begin VIDEO DECODERS + +A description of some of the currently available video decoders +follows. + +@section hevc + +HEVC / H.265 decoder. + +Note: the @option{skip_loop_filter} option has effect only at level +@code{all}. + +@section rawvideo + +Raw video decoder. + +This decoder decodes rawvideo streams. + +@subsection Options + +@table @option +@item top @var{top_field_first} +Specify the assumed field type of the input video. +@table @option +@item -1 +the video is assumed to be progressive (default) +@item 0 +bottom-field-first is assumed +@item 1 +top-field-first is assumed +@end table + +@end table + +@c man end VIDEO DECODERS + @chapter Audio Decoders @c man begin AUDIO DECODERS @@ -53,4 +90,205 @@ Loud sounds are fully compressed. Soft sounds are enhanced. @end table +@section flac + +FLAC audio decoder. + +This decoder aims to implement the complete FLAC specification from Xiph. + +@subsection FLAC Decoder options + +@table @option + +@item -use_buggy_lpc +The lavc FLAC encoder used to produce buggy streams with high lpc values +(like the default value). This option makes it possible to decode such streams +correctly by using lavc's old buggy lpc logic for decoding. + +@end table + +@section ffwavesynth + +Internal wave synthetizer. + +This decoder generates wave patterns according to predefined sequences. Its +use is purely internal and the format of the data it accepts is not publicly +documented. + +@section libcelt + +libcelt decoder wrapper. + +libcelt allows libavcodec to decode the Xiph CELT ultra-low delay audio codec. +Requires the presence of the libcelt headers and library during configuration. +You need to explicitly configure the build with @code{--enable-libcelt}. + +@section libgsm + +libgsm decoder wrapper. + +libgsm allows libavcodec to decode the GSM full rate audio codec. Requires +the presence of the libgsm headers and library during configuration. You need +to explicitly configure the build with @code{--enable-libgsm}. + +This decoder supports both the ordinary GSM and the Microsoft variant. + +@section libilbc + +libilbc decoder wrapper. + +libilbc allows libavcodec to decode the Internet Low Bitrate Codec (iLBC) +audio codec. Requires the presence of the libilbc headers and library during +configuration. You need to explicitly configure the build with +@code{--enable-libilbc}. + +@subsection Options + +The following option is supported by the libilbc wrapper. + +@table @option +@item enhance + +Enable the enhancement of the decoded audio when set to 1. The default +value is 0 (disabled). + +@end table + +@section libopencore-amrnb + +libopencore-amrnb decoder wrapper. + +libopencore-amrnb allows libavcodec to decode the Adaptive Multi-Rate +Narrowband audio codec. Using it requires the presence of the +libopencore-amrnb headers and library during configuration. You need to +explicitly configure the build with @code{--enable-libopencore-amrnb}. + +An FFmpeg native decoder for AMR-NB exists, so users can decode AMR-NB +without this library. + +@section libopencore-amrwb + +libopencore-amrwb decoder wrapper. + +libopencore-amrwb allows libavcodec to decode the Adaptive Multi-Rate +Wideband audio codec. Using it requires the presence of the +libopencore-amrwb headers and library during configuration. You need to +explicitly configure the build with @code{--enable-libopencore-amrwb}. + +An FFmpeg native decoder for AMR-WB exists, so users can decode AMR-WB +without this library. + +@section libopus + +libopus decoder wrapper. + +libopus allows libavcodec to decode the Opus Interactive Audio Codec. +Requires the presence of the libopus headers and library during +configuration. You need to explicitly configure the build with +@code{--enable-libopus}. + +An FFmpeg native decoder for Opus exists, so users can decode Opus +without this library. + @c man end AUDIO DECODERS + +@chapter Subtitles Decoders +@c man begin SUBTILES DECODERS + +@section dvbsub + +@subsection Options + +@table @option +@item compute_clut +@table @option +@item -1 +Compute clut if no matching CLUT is in the stream. +@item 0 +Never compute CLUT +@item 1 +Always compute CLUT and override the one provided in the stream. +@end table +@item dvb_substream +Selects the dvb substream, or all substreams if -1 which is default. + +@end table + +@section dvdsub + +This codec decodes the bitmap subtitles used in DVDs; the same subtitles can +also be found in VobSub file pairs and in some Matroska files. + +@subsection Options + +@table @option +@item palette +Specify the global palette used by the bitmaps. When stored in VobSub, the +palette is normally specified in the index file; in Matroska, the palette is +stored in the codec extra-data in the same format as in VobSub. In DVDs, the +palette is stored in the IFO file, and therefore not available when reading +from dumped VOB files. + +The format for this option is a string containing 16 24-bits hexadecimal +numbers (without 0x prefix) separated by comas, for example @code{0d00ee, +ee450d, 101010, eaeaea, 0ce60b, ec14ed, ebff0b, 0d617a, 7b7b7b, d1d1d1, +7b2a0e, 0d950c, 0f007b, cf0dec, cfa80c, 7c127b}. + +@item ifo_palette +Specify the IFO file from which the global palette is obtained. +(experimental) + +@item forced_subs_only +Only decode subtitle entries marked as forced. Some titles have forced +and non-forced subtitles in the same track. Setting this flag to @code{1} +will only keep the forced subtitles. Default value is @code{0}. +@end table + +@section libzvbi-teletext + +Libzvbi allows libavcodec to decode DVB teletext pages and DVB teletext +subtitles. Requires the presence of the libzvbi headers and library during +configuration. You need to explicitly configure the build with +@code{--enable-libzvbi}. + +@subsection Options + +@table @option +@item txt_page +List of teletext page numbers to decode. You may use the special * string to +match all pages. Pages that do not match the specified list are dropped. +Default value is *. +@item txt_chop_top +Discards the top teletext line. Default value is 1. +@item txt_format +Specifies the format of the decoded subtitles. The teletext decoder is capable +of decoding the teletext pages to bitmaps or to simple text, you should use +"bitmap" for teletext pages, because certain graphics and colors cannot be +expressed in simple text. You might use "text" for teletext based subtitles if +your application can handle simple text based subtitles. Default value is +bitmap. +@item txt_left +X offset of generated bitmaps, default is 0. +@item txt_top +Y offset of generated bitmaps, default is 0. +@item txt_chop_spaces +Chops leading and trailing spaces and removes empty lines from the generated +text. This option is useful for teletext based subtitles where empty spaces may +be present at the start or at the end of the lines or empty lines may be +present between the subtitle lines because of double-sized teletext charactes. +Default value is 1. +@item txt_duration +Sets the display duration of the decoded teletext pages or subtitles in +miliseconds. Default value is 30000 which is 30 seconds. +@item txt_transparent +Force transparent background of the generated teletext bitmaps. Default value +is 0 which means an opaque background. +@item txt_opacity +Sets the opacity (0-255) of the teletext background. If +@option{txt_transparent} is not set, it only affects characters between a start +box and an end box, typically subtitles. Default value is 0 if +@option{txt_transparent} is set, 255 otherwise. + +@end table + +@c man end SUBTILES DECODERS diff --git a/doc/default.css b/doc/default.css new file mode 100644 index 0000000000..bf50200c28 --- /dev/null +++ b/doc/default.css @@ -0,0 +1,165 @@ +a.summary-letter { + text-decoration: none; +} + +a { + color: #2D6198; +} + +a:visited { + color: #884488; +} + +#banner { + background-color: white; + position: relative; + text-align: center; +} + +#banner img { + margin-bottom: 1px; + margin-top: 5px; +} + +#body { + margin-left: 1em; + margin-right: 1em; +} + +body { + background-color: #313131; + margin: 0; + text-align: justify; +} + +.center { + margin-left: auto; + margin-right: auto; + text-align: center; +} + +#container { + background-color: white; + color: #202020; + margin-left: 1em; + margin-right: 1em; +} + +#footer { + text-align: center; +} + +h1 a, h2 a, h3 a, h4 a { + text-decoration: inherit; + color: inherit; +} + +h1, h2, h3, h4 { + padding-left: 0.4em; + border-radius: 4px; + padding-bottom: 0.25em; + padding-top: 0.25em; + border: 1px solid #6A996A; +} + +h1 { + background-color: #7BB37B; + color: #151515; + font-size: 1.2em; + padding-bottom: 0.3em; + padding-top: 0.3em; +} + +h2 { + color: #313131; + font-size: 1.0em; + background-color: #ABE3AB; +} + +h3 { + color: #313131; + font-size: 0.9em; + margin-bottom: -6px; + background-color: #BBF3BB; +} + +h4 { + color: #313131; + font-size: 0.8em; + margin-bottom: -8px; + background-color: #D1FDD1; +} + +img { + border: 0; +} + +#navbar { + background-color: #738073; + border-bottom: 1px solid #5C665C; + border-top: 1px solid #5C665C; + margin-top: 12px; + padding: 0.3em; + position: relative; + text-align: center; +} + +#navbar a, #navbar_secondary a { + color: white; + padding: 0.3em; + text-decoration: none; +} + +#navbar a:hover, #navbar_secondary a:hover { + background-color: #313131; + color: white; + text-decoration: none; +} + +#navbar_secondary { + background-color: #738073; + border-bottom: 1px solid #5C665C; + border-left: 1px solid #5C665C; + border-right: 1px solid #5C665C; + padding: 0.3em; + position: relative; + text-align: center; +} + +p { + margin-left: 1em; + margin-right: 1em; +} + +pre { + margin-left: 3em; + margin-right: 3em; + padding: 0.3em; + border: 1px solid #bbb; + background-color: #f7f7f7; +} + +dl dt { + font-weight: bold; +} + +#proj_desc { + font-size: 1.2em; +} + +#repos { + margin-left: 1em; + margin-right: 1em; + border-collapse: collapse; + border: solid 1px #6A996A; +} + +#repos th { + background-color: #7BB37B; + border: solid 1px #6A996A; +} + +#repos td { + padding: 0.2em; + border: solid 1px #6A996A; +} diff --git a/doc/demuxers.texi b/doc/demuxers.texi index 2f2f464404..6b5f8bba15 100644 --- a/doc/demuxers.texi +++ b/doc/demuxers.texi @@ -1,30 +1,343 @@ @chapter Demuxers @c man begin DEMUXERS -Demuxers are configured elements in Libav which allow to read the +Demuxers are configured elements in FFmpeg that can read the multimedia streams from a particular type of file. -When you configure your Libav build, all the supported demuxers +When you configure your FFmpeg build, all the supported demuxers are enabled by default. You can list all available ones using the -configure option "--list-demuxers". +configure option @code{--list-demuxers}. You can disable all the demuxers using the configure option -"--disable-demuxers", and selectively enable a single demuxer with -the option "--enable-demuxer=@var{DEMUXER}", or disable it -with the option "--disable-demuxer=@var{DEMUXER}". +@code{--disable-demuxers}, and selectively enable a single demuxer with +the option @code{--enable-demuxer=@var{DEMUXER}}, or disable it +with the option @code{--disable-demuxer=@var{DEMUXER}}. -The option "-formats" of the av* tools will display the list of +The option @code{-formats} of the ff* tools will display the list of enabled demuxers. The description of some of the currently available demuxers follows. +@section aa + +Audible Format 2, 3, and 4 demuxer. + +This demuxer is used to demux Audible Format 2, 3, and 4 (.aa) files. + +@section applehttp + +Apple HTTP Live Streaming demuxer. + +This demuxer presents all AVStreams from all variant streams. +The id field is set to the bitrate variant index number. By setting +the discard flags on AVStreams (by pressing 'a' or 'v' in ffplay), +the caller can decide which variant streams to actually receive. +The total bitrate of the variant that the stream belongs to is +available in a metadata key named "variant_bitrate". + +@section apng + +Animated Portable Network Graphics demuxer. + +This demuxer is used to demux APNG files. +All headers, but the PNG signature, up to (but not including) the first +fcTL chunk are transmitted as extradata. +Frames are then split as being all the chunks between two fcTL ones, or +between the last fcTL and IEND chunks. + +@table @option +@item -ignore_loop @var{bool} +Ignore the loop variable in the file if set. +@item -max_fps @var{int} +Maximum framerate in frames per second (0 for no limit). +@item -default_fps @var{int} +Default framerate in frames per second when none is specified in the file +(0 meaning as fast as possible). +@end table + +@section asf + +Advanced Systems Format demuxer. + +This demuxer is used to demux ASF files and MMS network streams. + +@table @option +@item -no_resync_search @var{bool} +Do not try to resynchronize by looking for a certain optional start code. +@end table + +@anchor{concat} +@section concat + +Virtual concatenation script demuxer. + +This demuxer reads a list of files and other directives from a text file and +demuxes them one after the other, as if all their packet had been muxed +together. + +The timestamps in the files are adjusted so that the first file starts at 0 +and each next file starts where the previous one finishes. Note that it is +done globally and may cause gaps if all streams do not have exactly the same +length. + +All files must have the same streams (same codecs, same time base, etc.). + +The duration of each file is used to adjust the timestamps of the next file: +if the duration is incorrect (because it was computed using the bit-rate or +because the file is truncated, for example), it can cause artifacts. The +@code{duration} directive can be used to override the duration stored in +each file. + +@subsection Syntax + +The script is a text file in extended-ASCII, with one directive per line. +Empty lines, leading spaces and lines starting with '#' are ignored. The +following directive is recognized: + +@table @option + +@item @code{file @var{path}} +Path to a file to read; special characters and spaces must be escaped with +backslash or single quotes. + +All subsequent file-related directives apply to that file. + +@item @code{ffconcat version 1.0} +Identify the script type and version. It also sets the @option{safe} option +to 1 if it was to its default -1. + +To make FFmpeg recognize the format automatically, this directive must +appears exactly as is (no extra space or byte-order-mark) on the very first +line of the script. + +@item @code{duration @var{dur}} +Duration of the file. This information can be specified from the file; +specifying it here may be more efficient or help if the information from the +file is not available or accurate. + +If the duration is set for all files, then it is possible to seek in the +whole concatenated video. + +@item @code{inpoint @var{timestamp}} +In point of the file. When the demuxer opens the file it instantly seeks to the +specified timestamp. Seeking is done so that all streams can be presented +successfully at In point. + +This directive works best with intra frame codecs, because for non-intra frame +ones you will usually get extra packets before the actual In point and the +decoded content will most likely contain frames before In point too. + +For each file, packets before the file In point will have timestamps less than +the calculated start timestamp of the file (negative in case of the first +file), and the duration of the files (if not specified by the @code{duration} +directive) will be reduced based on their specified In point. + +Because of potential packets before the specified In point, packet timestamps +may overlap between two concatenated files. + +@item @code{outpoint @var{timestamp}} +Out point of the file. When the demuxer reaches the specified decoding +timestamp in any of the streams, it handles it as an end of file condition and +skips the current and all the remaining packets from all streams. + +Out point is exclusive, which means that the demuxer will not output packets +with a decoding timestamp greater or equal to Out point. + +This directive works best with intra frame codecs and formats where all streams +are tightly interleaved. For non-intra frame codecs you will usually get +additional packets with presentation timestamp after Out point therefore the +decoded content will most likely contain frames after Out point too. If your +streams are not tightly interleaved you may not get all the packets from all +streams before Out point and you may only will be able to decode the earliest +stream until Out point. + +The duration of the files (if not specified by the @code{duration} +directive) will be reduced based on their specified Out point. + +@item @code{file_packet_metadata @var{key=value}} +Metadata of the packets of the file. The specified metadata will be set for +each file packet. You can specify this directive multiple times to add multiple +metadata entries. + +@item @code{stream} +Introduce a stream in the virtual file. +All subsequent stream-related directives apply to the last introduced +stream. +Some streams properties must be set in order to allow identifying the +matching streams in the subfiles. +If no streams are defined in the script, the streams from the first file are +copied. + +@item @code{exact_stream_id @var{id}} +Set the id of the stream. +If this directive is given, the string with the corresponding id in the +subfiles will be used. +This is especially useful for MPEG-PS (VOB) files, where the order of the +streams is not reliable. + +@end table + +@subsection Options + +This demuxer accepts the following option: + +@table @option + +@item safe +If set to 1, reject unsafe file paths. A file path is considered safe if it +does not contain a protocol specification and is relative and all components +only contain characters from the portable character set (letters, digits, +period, underscore and hyphen) and have no period at the beginning of a +component. + +If set to 0, any file name is accepted. + +The default is -1, it is equivalent to 1 if the format was automatically +probed and 0 otherwise. + +@item auto_convert +If set to 1, try to perform automatic conversions on packet data to make the +streams concatenable. +The default is 1. + +Currently, the only conversion is adding the h264_mp4toannexb bitstream +filter to H.264 streams in MP4 format. This is necessary in particular if +there are resolution changes. + +@item segment_time_metadata +If set to 1, every packet will contain the @var{lavf.concat.start_time} and the +@var{lavf.concat.duration} packet metadata values which are the start_time and +the duration of the respective file segments in the concatenated output +expressed in microseconds. The duration metadata is only set if it is known +based on the concat file. +The default is 0. + +@end table + +@subsection Examples + +@itemize +@item +Use absolute filenames and include some comments: +@example +# my first filename +file /mnt/share/file-1.wav +# my second filename including whitespace +file '/mnt/share/file 2.wav' +# my third filename including whitespace plus single quote +file '/mnt/share/file 3'\''.wav' +@end example + +@item +Allow for input format auto-probing, use safe filenames and set the duration of +the first file: +@example +ffconcat version 1.0 + +file file-1.wav +duration 20.0 + +file subdir/file-2.wav +@end example +@end itemize + +@section flv + +Adobe Flash Video Format demuxer. + +This demuxer is used to demux FLV files and RTMP network streams. + +@table @option +@item -flv_metadata @var{bool} +Allocate the streams according to the onMetaData array content. +@end table + +@section libgme + +The Game Music Emu library is a collection of video game music file emulators. + +See @url{http://code.google.com/p/game-music-emu/} for more information. + +Some files have multiple tracks. The demuxer will pick the first track by +default. The @option{track_index} option can be used to select a different +track. Track indexes start at 0. The demuxer exports the number of tracks as +@var{tracks} meta data entry. + +For very large files, the @option{max_size} option may have to be adjusted. + +@section gif + +Animated GIF demuxer. + +It accepts the following options: + +@table @option +@item min_delay +Set the minimum valid delay between frames in hundredths of seconds. +Range is 0 to 6000. Default value is 2. + +@item max_gif_delay +Set the maximum valid delay between frames in hundredth of seconds. +Range is 0 to 65535. Default value is 65535 (nearly eleven minutes), +the maximum value allowed by the specification. + +@item default_delay +Set the default delay between frames in hundredths of seconds. +Range is 0 to 6000. Default value is 10. + +@item ignore_loop +GIF files can contain information to loop a certain number of times (or +infinitely). If @option{ignore_loop} is set to 1, then the loop setting +from the input will be ignored and looping will not occur. If set to 0, +then looping will occur and will cycle the number of times according to +the GIF. Default value is 1. +@end table + +For example, with the overlay filter, place an infinitely looping GIF +over another video: +@example +ffmpeg -i input.mp4 -ignore_loop 0 -i input.gif -filter_complex overlay=shortest=1 out.mkv +@end example + +Note that in the above example the shortest option for overlay filter is +used to end the output video at the length of the shortest input file, +which in this case is @file{input.mp4} as the GIF in this example loops +infinitely. + @section image2 Image file demuxer. This demuxer reads from a list of image files specified by a pattern. +The syntax and meaning of the pattern is specified by the +option @var{pattern_type}. -The pattern may contain the string "%d" or "%0@var{N}d", which +The pattern may contain a suffix which is used to automatically +determine the format of the images contained in the files. + +The size, the pixel format, and the format of each image must be the +same for all the files in the sequence. + +This demuxer accepts the following options: +@table @option +@item framerate +Set the frame rate for the video stream. It defaults to 25. +@item loop +If set to 1, loop over the input. Default value is 0. +@item pattern_type +Select the pattern type used to interpret the provided filename. + +@var{pattern_type} accepts one of the following values. +@table @option +@item none +Disable pattern matching, therefore the video will only contain the specified +image. You should use this option if you do not want to create sequences from +multiple images and your filenames may contain special pattern characters. +@item sequence +Select a sequence pattern type, used to specify a sequence of files +indexed by sequential numbers. + +A sequence pattern may contain the string "%d" or "%0@var{N}d", which specifies the position of the characters representing a sequential number in each filename matched by the pattern. If the form "%d0@var{N}d" is used, the string representing the number in each @@ -32,13 +345,11 @@ filename is 0-padded and @var{N} is the total number of 0-padded digits representing the number. The literal character '%' can be specified in the pattern with the string "%%". -If the pattern contains "%d" or "%0@var{N}d", the first filename of +If the sequence pattern contains "%d" or "%0@var{N}d", the first filename of the file list specified by the pattern must contain a number -inclusively contained between 0 and 4, all the following numbers must -be sequential. This limitation may be hopefully fixed. - -The pattern may contain a suffix which is used to automatically -determine the format of the images contained in the files. +inclusively contained between @var{start_number} and +@var{start_number}+@var{start_number_range}-1, and all the following +numbers must be sequential. For example the pattern "img-%03d.bmp" will match a sequence of filenames of the form @file{img-001.bmp}, @file{img-002.bmp}, ..., @@ -46,68 +357,224 @@ filenames of the form @file{img-001.bmp}, @file{img-002.bmp}, ..., sequence of filenames of the form @file{i%m%g-1.jpg}, @file{i%m%g-2.jpg}, ..., @file{i%m%g-10.jpg}, etc. -The size, the pixel format, and the format of each image must be the -same for all the files in the sequence. +Note that the pattern must not necessarily contain "%d" or +"%0@var{N}d", for example to convert a single image file +@file{img.jpeg} you can employ the command: +@example +ffmpeg -i img.jpeg img.png +@end example + +@item glob +Select a glob wildcard pattern type. + +The pattern is interpreted like a @code{glob()} pattern. This is only +selectable if libavformat was compiled with globbing support. + +@item glob_sequence @emph{(deprecated, will be removed)} +Select a mixed glob wildcard/sequence pattern. + +If your version of libavformat was compiled with globbing support, and +the provided pattern contains at least one glob meta character among +@code{%*?[]@{@}} that is preceded by an unescaped "%", the pattern is +interpreted like a @code{glob()} pattern, otherwise it is interpreted +like a sequence pattern. -The following example shows how to use @command{avconv} for creating a -video from the images in the file sequence @file{img-001.jpeg}, -@file{img-002.jpeg}, ..., assuming an input framerate of 10 frames per -second: +All glob special characters @code{%*?[]@{@}} must be prefixed +with "%". To escape a literal "%" you shall use "%%". + +For example the pattern @code{foo-%*.jpeg} will match all the +filenames prefixed by "foo-" and terminating with ".jpeg", and +@code{foo-%?%?%?.jpeg} will match all the filenames prefixed with +"foo-", followed by a sequence of three characters, and terminating +with ".jpeg". + +This pattern type is deprecated in favor of @var{glob} and +@var{sequence}. +@end table + +Default value is @var{glob_sequence}. +@item pixel_format +Set the pixel format of the images to read. If not specified the pixel +format is guessed from the first image file in the sequence. +@item start_number +Set the index of the file matched by the image file pattern to start +to read from. Default value is 0. +@item start_number_range +Set the index interval range to check when looking for the first image +file in the sequence, starting from @var{start_number}. Default value +is 5. +@item ts_from_file +If set to 1, will set frame timestamp to modification time of image file. Note +that monotonity of timestamps is not provided: images go in the same order as +without this option. Default value is 0. +If set to 2, will set frame timestamp to the modification time of the image file in +nanosecond precision. +@item video_size +Set the video size of the images to read. If not specified the video +size is guessed from the first image file in the sequence. +@end table + +@subsection Examples + +@itemize +@item +Use @command{ffmpeg} for creating a video from the images in the file +sequence @file{img-001.jpeg}, @file{img-002.jpeg}, ..., assuming an +input frame rate of 10 frames per second: @example -avconv -i 'img-%03d.jpeg' -r 10 out.mkv +ffmpeg -framerate 10 -i 'img-%03d.jpeg' out.mkv @end example -Note that the pattern must not necessarily contain "%d" or -"%0@var{N}d", for example to convert a single image file -@file{img.jpeg} you can employ the command: +@item +As above, but start by reading from a file with index 100 in the sequence: +@example +ffmpeg -framerate 10 -start_number 100 -i 'img-%03d.jpeg' out.mkv +@end example + +@item +Read images matching the "*.png" glob pattern , that is all the files +terminating with the ".png" suffix: @example -avconv -i img.jpeg img.png +ffmpeg -framerate 10 -pattern_type glob -i "*.png" out.mkv @end example +@end itemize + +@section mov/mp4/3gp/Quicktme + +Quicktime / MP4 demuxer. +This demuxer accepts the following options: @table @option -@item -pixel_format @var{format} -Set the pixel format (for raw image) -@item -video_size @var{size} -Set the frame size (for raw image) -@item -framerate @var{rate} -Set the frame rate -@item -loop @var{bool} -Loop over the images -@item -start_number @var{start} -Specify the first number in the sequence +@item enable_drefs +Enable loading of external tracks, disabled by default. +Enabling this can theoretically leak information in some use cases. + +@item use_absolute_path +Allows loading of external tracks via absolute paths, disabled by default. +Enabling this poses a security risk. It should only be enabled if the source +is known to be non malicious. + @end table -@section applehttp +@section mpegts -Apple HTTP Live Streaming demuxer. +MPEG-2 transport stream demuxer. -This demuxer presents all AVStreams from all variant streams. -The id field is set to the bitrate variant index number. By setting -the discard flags on AVStreams (by pressing 'a' or 'v' in avplay), -the caller can decide which variant streams to actually receive. -The total bitrate of the variant that the stream belongs to is -available in a metadata key named "variant_bitrate". +This demuxer accepts the following options: +@table @option +@item resync_size +Set size limit for looking up a new synchronization. Default value is +65536. -@section flv +@item fix_teletext_pts +Override teletext packet PTS and DTS values with the timestamps calculated +from the PCR of the first program which the teletext stream is part of and is +not discarded. Default value is 1, set this option to 0 if you want your +teletext packet PTS and DTS values untouched. -Adobe Flash Video Format demuxer. +@item ts_packetsize +Output option carrying the raw packet size in bytes. +Show the detected raw packet size, cannot be set by the user. -This demuxer is used to demux FLV files and RTMP network streams. +@item scan_all_pmts +Scan and combine all PMTs. The value is an integer with value from -1 +to 1 (-1 means automatic setting, 1 means enabled, 0 means +disabled). Default value is -1. +@end table + +@section mpjpeg +MJPEG encapsulated in multi-part MIME demuxer. + +This demuxer allows reading of MJPEG, where each frame is represented as a part of +multipart/x-mixed-replace stream. @table @option -@item -flv_metadata @var{bool} -Allocate the streams according to the onMetaData array content. + +@item strict_mime_boundary +Default implementation applies a relaxed standard to multi-part MIME boundary detection, +to prevent regression with numerous existing endpoints not generating a proper MIME +MJPEG stream. Turning this option on by setting it to 1 will result in a stricter check +of the boundary value. @end table -@section asf +@section rawvideo -Advanced Systems Format demuxer. +Raw video demuxer. -This demuxer is used to demux ASF files and MMS network streams. +This demuxer allows one to read raw video data. Since there is no header +specifying the assumed video parameters, the user must specify them +in order to be able to decode the data correctly. +This demuxer accepts the following options: @table @option -@item -no_resync_search @var{bool} -Do not try to resynchronize by looking for a certain optional start code. + +@item framerate +Set input video frame rate. Default value is 25. + +@item pixel_format +Set the input video pixel format. Default value is @code{yuv420p}. + +@item video_size +Set the input video size. This value must be specified explicitly. +@end table + +For example to read a rawvideo file @file{input.raw} with +@command{ffplay}, assuming a pixel format of @code{rgb24}, a video +size of @code{320x240}, and a frame rate of 10 images per second, use +the command: +@example +ffplay -f rawvideo -pixel_format rgb24 -video_size 320x240 -framerate 10 input.raw +@end example + +@section sbg + +SBaGen script demuxer. + +This demuxer reads the script language used by SBaGen +@url{http://uazu.net/sbagen/} to generate binaural beats sessions. A SBG +script looks like that: +@example +-SE +a: 300-2.5/3 440+4.5/0 +b: 300-2.5/0 440+4.5/3 +off: - +NOW == a ++0:07:00 == b ++0:14:00 == a ++0:21:00 == b ++0:30:00 off +@end example + +A SBG script can mix absolute and relative timestamps. If the script uses +either only absolute timestamps (including the script start time) or only +relative ones, then its layout is fixed, and the conversion is +straightforward. On the other hand, if the script mixes both kind of +timestamps, then the @var{NOW} reference for relative timestamps will be +taken from the current time of day at the time the script is read, and the +script layout will be frozen according to that reference. That means that if +the script is directly played, the actual times will match the absolute +timestamps up to the sound controller's clock accuracy, but if the user +somehow pauses the playback or seeks, all times will be shifted accordingly. + +@section tedcaptions + +JSON captions used for @url{http://www.ted.com/, TED Talks}. + +TED does not provide links to the captions, but they can be guessed from the +page. The file @file{tools/bookmarklets.html} from the FFmpeg source tree +contains a bookmarklet to expose them. + +This demuxer accepts the following option: +@table @option +@item start_time +Set the start time of the TED talk, in milliseconds. The default is 15000 +(15s). It is used to sync the captions with the downloadable videos, because +they include a 15s intro. @end table -@c man end INPUT DEVICES +Example: convert the captions to a format most players understand: +@example +ffmpeg -i http://www.ted.com/talks/subtitles/id/1/lang/en talk1-en.srt +@end example + +@c man end DEMUXERS diff --git a/doc/developer.texi b/doc/developer.texi index 00e2b6028f..6db93cef70 100644 --- a/doc/developer.texi +++ b/doc/developer.texi @@ -1,4 +1,5 @@ \input texinfo @c -*- texinfo -*- +@documentencoding UTF-8 @settitle Developer Documentation @titlepage @@ -11,87 +12,44 @@ @chapter Developers Guide -@section API +@section Notes for external developers -@itemize @bullet -@item libavcodec is the library containing the codecs (both encoding and -decoding). Look at @file{libavcodec/apiexample.c} to see how to use it. - -@item libavformat is the library containing the file format handling (mux and -demux code for several formats). Look at @file{avplay.c} to use it in a -player. See @file{libavformat/output-example.c} to use it to generate -audio or video streams. -@end itemize +This document is mostly useful for internal FFmpeg developers. +External developers who need to use the API in their application should +refer to the API doxygen documentation in the public headers, and +check the examples in @file{doc/examples} and in the source code to +see how the public API is employed. -@section Integrating libav in your program - -Shared libraries should be used whenever is possible in order to reduce -the effort distributors have to pour to support programs and to ensure -only the public API is used. - -You can use Libav in your commercial program, but you must abide to the -license, LGPL or GPL depending on the specific features used, please refer -to @uref{http://libav.org/legal.html, our legal page} for a quick checklist and to -the following links for the exact text of each license: -@uref{http://git.libav.org/?p=libav.git;a=blob;f=COPYING.GPLv2, GPL version 2}, -@uref{http://git.libav.org/?p=libav.git;a=blob;f=COPYING.GPLv3, GPL version 3}, -@uref{http://git.libav.org/?p=libav.git;a=blob;f=COPYING.LGPLv2.1, LGPL version 2.1}, -@uref{http://git.libav.org/?p=libav.git;a=blob;f=COPYING.LGPLv3, LGPL version 3}. -Any modification to the source code can be suggested for inclusion. -The best way to proceed is to send your patches to the -@uref{https://lists.libav.org/mailman/listinfo/libav-devel, libav-devel} -mailing list. +You can use the FFmpeg libraries in your commercial program, but you +are encouraged to @emph{publish any patch you make}. In this case the +best way to proceed is to send your patches to the ffmpeg-devel +mailing list following the guidelines illustrated in the remainder of +this document. -@anchor{Coding Rules} -@section Coding Rules +For more detailed legal information about the use of FFmpeg in +external programs read the @file{LICENSE} file in the source tree and +consult @url{https://ffmpeg.org/legal.html}. -@subsection Code formatting conventions -The code is written in K&R C style. That means the following: +@section Contributing +There are 3 ways by which code gets into FFmpeg. @itemize @bullet -@item -The control statements are formatted by putting space between the statement -and parenthesis in the following way: -@example -for (i = 0; i < filter->input_count; i++) @{ -@end example - -@item -The case statement is always located at the same level as the switch itself: -@example -switch (link->init_state) @{ -case AVLINK_INIT: - continue; -case AVLINK_STARTINIT: - av_log(filter, AV_LOG_INFO, "circular filter chain detected"); - return 0; -@end example - -@item -Braces in function definitions are written on the new line: -@example -const char *avfilter_configuration(void) -@{ - return LIBAV_CONFIGURATION; -@} -@end example +@item Submitting patches to the main developer mailing list. + See @ref{Submitting patches} for details. +@item Directly committing changes to the main tree. +@item Committing changes to a git clone, for example on github.com or + gitorious.org. And asking us to merge these changes. +@end itemize -@item -Do not check for NULL values by comparison, @samp{if (p)} and -@samp{if (!p)} are correct; @samp{if (p == NULL)} and @samp{if (p != NULL)} -are not. +Whichever way, changes should be reviewed by the maintainer of the code +before they are committed. And they should follow the @ref{Coding Rules}. +The developer making the commit and the author are responsible for their changes +and should try to fix issues their commit causes. -@item -In case of a single-statement if, no curly braces are required: -@example -if (!pic || !picref) - goto fail; -@end example +@anchor{Coding Rules} +@section Coding Rules -@item -Do not put spaces immediately inside parentheses. @samp{if (ret)} is -a valid style; @samp{if ( ret )} is not. -@end itemize +@subsection Code formatting conventions There are the following guidelines regarding the indentation in files: @@ -107,10 +65,13 @@ rejected by the git repository. @item You should try to limit your code lines to 80 characters; however, do so if and only if this improves readability. + +@item +K&R coding style is used. @end itemize The presentation is one inspired by 'indent -i4 -kr -nut'. -The main priority in Libav is simplicity and small code size in order to +The main priority in FFmpeg is simplicity and small code size in order to minimize the bug count. @subsection Comments @@ -155,7 +116,7 @@ int myfunc(int my_parameter) @subsection C language features -Libav is programmed in the ISO C90 language with a few additional +FFmpeg is programmed in the ISO C90 language with a few additional features from ISO C99, namely: @itemize @bullet @@ -166,10 +127,10 @@ the @samp{inline} keyword; @samp{//} comments; @item -designated struct initializers (@samp{struct s x = @{ .i = 17 @};}) +designated struct initializers (@samp{struct s x = @{ .i = 17 @};}); @item -compound literals (@samp{x = (struct s) @{ 17, 23 @};}) +compound literals (@samp{x = (struct s) @{ 17, 23 @};}). @end itemize These features are supported by all compilers we care about, so we will not @@ -197,8 +158,8 @@ GCC statement expressions (@samp{(x = (@{ int y = 4; y; @})}). @subsection Naming conventions All names should be composed with underscores (_), not CamelCase. For example, @samp{avfilter_get_video_buffer} is an acceptable function name and -@samp{AVFilterGetVideo} is not. The only exception are structure -names; they should always be CamelCase. +@samp{AVFilterGetVideo} is not. The exception from this are type names, like +for example structs and enums; they should always be in CamelCase. There are the following conventions for naming variables and functions: @@ -221,8 +182,13 @@ across multiple libraries, use @code{avpriv_} as prefix, for example, @samp{avpriv_aac_parse_header}. @item -For externally visible symbols, each library has its own prefix. Check -the existing code and choose names accordingly. +Each library has its own prefix for public symbols, in addition to the +commonly used @code{av_} (@code{avformat_} for libavformat, +@code{avcodec_} for libavcodec, @code{swr_} for libswresample, etc). +Check the existing code and choose names accordingly. +Note that some symbols without these prefixes are also exported for +retro-compatibility reasons. These exceptions are declared in the +@code{lib<name>/lib<name>.v} files. @end itemize Furthermore, name space reserved for the system should not be invaded. @@ -246,10 +212,10 @@ should also be avoided if they don't make the code easier to understand. @end itemize @subsection Editor configuration -In order to configure Vim to follow Libav formatting conventions, paste +In order to configure Vim to follow FFmpeg formatting conventions, paste the following snippet into your @file{.vimrc}: @example -" Indentation rules for Libav: 4 spaces, no tabs. +" indentation rules for FFmpeg: 4 spaces, no tabs set expandtab set shiftwidth=4 set softtabstop=4 @@ -265,8 +231,8 @@ autocmd InsertEnter * match ForbiddenWhitespace /\t\|\s\+\%#\@@<!$/ @end example For Emacs, add these roughly equivalent lines to your @file{.emacs.d/init.el}: -@example -(c-add-style "libav" +@lisp +(c-add-style "ffmpeg" '("k&r" (c-basic-offset . 4) (indent-tabs-mode . nil) @@ -275,8 +241,8 @@ For Emacs, add these roughly equivalent lines to your @file{.emacs.d/init.el}: (statement-cont . (c-lineup-assignments +))) ) ) -(setq c-default-style "libav") -@end example +(setq c-default-style "ffmpeg") +@end lisp @section Development Policy @@ -291,20 +257,16 @@ a gift-style license, the @uref{http://www.gnu.org/licenses/gpl-2.0.html, GPL 2} including an "or any later version" clause is also acceptable, but LGPL is preferred. +If you add a new file, give it a proper license header. Do not copy and +paste it from a random place, use an existing file as template. @item -All the patches MUST be reviewed in the mailing list before they are -committed. - -@item -The Libav coding style should remain consistent. Changes to -conform will be suggested during the review or implemented on commit. - -@item -Patches should be generated using @code{git format-patch} or directly sent -using @code{git send-email}. -Please make sure you give the proper credit by setting the correct author -in the commit. +You must not commit code which breaks FFmpeg! (Meaning unfinished but +enabled code which breaks compilation or compiles but does not work or +breaks the regression tests) +You can commit unfinished stuff (for testing etc), but it must be disabled +(#ifdef etc) by default so it does not interfere with other developers' +work. @item The commit message should have a short first line in the form of @@ -313,24 +275,12 @@ from the body consisting of an explanation of why the change is necessary. If the commit fixes a known bug on the bug tracker, the commit message should include its bug ID. Referring to the issue on the bug tracker does not exempt you from writing an excerpt of the bug in the commit message. -If the patch is a bug fix which should be backported to stable releases, -i.e. a non-API/ABI-breaking bug fix, add @code{CC: libav-stable@@libav.org} -to the bottom of your commit message, and make sure to CC your patch to -this address, too. Some git setups will do this automatically. - -@item -Work in progress patches should be sent to the mailing list with the [WIP] -or the [RFC] tag. - -@item -Branches in public personal repos are advised as way to -work on issues collaboratively. @item -You do not have to over-test things. If it works for you and you think it -should work for others, send it to the mailing list for review. -If you have doubt about portability please state it in the submission so -people with specific hardware could test it. +You do not have to over-test things. If it works for you, and you think it +should work for others, then commit. If your code has problems +(portability, triggers compiler bugs, unusual environment etc) they will be +reported and eventually fixed. @item Do not commit unrelated changes together, split them into self-contained @@ -339,44 +289,90 @@ depend on B, then A can and should be committed first and separate from B. Keeping changes well split into self-contained parts makes reviewing and understanding them on the commit log mailing list easier. This also helps in case of debugging later on. +Also if you have doubts about splitting or not splitting, do not hesitate to +ask/discuss it on the developer mailing list. @item -Patches that change behavior of the programs (renaming options etc) or -public API or ABI should be discussed in depth and possible few days should -pass between discussion and commit. -Changes to the build system (Makefiles, configure script) which alter -the expected behavior should be considered in the same regard. +Do not change behavior of the programs (renaming options etc) or public +API or ABI without first discussing it on the ffmpeg-devel mailing list. +Do not remove functionality from the code. Just improve! + +Note: Redundant code can be removed. + +@item +Do not commit changes to the build system (Makefiles, configure script) +which change behavior, defaults etc, without asking first. The same +applies to compiler warning fixes, trivial looking fixes and to code +maintained by other developers. We usually have a reason for doing things +the way we do. Send your changes as patches to the ffmpeg-devel mailing +list, and if the code maintainers say OK, you may commit. This does not +apply to files you wrote and/or maintain. + +@item +We refuse source indentation and other cosmetic changes if they are mixed +with functional changes, such commits will be rejected and removed. Every +developer has his own indentation style, you should not change it. Of course +if you (re)write something, you can use your own style, even though we would +prefer if the indentation throughout FFmpeg was consistent (Many projects +force a given indentation style - we do not.). If you really need to make +indentation changes (try to avoid this), separate them strictly from real +changes. + +NOTE: If you had to put if()@{ .. @} over a large (> 5 lines) chunk of code, +then either do NOT change the indentation of the inner part within (do not +move it to the right)! or do so in a separate commit + +@item +Always fill out the commit log message. Describe in a few lines what you +changed and why. You can refer to mailing list postings if you fix a +particular bug. Comments such as "fixed!" or "Changed it." are unacceptable. +Recommended format: + +@example +area changed: Short 1 line description + +details describing what and why and giving references. +@end example + +@item +Make sure the author of the commit is set correctly. (see git commit --author) +If you apply a patch, send an +answer to ffmpeg-devel (or wherever you got the patch from) saying that +you applied the patch. @item When applying patches that have been discussed (at length) on the mailing list, reference the thread in the log message. @item -Subscribe to the -@uref{https://lists.libav.org/mailman/listinfo/libav-devel, libav-devel} and -@uref{https://lists.libav.org/mailman/listinfo/libav-commits, libav-commits} -mailing lists. -Bugs and possible improvements or general questions regarding commits -are discussed on libav-devel. We expect you to react if problems with -your code are uncovered. +Do NOT commit to code actively maintained by others without permission. +Send a patch to ffmpeg-devel instead. If no one answers within a reasonable +timeframe (12h for build failures and security fixes, 3 days small changes, +1 week for big patches) then commit your patch if you think it is OK. +Also note, the maintainer can simply ask for more time to review! + +@item +Subscribe to the ffmpeg-cvslog mailing list. The diffs of all commits +are sent there and reviewed by all the other developers. Bugs and possible +improvements or general questions regarding commits are discussed there. We +expect you to react if problems with your code are uncovered. @item Update the documentation if you change behavior or add features. If you are -unsure how best to do this, send an [RFC] patch to libav-devel. +unsure how best to do this, send a patch to ffmpeg-devel, the documentation +maintainer(s) will review and commit your stuff. @item -All discussions and decisions should be reported on the public developer -mailing list, so that there is a reference to them. -Other media (e.g. IRC) should be used for coordination and immediate -collaboration. +Try to keep important discussions and requests (also) on the public +developer mailing list, so that all developers can benefit from them. @item Never write to unallocated memory, never write over the end of arrays, always check values read from some untrusted source before using them -as array index or other risky things. Always use valgrind to double-check. +as array index or other risky things. @item -Remember to check if you need to bump versions for the specific libav +Remember to check if you need to bump versions for the specific libav* parts (libavutil, libavcodec, libavformat) you are changing. You need to change the version integer. Incrementing the first component means no backward compatibility to @@ -385,52 +381,59 @@ Incrementing the second component means backward compatible change (e.g. addition of a function to the public API or extension of an existing data structure). Incrementing the third component means a noteworthy binary compatible -change (e.g. encoder bug fix that matters for the decoder). +change (e.g. encoder bug fix that matters for the decoder). The third +component always starts at 100 to distinguish FFmpeg from Libav. @item -Compiler warnings indicate potential bugs or code with bad style. +Compiler warnings indicate potential bugs or code with bad style. If a type of +warning always points to correct and clean code, that warning should +be disabled, not the code changed. +Thus the remaining warnings can either be bugs or correct code. If it is a bug, the bug has to be fixed. If it is not, the code should be changed to not generate a warning unless that causes a slowdown or obfuscates the code. -If a type of warning leads to too many false positives, that warning -should be disabled, not the code changed. @item -If you add a new file, give it a proper license header. Do not copy and -paste it from a random place, use an existing file as template. +Make sure that no parts of the codebase that you maintain are missing from the +@file{MAINTAINERS} file. If something that you want to maintain is missing add it with +your name after it. +If at some point you no longer want to maintain some code, then please help in +finding a new maintainer and also don't forget to update the @file{MAINTAINERS} file. @end enumerate We think our rules are not too hard. If you have comments, contact us. +@anchor{Submitting patches} @section Submitting patches First, read the @ref{Coding Rules} above if you did not yet, in particular the rules regarding patch submission. -As stated already, please do not submit a patch which contains several -unrelated changes. +When you submit your patch, please use @code{git format-patch} or +@code{git send-email}. We cannot read other diffs :-). + +Also please do not submit a patch which contains several unrelated changes. Split it into separate, self-contained pieces. This does not mean splitting file by file. Instead, make the patch as small as possible while still keeping it as a logical unit that contains an individual change, even if it spans multiple files. This makes reviewing your patches much easier for us and greatly increases your chances of getting your patch applied. -Use the patcheck tool of Libav to check your patch. +Use the patcheck tool of FFmpeg to check your patch. The tool is located in the tools directory. -Run the @ref{Regression Tests} before submitting a patch in order to verify +Run the @ref{Regression tests} before submitting a patch in order to verify it does not cause unexpected problems. It also helps quite a bit if you tell us what the patch does (for example 'replaces lrint by lrintf'), and why (for example '*BSD isn't C99 compliant -and has no lrint()'). This kind of explanation should be the body of the -commit message. +and has no lrint()') Also please if you send several patches, send each patch as a separate mail, do not attach several unrelated patches to the same mail. Patches should be posted to the -@uref{https://lists.libav.org/mailman/listinfo/libav-devel, libav-devel} +@uref{https://lists.ffmpeg.org/mailman/listinfo/ffmpeg-devel, ffmpeg-devel} mailing list. Use @code{git send-email} when possible since it will properly send patches without requiring extra care. If you cannot, then send patches as base64-encoded attachments, so your patch is not trashed during @@ -439,8 +442,8 @@ transmission. Your patch will be reviewed on the mailing list. You will likely be asked to make some changes and are expected to send in an improved version that incorporates the requests from the review. This process may go through -several iterations. Once your patch is deemed good enough, it will be -committed to the official Libav tree. +several iterations. Once your patch is deemed good enough, some developer +will pick it up and commit it to the official FFmpeg tree. Give us a few days to react. But if some time passes without reaction, send a reminder by email. Your patch should eventually be dealt with. @@ -474,8 +477,8 @@ even if it is only a decoder? @item Did you add a rule to compile the appropriate files in the Makefile? -Remember to do this even if you are just adding a format to a file that -is already being compiled by some other rule, like a raw demuxer. +Remember to do this even if you're just adding a format to a file that is +already being compiled by some other rule, like a raw demuxer. @item Did you add an entry to the table of supported formats or codecs in @@ -502,15 +505,25 @@ Did you make sure it compiles standalone, i.e. with @enumerate @item -Does @code{make check} pass with the patch applied? +Does @code{make fate} pass with the patch applied? + +@item +Was the patch generated with git format-patch or send-email? @item -Is the patch against latest Libav git master branch? +Did you sign off your patch? (git commit -s) +See @url{http://git.kernel.org/?p=linux/kernel/git/torvalds/linux.git;a=blob_plain;f=Documentation/SubmittingPatches} for the meaning +of sign off. @item -Are you subscribed to the -@uref{https://lists.libav.org/mailman/listinfo/libav-devel, libav-devel} -mailing list? (Only list subscribers are allowed to post.) +Did you provide a clear git commit log message? + +@item +Is the patch against latest FFmpeg git master branch? + +@item +Are you subscribed to ffmpeg-devel? +(the list is subscribers only due to spam) @item Have you checked that the changes are minimal, so that the same cannot be @@ -534,6 +547,10 @@ should not crash, end in a (near) infinite loop, or allocate ridiculous amounts of memory when fed damaged data. @item +Did you test your decoder or demuxer against sample files? +Samples may be obtained at @url{https://samples.ffmpeg.org}. + +@item Does the patch not mix functional and cosmetic changes? @item @@ -553,7 +570,7 @@ If the patch fixes a bug, did you provide a verbose analysis of the bug? If the patch fixes a bug, did you provide enough information, including a sample, so the bug can be reproduced and the fix can be verified? Note please do not attach samples >100k to mails but rather provide a -URL, you can upload to ftp://upload.libav.org +URL, you can upload to ftp://upload.ffmpeg.org. @item Did you provide a verbose summary about what the patch does change? @@ -571,7 +588,7 @@ patch easily? @item If you added a new file, did you insert a license header? It should be -taken from Libav, not randomly copied and pasted from somewhere else. +taken from FFmpeg, not randomly copied and pasted from somewhere else. @item You should maintain alphabetical order in alphabetically ordered lists as @@ -582,16 +599,24 @@ Lines with similar content should be aligned vertically when doing so improves readability. @item +Consider adding a regression test for your code. + +@item +If you added YASM code please check that things still work with --disable-yasm. + +@item Make sure you check the return values of function and return appropriate -error codes. Especially memory allocation functions like @code{malloc()} +error codes. Especially memory allocation functions like @code{av_malloc()} are notoriously left unchecked, which is a serious problem. + +@item +Test your code with valgrind and or Address Sanitizer to ensure it's free +of leaks, out of array accesses, etc. @end enumerate @section Patch review process -All patches posted to the -@uref{https://lists.libav.org/mailman/listinfo/libav-devel, libav-devel} -mailing list will be reviewed, unless they contain a +All patches posted to ffmpeg-devel will be reviewed, unless they contain a clear note that the patch is not for the git master branch. Reviews and comments will be posted as replies to the patch on the mailing list. The patch submitter then has to take care of every comment, @@ -605,27 +630,46 @@ After a patch is approved it will be committed to the repository. We will review all submitted patches, but sometimes we are quite busy so especially for large patches this can take several weeks. -When resubmitting patches, if their size grew or during the review different -issues arisen please split the patch so each issue has a specific patch. +If you feel that the review process is too slow and you are willing to try to +take over maintainership of the area of code you change then just clone +git master and maintain the area of code there. We will merge each area from +where its best maintained. + +When resubmitting patches, please do not make any significant changes +not related to the comments received during review. Such patches will +be rejected. Instead, submit significant changes or new features as +separate patches. + +Everyone is welcome to review patches. Also if you are waiting for your patch +to be reviewed, please consider helping to review other patches, that is a great +way to get everyone's patches reviewed sooner. + +@anchor{Regression tests} +@section Regression tests -@anchor{Regression Tests} -@section Regression Tests +Before submitting a patch (or committing to the repository), you should at least +test that you did not break anything. -Before submitting a patch (or committing to the repository), you should at -least make sure that it does not break anything. +Running 'make fate' accomplishes this, please see @url{fate.html} for details. -If the code changed has already a test present in FATE you should run it, -otherwise it is advised to add it. +[Of course, some patches may change the results of the regression tests. In +this case, the reference results of the regression tests shall be modified +accordingly]. -Improvements to codec or demuxer might change the FATE results. Make sure -to commit the update reference with the change and to explain in the comment -why the expected result changed. +@subsection Adding files to the fate-suite dataset -Please refer to @url{fate.html}. +When there is no muxer or encoder available to generate test media for a +specific test then the media has to be included in the fate-suite. +First please make sure that the sample file is as small as possible to test the +respective decoder or demuxer sufficiently. Large files increase network +bandwidth and disk space requirements. +Once you have a working fate test and fate sample, provide in the commit +message or introductory message for the patch series that you post to +the ffmpeg-devel mailing list, a direct link to download the sample media. @subsection Visualizing Test Coverage -The Libav build system allows visualizing the test coverage in an easy +The FFmpeg build system allows visualizing the test coverage in an easy manner with the coverage tools @code{gcov}/@code{lcov}. This involves the following steps: @@ -637,7 +681,7 @@ the following steps: @item Run your test case, either manually or via FATE. This can be either the full FATE regression suite, or any arbitrary invocation of any - front-end tool provided by Libav, in any combination. + front-end tool provided by FFmpeg, in any combination. @item Run @code{make lcov} to generate coverage data in HTML format. @@ -666,11 +710,11 @@ your configure line instead. @anchor{Release process} @section Release process -Libav maintains a set of @strong{release branches}, which are the +FFmpeg maintains a set of @strong{release branches}, which are the recommended deliverable for system integrators and distributors (such as -Linux distributions, etc.). At irregular times, a @strong{release +Linux distributions, etc.). At regular times, a @strong{release manager} prepares, tests and publishes tarballs on the -@url{http://libav.org} website. +@url{https://ffmpeg.org} website. There are two kinds of releases: @@ -685,7 +729,7 @@ which are named @code{release/X}, with @code{X} being the release version number. @end enumerate -Note that we promise to our users that shared libraries from any Libav +Note that we promise to our users that shared libraries from any FFmpeg release never break programs that have been @strong{compiled} against previous versions of @strong{the same release series} in any case! @@ -693,7 +737,7 @@ However, from time to time, we do make API changes that require adaptations in applications. Such changes are only allowed in (new) major releases and require further steps such as bumping library version numbers and/or adjustments to the symbol versioning file. Please discuss such changes -on the @strong{libav-devel} mailing list in time to allow forward planning. +on the @strong{ffmpeg-devel} mailing list in time to allow forward planning. @anchor{Criteria for Point Releases} @subsection Criteria for Point Releases @@ -707,7 +751,7 @@ Fixes a security issue, preferably identified by a @strong{CVE number} issued by @url{http://cve.mitre.org/}. @item -Fixes a documented bug in @url{http://bugzilla.libav.org}. +Fixes a documented bug in @url{https://trac.ffmpeg.org}. @item Improves the included documentation. @@ -719,11 +763,6 @@ point releases of the same release branch. The order for checking the rules is (1 OR 2 OR 3) AND 4. -All Libav developers are welcome to nominate commits that they push to -@code{master} by mailing the @strong{libav-stable} mailing list. The -easiest way to do so is to include @code{CC: libav-stable@@libav.org} in -the commit message. - @subsection Release Checklist @@ -735,50 +774,38 @@ Ensure that the @file{RELEASE} file contains the version number for the upcoming release. @item -File a release tracking bug in @url{http://bugzilla.libav.org}. Make -sure that the bug has an alias named @code{ReleaseX.Y} for the -@code{X.Y} release. +Add the release at @url{https://trac.ffmpeg.org/admin/ticket/versions}. @item Announce the intent to do a release to the mailing list. @item -Reassign unresolved blocking bugs from previous release -tracking bugs to the new bug. - -@item -Review patch nominations that reach the @strong{libav-stable} -mailing list, and push patches that fulfill the stable release -criteria to the release branch. +Make sure all relevant security fixes have been backported. See +@url{https://ffmpeg.org/security.html}. @item Ensure that the FATE regression suite still passes in the release branch on at least @strong{i386} and @strong{amd64} -(cf. @ref{Regression Tests}). +(cf. @ref{Regression tests}). @item -Prepare the release tarballs in @code{xz} and @code{gz} formats, and -supplementing files that contain @code{md5} and @code{sha1} -checksums. +Prepare the release tarballs in @code{bz2} and @code{gz} formats, and +supplementing files that contain @code{gpg} signatures @item -Publish the tarballs at @url{http://libav.org/releases}. Create and -push an annotated tag in the form @code{vX}, with @code{X} +Publish the tarballs at @url{https://ffmpeg.org/releases}. Create and +push an annotated tag in the form @code{nX}, with @code{X} containing the version number. @item -Build the tarballs with the Windows binaries, and publish them at -@url{http://win32.libav.org/releases}. - -@item -Propose and send a patch to the @strong{libav-devel} mailing list +Propose and send a patch to the @strong{ffmpeg-devel} mailing list with a news entry for the website. @item Publish the news entry. @item -Send announcement to the mailing list. +Send an announcement to the mailing list. @end enumerate @bye diff --git a/doc/devices.texi b/doc/devices.texi new file mode 100644 index 0000000000..5e74a962d7 --- /dev/null +++ b/doc/devices.texi @@ -0,0 +1,25 @@ +@chapter Device Options +@c man begin DEVICE OPTIONS + +The libavdevice library provides the same interface as +libavformat. Namely, an input device is considered like a demuxer, and +an output device like a muxer, and the interface and generic device +options are the same provided by libavformat (see the ffmpeg-formats +manual). + +In addition each input or output device may support so-called private +options, which are specific for that component. + +Options may be set by specifying -@var{option} @var{value} in the +FFmpeg tools, or by setting the value explicitly in the device +@code{AVFormatContext} options or using the @file{libavutil/opt.h} API +for programmatic use. + +@c man end DEVICE OPTIONS + +@ifclear config-writeonly +@include indevs.texi +@end ifclear +@ifclear config-readonly +@include outdevs.texi +@end ifclear diff --git a/doc/doxy-wrapper.sh b/doc/doxy-wrapper.sh index d38dd0bcdd..fe0102b5bf 100755 --- a/doc/doxy-wrapper.sh +++ b/doc/doxy-wrapper.sh @@ -1,15 +1,21 @@ #!/bin/sh -SRC_PATH="${1}" +OUT_DIR="${1}" DOXYFILE="${2}" +DOXYGEN="${3}" -shift 2 +shift 3 -doxygen - <<EOF +if [ -e "VERSION" ]; then + VERSION=`cat "VERSION"` +else + VERSION=`git describe` +fi + +$DOXYGEN - <<EOF @INCLUDE = ${DOXYFILE} INPUT = $@ -EXAMPLE_PATH = ${SRC_PATH}/doc/examples -HTML_HEADER = ${SRC_PATH}/doc/doxy/header.html -HTML_FOOTER = ${SRC_PATH}/doc/doxy/footer.html -HTML_STYLESHEET = ${SRC_PATH}/doc/doxy/doxy_stylesheet.css +HTML_TIMESTAMP = NO +PROJECT_NUMBER = $VERSION +OUTPUT_DIRECTORY = $OUT_DIR EOF diff --git a/doc/doxy/doxy_stylesheet.css b/doc/doxy/doxy_stylesheet.css deleted file mode 100644 index d6dadded57..0000000000 --- a/doc/doxy/doxy_stylesheet.css +++ /dev/null @@ -1,2021 +0,0 @@ -/*! - * Bootstrap v2.1.1 - * - * Copyright 2012 Twitter, Inc - * Licensed under the Apache License v2.0 - * http://www.apache.org/licenses/LICENSE-2.0 - * - * Designed and built with all the love in the world @twitter by @mdo and @fat. - */ - -html { - font-size: 100%; - -webkit-text-size-adjust: 100%; - -ms-text-size-adjust: 100%; -} -a:focus { - outline: thin dotted #333; - outline: 5px auto -webkit-focus-ring-color; 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-} - -.list-2panes li { - width: 470px; - width: 470px; - float: left; - margin-left: 30px; - min-height: 1px; -} -/* The standard CSS for doxygen */ - -/* @group Heading Levels */ - - -dt { - font-weight: bold; -} - -div.multicol { - -moz-column-gap: 1em; - -webkit-column-gap: 1em; - -moz-column-count: 3; - -webkit-column-count: 3; -} - -p.startli, p.startdd, p.starttd { - margin-top: 2px; -} - -p.endli { - margin-bottom: 0px; -} - -p.enddd { - margin-bottom: 4px; -} - -p.endtd { - margin-bottom: 2px; -} - -/* @end */ - -caption { - font-weight: bold; -} - -span.legend { - font-size: 70%; - text-align: center; -} - -h3.version { - font-size: 90%; - text-align: center; -} - -div.qindex, div.tablisttab{ - background-color: #EBF6EB; - border: 1px solid #A3D7A3; - text-align: center; -} - -div.qindex, div.tablistpath { - width: 100%; - line-height: 140%; -} - -div.tablisttab { - margin-right: 15px; -} - -/* @group Link Styling */ - -a { - color: #3D8C3D; - font-weight: normal; - text-decoration: none; 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- /* firefox specific markup */ - -moz-box-shadow: rgba(0, 0, 0, 0.15) 5px 5px 5px; - -moz-border-radius-topright: 8px; - -moz-border-radius-topleft: 8px; - /* webkit specific markup */ - -webkit-box-shadow: 5px 5px 5px rgba(0, 0, 0, 0.15); - -webkit-border-top-right-radius: 8px; - -webkit-border-top-left-radius: 8px; - background-repeat:repeat-x; - background-color: #E2F2E2; - -} - -.memdoc, dl.reflist dd { - border-bottom: 1px solid #A8D9A8; - border-left: 1px solid #A8D9A8; - border-right: 1px solid #A8D9A8; - padding: 2px 5px; - background-color: #FBFDFB; - border-top-width: 0; - /* opera specific markup */ - border-bottom-left-radius: 8px; - border-bottom-right-radius: 8px; - box-shadow: 5px 5px 5px rgba(0, 0, 0, 0.15); - /* firefox specific markup */ - -moz-border-radius-bottomleft: 8px; - -moz-border-radius-bottomright: 8px; - -moz-box-shadow: rgba(0, 0, 0, 0.15) 5px 5px 5px; - background-image: -moz-linear-gradient(center top, #FFFFFF 0%, #FFFFFF 60%, #F7FBF7 95%, #EEF7EE); - /* webkit specific markup */ - -webkit-border-bottom-left-radius: 8px; 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- color: #2A612A; -} - -table.doxtable { - border-collapse:collapse; -} - -table.doxtable td, table.doxtable th { - border: 1px solid #2D682D; - padding: 3px 7px 2px; -} - -table.doxtable th { - background-color: #377F37; - color: #FFFFFF; - font-size: 110%; - padding-bottom: 4px; - padding-top: 5px; - text-align:left; -} - -table.fieldtable { - width: 100%; - margin-bottom: 10px; - border: 1px solid #A8D9A8; - border-spacing: 0px; - -moz-border-radius: 4px; - -webkit-border-radius: 4px; - border-radius: 4px; - -moz-box-shadow: rgba(0, 0, 0, 0.15) 2px 2px 2px; - -webkit-box-shadow: 2px 2px 2px rgba(0, 0, 0, 0.15); - box-shadow: 2px 2px 2px rgba(0, 0, 0, 0.15); -} - -.fieldtable td, .fieldtable th { - padding: 3px 7px 2px; -} - -.fieldtable td.fieldtype, .fieldtable td.fieldname { - white-space: nowrap; - border-right: 1px solid #A8D9A8; - border-bottom: 1px solid #A8D9A8; - vertical-align: top; -} - -.fieldtable td.fielddoc { - border-bottom: 1px solid #A8D9A8; - width: 100%; -} - -.fieldtable tr:last-child td { - border-bottom: none; 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- font-weight:bold; - margin-right:10px; - padding:5px; -} - -dl.citelist dd { - margin:2px 0; - padding:5px 0; -} - -@media print -{ - #top { display: none; } - #side-nav { display: none; } - #nav-path { display: none; } - body { overflow:visible; } - h1, h2, h3, h4, h5, h6 { page-break-after: avoid; } - .summary { display: none; } - .memitem { page-break-inside: avoid; } - #doc-content - { - margin-left:0 !important; - height:auto !important; - width:auto !important; - overflow:inherit; - display:inline; - } - pre.fragment - { - overflow: visible; - text-wrap: unrestricted; - white-space: -moz-pre-wrap; /* Moz */ - white-space: -pre-wrap; /* Opera 4-6 */ - white-space: -o-pre-wrap; /* Opera 7 */ - white-space: pre-wrap; /* CSS3 */ - word-wrap: break-word; /* IE 5.5+ */ - } -} - -#proj_desc { - font-size: 1.2em; -} diff --git a/doc/doxy/footer.html b/doc/doxy/footer.html deleted file mode 100644 index 101e6fe70b..0000000000 --- a/doc/doxy/footer.html +++ /dev/null @@ -1,9 +0,0 @@ - - <footer class="footer pagination-right"> - <span class="label label-info"> - Generated on $datetime for $projectname by <a href="http://www.doxygen.org/index.html">doxygen</a> $doxygenversion - </span> - </footer> -</div> -</body> -</html> diff --git a/doc/doxy/header.html b/doc/doxy/header.html deleted file mode 100644 index 312990cdbc..0000000000 --- a/doc/doxy/header.html +++ /dev/null @@ -1,16 +0,0 @@ -<!DOCTYPE html> -<html> -<head> -<meta http-equiv="Content-Type" content="text/html; charset=UTF-8"/> -<meta http-equiv="X-UA-Compatible" content="IE=9"/> -<!--BEGIN PROJECT_NAME--><title>$projectname: $title</title><!--END PROJECT_NAME--> -<!--BEGIN !PROJECT_NAME--><title>$title</title><!--END !PROJECT_NAME--> -<link href="$relpath$doxy_stylesheet.css" rel="stylesheet" type="text/css" /> -<!--Header replace --> - -</head> - -<div class="container"> - -<!--Header replace --> -<div class="menu"> diff --git a/doc/encoders.texi b/doc/encoders.texi index 315c901ecb..e9311eb432 100644 --- a/doc/encoders.texi +++ b/doc/encoders.texi @@ -1,10 +1,10 @@ @chapter Encoders @c man begin ENCODERS -Encoders are configured elements in Libav which allow the encoding of +Encoders are configured elements in FFmpeg which allow the encoding of multimedia streams. -When you configure your Libav build, all the supported native encoders +When you configure your FFmpeg build, all the supported native encoders are enabled by default. Encoders requiring an external library must be enabled manually via the corresponding @code{--enable-lib} option. You can list all available encoders using the configure option @code{--list-encoders}. @@ -14,7 +14,7 @@ You can disable all the encoders with the configure option with the options @code{--enable-encoder=@var{ENCODER}} / @code{--disable-encoder=@var{ENCODER}}. -The option @code{-encoders} of the av* tools will display the list of +The option @code{-encoders} of the ff* tools will display the list of enabled encoders. @c man end ENCODERS @@ -25,6 +25,126 @@ enabled encoders. A description of some of the currently available audio encoders follows. +@anchor{aacenc} +@section aac + +Advanced Audio Coding (AAC) encoder. + +This encoder is the default AAC encoder, natively implemented into FFmpeg. Its +quality is on par or better than libfdk_aac at the default bitrate of 128kbps. +This encoder also implements more options, profiles and samplerates than +other encoders (with only the AAC-HE profile pending to be implemented) so this +encoder has become the default and is the recommended choice. + +@subsection Options + +@table @option +@item b +Set bit rate in bits/s. Setting this automatically activates constant bit rate +(CBR) mode. If this option is unspecified it is set to 128kbps. + +@item q +Set quality for variable bit rate (VBR) mode. This option is valid only using +the @command{ffmpeg} command-line tool. For library interface users, use +@option{global_quality}. + +@item cutoff +Set cutoff frequency. If unspecified will allow the encoder to dynamically +adjust the cutoff to improve clarity on low bitrates. + +@item aac_coder +Set AAC encoder coding method. Possible values: + +@table @samp +@item twoloop +Two loop searching (TLS) method. + +This method first sets quantizers depending on band thresholds and then tries +to find an optimal combination by adding or subtracting a specific value from +all quantizers and adjusting some individual quantizer a little. +Will tune itself based on whether aac_is/aac_ms/aac_pns are enabled. +This is the default choice for a coder. + +@item anmr +Average noise to mask ratio (ANMR) trellis-based solution. + +This is an experimental coder which currently produces a lower quality, is more +unstable and is slower than the default twoloop coder but has potential. +Currently has no support for the @option{aac_is} or @option{aac_pns} options. +Not currently recommended. + +@item fast +Constant quantizer method. + +This method sets a constant quantizer for all bands. This is the fastest of all +the methods and has no rate control or support for @option{aac_is} or +@option{aac_pns}. +Not recommended. + +@end table + +@item aac_ms +Sets mid/side coding mode. The default value of auto will automatically use +M/S with bands which will benefit from such coding. Can be forced for all bands +using the value "enable", which is mainly useful for debugging or disabled using +"disable". + +@item aac_is +Sets intensity stereo coding tool usage. By default, it's enabled and will +automatically toggle IS for similar pairs of stereo bands if it's benefitial. +Can be disabled for debugging by setting the value to "disable". + +@item aac_pns +Uses perceptual noise substitution to replace low entropy high frequency bands +with imperceivable white noise during the decoding process. By default, it's +enabled, but can be disabled for debugging purposes by using "disable". + +@item aac_tns +Enables the use of a multitap FIR filter which spans through the high frequency +bands to hide quantization noise during the encoding process and is reverted +by the decoder. As well as decreasing unpleasant artifacts in the high range +this also reduces the entropy in the high bands and allows for more bits to +be used by the mid-low bands. By default it's enabled but can be disabled for +debugging by setting the option to "disable". + +@item aac_ltp +Enables the use of the long term prediction extension which increases coding +efficiency in very low bandwidth situations such as encoding of voice or +solo piano music by extending constant harmonic peaks in bands throughout +frames. This option is implied by profile:a aac_low and is incompatible with +aac_pred. Use in conjunction with @option{-ar} to decrease the samplerate. + +@item aac_pred +Enables the use of a more traditional style of prediction where the spectral +coefficients transmitted are replaced by the difference of the current +coefficients minus the previous "predicted" coefficients. In theory and sometimes +in practice this can improve quality for low to mid bitrate audio. +This option implies the aac_main profile and is incompatible with aac_ltp. + +@item profile +Sets the encoding profile, possible values: + +@table @samp +@item aac_low +The default, AAC "Low-complexity" profile. Is the most compatible and produces +decent quality. + +@item mpeg2_aac_low +Equivalent to -profile:a aac_low -aac_pns 0. PNS was introduced with the MPEG4 +specifications. + +@item aac_ltp +Long term prediction profile, is enabled by and will enable the aac_ltp option. +Introduced in MPEG4. + +@item aac_main +Main-type prediction profile, is enabled by and will enable the aac_pred option. +Introduced in MPEG2. + +If this option is unspecified it is set to @samp{aac_low}. +@end table +@end table + @section ac3 and ac3_fixed AC-3 audio encoders. @@ -369,7 +489,7 @@ is highly recommended that it be left as enabled except for testing purposes. @end table -@subheading Floating-Point-Only AC-3 Encoding Options +@subsection Floating-Point-Only AC-3 Encoding Options These options are only valid for the floating-point encoder and do not exist for the fixed-point encoder due to the corresponding features not being @@ -412,32 +532,817 @@ Selected by Encoder (default) @end table +@anchor{flac} +@section flac + +FLAC (Free Lossless Audio Codec) Encoder + +@subsection Options + +The following options are supported by FFmpeg's flac encoder. + +@table @option +@item compression_level +Sets the compression level, which chooses defaults for many other options +if they are not set explicitly. + +@item frame_size +Sets the size of the frames in samples per channel. + +@item lpc_coeff_precision +Sets the LPC coefficient precision, valid values are from 1 to 15, 15 is the +default. + +@item lpc_type +Sets the first stage LPC algorithm +@table @samp +@item none +LPC is not used + +@item fixed +fixed LPC coefficients + +@item levinson + +@item cholesky +@end table + +@item lpc_passes +Number of passes to use for Cholesky factorization during LPC analysis + +@item min_partition_order +The minimum partition order + +@item max_partition_order +The maximum partition order + +@item prediction_order_method +@table @samp +@item estimation +@item 2level +@item 4level +@item 8level +@item search +Bruteforce search +@item log +@end table + +@item ch_mode +Channel mode +@table @samp +@item auto +The mode is chosen automatically for each frame +@item indep +Chanels are independently coded +@item left_side +@item right_side +@item mid_side +@end table + +@item exact_rice_parameters +Chooses if rice parameters are calculated exactly or approximately. +if set to 1 then they are chosen exactly, which slows the code down slightly and +improves compression slightly. + +@item multi_dim_quant +Multi Dimensional Quantization. If set to 1 then a 2nd stage LPC algorithm is +applied after the first stage to finetune the coefficients. This is quite slow +and slightly improves compression. + +@end table + +@anchor{libfaac} +@section libfaac + +libfaac AAC (Advanced Audio Coding) encoder wrapper. + +This encoder is of much lower quality and is more unstable than any other AAC +encoders, so it's highly recommended to instead use other encoders, like +@ref{aacenc,,the native FFmpeg AAC encoder}. + +This encoder also requires the presence of the libfaac headers and library +during configuration. You need to explicitly configure the build with +@code{--enable-libfaac --enable-nonfree}. + +@subsection Options + +The following shared FFmpeg codec options are recognized. + +The following options are supported by the libfaac wrapper. The +@command{faac}-equivalent of the options are listed in parentheses. + +@table @option +@item b (@emph{-b}) +Set bit rate in bits/s for ABR (Average Bit Rate) mode. If the bit rate +is not explicitly specified, it is automatically set to a suitable +value depending on the selected profile. @command{faac} bitrate is +expressed in kilobits/s. + +Note that libfaac does not support CBR (Constant Bit Rate) but only +ABR (Average Bit Rate). + +If VBR mode is enabled this option is ignored. + +@item ar (@emph{-R}) +Set audio sampling rate (in Hz). + +@item ac (@emph{-c}) +Set the number of audio channels. + +@item cutoff (@emph{-C}) +Set cutoff frequency. If not specified (or explicitly set to 0) it +will use a value automatically computed by the library. Default value +is 0. + +@item profile +Set audio profile. + +The following profiles are recognized: +@table @samp +@item aac_main +Main AAC (Main) + +@item aac_low +Low Complexity AAC (LC) + +@item aac_ssr +Scalable Sample Rate (SSR) + +@item aac_ltp +Long Term Prediction (LTP) +@end table + +If not specified it is set to @samp{aac_low}. + +@item flags +qscale +Set constant quality VBR (Variable Bit Rate) mode. + +@item global_quality +Set quality in VBR mode as an integer number of lambda units. + +Only relevant when VBR mode is enabled with @code{flags +qscale}. The +value is converted to QP units by dividing it by @code{FF_QP2LAMBDA}, +and used to set the quality value used by libfaac. A reasonable range +for the option value in QP units is [10-500], the higher the value the +higher the quality. + +@item q (@emph{-q}) +Enable VBR mode when set to a non-negative value, and set constant +quality value as a double floating point value in QP units. + +The value sets the quality value used by libfaac. A reasonable range +for the option value is [10-500], the higher the value the higher the +quality. + +This option is valid only using the @command{ffmpeg} command-line +tool. For library interface users, use @option{global_quality}. +@end table + +@subsection Examples + +@itemize +@item +Use @command{ffmpeg} to convert an audio file to ABR 128 kbps AAC in an M4A (MP4) +container: +@example +ffmpeg -i input.wav -codec:a libfaac -b:a 128k -output.m4a +@end example + +@item +Use @command{ffmpeg} to convert an audio file to VBR AAC, using the +LTP AAC profile: +@example +ffmpeg -i input.wav -c:a libfaac -profile:a aac_ltp -q:a 100 output.m4a +@end example +@end itemize + +@anchor{libfdk-aac-enc} +@section libfdk_aac + +libfdk-aac AAC (Advanced Audio Coding) encoder wrapper. + +The libfdk-aac library is based on the Fraunhofer FDK AAC code from +the Android project. + +Requires the presence of the libfdk-aac headers and library during +configuration. You need to explicitly configure the build with +@code{--enable-libfdk-aac}. The library is also incompatible with GPL, +so if you allow the use of GPL, you should configure with +@code{--enable-gpl --enable-nonfree --enable-libfdk-aac}. + +This encoder is considered to produce output on par or worse at 128kbps to the +@ref{aacenc,,the native FFmpeg AAC encoder} but can often produce better +sounding audio at identical or lower bitrates and has support for the +AAC-HE profiles. + +VBR encoding, enabled through the @option{vbr} or @option{flags ++qscale} options, is experimental and only works with some +combinations of parameters. + +Support for encoding 7.1 audio is only available with libfdk-aac 0.1.3 or +higher. + +For more information see the fdk-aac project at +@url{http://sourceforge.net/p/opencore-amr/fdk-aac/}. + +@subsection Options + +The following options are mapped on the shared FFmpeg codec options. + +@table @option +@item b +Set bit rate in bits/s. If the bitrate is not explicitly specified, it +is automatically set to a suitable value depending on the selected +profile. + +In case VBR mode is enabled the option is ignored. + +@item ar +Set audio sampling rate (in Hz). + +@item channels +Set the number of audio channels. + +@item flags +qscale +Enable fixed quality, VBR (Variable Bit Rate) mode. +Note that VBR is implicitly enabled when the @option{vbr} value is +positive. + +@item cutoff +Set cutoff frequency. If not specified (or explicitly set to 0) it +will use a value automatically computed by the library. Default value +is 0. + +@item profile +Set audio profile. + +The following profiles are recognized: +@table @samp +@item aac_low +Low Complexity AAC (LC) + +@item aac_he +High Efficiency AAC (HE-AAC) + +@item aac_he_v2 +High Efficiency AAC version 2 (HE-AACv2) + +@item aac_ld +Low Delay AAC (LD) + +@item aac_eld +Enhanced Low Delay AAC (ELD) +@end table + +If not specified it is set to @samp{aac_low}. +@end table + +The following are private options of the libfdk_aac encoder. + +@table @option +@item afterburner +Enable afterburner feature if set to 1, disabled if set to 0. This +improves the quality but also the required processing power. + +Default value is 1. + +@item eld_sbr +Enable SBR (Spectral Band Replication) for ELD if set to 1, disabled +if set to 0. + +Default value is 0. + +@item signaling +Set SBR/PS signaling style. + +It can assume one of the following values: +@table @samp +@item default +choose signaling implicitly (explicit hierarchical by default, +implicit if global header is disabled) + +@item implicit +implicit backwards compatible signaling + +@item explicit_sbr +explicit SBR, implicit PS signaling + +@item explicit_hierarchical +explicit hierarchical signaling +@end table + +Default value is @samp{default}. + +@item latm +Output LATM/LOAS encapsulated data if set to 1, disabled if set to 0. + +Default value is 0. + +@item header_period +Set StreamMuxConfig and PCE repetition period (in frames) for sending +in-band configuration buffers within LATM/LOAS transport layer. + +Must be a 16-bits non-negative integer. + +Default value is 0. + +@item vbr +Set VBR mode, from 1 to 5. 1 is lowest quality (though still pretty +good) and 5 is highest quality. A value of 0 will disable VBR, and CBR +(Constant Bit Rate) is enabled. + +Currently only the @samp{aac_low} profile supports VBR encoding. + +VBR modes 1-5 correspond to roughly the following average bit rates: + +@table @samp +@item 1 +32 kbps/channel +@item 2 +40 kbps/channel +@item 3 +48-56 kbps/channel +@item 4 +64 kbps/channel +@item 5 +about 80-96 kbps/channel +@end table + +Default value is 0. +@end table + +@subsection Examples + +@itemize +@item +Use @command{ffmpeg} to convert an audio file to VBR AAC in an M4A (MP4) +container: +@example +ffmpeg -i input.wav -codec:a libfdk_aac -vbr 3 output.m4a +@end example + +@item +Use @command{ffmpeg} to convert an audio file to CBR 64k kbps AAC, using the +High-Efficiency AAC profile: +@example +ffmpeg -i input.wav -c:a libfdk_aac -profile:a aac_he -b:a 64k output.m4a +@end example +@end itemize + +@anchor{libmp3lame} +@section libmp3lame + +LAME (Lame Ain't an MP3 Encoder) MP3 encoder wrapper. + +Requires the presence of the libmp3lame headers and library during +configuration. You need to explicitly configure the build with +@code{--enable-libmp3lame}. + +See @ref{libshine} for a fixed-point MP3 encoder, although with a +lower quality. + +@subsection Options + +The following options are supported by the libmp3lame wrapper. The +@command{lame}-equivalent of the options are listed in parentheses. + +@table @option +@item b (@emph{-b}) +Set bitrate expressed in bits/s for CBR or ABR. LAME @code{bitrate} is +expressed in kilobits/s. + +@item q (@emph{-V}) +Set constant quality setting for VBR. This option is valid only +using the @command{ffmpeg} command-line tool. For library interface +users, use @option{global_quality}. + +@item compression_level (@emph{-q}) +Set algorithm quality. Valid arguments are integers in the 0-9 range, +with 0 meaning highest quality but slowest, and 9 meaning fastest +while producing the worst quality. + +@item reservoir +Enable use of bit reservoir when set to 1. Default value is 1. LAME +has this enabled by default, but can be overridden by use +@option{--nores} option. + +@item joint_stereo (@emph{-m j}) +Enable the encoder to use (on a frame by frame basis) either L/R +stereo or mid/side stereo. Default value is 1. + +@item abr (@emph{--abr}) +Enable the encoder to use ABR when set to 1. The @command{lame} +@option{--abr} sets the target bitrate, while this options only +tells FFmpeg to use ABR still relies on @option{b} to set bitrate. + +@end table + +@section libopencore-amrnb + +OpenCORE Adaptive Multi-Rate Narrowband encoder. + +Requires the presence of the libopencore-amrnb headers and library during +configuration. You need to explicitly configure the build with +@code{--enable-libopencore-amrnb --enable-version3}. + +This is a mono-only encoder. Officially it only supports 8000Hz sample rate, +but you can override it by setting @option{strict} to @samp{unofficial} or +lower. + +@subsection Options + +@table @option + +@item b +Set bitrate in bits per second. Only the following bitrates are supported, +otherwise libavcodec will round to the nearest valid bitrate. + +@table @option +@item 4750 +@item 5150 +@item 5900 +@item 6700 +@item 7400 +@item 7950 +@item 10200 +@item 12200 +@end table + +@item dtx +Allow discontinuous transmission (generate comfort noise) when set to 1. The +default value is 0 (disabled). + +@end table + +@anchor{libshine} +@section libshine + +Shine Fixed-Point MP3 encoder wrapper. + +Shine is a fixed-point MP3 encoder. It has a far better performance on +platforms without an FPU, e.g. armel CPUs, and some phones and tablets. +However, as it is more targeted on performance than quality, it is not on par +with LAME and other production-grade encoders quality-wise. Also, according to +the project's homepage, this encoder may not be free of bugs as the code was +written a long time ago and the project was dead for at least 5 years. + +This encoder only supports stereo and mono input. This is also CBR-only. + +The original project (last updated in early 2007) is at +@url{http://sourceforge.net/projects/libshine-fxp/}. We only support the +updated fork by the Savonet/Liquidsoap project at @url{https://github.com/savonet/shine}. + +Requires the presence of the libshine headers and library during +configuration. You need to explicitly configure the build with +@code{--enable-libshine}. + +See also @ref{libmp3lame}. + +@subsection Options + +The following options are supported by the libshine wrapper. The +@command{shineenc}-equivalent of the options are listed in parentheses. + +@table @option +@item b (@emph{-b}) +Set bitrate expressed in bits/s for CBR. @command{shineenc} @option{-b} option +is expressed in kilobits/s. + +@end table + +@section libtwolame + +TwoLAME MP2 encoder wrapper. + +Requires the presence of the libtwolame headers and library during +configuration. You need to explicitly configure the build with +@code{--enable-libtwolame}. + +@subsection Options + +The following options are supported by the libtwolame wrapper. The +@command{twolame}-equivalent options follow the FFmpeg ones and are in +parentheses. + +@table @option +@item b (@emph{-b}) +Set bitrate expressed in bits/s for CBR. @command{twolame} @option{b} +option is expressed in kilobits/s. Default value is 128k. + +@item q (@emph{-V}) +Set quality for experimental VBR support. Maximum value range is +from -50 to 50, useful range is from -10 to 10. The higher the +value, the better the quality. This option is valid only using the +@command{ffmpeg} command-line tool. For library interface users, +use @option{global_quality}. + +@item mode (@emph{--mode}) +Set the mode of the resulting audio. Possible values: + +@table @samp +@item auto +Choose mode automatically based on the input. This is the default. +@item stereo +Stereo +@item joint_stereo +Joint stereo +@item dual_channel +Dual channel +@item mono +Mono +@end table + +@item psymodel (@emph{--psyc-mode}) +Set psychoacoustic model to use in encoding. The argument must be +an integer between -1 and 4, inclusive. The higher the value, the +better the quality. The default value is 3. + +@item energy_levels (@emph{--energy}) +Enable energy levels extensions when set to 1. The default value is +0 (disabled). + +@item error_protection (@emph{--protect}) +Enable CRC error protection when set to 1. The default value is 0 +(disabled). + +@item copyright (@emph{--copyright}) +Set MPEG audio copyright flag when set to 1. The default value is 0 +(disabled). + +@item original (@emph{--original}) +Set MPEG audio original flag when set to 1. The default value is 0 +(disabled). + +@end table + +@section libvo-amrwbenc + +VisualOn Adaptive Multi-Rate Wideband encoder. + +Requires the presence of the libvo-amrwbenc headers and library during +configuration. You need to explicitly configure the build with +@code{--enable-libvo-amrwbenc --enable-version3}. + +This is a mono-only encoder. Officially it only supports 16000Hz sample +rate, but you can override it by setting @option{strict} to +@samp{unofficial} or lower. + +@subsection Options + +@table @option + +@item b +Set bitrate in bits/s. Only the following bitrates are supported, otherwise +libavcodec will round to the nearest valid bitrate. + +@table @samp +@item 6600 +@item 8850 +@item 12650 +@item 14250 +@item 15850 +@item 18250 +@item 19850 +@item 23050 +@item 23850 +@end table + +@item dtx +Allow discontinuous transmission (generate comfort noise) when set to 1. The +default value is 0 (disabled). + +@end table + +@section libopus + +libopus Opus Interactive Audio Codec encoder wrapper. + +Requires the presence of the libopus headers and library during +configuration. You need to explicitly configure the build with +@code{--enable-libopus}. + +@subsection Option Mapping + +Most libopus options are modelled after the @command{opusenc} utility from +opus-tools. The following is an option mapping chart describing options +supported by the libopus wrapper, and their @command{opusenc}-equivalent +in parentheses. + +@table @option + +@item b (@emph{bitrate}) +Set the bit rate in bits/s. FFmpeg's @option{b} option is +expressed in bits/s, while @command{opusenc}'s @option{bitrate} in +kilobits/s. + +@item vbr (@emph{vbr}, @emph{hard-cbr}, and @emph{cvbr}) +Set VBR mode. The FFmpeg @option{vbr} option has the following +valid arguments, with the @command{opusenc} equivalent options +in parentheses: + +@table @samp +@item off (@emph{hard-cbr}) +Use constant bit rate encoding. + +@item on (@emph{vbr}) +Use variable bit rate encoding (the default). + +@item constrained (@emph{cvbr}) +Use constrained variable bit rate encoding. +@end table + +@item compression_level (@emph{comp}) +Set encoding algorithm complexity. Valid options are integers in +the 0-10 range. 0 gives the fastest encodes but lower quality, while 10 +gives the highest quality but slowest encoding. The default is 10. + +@item frame_duration (@emph{framesize}) +Set maximum frame size, or duration of a frame in milliseconds. The +argument must be exactly the following: 2.5, 5, 10, 20, 40, 60. Smaller +frame sizes achieve lower latency but less quality at a given bitrate. +Sizes greater than 20ms are only interesting at fairly low bitrates. +The default is 20ms. + +@item packet_loss (@emph{expect-loss}) +Set expected packet loss percentage. The default is 0. + +@item application (N.A.) +Set intended application type. Valid options are listed below: + +@table @samp +@item voip +Favor improved speech intelligibility. +@item audio +Favor faithfulness to the input (the default). +@item lowdelay +Restrict to only the lowest delay modes. +@end table + +@item cutoff (N.A.) +Set cutoff bandwidth in Hz. The argument must be exactly one of the +following: 4000, 6000, 8000, 12000, or 20000, corresponding to +narrowband, mediumband, wideband, super wideband, and fullband +respectively. The default is 0 (cutoff disabled). + +@end table + +@section libvorbis + +libvorbis encoder wrapper. + +Requires the presence of the libvorbisenc headers and library during +configuration. You need to explicitly configure the build with +@code{--enable-libvorbis}. + +@subsection Options + +The following options are supported by the libvorbis wrapper. The +@command{oggenc}-equivalent of the options are listed in parentheses. + +To get a more accurate and extensive documentation of the libvorbis +options, consult the libvorbisenc's and @command{oggenc}'s documentations. +See @url{http://xiph.org/vorbis/}, +@url{http://wiki.xiph.org/Vorbis-tools}, and oggenc(1). + +@table @option +@item b (@emph{-b}) +Set bitrate expressed in bits/s for ABR. @command{oggenc} @option{-b} is +expressed in kilobits/s. + +@item q (@emph{-q}) +Set constant quality setting for VBR. The value should be a float +number in the range of -1.0 to 10.0. The higher the value, the better +the quality. The default value is @samp{3.0}. + +This option is valid only using the @command{ffmpeg} command-line tool. +For library interface users, use @option{global_quality}. + +@item cutoff (@emph{--advanced-encode-option lowpass_frequency=N}) +Set cutoff bandwidth in Hz, a value of 0 disables cutoff. @command{oggenc}'s +related option is expressed in kHz. The default value is @samp{0} (cutoff +disabled). + +@item minrate (@emph{-m}) +Set minimum bitrate expressed in bits/s. @command{oggenc} @option{-m} is +expressed in kilobits/s. + +@item maxrate (@emph{-M}) +Set maximum bitrate expressed in bits/s. @command{oggenc} @option{-M} is +expressed in kilobits/s. This only has effect on ABR mode. + +@item iblock (@emph{--advanced-encode-option impulse_noisetune=N}) +Set noise floor bias for impulse blocks. The value is a float number from +-15.0 to 0.0. A negative bias instructs the encoder to pay special attention +to the crispness of transients in the encoded audio. The tradeoff for better +transient response is a higher bitrate. + +@end table + +@anchor{libwavpack} @section libwavpack A wrapper providing WavPack encoding through libwavpack. Only lossless mode using 32-bit integer samples is supported currently. -The @option{compression_level} option can be used to control speed vs. -compression tradeoff, with the values mapped to libwavpack as follows: + +Requires the presence of the libwavpack headers and library during +configuration. You need to explicitly configure the build with +@code{--enable-libwavpack}. + +Note that a libavcodec-native encoder for the WavPack codec exists so users can +encode audios with this codec without using this encoder. See @ref{wavpackenc}. + +@subsection Options + +@command{wavpack} command line utility's corresponding options are listed in +parentheses, if any. @table @option +@item frame_size (@emph{--blocksize}) +Default is 32768. -@item 0 -Fast mode - corresponding to the wavpack @option{-f} option. +@item compression_level +Set speed vs. compression tradeoff. Acceptable arguments are listed below: + +@table @samp +@item 0 (@emph{-f}) +Fast mode. @item 1 Normal (default) settings. -@item 2 -High quality - corresponding to the wavpack @option{-h} option. +@item 2 (@emph{-h}) +High quality. -@item 3 -Very high quality - corresponding to the wavpack @option{-hh} option. +@item 3 (@emph{-hh}) +Very high quality. + +@item 4-8 (@emph{-hh -x}@var{EXTRAPROC}) +Same as @samp{3}, but with extra processing enabled. + +@samp{4} is the same as @option{-x2} and @samp{8} is the same as @option{-x6}. + +@end table +@end table + +@anchor{wavpackenc} +@section wavpack + +WavPack lossless audio encoder. + +This is a libavcodec-native WavPack encoder. There is also an encoder based on +libwavpack, but there is virtually no reason to use that encoder. + +See also @ref{libwavpack}. + +@subsection Options + +The equivalent options for @command{wavpack} command line utility are listed in +parentheses. + +@subsubsection Shared options + +The following shared options are effective for this encoder. Only special notes +about this particular encoder will be documented here. For the general meaning +of the options, see @ref{codec-options,,the Codec Options chapter}. + +@table @option +@item frame_size (@emph{--blocksize}) +For this encoder, the range for this option is between 128 and 131072. Default +is automatically decided based on sample rate and number of channel. + +For the complete formula of calculating default, see +@file{libavcodec/wavpackenc.c}. + +@item compression_level (@emph{-f}, @emph{-h}, @emph{-hh}, and @emph{-x}) +This option's syntax is consistent with @ref{libwavpack}'s. +@end table + +@subsubsection Private options + +@table @option +@item joint_stereo (@emph{-j}) +Set whether to enable joint stereo. Valid values are: + +@table @samp +@item on (@emph{1}) +Force mid/side audio encoding. +@item off (@emph{0}) +Force left/right audio encoding. +@item auto +Let the encoder decide automatically. +@end table -@item 4-8 -Same as 3, but with extra processing enabled - corresponding to the wavpack -@option{-x} option. I.e. 4 is the same as @option{-x2} and 8 is the same as -@option{-x6}. +@item optimize_mono +Set whether to enable optimization for mono. This option is only effective for +non-mono streams. Available values: + +@table @samp +@item on +enabled +@item off +disabled +@end table @end table @@ -446,6 +1351,333 @@ Same as 3, but with extra processing enabled - corresponding to the wavpack @chapter Video Encoders @c man begin VIDEO ENCODERS +A description of some of the currently available video encoders +follows. + +@section libopenh264 + +Cisco libopenh264 H.264/MPEG-4 AVC encoder wrapper. + +This encoder requires the presence of the libopenh264 headers and +library during configuration. You need to explicitly configure the +build with @code{--enable-libopenh264}. The library is detected using +@command{pkg-config}. + +For more information about the library see +@url{http://www.openh264.org}. + +@subsection Options + +The following FFmpeg global options affect the configurations of the +libopenh264 encoder. + +@table @option +@item b +Set the bitrate (as a number of bits per second). + +@item g +Set the GOP size. + +@item maxrate +Set the max bitrate (as a number of bits per second). + +@item flags +global_header +Set global header in the bitstream. + +@item slices +Set the number of slices, used in parallelized encoding. Default value +is 0. This is only used when @option{slice_mode} is set to +@samp{fixed}. + +@item slice_mode +Set slice mode. Can assume one of the follwing possible values: + +@table @samp +@item fixed +a fixed number of slices +@item rowmb +one slice per row of macroblocks +@item auto +automatic number of slices according to number of threads +@item dyn +dynamic slicing +@end table + +Default value is @samp{auto}. + +@item loopfilter +Enable loop filter, if set to 1 (automatically enabled). To disable +set a value of 0. + +@item profile +Set profile restrictions. If set to the value of @samp{main} enable +CABAC (set the @code{SEncParamExt.iEntropyCodingModeFlag} flag to 1). + +@item max_nal_size +Set maximum NAL size in bytes. + +@item allow_skip_frames +Allow skipping frames to hit the target bitrate if set to 1. +@end table + +@section jpeg2000 + +The native jpeg 2000 encoder is lossy by default, the @code{-q:v} +option can be used to set the encoding quality. Lossless encoding +can be selected with @code{-pred 1}. + +@subsection Options + +@table @option +@item format +Can be set to either @code{j2k} or @code{jp2} (the default) that +makes it possible to store non-rgb pix_fmts. + +@end table + +@section snow + +@subsection Options + +@table @option +@item iterative_dia_size +dia size for the iterative motion estimation +@end table + +@section libtheora + +libtheora Theora encoder wrapper. + +Requires the presence of the libtheora headers and library during +configuration. You need to explicitly configure the build with +@code{--enable-libtheora}. + +For more information about the libtheora project see +@url{http://www.theora.org/}. + +@subsection Options + +The following global options are mapped to internal libtheora options +which affect the quality and the bitrate of the encoded stream. + +@table @option +@item b +Set the video bitrate in bit/s for CBR (Constant Bit Rate) mode. In +case VBR (Variable Bit Rate) mode is enabled this option is ignored. + +@item flags +Used to enable constant quality mode (VBR) encoding through the +@option{qscale} flag, and to enable the @code{pass1} and @code{pass2} +modes. + +@item g +Set the GOP size. + +@item global_quality +Set the global quality as an integer in lambda units. + +Only relevant when VBR mode is enabled with @code{flags +qscale}. The +value is converted to QP units by dividing it by @code{FF_QP2LAMBDA}, +clipped in the [0 - 10] range, and then multiplied by 6.3 to get a +value in the native libtheora range [0-63]. A higher value corresponds +to a higher quality. + +@item q +Enable VBR mode when set to a non-negative value, and set constant +quality value as a double floating point value in QP units. + +The value is clipped in the [0-10] range, and then multiplied by 6.3 +to get a value in the native libtheora range [0-63]. + +This option is valid only using the @command{ffmpeg} command-line +tool. For library interface users, use @option{global_quality}. +@end table + +@subsection Examples + +@itemize +@item +Set maximum constant quality (VBR) encoding with @command{ffmpeg}: +@example +ffmpeg -i INPUT -codec:v libtheora -q:v 10 OUTPUT.ogg +@end example + +@item +Use @command{ffmpeg} to convert a CBR 1000 kbps Theora video stream: +@example +ffmpeg -i INPUT -codec:v libtheora -b:v 1000k OUTPUT.ogg +@end example +@end itemize + +@section libvpx + +VP8/VP9 format supported through libvpx. + +Requires the presence of the libvpx headers and library during configuration. +You need to explicitly configure the build with @code{--enable-libvpx}. + +@subsection Options + +The following options are supported by the libvpx wrapper. The +@command{vpxenc}-equivalent options or values are listed in parentheses +for easy migration. + +To reduce the duplication of documentation, only the private options +and some others requiring special attention are documented here. For +the documentation of the undocumented generic options, see +@ref{codec-options,,the Codec Options chapter}. + +To get more documentation of the libvpx options, invoke the command +@command{ffmpeg -h encoder=libvpx}, @command{ffmpeg -h encoder=libvpx-vp9} or +@command{vpxenc --help}. Further information is available in the libvpx API +documentation. + +@table @option + +@item b (@emph{target-bitrate}) +Set bitrate in bits/s. Note that FFmpeg's @option{b} option is +expressed in bits/s, while @command{vpxenc}'s @option{target-bitrate} is in +kilobits/s. + +@item g (@emph{kf-max-dist}) + +@item keyint_min (@emph{kf-min-dist}) + +@item qmin (@emph{min-q}) + +@item qmax (@emph{max-q}) + +@item bufsize (@emph{buf-sz}, @emph{buf-optimal-sz}) +Set ratecontrol buffer size (in bits). Note @command{vpxenc}'s options are +specified in milliseconds, the libvpx wrapper converts this value as follows: +@code{buf-sz = bufsize * 1000 / bitrate}, +@code{buf-optimal-sz = bufsize * 1000 / bitrate * 5 / 6}. + +@item rc_init_occupancy (@emph{buf-initial-sz}) +Set number of bits which should be loaded into the rc buffer before decoding +starts. Note @command{vpxenc}'s option is specified in milliseconds, the libvpx +wrapper converts this value as follows: +@code{rc_init_occupancy * 1000 / bitrate}. + +@item undershoot-pct +Set datarate undershoot (min) percentage of the target bitrate. + +@item overshoot-pct +Set datarate overshoot (max) percentage of the target bitrate. + +@item skip_threshold (@emph{drop-frame}) + +@item qcomp (@emph{bias-pct}) + +@item maxrate (@emph{maxsection-pct}) +Set GOP max bitrate in bits/s. Note @command{vpxenc}'s option is specified as a +percentage of the target bitrate, the libvpx wrapper converts this value as +follows: @code{(maxrate * 100 / bitrate)}. + +@item minrate (@emph{minsection-pct}) +Set GOP min bitrate in bits/s. Note @command{vpxenc}'s option is specified as a +percentage of the target bitrate, the libvpx wrapper converts this value as +follows: @code{(minrate * 100 / bitrate)}. + +@item minrate, maxrate, b @emph{end-usage=cbr} +@code{(minrate == maxrate == bitrate)}. + +@item crf (@emph{end-usage=cq}, @emph{cq-level}) + +@item tune (@emph{tune}) +@table @samp +@item psnr (@emph{psnr}) +@item ssim (@emph{ssim}) +@end table + +@item quality, deadline (@emph{deadline}) +@table @samp +@item best +Use best quality deadline. Poorly named and quite slow, this option should be +avoided as it may give worse quality output than good. +@item good +Use good quality deadline. This is a good trade-off between speed and quality +when used with the @option{cpu-used} option. +@item realtime +Use realtime quality deadline. +@end table + +@item speed, cpu-used (@emph{cpu-used}) +Set quality/speed ratio modifier. Higher values speed up the encode at the cost +of quality. + +@item nr (@emph{noise-sensitivity}) + +@item static-thresh +Set a change threshold on blocks below which they will be skipped by the +encoder. + +@item slices (@emph{token-parts}) +Note that FFmpeg's @option{slices} option gives the total number of partitions, +while @command{vpxenc}'s @option{token-parts} is given as +@code{log2(partitions)}. + +@item max-intra-rate +Set maximum I-frame bitrate as a percentage of the target bitrate. A value of 0 +means unlimited. + +@item force_key_frames +@code{VPX_EFLAG_FORCE_KF} + +@item Alternate reference frame related +@table @option +@item auto-alt-ref +Enable use of alternate reference frames (2-pass only). +@item arnr-max-frames +Set altref noise reduction max frame count. +@item arnr-type +Set altref noise reduction filter type: backward, forward, centered. +@item arnr-strength +Set altref noise reduction filter strength. +@item rc-lookahead, lag-in-frames (@emph{lag-in-frames}) +Set number of frames to look ahead for frametype and ratecontrol. +@end table + +@item error-resilient +Enable error resiliency features. + +@item VP9-specific options +@table @option +@item lossless +Enable lossless mode. +@item tile-columns +Set number of tile columns to use. Note this is given as +@code{log2(tile_columns)}. For example, 8 tile columns would be requested by +setting the @option{tile-columns} option to 3. +@item tile-rows +Set number of tile rows to use. Note this is given as @code{log2(tile_rows)}. +For example, 4 tile rows would be requested by setting the @option{tile-rows} +option to 2. +@item frame-parallel +Enable frame parallel decodability features. +@item aq-mode +Set adaptive quantization mode (0: off (default), 1: variance 2: complexity, 3: +cyclic refresh). +@item colorspace @emph{color-space} +Set input color space. The VP9 bitstream supports signaling the following +colorspaces: +@table @option +@item @samp{rgb} @emph{sRGB} +@item @samp{bt709} @emph{bt709} +@item @samp{unspecified} @emph{unknown} +@item @samp{bt470bg} @emph{bt601} +@item @samp{smpte170m} @emph{smpte170} +@item @samp{smpte240m} @emph{smpte240} +@item @samp{bt2020_ncl} @emph{bt2020} +@end table +@end table + +@end table + +For more information about libvpx see: +@url{http://www.webmproject.org/} + + @section libwebp libwebp WebP Image encoder wrapper @@ -505,6 +1737,525 @@ Small-sized colorful images Text-like @end table +@end table + +@section libx264, libx264rgb + +x264 H.264/MPEG-4 AVC encoder wrapper. + +This encoder requires the presence of the libx264 headers and library +during configuration. You need to explicitly configure the build with +@code{--enable-libx264}. + +libx264 supports an impressive number of features, including 8x8 and +4x4 adaptive spatial transform, adaptive B-frame placement, CAVLC/CABAC +entropy coding, interlacing (MBAFF), lossless mode, psy optimizations +for detail retention (adaptive quantization, psy-RD, psy-trellis). + +Many libx264 encoder options are mapped to FFmpeg global codec +options, while unique encoder options are provided through private +options. Additionally the @option{x264opts} and @option{x264-params} +private options allows one to pass a list of key=value tuples as accepted +by the libx264 @code{x264_param_parse} function. + +The x264 project website is at +@url{http://www.videolan.org/developers/x264.html}. + +The libx264rgb encoder is the same as libx264, except it accepts packed RGB +pixel formats as input instead of YUV. + +@subsection Supported Pixel Formats + +x264 supports 8- to 10-bit color spaces. The exact bit depth is controlled at +x264's configure time. FFmpeg only supports one bit depth in one particular +build. In other words, it is not possible to build one FFmpeg with multiple +versions of x264 with different bit depths. + +@subsection Options + +The following options are supported by the libx264 wrapper. The +@command{x264}-equivalent options or values are listed in parentheses +for easy migration. + +To reduce the duplication of documentation, only the private options +and some others requiring special attention are documented here. For +the documentation of the undocumented generic options, see +@ref{codec-options,,the Codec Options chapter}. + +To get a more accurate and extensive documentation of the libx264 +options, invoke the command @command{x264 --full-help} or consult +the libx264 documentation. + +@table @option +@item b (@emph{bitrate}) +Set bitrate in bits/s. Note that FFmpeg's @option{b} option is +expressed in bits/s, while @command{x264}'s @option{bitrate} is in +kilobits/s. + +@item bf (@emph{bframes}) + +@item g (@emph{keyint}) + +@item qmin (@emph{qpmin}) +Minimum quantizer scale. + +@item qmax (@emph{qpmax}) +Maximum quantizer scale. + +@item qdiff (@emph{qpstep}) +Maximum difference between quantizer scales. + +@item qblur (@emph{qblur}) +Quantizer curve blur + +@item qcomp (@emph{qcomp}) +Quantizer curve compression factor + +@item refs (@emph{ref}) +Number of reference frames each P-frame can use. The range is from @var{0-16}. + +@item sc_threshold (@emph{scenecut}) +Sets the threshold for the scene change detection. + +@item trellis (@emph{trellis}) +Performs Trellis quantization to increase efficiency. Enabled by default. + +@item nr (@emph{nr}) + +@item me_range (@emph{merange}) +Maximum range of the motion search in pixels. + +@item me_method (@emph{me}) +Set motion estimation method. Possible values in the decreasing order +of speed: + +@table @samp +@item dia (@emph{dia}) +@item epzs (@emph{dia}) +Diamond search with radius 1 (fastest). @samp{epzs} is an alias for +@samp{dia}. +@item hex (@emph{hex}) +Hexagonal search with radius 2. +@item umh (@emph{umh}) +Uneven multi-hexagon search. +@item esa (@emph{esa}) +Exhaustive search. +@item tesa (@emph{tesa}) +Hadamard exhaustive search (slowest). +@end table + +@item subq (@emph{subme}) +Sub-pixel motion estimation method. + +@item b_strategy (@emph{b-adapt}) +Adaptive B-frame placement decision algorithm. Use only on first-pass. + +@item keyint_min (@emph{min-keyint}) +Minimum GOP size. + +@item coder +Set entropy encoder. Possible values: + +@table @samp +@item ac +Enable CABAC. + +@item vlc +Enable CAVLC and disable CABAC. It generates the same effect as +@command{x264}'s @option{--no-cabac} option. +@end table + +@item cmp +Set full pixel motion estimation comparation algorithm. Possible values: + +@table @samp +@item chroma +Enable chroma in motion estimation. + +@item sad +Ignore chroma in motion estimation. It generates the same effect as +@command{x264}'s @option{--no-chroma-me} option. +@end table + +@item threads (@emph{threads}) +Number of encoding threads. + +@item thread_type +Set multithreading technique. Possible values: + +@table @samp +@item slice +Slice-based multithreading. It generates the same effect as +@command{x264}'s @option{--sliced-threads} option. +@item frame +Frame-based multithreading. +@end table + +@item flags +Set encoding flags. It can be used to disable closed GOP and enable +open GOP by setting it to @code{-cgop}. The result is similar to +the behavior of @command{x264}'s @option{--open-gop} option. + +@item rc_init_occupancy (@emph{vbv-init}) + +@item preset (@emph{preset}) +Set the encoding preset. + +@item tune (@emph{tune}) +Set tuning of the encoding params. + +@item profile (@emph{profile}) +Set profile restrictions. + +@item fastfirstpass +Enable fast settings when encoding first pass, when set to 1. When set +to 0, it has the same effect of @command{x264}'s +@option{--slow-firstpass} option. + +@item crf (@emph{crf}) +Set the quality for constant quality mode. + +@item crf_max (@emph{crf-max}) +In CRF mode, prevents VBV from lowering quality beyond this point. + +@item qp (@emph{qp}) +Set constant quantization rate control method parameter. + +@item aq-mode (@emph{aq-mode}) +Set AQ method. Possible values: + +@table @samp +@item none (@emph{0}) +Disabled. + +@item variance (@emph{1}) +Variance AQ (complexity mask). + +@item autovariance (@emph{2}) +Auto-variance AQ (experimental). +@end table + +@item aq-strength (@emph{aq-strength}) +Set AQ strength, reduce blocking and blurring in flat and textured areas. + +@item psy +Use psychovisual optimizations when set to 1. When set to 0, it has the +same effect as @command{x264}'s @option{--no-psy} option. + +@item psy-rd (@emph{psy-rd}) +Set strength of psychovisual optimization, in +@var{psy-rd}:@var{psy-trellis} format. + +@item rc-lookahead (@emph{rc-lookahead}) +Set number of frames to look ahead for frametype and ratecontrol. + +@item weightb +Enable weighted prediction for B-frames when set to 1. When set to 0, +it has the same effect as @command{x264}'s @option{--no-weightb} option. + +@item weightp (@emph{weightp}) +Set weighted prediction method for P-frames. Possible values: + +@table @samp +@item none (@emph{0}) +Disabled +@item simple (@emph{1}) +Enable only weighted refs +@item smart (@emph{2}) +Enable both weighted refs and duplicates +@end table + +@item ssim (@emph{ssim}) +Enable calculation and printing SSIM stats after the encoding. + +@item intra-refresh (@emph{intra-refresh}) +Enable the use of Periodic Intra Refresh instead of IDR frames when set +to 1. + +@item avcintra-class (@emph{class}) +Configure the encoder to generate AVC-Intra. +Valid values are 50,100 and 200 + +@item bluray-compat (@emph{bluray-compat}) +Configure the encoder to be compatible with the bluray standard. +It is a shorthand for setting "bluray-compat=1 force-cfr=1". + +@item b-bias (@emph{b-bias}) +Set the influence on how often B-frames are used. + +@item b-pyramid (@emph{b-pyramid}) +Set method for keeping of some B-frames as references. Possible values: + +@table @samp +@item none (@emph{none}) +Disabled. +@item strict (@emph{strict}) +Strictly hierarchical pyramid. +@item normal (@emph{normal}) +Non-strict (not Blu-ray compatible). +@end table + +@item mixed-refs +Enable the use of one reference per partition, as opposed to one +reference per macroblock when set to 1. When set to 0, it has the +same effect as @command{x264}'s @option{--no-mixed-refs} option. + +@item 8x8dct +Enable adaptive spatial transform (high profile 8x8 transform) +when set to 1. When set to 0, it has the same effect as +@command{x264}'s @option{--no-8x8dct} option. + +@item fast-pskip +Enable early SKIP detection on P-frames when set to 1. When set +to 0, it has the same effect as @command{x264}'s +@option{--no-fast-pskip} option. + +@item aud (@emph{aud}) +Enable use of access unit delimiters when set to 1. + +@item mbtree +Enable use macroblock tree ratecontrol when set to 1. When set +to 0, it has the same effect as @command{x264}'s +@option{--no-mbtree} option. + +@item deblock (@emph{deblock}) +Set loop filter parameters, in @var{alpha}:@var{beta} form. + +@item cplxblur (@emph{cplxblur}) +Set fluctuations reduction in QP (before curve compression). + +@item partitions (@emph{partitions}) +Set partitions to consider as a comma-separated list of. Possible +values in the list: + +@table @samp +@item p8x8 +8x8 P-frame partition. +@item p4x4 +4x4 P-frame partition. +@item b8x8 +4x4 B-frame partition. +@item i8x8 +8x8 I-frame partition. +@item i4x4 +4x4 I-frame partition. +(Enabling @samp{p4x4} requires @samp{p8x8} to be enabled. Enabling +@samp{i8x8} requires adaptive spatial transform (@option{8x8dct} +option) to be enabled.) +@item none (@emph{none}) +Do not consider any partitions. +@item all (@emph{all}) +Consider every partition. +@end table + +@item direct-pred (@emph{direct}) +Set direct MV prediction mode. Possible values: + +@table @samp +@item none (@emph{none}) +Disable MV prediction. +@item spatial (@emph{spatial}) +Enable spatial predicting. +@item temporal (@emph{temporal}) +Enable temporal predicting. +@item auto (@emph{auto}) +Automatically decided. +@end table + +@item slice-max-size (@emph{slice-max-size}) +Set the limit of the size of each slice in bytes. If not specified +but RTP payload size (@option{ps}) is specified, that is used. + +@item stats (@emph{stats}) +Set the file name for multi-pass stats. + +@item nal-hrd (@emph{nal-hrd}) +Set signal HRD information (requires @option{vbv-bufsize} to be set). +Possible values: + +@table @samp +@item none (@emph{none}) +Disable HRD information signaling. +@item vbr (@emph{vbr}) +Variable bit rate. +@item cbr (@emph{cbr}) +Constant bit rate (not allowed in MP4 container). +@end table + +@item x264opts (N.A.) +Set any x264 option, see @command{x264 --fullhelp} for a list. + +Argument is a list of @var{key}=@var{value} couples separated by +":". In @var{filter} and @var{psy-rd} options that use ":" as a separator +themselves, use "," instead. They accept it as well since long ago but this +is kept undocumented for some reason. + +For example to specify libx264 encoding options with @command{ffmpeg}: +@example +ffmpeg -i foo.mpg -vcodec libx264 -x264opts keyint=123:min-keyint=20 -an out.mkv +@end example + +@item a53cc @var{boolean} +Import closed captions (which must be ATSC compatible format) into output. +Only the mpeg2 and h264 decoders provide these. Default is 0 (off). + +@item x264-params (N.A.) +Override the x264 configuration using a :-separated list of key=value +parameters. + +This option is functionally the same as the @option{x264opts}, but is +duplicated for compatibility with the Libav fork. + +For example to specify libx264 encoding options with @command{ffmpeg}: +@example +ffmpeg -i INPUT -c:v libx264 -x264-params level=30:bframes=0:weightp=0:\ +cabac=0:ref=1:vbv-maxrate=768:vbv-bufsize=2000:analyse=all:me=umh:\ +no-fast-pskip=1:subq=6:8x8dct=0:trellis=0 OUTPUT +@end example +@end table + +Encoding ffpresets for common usages are provided so they can be used with the +general presets system (e.g. passing the @option{pre} option). + +@section libx265 + +x265 H.265/HEVC encoder wrapper. + +This encoder requires the presence of the libx265 headers and library +during configuration. You need to explicitly configure the build with +@option{--enable-libx265}. + +@subsection Options + +@table @option +@item preset +Set the x265 preset. + +@item tune +Set the x265 tune parameter. + +@item x265-params +Set x265 options using a list of @var{key}=@var{value} couples separated +by ":". See @command{x265 --help} for a list of options. + +For example to specify libx265 encoding options with @option{-x265-params}: + +@example +ffmpeg -i input -c:v libx265 -x265-params crf=26:psy-rd=1 output.mp4 +@end example +@end table + +@section libxvid + +Xvid MPEG-4 Part 2 encoder wrapper. + +This encoder requires the presence of the libxvidcore headers and library +during configuration. You need to explicitly configure the build with +@code{--enable-libxvid --enable-gpl}. + +The native @code{mpeg4} encoder supports the MPEG-4 Part 2 format, so +users can encode to this format without this library. + +@subsection Options + +The following options are supported by the libxvid wrapper. Some of +the following options are listed but are not documented, and +correspond to shared codec options. See @ref{codec-options,,the Codec +Options chapter} for their documentation. The other shared options +which are not listed have no effect for the libxvid encoder. + +@table @option +@item b + +@item g + +@item qmin + +@item qmax + +@item mpeg_quant + +@item threads + +@item bf + +@item b_qfactor + +@item b_qoffset + +@item flags +Set specific encoding flags. Possible values: + +@table @samp + +@item mv4 +Use four motion vector by macroblock. + +@item aic +Enable high quality AC prediction. + +@item gray +Only encode grayscale. + +@item gmc +Enable the use of global motion compensation (GMC). + +@item qpel +Enable quarter-pixel motion compensation. + +@item cgop +Enable closed GOP. + +@item global_header +Place global headers in extradata instead of every keyframe. + +@end table + +@item trellis + +@item me_method +Set motion estimation method. Possible values in decreasing order of +speed and increasing order of quality: + +@table @samp +@item zero +Use no motion estimation (default). + +@item phods +@item x1 +@item log +Enable advanced diamond zonal search for 16x16 blocks and half-pixel +refinement for 16x16 blocks. @samp{x1} and @samp{log} are aliases for +@samp{phods}. + +@item epzs +Enable all of the things described above, plus advanced diamond zonal +search for 8x8 blocks, half-pixel refinement for 8x8 blocks, and motion +estimation on chroma planes. + +@item full +Enable all of the things described above, plus extended 16x16 and 8x8 +blocks search. +@end table + +@item mbd +Set macroblock decision algorithm. Possible values in the increasing +order of quality: + +@table @samp +@item simple +Use macroblock comparing function algorithm (default). + +@item bits +Enable rate distortion-based half pixel and quarter pixel refinement for +16x16 blocks. + +@item rd +Enable all of the things described above, plus rate distortion-based +half pixel and quarter pixel refinement for 8x8 blocks, and rate +distortion-based search using square pattern. +@end table + @item lumi_aq Enable lumi masking adaptive quantization when set to 1. Default is 0 (disabled). @@ -557,203 +2308,51 @@ fastest. @end table -@section libx264 - -x264 H.264/MPEG-4 AVC encoder wrapper - -x264 supports an impressive number of features, including 8x8 and 4x4 adaptive -spatial transform, adaptive B-frame placement, CAVLC/CABAC entropy coding, -interlacing (MBAFF), lossless mode, psy optimizations for detail retention -(adaptive quantization, psy-RD, psy-trellis). - -The Libav wrapper provides a mapping for most of them using global options -that match those of the encoders and provides private options for the unique -encoder options. Additionally an expert override is provided to directly pass -a list of key=value tuples as accepted by x264_param_parse. - -@subsection Option Mapping - -The following options are supported by the x264 wrapper, the x264-equivalent -options follow the Libav ones. - -@multitable { } { } { } -@item b @tab bitrate -@tab Libav @code{b} option is expressed in bits/s, x264 @code{bitrate} in kilobits/s. -@item bf @tab bframes -@tab Maximum number of B-frames. -@item g @tab keyint -@tab Maximum GOP size. -@item qmin @tab qpmin -@tab Minimum quantizer scale. -@item qmax @tab qpmax -@tab Maximum quantizer scale. -@item qdiff @tab qpstep -@tab Maximum difference between quantizer scales. -@item qblur @tab qblur -@tab Quantizer curve blur -@item qcomp @tab qcomp -@tab Quantizer curve compression factor -@item refs @tab ref -@tab Number of reference frames each P-frame can use. The range is from @var{0-16}. -@item sc_threshold @tab scenecut -@tab Sets the threshold for the scene change detection. -@item trellis @tab trellis -@tab Performs Trellis quantization to increase efficiency. Enabled by default. -@item nr @tab nr -@tab Noise reduction. -@item me_range @tab merange -@tab Maximum range of the motion search in pixels. -@item subq @tab subme -@tab Sub-pixel motion estimation method. -@item b_strategy @tab b-adapt -@tab Adaptive B-frame placement decision algorithm. Use only on first-pass. -@item keyint_min @tab min-keyint -@tab Minimum GOP size. -@item coder @tab cabac -@tab Set coder to @code{ac} to use CABAC. -@item cmp @tab chroma-me -@tab Set to @code{chroma} to use chroma motion estimation. -@item threads @tab threads -@tab Number of encoding threads. -@item thread_type @tab sliced_threads -@tab Set to @code{slice} to use sliced threading instead of frame threading. -@item flags -cgop @tab open-gop -@tab Set @code{-cgop} to use recovery points to close GOPs. -@item rc_init_occupancy @tab vbv-init -@tab Initial buffer occupancy. -@end multitable - -@subsection Private Options -@table @option -@item -preset @var{string} -Set the encoding preset (cf. x264 --fullhelp). -@item -tune @var{string} -Tune the encoding params (cf. x264 --fullhelp). -@item -profile @var{string} -Set profile restrictions (cf. x264 --fullhelp). -@item -fastfirstpass @var{integer} -Use fast settings when encoding first pass. -@item -crf @var{float} -Select the quality for constant quality mode. -@item -crf_max @var{float} -In CRF mode, prevents VBV from lowering quality beyond this point. -@item -qp @var{integer} -Constant quantization parameter rate control method. -@item -aq-mode @var{integer} -AQ method - -Possible values: -@table @samp -@item none - -@item variance -Variance AQ (complexity mask). -@item autovariance -Auto-variance AQ (experimental). -@end table -@item -aq-strength @var{float} -AQ strength, reduces blocking and blurring in flat and textured areas. -@item -psy @var{integer} -Use psychovisual optimizations. -@item -psy-rd @var{string} -Strength of psychovisual optimization, in <psy-rd>:<psy-trellis> format. -@item -rc-lookahead @var{integer} -Number of frames to look ahead for frametype and ratecontrol. -@item -weightb @var{integer} -Weighted prediction for B-frames. -@item -weightp @var{integer} -Weighted prediction analysis method. - -Possible values: -@table @samp -@item none - -@item simple +@section mpeg2 -@item smart +MPEG-2 video encoder. -@end table -@item -ssim @var{integer} -Calculate and print SSIM stats. -@item -intra-refresh @var{integer} -Use Periodic Intra Refresh instead of IDR frames. -@item -bluray-compat @var{integer} -Configure the encoder to be compatible with the bluray standard. -It is a shorthand for setting "bluray-compat=1 force-cfr=1". -@item -b-bias @var{integer} -Influences how often B-frames are used. -@item -b-pyramid @var{integer} -Keep some B-frames as references. - -Possible values: -@table @samp -@item none +@subsection Options -@item strict -Strictly hierarchical pyramid. -@item normal -Non-strict (not Blu-ray compatible). +@table @option +@item seq_disp_ext @var{integer} +Specifies if the encoder should write a sequence_display_extension to the +output. +@table @option +@item -1 +@itemx auto +Decide automatically to write it or not (this is the default) by checking if +the data to be written is different from the default or unspecified values. +@item 0 +@itemx never +Never write it. +@item 1 +@itemx always +Always write it. @end table -@item -mixed-refs @var{integer} -One reference per partition, as opposed to one reference per macroblock. -@item -8x8dct @var{integer} -High profile 8x8 transform. -@item -fast-pskip @var{integer} -@item -aud @var{integer} -Use access unit delimiters. -@item -mbtree @var{integer} -Use macroblock tree ratecontrol. -@item -deblock @var{string} -Loop filter parameters, in <alpha:beta> form. -@item -cplxblur @var{float} -Reduce fluctuations in QP (before curve compression). -@item -partitions @var{string} -A comma-separated list of partitions to consider, possible values: p8x8, p4x4, b8x8, i8x8, i4x4, none, all. -@item -direct-pred @var{integer} -Direct MV prediction mode - -Possible values: -@table @samp -@item none - -@item spatial - -@item temporal - -@item auto - @end table -@item -slice-max-size @var{integer} -Limit the size of each slice in bytes. -@item -stats @var{string} -Filename for 2 pass stats. -@item -nal-hrd @var{integer} -Signal HRD information (requires vbv-bufsize; cbr not allowed in .mp4). -Possible values: -@table @samp -@item none +@section png -@item vbr +PNG image encoder. -@item cbr +@subsection Private options -@end table -@item -x264-params @var{string} -Override the x264 configuration using a :-separated list of key=value parameters. -@example --x264-params level=30:bframes=0:weightp=0:cabac=0:ref=1:vbv-maxrate=768:vbv-bufsize=2000:analyse=all:me=umh:no-fast-pskip=1:subq=6:8x8dct=0:trellis=0 -@end example +@table @option +@item dpi @var{integer} +Set physical density of pixels, in dots per inch, unset by default +@item dpm @var{integer} +Set physical density of pixels, in dots per meter, unset by default @end table -Encoding avpresets for common usages are provided so they can be used with the -general presets system (e.g. passing the @code{-pre} option). - @section ProRes Apple ProRes encoder. -@subsection Private Options +FFmpeg contains 2 ProRes encoders, the prores-aw and prores-ks encoder. +The used encoder can be chosen with the @code{-vcodec} option. + +@subsection Private Options for prores-ks @table @option @item profile @var{integer} @@ -803,7 +2402,7 @@ Use @var{0} to disable alpha plane coding. @subsection Speed considerations In the default mode of operation the encoder has to honor frame constraints -(i.e. not produc frames with size bigger than requested) while still making +(i.e. not produce frames with size bigger than requested) while still making output picture as good as possible. A frame containing a lot of small details is harder to compress and the encoder would spend more time searching for appropriate quantizers for each slice. @@ -847,11 +2446,11 @@ Specifically this means either @itemize @minus @item @var{CQP} - constant quantizer scale, when the @option{qscale} codec flag is -also set (the @option{-qscale} avconv option). +also set (the @option{-qscale} ffmpeg option). @item @var{LA_ICQ} - intelligent constant quality with lookahead, when the -@option{la_depth} option is also set. +@option{look_ahead} option is also set. @item @var{ICQ} -- intelligent constant quality otherwise. @@ -862,7 +2461,7 @@ Otherwise, a bitrate-based mode is used. For all of those, you should specify at least the desired average bitrate with the @option{b} option. @itemize @minus @item -@var{LA} - VBR with lookahead, when the @option{la_depth} option is specified. +@var{LA} - VBR with lookahead, when the @option{look_ahead} option is specified. @item @var{VCM} - video conferencing mode, when the @option{vcm} option is set. @@ -923,3 +2522,27 @@ encoder use CAVLC instead of CABAC. @end itemize @c man end VIDEO ENCODERS + +@chapter Subtitles Encoders +@c man begin SUBTITLES ENCODERS + +@section dvdsub + +This codec encodes the bitmap subtitle format that is used in DVDs. +Typically they are stored in VOBSUB file pairs (*.idx + *.sub), +and they can also be used in Matroska files. + +@subsection Options + +@table @option +@item even_rows_fix +When set to 1, enable a work-around that makes the number of pixel rows +even in all subtitles. This fixes a problem with some players that +cut off the bottom row if the number is odd. The work-around just adds +a fully transparent row if needed. The overhead is low, typically +one byte per subtitle on average. + +By default, this work-around is disabled. +@end table + +@c man end SUBTITLES ENCODERS diff --git a/doc/errno.txt b/doc/errno.txt new file mode 100644 index 0000000000..933a4de51e --- /dev/null +++ b/doc/errno.txt @@ -0,0 +1,174 @@ +The following table lists most error codes found in various operating +systems supported by FFmpeg. + + OS +Code Std F LBMWwb Text (YMMV) + +E2BIG POSIX ++++++ Argument list too long +EACCES POSIX ++++++ Permission denied +EADDRINUSE POSIX +++..+ Address in use +EADDRNOTAVAIL POSIX +++..+ Cannot assign requested address +EADV +..... Advertise error +EAFNOSUPPORT POSIX +++..+ Address family not supported +EAGAIN POSIX + ++++++ Resource temporarily unavailable +EALREADY POSIX +++..+ Operation already in progress +EAUTH .++... Authentication error +EBADARCH ..+... Bad CPU type in executable +EBADE +..... Invalid exchange +EBADEXEC ..+... Bad executable +EBADF POSIX ++++++ Bad file descriptor +EBADFD +..... File descriptor in bad state +EBADMACHO ..+... Malformed Macho file +EBADMSG POSIX ++4... Bad message +EBADR +..... Invalid request descriptor +EBADRPC .++... RPC struct is bad +EBADRQC +..... Invalid request code +EBADSLT +..... Invalid slot +EBFONT +..... Bad font file format +EBUSY POSIX - ++++++ Device or resource busy +ECANCELED POSIX +++... Operation canceled +ECHILD POSIX ++++++ No child processes +ECHRNG +..... Channel number out of range +ECOMM +..... Communication error on send +ECONNABORTED POSIX +++..+ Software caused connection abort +ECONNREFUSED POSIX - +++ss+ Connection refused +ECONNRESET POSIX +++..+ Connection reset +EDEADLK POSIX ++++++ Resource deadlock avoided +EDEADLOCK +..++. File locking deadlock error +EDESTADDRREQ POSIX +++... Destination address required +EDEVERR ..+... Device error +EDOM C89 - ++++++ Numerical argument out of domain +EDOOFUS .F.... Programming error +EDOTDOT +..... RFS specific error +EDQUOT POSIX +++... Disc quota exceeded +EEXIST POSIX ++++++ File exists +EFAULT POSIX - ++++++ Bad address +EFBIG POSIX - ++++++ File too large +EFTYPE .++... Inappropriate file type or format +EHOSTDOWN +++... Host is down +EHOSTUNREACH POSIX +++..+ No route to host +EHWPOISON +..... Memory page has hardware error +EIDRM POSIX +++... Identifier removed +EILSEQ C99 ++++++ Illegal byte sequence +EINPROGRESS POSIX - +++ss+ Operation in progress +EINTR POSIX - ++++++ Interrupted system call +EINVAL POSIX + ++++++ Invalid argument +EIO POSIX + ++++++ I/O error +EISCONN POSIX +++..+ Socket is already connected +EISDIR POSIX ++++++ Is a directory +EISNAM +..... Is a named type file +EKEYEXPIRED +..... Key has expired +EKEYREJECTED +..... Key was rejected by service +EKEYREVOKED +..... Key has been revoked +EL2HLT +..... Level 2 halted +EL2NSYNC +..... Level 2 not synchronized +EL3HLT +..... Level 3 halted +EL3RST +..... Level 3 reset +ELIBACC +..... Can not access a needed shared library +ELIBBAD +..... Accessing a corrupted shared library +ELIBEXEC +..... Cannot exec a shared library directly +ELIBMAX +..... Too many shared libraries +ELIBSCN +..... .lib section in a.out corrupted +ELNRNG +..... Link number out of range +ELOOP POSIX +++..+ Too many levels of symbolic links +EMEDIUMTYPE +..... Wrong medium type +EMFILE POSIX ++++++ Too many open files +EMLINK POSIX ++++++ Too many links +EMSGSIZE POSIX +++..+ Message too long +EMULTIHOP POSIX ++4... Multihop attempted +ENAMETOOLONG POSIX - ++++++ File name too long +ENAVAIL +..... No XENIX semaphores available +ENEEDAUTH .++... Need authenticator +ENETDOWN POSIX +++..+ Network is down +ENETRESET SUSv3 +++..+ Network dropped connection on reset +ENETUNREACH POSIX +++..+ Network unreachable +ENFILE POSIX ++++++ Too many open files in system +ENOANO +..... No anode +ENOATTR .++... Attribute not found +ENOBUFS POSIX - +++..+ No buffer space available +ENOCSI +..... No CSI structure available +ENODATA XSR +N4... No message available +ENODEV POSIX - ++++++ No such device +ENOENT POSIX - ++++++ No such file or directory +ENOEXEC POSIX ++++++ Exec format error +ENOFILE ...++. No such file or directory +ENOKEY +..... Required key not available +ENOLCK POSIX ++++++ No locks available +ENOLINK POSIX ++4... Link has been severed +ENOMEDIUM +..... No medium found +ENOMEM POSIX ++++++ Not enough space +ENOMSG POSIX +++..+ No message of desired type +ENONET +..... Machine is not on the network +ENOPKG +..... Package not installed +ENOPROTOOPT POSIX +++..+ Protocol not available +ENOSPC POSIX ++++++ No space left on device +ENOSR XSR +N4... No STREAM resources +ENOSTR XSR +N4... Not a STREAM +ENOSYS POSIX + ++++++ Function not implemented +ENOTBLK +++... Block device required +ENOTCONN POSIX +++..+ Socket is not connected +ENOTDIR POSIX ++++++ Not a directory +ENOTEMPTY POSIX ++++++ Directory not empty +ENOTNAM +..... Not a XENIX named type file +ENOTRECOVERABLE SUSv4 - +..... State not recoverable +ENOTSOCK POSIX +++..+ Socket operation on non-socket +ENOTSUP POSIX +++... Operation not supported +ENOTTY POSIX ++++++ Inappropriate I/O control operation +ENOTUNIQ +..... Name not unique on network +ENXIO POSIX ++++++ No such device or address +EOPNOTSUPP POSIX +++..+ Operation not supported (on socket) +EOVERFLOW POSIX +++..+ Value too large to be stored in data type +EOWNERDEAD SUSv4 +..... Owner died +EPERM POSIX - ++++++ Operation not permitted +EPFNOSUPPORT +++..+ Protocol family not supported +EPIPE POSIX - ++++++ Broken pipe +EPROCLIM .++... Too many processes +EPROCUNAVAIL .++... Bad procedure for program +EPROGMISMATCH .++... Program version wrong +EPROGUNAVAIL .++... RPC prog. not avail +EPROTO POSIX ++4... Protocol error +EPROTONOSUPPORT POSIX - +++ss+ Protocol not supported +EPROTOTYPE POSIX +++..+ Protocol wrong type for socket +EPWROFF ..+... Device power is off +ERANGE C89 - ++++++ Result too large +EREMCHG +..... Remote address changed +EREMOTE +++... Object is remote +EREMOTEIO +..... Remote I/O error +ERESTART +..... Interrupted system call should be restarted +ERFKILL +..... Operation not possible due to RF-kill +EROFS POSIX ++++++ Read-only file system +ERPCMISMATCH .++... RPC version wrong +ESHLIBVERS ..+... Shared library version mismatch +ESHUTDOWN +++..+ Cannot send after socket shutdown +ESOCKTNOSUPPORT +++... Socket type not supported +ESPIPE POSIX ++++++ Illegal seek +ESRCH POSIX ++++++ No such process +ESRMNT +..... Srmount error +ESTALE POSIX +++..+ Stale NFS file handle +ESTRPIPE +..... Streams pipe error +ETIME XSR +N4... Stream ioctl timeout +ETIMEDOUT POSIX - +++ss+ Connection timed out +ETOOMANYREFS +++... Too many references: cannot splice +ETXTBSY POSIX +++... Text file busy +EUCLEAN +..... Structure needs cleaning +EUNATCH +..... Protocol driver not attached +EUSERS +++... Too many users +EWOULDBLOCK POSIX +++..+ Operation would block +EXDEV POSIX ++++++ Cross-device link +EXFULL +..... Exchange full + +Notations: + +F: used in FFmpeg (-: a few times, +: a lot) + +SUSv3: Single Unix Specification, version 3 +SUSv4: Single Unix Specification, version 4 +XSR: XSI STREAMS (obsolete) + +OS: availability on some supported operating systems +L: GNU/Linux +B: BSD (F: FreeBSD, N: NetBSD) +M: MacOS X +W: Microsoft Windows (s: emulated with winsock, see libavformat/network.h) +w: Mingw32 (3.17) and Mingw64 (2.0.1) +b: BeOS diff --git a/doc/eval.texi b/doc/eval.texi deleted file mode 100644 index e1fd7ee484..0000000000 --- a/doc/eval.texi +++ /dev/null @@ -1,156 +0,0 @@ -@chapter Expression Evaluation -@c man begin EXPRESSION EVALUATION - -When evaluating an arithmetic expression, Libav uses an internal -formula evaluator, implemented through the @file{libavutil/eval.h} -interface. - -An expression may contain unary, binary operators, constants, and -functions. - -Two expressions @var{expr1} and @var{expr2} can be combined to form -another expression "@var{expr1};@var{expr2}". -@var{expr1} and @var{expr2} are evaluated in turn, and the new -expression evaluates to the value of @var{expr2}. - -The following binary operators are available: @code{+}, @code{-}, -@code{*}, @code{/}, @code{^}. - -The following unary operators are available: @code{+}, @code{-}. - -The following functions are available: -@table @option -@item sinh(x) -@item cosh(x) -@item tanh(x) -@item sin(x) -@item cos(x) -@item tan(x) -@item atan(x) -@item asin(x) -@item acos(x) -@item exp(x) -@item log(x) -@item abs(x) -@item squish(x) -@item gauss(x) -@item isinf(x) -Return 1.0 if @var{x} is +/-INFINITY, 0.0 otherwise. -@item isnan(x) -Return 1.0 if @var{x} is NAN, 0.0 otherwise. - -@item mod(x, y) -@item max(x, y) -@item min(x, y) -@item eq(x, y) -@item gte(x, y) -@item gt(x, y) -@item lte(x, y) -@item lt(x, y) -@item st(var, expr) -Allow to store the value of the expression @var{expr} in an internal -variable. @var{var} specifies the number of the variable where to -store the value, and it is a value ranging from 0 to 9. The function -returns the value stored in the internal variable. - -@item ld(var) -Allow to load the value of the internal variable with number -@var{var}, which was previously stored with st(@var{var}, @var{expr}). -The function returns the loaded value. - -@item while(cond, expr) -Evaluate expression @var{expr} while the expression @var{cond} is -non-zero, and returns the value of the last @var{expr} evaluation, or -NAN if @var{cond} was always false. - -@item ceil(expr) -Round the value of expression @var{expr} upwards to the nearest -integer. For example, "ceil(1.5)" is "2.0". - -@item floor(expr) -Round the value of expression @var{expr} downwards to the nearest -integer. For example, "floor(-1.5)" is "-2.0". - -@item trunc(expr) -Round the value of expression @var{expr} towards zero to the nearest -integer. For example, "trunc(-1.5)" is "-1.0". - -@item sqrt(expr) -Compute the square root of @var{expr}. This is equivalent to -"(@var{expr})^.5". - -@item not(expr) -Return 1.0 if @var{expr} is zero, 0.0 otherwise. -@end table - -Note that: - -@code{*} works like AND - -@code{+} works like OR - -thus -@example -if A then B else C -@end example -is equivalent to -@example -A*B + not(A)*C -@end example - -In your C code, you can extend the list of unary and binary functions, -and define recognized constants, so that they are available for your -expressions. - -The evaluator also recognizes the International System number -postfixes. If 'i' is appended after the postfix, powers of 2 are used -instead of powers of 10. The 'B' postfix multiplies the value for 8, -and can be appended after another postfix or used alone. This allows -using for example 'KB', 'MiB', 'G' and 'B' as postfix. - -Follows the list of available International System postfixes, with -indication of the corresponding powers of 10 and of 2. -@table @option -@item y --24 / -80 -@item z --21 / -70 -@item a --18 / -60 -@item f --15 / -50 -@item p --12 / -40 -@item n --9 / -30 -@item u --6 / -20 -@item m --3 / -10 -@item c --2 -@item d --1 -@item h -2 -@item k -3 / 10 -@item K -3 / 10 -@item M -6 / 20 -@item G -9 / 30 -@item T -12 / 40 -@item P -15 / 40 -@item E -18 / 50 -@item Z -21 / 60 -@item Y -24 / 70 -@end table - -@c man end diff --git a/doc/examples/Makefile b/doc/examples/Makefile new file mode 100644 index 0000000000..af3815995a --- /dev/null +++ b/doc/examples/Makefile @@ -0,0 +1,46 @@ +# use pkg-config for getting CFLAGS and LDLIBS +FFMPEG_LIBS= libavdevice \ + libavformat \ + libavfilter \ + libavcodec \ + libswresample \ + libswscale \ + libavutil \ + +CFLAGS += -Wall -g +CFLAGS := $(shell pkg-config --cflags $(FFMPEG_LIBS)) $(CFLAGS) +LDLIBS := $(shell pkg-config --libs $(FFMPEG_LIBS)) $(LDLIBS) + +EXAMPLES= avio_dir_cmd \ + avio_reading \ + decoding_encoding \ + demuxing_decoding \ + extract_mvs \ + filtering_video \ + filtering_audio \ + http_multiclient \ + metadata \ + muxing \ + remuxing \ + resampling_audio \ + scaling_video \ + transcode_aac \ + transcoding \ + +OBJS=$(addsuffix .o,$(EXAMPLES)) + +# the following examples make explicit use of the math library +avcodec: LDLIBS += -lm +decoding_encoding: LDLIBS += -lm +muxing: LDLIBS += -lm +resampling_audio: LDLIBS += -lm + +.phony: all clean-test clean + +all: $(OBJS) $(EXAMPLES) + +clean-test: + $(RM) test*.pgm test.h264 test.mp2 test.sw test.mpg + +clean: clean-test + $(RM) $(EXAMPLES) $(OBJS) diff --git a/doc/examples/README b/doc/examples/README new file mode 100644 index 0000000000..c1ce619d35 --- /dev/null +++ b/doc/examples/README @@ -0,0 +1,23 @@ +FFmpeg examples README +---------------------- + +Both following use cases rely on pkg-config and make, thus make sure +that you have them installed and working on your system. + + +Method 1: build the installed examples in a generic read/write user directory + +Copy to a read/write user directory and just use "make", it will link +to the libraries on your system, assuming the PKG_CONFIG_PATH is +correctly configured. + +Method 2: build the examples in-tree + +Assuming you are in the source FFmpeg checkout directory, you need to build +FFmpeg (no need to make install in any prefix). Then just run "make examples". +This will build the examples using the FFmpeg build system. You can clean those +examples using "make examplesclean" + +If you want to try the dedicated Makefile examples (to emulate the first +method), go into doc/examples and run a command such as +PKG_CONFIG_PATH=pc-uninstalled make. diff --git a/doc/examples/avio_dir_cmd.c b/doc/examples/avio_dir_cmd.c new file mode 100644 index 0000000000..50c435cf8f --- /dev/null +++ b/doc/examples/avio_dir_cmd.c @@ -0,0 +1,180 @@ +/* + * Copyright (c) 2014 Lukasz Marek + * + * Permission is hereby granted, free of charge, to any person obtaining a copy + * of this software and associated documentation files (the "Software"), to deal + * in the Software without restriction, including without limitation the rights + * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell + * copies of the Software, and to permit persons to whom the Software is + * furnished to do so, subject to the following conditions: + * + * The above copyright notice and this permission notice shall be included in + * all copies or substantial portions of the Software. + * + * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR + * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, + * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL + * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER + * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, + * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN + * THE SOFTWARE. + */ + +#include <libavcodec/avcodec.h> +#include <libavformat/avformat.h> +#include <libavformat/avio.h> + +static const char *type_string(int type) +{ + switch (type) { + case AVIO_ENTRY_DIRECTORY: + return "<DIR>"; + case AVIO_ENTRY_FILE: + return "<FILE>"; + case AVIO_ENTRY_BLOCK_DEVICE: + return "<BLOCK DEVICE>"; + case AVIO_ENTRY_CHARACTER_DEVICE: + return "<CHARACTER DEVICE>"; + case AVIO_ENTRY_NAMED_PIPE: + return "<PIPE>"; + case AVIO_ENTRY_SYMBOLIC_LINK: + return "<LINK>"; + case AVIO_ENTRY_SOCKET: + return "<SOCKET>"; + case AVIO_ENTRY_SERVER: + return "<SERVER>"; + case AVIO_ENTRY_SHARE: + return "<SHARE>"; + case AVIO_ENTRY_WORKGROUP: + return "<WORKGROUP>"; + case AVIO_ENTRY_UNKNOWN: + default: + break; + } + return "<UNKNOWN>"; +} + +static int list_op(const char *input_dir) +{ + AVIODirEntry *entry = NULL; + AVIODirContext *ctx = NULL; + int cnt, ret; + char filemode[4], uid_and_gid[20]; + + if ((ret = avio_open_dir(&ctx, input_dir, NULL)) < 0) { + av_log(NULL, AV_LOG_ERROR, "Cannot open directory: %s.\n", av_err2str(ret)); + goto fail; + } + + cnt = 0; + for (;;) { + if ((ret = avio_read_dir(ctx, &entry)) < 0) { + av_log(NULL, AV_LOG_ERROR, "Cannot list directory: %s.\n", av_err2str(ret)); + goto fail; + } + if (!entry) + break; + if (entry->filemode == -1) { + snprintf(filemode, 4, "???"); + } else { + snprintf(filemode, 4, "%3"PRIo64, entry->filemode); + } + snprintf(uid_and_gid, 20, "%"PRId64"(%"PRId64")", entry->user_id, entry->group_id); + if (cnt == 0) + av_log(NULL, AV_LOG_INFO, "%-9s %12s %30s %10s %s %16s %16s %16s\n", + "TYPE", "SIZE", "NAME", "UID(GID)", "UGO", "MODIFIED", + "ACCESSED", "STATUS_CHANGED"); + av_log(NULL, AV_LOG_INFO, "%-9s %12"PRId64" %30s %10s %s %16"PRId64" %16"PRId64" %16"PRId64"\n", + type_string(entry->type), + entry->size, + entry->name, + uid_and_gid, + filemode, + entry->modification_timestamp, + entry->access_timestamp, + entry->status_change_timestamp); + avio_free_directory_entry(&entry); + cnt++; + }; + + fail: + avio_close_dir(&ctx); + return ret; +} + +static int del_op(const char *url) +{ + int ret = avpriv_io_delete(url); + if (ret < 0) + av_log(NULL, AV_LOG_ERROR, "Cannot delete '%s': %s.\n", url, av_err2str(ret)); + return ret; +} + +static int move_op(const char *src, const char *dst) +{ + int ret = avpriv_io_move(src, dst); + if (ret < 0) + av_log(NULL, AV_LOG_ERROR, "Cannot move '%s' into '%s': %s.\n", src, dst, av_err2str(ret)); + return ret; +} + + +static void usage(const char *program_name) +{ + fprintf(stderr, "usage: %s OPERATION entry1 [entry2]\n" + "API example program to show how to manipulate resources " + "accessed through AVIOContext.\n" + "OPERATIONS:\n" + "list list content of the directory\n" + "move rename content in directory\n" + "del delete content in directory\n", + program_name); +} + +int main(int argc, char *argv[]) +{ + const char *op = NULL; + int ret; + + av_log_set_level(AV_LOG_DEBUG); + + if (argc < 2) { + usage(argv[0]); + return 1; + } + + /* register codecs and formats and other lavf/lavc components*/ + av_register_all(); + avformat_network_init(); + + op = argv[1]; + if (strcmp(op, "list") == 0) { + if (argc < 3) { + av_log(NULL, AV_LOG_INFO, "Missing argument for list operation.\n"); + ret = AVERROR(EINVAL); + } else { + ret = list_op(argv[2]); + } + } else if (strcmp(op, "del") == 0) { + if (argc < 3) { + av_log(NULL, AV_LOG_INFO, "Missing argument for del operation.\n"); + ret = AVERROR(EINVAL); + } else { + ret = del_op(argv[2]); + } + } else if (strcmp(op, "move") == 0) { + if (argc < 4) { + av_log(NULL, AV_LOG_INFO, "Missing argument for move operation.\n"); + ret = AVERROR(EINVAL); + } else { + ret = move_op(argv[2], argv[3]); + } + } else { + av_log(NULL, AV_LOG_INFO, "Invalid operation %s\n", op); + ret = AVERROR(EINVAL); + } + + avformat_network_deinit(); + + return ret < 0 ? 1 : 0; +} diff --git a/doc/examples/avio_reading.c b/doc/examples/avio_reading.c new file mode 100644 index 0000000000..02474e907a --- /dev/null +++ b/doc/examples/avio_reading.c @@ -0,0 +1,134 @@ +/* + * Copyright (c) 2014 Stefano Sabatini + * + * Permission is hereby granted, free of charge, to any person obtaining a copy + * of this software and associated documentation files (the "Software"), to deal + * in the Software without restriction, including without limitation the rights + * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell + * copies of the Software, and to permit persons to whom the Software is + * furnished to do so, subject to the following conditions: + * + * The above copyright notice and this permission notice shall be included in + * all copies or substantial portions of the Software. + * + * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR + * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, + * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL + * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER + * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, + * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN + * THE SOFTWARE. + */ + +/** + * @file + * libavformat AVIOContext API example. + * + * Make libavformat demuxer access media content through a custom + * AVIOContext read callback. + * @example avio_reading.c + */ + +#include <libavcodec/avcodec.h> +#include <libavformat/avformat.h> +#include <libavformat/avio.h> +#include <libavutil/file.h> + +struct buffer_data { + uint8_t *ptr; + size_t size; ///< size left in the buffer +}; + +static int read_packet(void *opaque, uint8_t *buf, int buf_size) +{ + struct buffer_data *bd = (struct buffer_data *)opaque; + buf_size = FFMIN(buf_size, bd->size); + + printf("ptr:%p size:%zu\n", bd->ptr, bd->size); + + /* copy internal buffer data to buf */ + memcpy(buf, bd->ptr, buf_size); + bd->ptr += buf_size; + bd->size -= buf_size; + + return buf_size; +} + +int main(int argc, char *argv[]) +{ + AVFormatContext *fmt_ctx = NULL; + AVIOContext *avio_ctx = NULL; + uint8_t *buffer = NULL, *avio_ctx_buffer = NULL; + size_t buffer_size, avio_ctx_buffer_size = 4096; + char *input_filename = NULL; + int ret = 0; + struct buffer_data bd = { 0 }; + + if (argc != 2) { + fprintf(stderr, "usage: %s input_file\n" + "API example program to show how to read from a custom buffer " + "accessed through AVIOContext.\n", argv[0]); + return 1; + } + input_filename = argv[1]; + + /* register codecs and formats and other lavf/lavc components*/ + av_register_all(); + + /* slurp file content into buffer */ + ret = av_file_map(input_filename, &buffer, &buffer_size, 0, NULL); + if (ret < 0) + goto end; + + /* fill opaque structure used by the AVIOContext read callback */ + bd.ptr = buffer; + bd.size = buffer_size; + + if (!(fmt_ctx = avformat_alloc_context())) { + ret = AVERROR(ENOMEM); + goto end; + } + + avio_ctx_buffer = av_malloc(avio_ctx_buffer_size); + if (!avio_ctx_buffer) { + ret = AVERROR(ENOMEM); + goto end; + } + avio_ctx = avio_alloc_context(avio_ctx_buffer, avio_ctx_buffer_size, + 0, &bd, &read_packet, NULL, NULL); + if (!avio_ctx) { + ret = AVERROR(ENOMEM); + goto end; + } + fmt_ctx->pb = avio_ctx; + + ret = avformat_open_input(&fmt_ctx, NULL, NULL, NULL); + if (ret < 0) { + fprintf(stderr, "Could not open input\n"); + goto end; + } + + ret = avformat_find_stream_info(fmt_ctx, NULL); + if (ret < 0) { + fprintf(stderr, "Could not find stream information\n"); + goto end; + } + + av_dump_format(fmt_ctx, 0, input_filename, 0); + +end: + avformat_close_input(&fmt_ctx); + /* note: the internal buffer could have changed, and be != avio_ctx_buffer */ + if (avio_ctx) { + av_freep(&avio_ctx->buffer); + av_freep(&avio_ctx); + } + av_file_unmap(buffer, buffer_size); + + if (ret < 0) { + fprintf(stderr, "Error occurred: %s\n", av_err2str(ret)); + return 1; + } + + return 0; +} diff --git a/doc/examples/avcodec.c b/doc/examples/decoding_encoding.c index df0af4b1ea..06a98a630e 100644 --- a/doc/examples/avcodec.c +++ b/doc/examples/decoding_encoding.c @@ -1,47 +1,44 @@ /* - * copyright (c) 2001 Fabrice Bellard + * Copyright (c) 2001 Fabrice Bellard * - * This file is part of Libav. + * Permission is hereby granted, free of charge, to any person obtaining a copy + * of this software and associated documentation files (the "Software"), to deal + * in the Software without restriction, including without limitation the rights + * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell + * copies of the Software, and to permit persons to whom the Software is + * furnished to do so, subject to the following conditions: * - * Libav is free software; you can redistribute it and/or - * modify it under the terms of the GNU Lesser General Public - * License as published by the Free Software Foundation; either - * version 2.1 of the License, or (at your option) any later version. + * The above copyright notice and this permission notice shall be included in + * all copies or substantial portions of the Software. * - * Libav is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU - * Lesser General Public License for more details. - * - * You should have received a copy of the GNU Lesser General Public - * License along with Libav; if not, write to the Free Software - * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA + * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR + * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, + * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL + * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER + * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, + * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN + * THE SOFTWARE. */ /** * @file * libavcodec API use example. * - * @example avcodec.c - * Note that this library only handles codecs (mpeg, mpeg4, etc...), - * not file formats (avi, vob, etc...). See library 'libavformat' for the + * @example decoding_encoding.c + * Note that libavcodec only handles codecs (mpeg, mpeg4, etc...), + * not file formats (avi, vob, mp4, mov, mkv, mxf, flv, mpegts, mpegps, etc...). See library 'libavformat' for the * format handling */ -#include <stdlib.h> -#include <stdio.h> -#include <string.h> - -#ifdef HAVE_AV_CONFIG_H -#undef HAVE_AV_CONFIG_H -#endif +#include <math.h> -#include "libavcodec/avcodec.h" -#include "libavutil/channel_layout.h" -#include "libavutil/common.h" -#include "libavutil/imgutils.h" -#include "libavutil/mathematics.h" -#include "libavutil/samplefmt.h" +#include <libavutil/opt.h> +#include <libavcodec/avcodec.h> +#include <libavutil/channel_layout.h> +#include <libavutil/common.h> +#include <libavutil/imgutils.h> +#include <libavutil/mathematics.h> +#include <libavutil/samplefmt.h> #define INBUF_SIZE 4096 #define AUDIO_INBUF_SIZE 20480 @@ -115,16 +112,20 @@ static void audio_encode_example(const char *filename) uint16_t *samples; float t, tincr; - printf("Audio encoding\n"); + printf("Encode audio file %s\n", filename); /* find the MP2 encoder */ codec = avcodec_find_encoder(AV_CODEC_ID_MP2); if (!codec) { - fprintf(stderr, "codec not found\n"); + fprintf(stderr, "Codec not found\n"); exit(1); } c = avcodec_alloc_context3(codec); + if (!c) { + fprintf(stderr, "Could not allocate audio codec context\n"); + exit(1); + } /* put sample parameters */ c->bit_rate = 64000; @@ -132,7 +133,7 @@ static void audio_encode_example(const char *filename) /* check that the encoder supports s16 pcm input */ c->sample_fmt = AV_SAMPLE_FMT_S16; if (!check_sample_fmt(codec, c->sample_fmt)) { - fprintf(stderr, "encoder does not support %s", + fprintf(stderr, "Encoder does not support sample format %s", av_get_sample_fmt_name(c->sample_fmt)); exit(1); } @@ -144,20 +145,20 @@ static void audio_encode_example(const char *filename) /* open it */ if (avcodec_open2(c, codec, NULL) < 0) { - fprintf(stderr, "could not open codec\n"); + fprintf(stderr, "Could not open codec\n"); exit(1); } f = fopen(filename, "wb"); if (!f) { - fprintf(stderr, "could not open %s\n", filename); + fprintf(stderr, "Could not open %s\n", filename); exit(1); } /* frame containing input raw audio */ frame = av_frame_alloc(); if (!frame) { - fprintf(stderr, "could not allocate audio frame\n"); + fprintf(stderr, "Could not allocate audio frame\n"); exit(1); } @@ -169,9 +170,13 @@ static void audio_encode_example(const char *filename) * we calculate the size of the samples buffer in bytes */ buffer_size = av_samples_get_buffer_size(NULL, c->channels, c->frame_size, c->sample_fmt, 0); + if (buffer_size < 0) { + fprintf(stderr, "Could not get sample buffer size\n"); + exit(1); + } samples = av_malloc(buffer_size); if (!samples) { - fprintf(stderr, "could not allocate %d bytes for samples buffer\n", + fprintf(stderr, "Could not allocate %d bytes for samples buffer\n", buffer_size); exit(1); } @@ -179,14 +184,14 @@ static void audio_encode_example(const char *filename) ret = avcodec_fill_audio_frame(frame, c->channels, c->sample_fmt, (const uint8_t*)samples, buffer_size, 0); if (ret < 0) { - fprintf(stderr, "could not setup audio frame\n"); + fprintf(stderr, "Could not setup audio frame\n"); exit(1); } /* encode a single tone sound */ t = 0; tincr = 2 * M_PI * 440.0 / c->sample_rate; - for(i=0;i<200;i++) { + for (i = 0; i < 200; i++) { av_init_packet(&pkt); pkt.data = NULL; // packet data will be allocated by the encoder pkt.size = 0; @@ -201,9 +206,23 @@ static void audio_encode_example(const char *filename) /* encode the samples */ ret = avcodec_encode_audio2(c, &pkt, frame, &got_output); if (ret < 0) { - fprintf(stderr, "error encoding audio frame\n"); + fprintf(stderr, "Error encoding audio frame\n"); + exit(1); + } + if (got_output) { + fwrite(pkt.data, 1, pkt.size, f); + av_packet_unref(&pkt); + } + } + + /* get the delayed frames */ + for (got_output = 1; got_output; i++) { + ret = avcodec_encode_audio2(c, &pkt, NULL, &got_output); + if (ret < 0) { + fprintf(stderr, "Error encoding frame\n"); exit(1); } + if (got_output) { fwrite(pkt.data, 1, pkt.size, f); av_packet_unref(&pkt); @@ -232,26 +251,30 @@ static void audio_decode_example(const char *outfilename, const char *filename) av_init_packet(&avpkt); - printf("Audio decoding\n"); + printf("Decode audio file %s to %s\n", filename, outfilename); /* find the mpeg audio decoder */ codec = avcodec_find_decoder(AV_CODEC_ID_MP2); if (!codec) { - fprintf(stderr, "codec not found\n"); + fprintf(stderr, "Codec not found\n"); exit(1); } c = avcodec_alloc_context3(codec); + if (!c) { + fprintf(stderr, "Could not allocate audio codec context\n"); + exit(1); + } /* open it */ if (avcodec_open2(c, codec, NULL) < 0) { - fprintf(stderr, "could not open codec\n"); + fprintf(stderr, "Could not open codec\n"); exit(1); } f = fopen(filename, "rb"); if (!f) { - fprintf(stderr, "could not open %s\n", filename); + fprintf(stderr, "Could not open %s\n", filename); exit(1); } outfile = fopen(outfilename, "wb"); @@ -265,11 +288,12 @@ static void audio_decode_example(const char *outfilename, const char *filename) avpkt.size = fread(inbuf, 1, AUDIO_INBUF_SIZE, f); while (avpkt.size > 0) { + int i, ch; int got_frame = 0; if (!decoded_frame) { if (!(decoded_frame = av_frame_alloc())) { - fprintf(stderr, "out of memory\n"); + fprintf(stderr, "Could not allocate audio frame\n"); exit(1); } } @@ -281,13 +305,20 @@ static void audio_decode_example(const char *outfilename, const char *filename) } if (got_frame) { /* if a frame has been decoded, output it */ - int data_size = av_samples_get_buffer_size(NULL, c->channels, - decoded_frame->nb_samples, - c->sample_fmt, 1); - fwrite(decoded_frame->data[0], 1, data_size, outfile); + int data_size = av_get_bytes_per_sample(c->sample_fmt); + if (data_size < 0) { + /* This should not occur, checking just for paranoia */ + fprintf(stderr, "Failed to calculate data size\n"); + exit(1); + } + for (i=0; i<decoded_frame->nb_samples; i++) + for (ch=0; ch<c->channels; ch++) + fwrite(decoded_frame->data[ch] + data_size*i, 1, data_size, outfile); } avpkt.size -= len; avpkt.data += len; + avpkt.dts = + avpkt.pts = AV_NOPTS_VALUE; if (avpkt.size < AUDIO_REFILL_THRESH) { /* Refill the input buffer, to avoid trying to decode * incomplete frames. Instead of this, one could also use @@ -313,27 +344,30 @@ static void audio_decode_example(const char *outfilename, const char *filename) /* * Video encoding example */ -static void video_encode_example(const char *filename) +static void video_encode_example(const char *filename, int codec_id) { AVCodec *codec; AVCodecContext *c= NULL; int i, ret, x, y, got_output; FILE *f; - AVFrame *picture; + AVFrame *frame; AVPacket pkt; uint8_t endcode[] = { 0, 0, 1, 0xb7 }; - printf("Video encoding\n"); + printf("Encode video file %s\n", filename); /* find the mpeg1 video encoder */ - codec = avcodec_find_encoder(AV_CODEC_ID_MPEG1VIDEO); + codec = avcodec_find_encoder(codec_id); if (!codec) { - fprintf(stderr, "codec not found\n"); + fprintf(stderr, "Codec not found\n"); exit(1); } c = avcodec_alloc_context3(codec); - picture = av_frame_alloc(); + if (!c) { + fprintf(stderr, "Could not allocate video codec context\n"); + exit(1); + } /* put sample parameters */ c->bit_rate = 400000; @@ -341,35 +375,52 @@ static void video_encode_example(const char *filename) c->width = 352; c->height = 288; /* frames per second */ - c->time_base= (AVRational){1,25}; - c->gop_size = 10; /* emit one intra frame every ten frames */ - c->max_b_frames=1; + c->time_base = (AVRational){1,25}; + /* emit one intra frame every ten frames + * check frame pict_type before passing frame + * to encoder, if frame->pict_type is AV_PICTURE_TYPE_I + * then gop_size is ignored and the output of encoder + * will always be I frame irrespective to gop_size + */ + c->gop_size = 10; + c->max_b_frames = 1; c->pix_fmt = AV_PIX_FMT_YUV420P; + if (codec_id == AV_CODEC_ID_H264) + av_opt_set(c->priv_data, "preset", "slow", 0); + /* open it */ if (avcodec_open2(c, codec, NULL) < 0) { - fprintf(stderr, "could not open codec\n"); + fprintf(stderr, "Could not open codec\n"); exit(1); } f = fopen(filename, "wb"); if (!f) { - fprintf(stderr, "could not open %s\n", filename); + fprintf(stderr, "Could not open %s\n", filename); + exit(1); + } + + frame = av_frame_alloc(); + if (!frame) { + fprintf(stderr, "Could not allocate video frame\n"); exit(1); } + frame->format = c->pix_fmt; + frame->width = c->width; + frame->height = c->height; - ret = av_image_alloc(picture->data, picture->linesize, c->width, c->height, + /* the image can be allocated by any means and av_image_alloc() is + * just the most convenient way if av_malloc() is to be used */ + ret = av_image_alloc(frame->data, frame->linesize, c->width, c->height, c->pix_fmt, 32); if (ret < 0) { - fprintf(stderr, "could not alloc raw picture buffer\n"); + fprintf(stderr, "Could not allocate raw picture buffer\n"); exit(1); } - picture->format = c->pix_fmt; - picture->width = c->width; - picture->height = c->height; /* encode 1 second of video */ - for(i=0;i<25;i++) { + for (i = 0; i < 25; i++) { av_init_packet(&pkt); pkt.data = NULL; // packet data will be allocated by the encoder pkt.size = 0; @@ -377,31 +428,31 @@ static void video_encode_example(const char *filename) fflush(stdout); /* prepare a dummy image */ /* Y */ - for(y=0;y<c->height;y++) { - for(x=0;x<c->width;x++) { - picture->data[0][y * picture->linesize[0] + x] = x + y + i * 3; + for (y = 0; y < c->height; y++) { + for (x = 0; x < c->width; x++) { + frame->data[0][y * frame->linesize[0] + x] = x + y + i * 3; } } /* Cb and Cr */ - for(y=0;y<c->height/2;y++) { - for(x=0;x<c->width/2;x++) { - picture->data[1][y * picture->linesize[1] + x] = 128 + y + i * 2; - picture->data[2][y * picture->linesize[2] + x] = 64 + x + i * 5; + for (y = 0; y < c->height/2; y++) { + for (x = 0; x < c->width/2; x++) { + frame->data[1][y * frame->linesize[1] + x] = 128 + y + i * 2; + frame->data[2][y * frame->linesize[2] + x] = 64 + x + i * 5; } } - picture->pts = i; + frame->pts = i; /* encode the image */ - ret = avcodec_encode_video2(c, &pkt, picture, &got_output); + ret = avcodec_encode_video2(c, &pkt, frame, &got_output); if (ret < 0) { - fprintf(stderr, "error encoding frame\n"); + fprintf(stderr, "Error encoding frame\n"); exit(1); } if (got_output) { - printf("encoding frame %3d (size=%5d)\n", i, pkt.size); + printf("Write frame %3d (size=%5d)\n", i, pkt.size); fwrite(pkt.data, 1, pkt.size, f); av_packet_unref(&pkt); } @@ -413,12 +464,12 @@ static void video_encode_example(const char *filename) ret = avcodec_encode_video2(c, &pkt, NULL, &got_output); if (ret < 0) { - fprintf(stderr, "error encoding frame\n"); + fprintf(stderr, "Error encoding frame\n"); exit(1); } if (got_output) { - printf("encoding frame %3d (size=%5d)\n", i, pkt.size); + printf("Write frame %3d (size=%5d)\n", i, pkt.size); fwrite(pkt.data, 1, pkt.size, f); av_packet_unref(&pkt); } @@ -430,8 +481,8 @@ static void video_encode_example(const char *filename) avcodec_close(c); av_free(c); - av_freep(&picture->data[0]); - av_frame_free(&picture); + av_freep(&frame->data[0]); + av_frame_free(&frame); printf("\n"); } @@ -445,22 +496,49 @@ static void pgm_save(unsigned char *buf, int wrap, int xsize, int ysize, FILE *f; int i; - f=fopen(filename,"w"); - fprintf(f,"P5\n%d %d\n%d\n",xsize,ysize,255); - for(i=0;i<ysize;i++) - fwrite(buf + i * wrap,1,xsize,f); + f = fopen(filename,"w"); + fprintf(f, "P5\n%d %d\n%d\n", xsize, ysize, 255); + for (i = 0; i < ysize; i++) + fwrite(buf + i * wrap, 1, xsize, f); fclose(f); } +static int decode_write_frame(const char *outfilename, AVCodecContext *avctx, + AVFrame *frame, int *frame_count, AVPacket *pkt, int last) +{ + int len, got_frame; + char buf[1024]; + + len = avcodec_decode_video2(avctx, frame, &got_frame, pkt); + if (len < 0) { + fprintf(stderr, "Error while decoding frame %d\n", *frame_count); + return len; + } + if (got_frame) { + printf("Saving %sframe %3d\n", last ? "last " : "", *frame_count); + fflush(stdout); + + /* the picture is allocated by the decoder, no need to free it */ + snprintf(buf, sizeof(buf), outfilename, *frame_count); + pgm_save(frame->data[0], frame->linesize[0], + frame->width, frame->height, buf); + (*frame_count)++; + } + if (pkt->data) { + pkt->size -= len; + pkt->data += len; + } + return 0; +} + static void video_decode_example(const char *outfilename, const char *filename) { AVCodec *codec; AVCodecContext *c= NULL; - int frame, got_picture, len; + int frame_count; FILE *f; - AVFrame *picture; + AVFrame *frame; uint8_t inbuf[INBUF_SIZE + AV_INPUT_BUFFER_PADDING_SIZE]; - char buf[1024]; AVPacket avpkt; av_init_packet(&avpkt); @@ -468,17 +546,20 @@ static void video_decode_example(const char *outfilename, const char *filename) /* set end of buffer to 0 (this ensures that no overreading happens for damaged mpeg streams) */ memset(inbuf + INBUF_SIZE, 0, AV_INPUT_BUFFER_PADDING_SIZE); - printf("Video decoding\n"); + printf("Decode video file %s to %s\n", filename, outfilename); /* find the mpeg1 video decoder */ codec = avcodec_find_decoder(AV_CODEC_ID_MPEG1VIDEO); if (!codec) { - fprintf(stderr, "codec not found\n"); + fprintf(stderr, "Codec not found\n"); exit(1); } c = avcodec_alloc_context3(codec); - picture = av_frame_alloc(); + if (!c) { + fprintf(stderr, "Could not allocate video codec context\n"); + exit(1); + } if (codec->capabilities & AV_CODEC_CAP_TRUNCATED) c->flags |= AV_CODEC_FLAG_TRUNCATED; // we do not send complete frames @@ -489,20 +570,24 @@ static void video_decode_example(const char *outfilename, const char *filename) /* open it */ if (avcodec_open2(c, codec, NULL) < 0) { - fprintf(stderr, "could not open codec\n"); + fprintf(stderr, "Could not open codec\n"); exit(1); } - /* the codec gives us the frame size, in samples */ - f = fopen(filename, "rb"); if (!f) { - fprintf(stderr, "could not open %s\n", filename); + fprintf(stderr, "Could not open %s\n", filename); exit(1); } - frame = 0; - for(;;) { + frame = av_frame_alloc(); + if (!frame) { + fprintf(stderr, "Could not allocate video frame\n"); + exit(1); + } + + frame_count = 0; + for (;;) { avpkt.size = fread(inbuf, 1, INBUF_SIZE, f); if (avpkt.size == 0) break; @@ -523,26 +608,9 @@ static void video_decode_example(const char *outfilename, const char *filename) /* here, we use a stream based decoder (mpeg1video), so we feed decoder and see if it could decode a frame */ avpkt.data = inbuf; - while (avpkt.size > 0) { - len = avcodec_decode_video2(c, picture, &got_picture, &avpkt); - if (len < 0) { - fprintf(stderr, "Error while decoding frame %d\n", frame); + while (avpkt.size > 0) + if (decode_write_frame(outfilename, c, frame, &frame_count, &avpkt, 0) < 0) exit(1); - } - if (got_picture) { - printf("saving frame %3d\n", frame); - fflush(stdout); - - /* the picture is allocated by the decoder. no need to - free it */ - snprintf(buf, sizeof(buf), outfilename, frame); - pgm_save(picture->data[0], picture->linesize[0], - c->width, c->height, buf); - frame++; - } - avpkt.size -= len; - avpkt.data += len; - } } /* some codecs, such as MPEG, transmit the I and P frame with a @@ -550,46 +618,48 @@ static void video_decode_example(const char *outfilename, const char *filename) chance to get the last frame of the video */ avpkt.data = NULL; avpkt.size = 0; - len = avcodec_decode_video2(c, picture, &got_picture, &avpkt); - if (got_picture) { - printf("saving last frame %3d\n", frame); - fflush(stdout); - - /* the picture is allocated by the decoder. no need to - free it */ - snprintf(buf, sizeof(buf), outfilename, frame); - pgm_save(picture->data[0], picture->linesize[0], - c->width, c->height, buf); - frame++; - } + decode_write_frame(outfilename, c, frame, &frame_count, &avpkt, 1); fclose(f); avcodec_close(c); av_free(c); - av_frame_free(&picture); + av_frame_free(&frame); printf("\n"); } int main(int argc, char **argv) { - const char *filename; + const char *output_type; /* register all the codecs */ avcodec_register_all(); - if (argc <= 1) { - audio_encode_example("/tmp/test.mp2"); - audio_decode_example("/tmp/test.sw", "/tmp/test.mp2"); - - video_encode_example("/tmp/test.mpg"); - filename = "/tmp/test.mpg"; + if (argc < 2) { + printf("usage: %s output_type\n" + "API example program to decode/encode a media stream with libavcodec.\n" + "This program generates a synthetic stream and encodes it to a file\n" + "named test.h264, test.mp2 or test.mpg depending on output_type.\n" + "The encoded stream is then decoded and written to a raw data output.\n" + "output_type must be chosen between 'h264', 'mp2', 'mpg'.\n", + argv[0]); + return 1; + } + output_type = argv[1]; + + if (!strcmp(output_type, "h264")) { + video_encode_example("test.h264", AV_CODEC_ID_H264); + } else if (!strcmp(output_type, "mp2")) { + audio_encode_example("test.mp2"); + audio_decode_example("test.pcm", "test.mp2"); + } else if (!strcmp(output_type, "mpg")) { + video_encode_example("test.mpg", AV_CODEC_ID_MPEG1VIDEO); + video_decode_example("test%02d.pgm", "test.mpg"); } else { - filename = argv[1]; + fprintf(stderr, "Invalid output type '%s', choose between 'h264', 'mp2', or 'mpg'\n", + output_type); + return 1; } - // audio_decode_example("/tmp/test.sw", filename); - video_decode_example("/tmp/test%d.pgm", filename); - return 0; } diff --git a/doc/examples/demuxing_decoding.c b/doc/examples/demuxing_decoding.c new file mode 100644 index 0000000000..59e0ccc986 --- /dev/null +++ b/doc/examples/demuxing_decoding.c @@ -0,0 +1,383 @@ +/* + * Copyright (c) 2012 Stefano Sabatini + * + * Permission is hereby granted, free of charge, to any person obtaining a copy + * of this software and associated documentation files (the "Software"), to deal + * in the Software without restriction, including without limitation the rights + * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell + * copies of the Software, and to permit persons to whom the Software is + * furnished to do so, subject to the following conditions: + * + * The above copyright notice and this permission notice shall be included in + * all copies or substantial portions of the Software. + * + * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR + * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, + * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL + * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER + * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, + * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN + * THE SOFTWARE. + */ + +/** + * @file + * Demuxing and decoding example. + * + * Show how to use the libavformat and libavcodec API to demux and + * decode audio and video data. + * @example demuxing_decoding.c + */ + +#include <libavutil/imgutils.h> +#include <libavutil/samplefmt.h> +#include <libavutil/timestamp.h> +#include <libavformat/avformat.h> + +static AVFormatContext *fmt_ctx = NULL; +static AVCodecContext *video_dec_ctx = NULL, *audio_dec_ctx; +static int width, height; +static enum AVPixelFormat pix_fmt; +static AVStream *video_stream = NULL, *audio_stream = NULL; +static const char *src_filename = NULL; +static const char *video_dst_filename = NULL; +static const char *audio_dst_filename = NULL; +static FILE *video_dst_file = NULL; +static FILE *audio_dst_file = NULL; + +static uint8_t *video_dst_data[4] = {NULL}; +static int video_dst_linesize[4]; +static int video_dst_bufsize; + +static int video_stream_idx = -1, audio_stream_idx = -1; +static AVFrame *frame = NULL; +static AVPacket pkt; +static int video_frame_count = 0; +static int audio_frame_count = 0; + +/* Enable or disable frame reference counting. You are not supposed to support + * both paths in your application but pick the one most appropriate to your + * needs. Look for the use of refcount in this example to see what are the + * differences of API usage between them. */ +static int refcount = 0; + +static int decode_packet(int *got_frame, int cached) +{ + int ret = 0; + int decoded = pkt.size; + + *got_frame = 0; + + if (pkt.stream_index == video_stream_idx) { + /* decode video frame */ + ret = avcodec_decode_video2(video_dec_ctx, frame, got_frame, &pkt); + if (ret < 0) { + fprintf(stderr, "Error decoding video frame (%s)\n", av_err2str(ret)); + return ret; + } + + if (*got_frame) { + + if (frame->width != width || frame->height != height || + frame->format != pix_fmt) { + /* To handle this change, one could call av_image_alloc again and + * decode the following frames into another rawvideo file. */ + fprintf(stderr, "Error: Width, height and pixel format have to be " + "constant in a rawvideo file, but the width, height or " + "pixel format of the input video changed:\n" + "old: width = %d, height = %d, format = %s\n" + "new: width = %d, height = %d, format = %s\n", + width, height, av_get_pix_fmt_name(pix_fmt), + frame->width, frame->height, + av_get_pix_fmt_name(frame->format)); + return -1; + } + + printf("video_frame%s n:%d coded_n:%d pts:%s\n", + cached ? "(cached)" : "", + video_frame_count++, frame->coded_picture_number, + av_ts2timestr(frame->pts, &video_dec_ctx->time_base)); + + /* copy decoded frame to destination buffer: + * this is required since rawvideo expects non aligned data */ + av_image_copy(video_dst_data, video_dst_linesize, + (const uint8_t **)(frame->data), frame->linesize, + pix_fmt, width, height); + + /* write to rawvideo file */ + fwrite(video_dst_data[0], 1, video_dst_bufsize, video_dst_file); + } + } else if (pkt.stream_index == audio_stream_idx) { + /* decode audio frame */ + ret = avcodec_decode_audio4(audio_dec_ctx, frame, got_frame, &pkt); + if (ret < 0) { + fprintf(stderr, "Error decoding audio frame (%s)\n", av_err2str(ret)); + return ret; + } + /* Some audio decoders decode only part of the packet, and have to be + * called again with the remainder of the packet data. + * Sample: fate-suite/lossless-audio/luckynight-partial.shn + * Also, some decoders might over-read the packet. */ + decoded = FFMIN(ret, pkt.size); + + if (*got_frame) { + size_t unpadded_linesize = frame->nb_samples * av_get_bytes_per_sample(frame->format); + printf("audio_frame%s n:%d nb_samples:%d pts:%s\n", + cached ? "(cached)" : "", + audio_frame_count++, frame->nb_samples, + av_ts2timestr(frame->pts, &audio_dec_ctx->time_base)); + + /* Write the raw audio data samples of the first plane. This works + * fine for packed formats (e.g. AV_SAMPLE_FMT_S16). However, + * most audio decoders output planar audio, which uses a separate + * plane of audio samples for each channel (e.g. AV_SAMPLE_FMT_S16P). + * In other words, this code will write only the first audio channel + * in these cases. + * You should use libswresample or libavfilter to convert the frame + * to packed data. */ + fwrite(frame->extended_data[0], 1, unpadded_linesize, audio_dst_file); + } + } + + /* If we use frame reference counting, we own the data and need + * to de-reference it when we don't use it anymore */ + if (*got_frame && refcount) + av_frame_unref(frame); + + return decoded; +} + +static int open_codec_context(int *stream_idx, + AVFormatContext *fmt_ctx, enum AVMediaType type) +{ + int ret, stream_index; + AVStream *st; + AVCodecContext *dec_ctx = NULL; + AVCodec *dec = NULL; + AVDictionary *opts = NULL; + + ret = av_find_best_stream(fmt_ctx, type, -1, -1, NULL, 0); + if (ret < 0) { + fprintf(stderr, "Could not find %s stream in input file '%s'\n", + av_get_media_type_string(type), src_filename); + return ret; + } else { + stream_index = ret; + st = fmt_ctx->streams[stream_index]; + + /* find decoder for the stream */ + dec_ctx = st->codec; + dec = avcodec_find_decoder(dec_ctx->codec_id); + if (!dec) { + fprintf(stderr, "Failed to find %s codec\n", + av_get_media_type_string(type)); + return AVERROR(EINVAL); + } + + /* Init the decoders, with or without reference counting */ + av_dict_set(&opts, "refcounted_frames", refcount ? "1" : "0", 0); + if ((ret = avcodec_open2(dec_ctx, dec, &opts)) < 0) { + fprintf(stderr, "Failed to open %s codec\n", + av_get_media_type_string(type)); + return ret; + } + *stream_idx = stream_index; + } + + return 0; +} + +static int get_format_from_sample_fmt(const char **fmt, + enum AVSampleFormat sample_fmt) +{ + int i; + struct sample_fmt_entry { + enum AVSampleFormat sample_fmt; const char *fmt_be, *fmt_le; + } sample_fmt_entries[] = { + { AV_SAMPLE_FMT_U8, "u8", "u8" }, + { AV_SAMPLE_FMT_S16, "s16be", "s16le" }, + { AV_SAMPLE_FMT_S32, "s32be", "s32le" }, + { AV_SAMPLE_FMT_FLT, "f32be", "f32le" }, + { AV_SAMPLE_FMT_DBL, "f64be", "f64le" }, + }; + *fmt = NULL; + + for (i = 0; i < FF_ARRAY_ELEMS(sample_fmt_entries); i++) { + struct sample_fmt_entry *entry = &sample_fmt_entries[i]; + if (sample_fmt == entry->sample_fmt) { + *fmt = AV_NE(entry->fmt_be, entry->fmt_le); + return 0; + } + } + + fprintf(stderr, + "sample format %s is not supported as output format\n", + av_get_sample_fmt_name(sample_fmt)); + return -1; +} + +int main (int argc, char **argv) +{ + int ret = 0, got_frame; + + if (argc != 4 && argc != 5) { + fprintf(stderr, "usage: %s [-refcount] input_file video_output_file audio_output_file\n" + "API example program to show how to read frames from an input file.\n" + "This program reads frames from a file, decodes them, and writes decoded\n" + "video frames to a rawvideo file named video_output_file, and decoded\n" + "audio frames to a rawaudio file named audio_output_file.\n\n" + "If the -refcount option is specified, the program use the\n" + "reference counting frame system which allows keeping a copy of\n" + "the data for longer than one decode call.\n" + "\n", argv[0]); + exit(1); + } + if (argc == 5 && !strcmp(argv[1], "-refcount")) { + refcount = 1; + argv++; + } + src_filename = argv[1]; + video_dst_filename = argv[2]; + audio_dst_filename = argv[3]; + + /* register all formats and codecs */ + av_register_all(); + + /* open input file, and allocate format context */ + if (avformat_open_input(&fmt_ctx, src_filename, NULL, NULL) < 0) { + fprintf(stderr, "Could not open source file %s\n", src_filename); + exit(1); + } + + /* retrieve stream information */ + if (avformat_find_stream_info(fmt_ctx, NULL) < 0) { + fprintf(stderr, "Could not find stream information\n"); + exit(1); + } + + if (open_codec_context(&video_stream_idx, fmt_ctx, AVMEDIA_TYPE_VIDEO) >= 0) { + video_stream = fmt_ctx->streams[video_stream_idx]; + video_dec_ctx = video_stream->codec; + + video_dst_file = fopen(video_dst_filename, "wb"); + if (!video_dst_file) { + fprintf(stderr, "Could not open destination file %s\n", video_dst_filename); + ret = 1; + goto end; + } + + /* allocate image where the decoded image will be put */ + width = video_dec_ctx->width; + height = video_dec_ctx->height; + pix_fmt = video_dec_ctx->pix_fmt; + ret = av_image_alloc(video_dst_data, video_dst_linesize, + width, height, pix_fmt, 1); + if (ret < 0) { + fprintf(stderr, "Could not allocate raw video buffer\n"); + goto end; + } + video_dst_bufsize = ret; + } + + if (open_codec_context(&audio_stream_idx, fmt_ctx, AVMEDIA_TYPE_AUDIO) >= 0) { + audio_stream = fmt_ctx->streams[audio_stream_idx]; + audio_dec_ctx = audio_stream->codec; + audio_dst_file = fopen(audio_dst_filename, "wb"); + if (!audio_dst_file) { + fprintf(stderr, "Could not open destination file %s\n", audio_dst_filename); + ret = 1; + goto end; + } + } + + /* dump input information to stderr */ + av_dump_format(fmt_ctx, 0, src_filename, 0); + + if (!audio_stream && !video_stream) { + fprintf(stderr, "Could not find audio or video stream in the input, aborting\n"); + ret = 1; + goto end; + } + + frame = av_frame_alloc(); + if (!frame) { + fprintf(stderr, "Could not allocate frame\n"); + ret = AVERROR(ENOMEM); + goto end; + } + + /* initialize packet, set data to NULL, let the demuxer fill it */ + av_init_packet(&pkt); + pkt.data = NULL; + pkt.size = 0; + + if (video_stream) + printf("Demuxing video from file '%s' into '%s'\n", src_filename, video_dst_filename); + if (audio_stream) + printf("Demuxing audio from file '%s' into '%s'\n", src_filename, audio_dst_filename); + + /* read frames from the file */ + while (av_read_frame(fmt_ctx, &pkt) >= 0) { + AVPacket orig_pkt = pkt; + do { + ret = decode_packet(&got_frame, 0); + if (ret < 0) + break; + pkt.data += ret; + pkt.size -= ret; + } while (pkt.size > 0); + av_packet_unref(&orig_pkt); + } + + /* flush cached frames */ + pkt.data = NULL; + pkt.size = 0; + do { + decode_packet(&got_frame, 1); + } while (got_frame); + + printf("Demuxing succeeded.\n"); + + if (video_stream) { + printf("Play the output video file with the command:\n" + "ffplay -f rawvideo -pix_fmt %s -video_size %dx%d %s\n", + av_get_pix_fmt_name(pix_fmt), width, height, + video_dst_filename); + } + + if (audio_stream) { + enum AVSampleFormat sfmt = audio_dec_ctx->sample_fmt; + int n_channels = audio_dec_ctx->channels; + const char *fmt; + + if (av_sample_fmt_is_planar(sfmt)) { + const char *packed = av_get_sample_fmt_name(sfmt); + printf("Warning: the sample format the decoder produced is planar " + "(%s). This example will output the first channel only.\n", + packed ? packed : "?"); + sfmt = av_get_packed_sample_fmt(sfmt); + n_channels = 1; + } + + if ((ret = get_format_from_sample_fmt(&fmt, sfmt)) < 0) + goto end; + + printf("Play the output audio file with the command:\n" + "ffplay -f %s -ac %d -ar %d %s\n", + fmt, n_channels, audio_dec_ctx->sample_rate, + audio_dst_filename); + } + +end: + avcodec_close(video_dec_ctx); + avcodec_close(audio_dec_ctx); + avformat_close_input(&fmt_ctx); + if (video_dst_file) + fclose(video_dst_file); + if (audio_dst_file) + fclose(audio_dst_file); + av_frame_free(&frame); + av_free(video_dst_data[0]); + + return ret < 0; +} diff --git a/doc/examples/extract_mvs.c b/doc/examples/extract_mvs.c new file mode 100644 index 0000000000..975189c77d --- /dev/null +++ b/doc/examples/extract_mvs.c @@ -0,0 +1,185 @@ +/* + * Copyright (c) 2012 Stefano Sabatini + * Copyright (c) 2014 Clément Bœsch + * + * Permission is hereby granted, free of charge, to any person obtaining a copy + * of this software and associated documentation files (the "Software"), to deal + * in the Software without restriction, including without limitation the rights + * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell + * copies of the Software, and to permit persons to whom the Software is + * furnished to do so, subject to the following conditions: + * + * The above copyright notice and this permission notice shall be included in + * all copies or substantial portions of the Software. + * + * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR + * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, + * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL + * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER + * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, + * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN + * THE SOFTWARE. + */ + +#include <libavutil/motion_vector.h> +#include <libavformat/avformat.h> + +static AVFormatContext *fmt_ctx = NULL; +static AVCodecContext *video_dec_ctx = NULL; +static AVStream *video_stream = NULL; +static const char *src_filename = NULL; + +static int video_stream_idx = -1; +static AVFrame *frame = NULL; +static AVPacket pkt; +static int video_frame_count = 0; + +static int decode_packet(int *got_frame, int cached) +{ + int decoded = pkt.size; + + *got_frame = 0; + + if (pkt.stream_index == video_stream_idx) { + int ret = avcodec_decode_video2(video_dec_ctx, frame, got_frame, &pkt); + if (ret < 0) { + fprintf(stderr, "Error decoding video frame (%s)\n", av_err2str(ret)); + return ret; + } + + if (*got_frame) { + int i; + AVFrameSideData *sd; + + video_frame_count++; + sd = av_frame_get_side_data(frame, AV_FRAME_DATA_MOTION_VECTORS); + if (sd) { + const AVMotionVector *mvs = (const AVMotionVector *)sd->data; + for (i = 0; i < sd->size / sizeof(*mvs); i++) { + const AVMotionVector *mv = &mvs[i]; + printf("%d,%2d,%2d,%2d,%4d,%4d,%4d,%4d,0x%"PRIx64"\n", + video_frame_count, mv->source, + mv->w, mv->h, mv->src_x, mv->src_y, + mv->dst_x, mv->dst_y, mv->flags); + } + } + } + } + + return decoded; +} + +static int open_codec_context(int *stream_idx, + AVFormatContext *fmt_ctx, enum AVMediaType type) +{ + int ret; + AVStream *st; + AVCodecContext *dec_ctx = NULL; + AVCodec *dec = NULL; + AVDictionary *opts = NULL; + + ret = av_find_best_stream(fmt_ctx, type, -1, -1, NULL, 0); + if (ret < 0) { + fprintf(stderr, "Could not find %s stream in input file '%s'\n", + av_get_media_type_string(type), src_filename); + return ret; + } else { + *stream_idx = ret; + st = fmt_ctx->streams[*stream_idx]; + + /* find decoder for the stream */ + dec_ctx = st->codec; + dec = avcodec_find_decoder(dec_ctx->codec_id); + if (!dec) { + fprintf(stderr, "Failed to find %s codec\n", + av_get_media_type_string(type)); + return AVERROR(EINVAL); + } + + /* Init the video decoder */ + av_dict_set(&opts, "flags2", "+export_mvs", 0); + if ((ret = avcodec_open2(dec_ctx, dec, &opts)) < 0) { + fprintf(stderr, "Failed to open %s codec\n", + av_get_media_type_string(type)); + return ret; + } + } + + return 0; +} + +int main(int argc, char **argv) +{ + int ret = 0, got_frame; + + if (argc != 2) { + fprintf(stderr, "Usage: %s <video>\n", argv[0]); + exit(1); + } + src_filename = argv[1]; + + av_register_all(); + + if (avformat_open_input(&fmt_ctx, src_filename, NULL, NULL) < 0) { + fprintf(stderr, "Could not open source file %s\n", src_filename); + exit(1); + } + + if (avformat_find_stream_info(fmt_ctx, NULL) < 0) { + fprintf(stderr, "Could not find stream information\n"); + exit(1); + } + + if (open_codec_context(&video_stream_idx, fmt_ctx, AVMEDIA_TYPE_VIDEO) >= 0) { + video_stream = fmt_ctx->streams[video_stream_idx]; + video_dec_ctx = video_stream->codec; + } + + av_dump_format(fmt_ctx, 0, src_filename, 0); + + if (!video_stream) { + fprintf(stderr, "Could not find video stream in the input, aborting\n"); + ret = 1; + goto end; + } + + frame = av_frame_alloc(); + if (!frame) { + fprintf(stderr, "Could not allocate frame\n"); + ret = AVERROR(ENOMEM); + goto end; + } + + printf("framenum,source,blockw,blockh,srcx,srcy,dstx,dsty,flags\n"); + + /* initialize packet, set data to NULL, let the demuxer fill it */ + av_init_packet(&pkt); + pkt.data = NULL; + pkt.size = 0; + + /* read frames from the file */ + while (av_read_frame(fmt_ctx, &pkt) >= 0) { + AVPacket orig_pkt = pkt; + do { + ret = decode_packet(&got_frame, 0); + if (ret < 0) + break; + pkt.data += ret; + pkt.size -= ret; + } while (pkt.size > 0); + av_packet_unref(&orig_pkt); + } + + /* flush cached frames */ + pkt.data = NULL; + pkt.size = 0; + do { + decode_packet(&got_frame, 1); + } while (got_frame); + +end: + avcodec_close(video_dec_ctx); + avformat_close_input(&fmt_ctx); + av_frame_free(&frame); + return ret < 0; +} diff --git a/doc/examples/filter_audio.c b/doc/examples/filter_audio.c index 60fe107dda..01761dcee4 100644 --- a/doc/examples/filter_audio.c +++ b/doc/examples/filter_audio.c @@ -1,20 +1,20 @@ /* * copyright (c) 2013 Andrew Kelley * - * This file is part of Libav. + * This file is part of FFmpeg. * - * Libav is free software; you can redistribute it and/or + * FFmpeg is free software; you can redistribute it and/or * modify it under the terms of the GNU Lesser General Public * License as published by the Free Software Foundation; either * version 2.1 of the License, or (at your option) any later version. * - * Libav is distributed in the hope that it will be useful, + * FFmpeg is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Lesser General Public License for more details. * * You should have received a copy of the GNU Lesser General Public - * License along with Libav; if not, write to the Free Software + * License along with FFmpeg; if not, write to the Free Software * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ diff --git a/doc/examples/filtering_audio.c b/doc/examples/filtering_audio.c new file mode 100644 index 0000000000..89c80cfd55 --- /dev/null +++ b/doc/examples/filtering_audio.c @@ -0,0 +1,295 @@ +/* + * Copyright (c) 2010 Nicolas George + * Copyright (c) 2011 Stefano Sabatini + * Copyright (c) 2012 Clément Bœsch + * + * Permission is hereby granted, free of charge, to any person obtaining a copy + * of this software and associated documentation files (the "Software"), to deal + * in the Software without restriction, including without limitation the rights + * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell + * copies of the Software, and to permit persons to whom the Software is + * furnished to do so, subject to the following conditions: + * + * The above copyright notice and this permission notice shall be included in + * all copies or substantial portions of the Software. + * + * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR + * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, + * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL + * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER + * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, + * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN + * THE SOFTWARE. + */ + +/** + * @file + * API example for audio decoding and filtering + * @example filtering_audio.c + */ + +#include <unistd.h> + +#include <libavcodec/avcodec.h> +#include <libavformat/avformat.h> +#include <libavfilter/avfiltergraph.h> +#include <libavfilter/buffersink.h> +#include <libavfilter/buffersrc.h> +#include <libavutil/opt.h> + +static const char *filter_descr = "aresample=8000,aformat=sample_fmts=s16:channel_layouts=mono"; +static const char *player = "ffplay -f s16le -ar 8000 -ac 1 -"; + +static AVFormatContext *fmt_ctx; +static AVCodecContext *dec_ctx; +AVFilterContext *buffersink_ctx; +AVFilterContext *buffersrc_ctx; +AVFilterGraph *filter_graph; +static int audio_stream_index = -1; + +static int open_input_file(const char *filename) +{ + int ret; + AVCodec *dec; + + if ((ret = avformat_open_input(&fmt_ctx, filename, NULL, NULL)) < 0) { + av_log(NULL, AV_LOG_ERROR, "Cannot open input file\n"); + return ret; + } + + if ((ret = avformat_find_stream_info(fmt_ctx, NULL)) < 0) { + av_log(NULL, AV_LOG_ERROR, "Cannot find stream information\n"); + return ret; + } + + /* select the audio stream */ + ret = av_find_best_stream(fmt_ctx, AVMEDIA_TYPE_AUDIO, -1, -1, &dec, 0); + if (ret < 0) { + av_log(NULL, AV_LOG_ERROR, "Cannot find a audio stream in the input file\n"); + return ret; + } + audio_stream_index = ret; + dec_ctx = fmt_ctx->streams[audio_stream_index]->codec; + av_opt_set_int(dec_ctx, "refcounted_frames", 1, 0); + + /* init the audio decoder */ + if ((ret = avcodec_open2(dec_ctx, dec, NULL)) < 0) { + av_log(NULL, AV_LOG_ERROR, "Cannot open audio decoder\n"); + return ret; + } + + return 0; +} + +static int init_filters(const char *filters_descr) +{ + char args[512]; + int ret = 0; + AVFilter *abuffersrc = avfilter_get_by_name("abuffer"); + AVFilter *abuffersink = avfilter_get_by_name("abuffersink"); + AVFilterInOut *outputs = avfilter_inout_alloc(); + AVFilterInOut *inputs = avfilter_inout_alloc(); + static const enum AVSampleFormat out_sample_fmts[] = { AV_SAMPLE_FMT_S16, -1 }; + static const int64_t out_channel_layouts[] = { AV_CH_LAYOUT_MONO, -1 }; + static const int out_sample_rates[] = { 8000, -1 }; + const AVFilterLink *outlink; + AVRational time_base = fmt_ctx->streams[audio_stream_index]->time_base; + + filter_graph = avfilter_graph_alloc(); + if (!outputs || !inputs || !filter_graph) { + ret = AVERROR(ENOMEM); + goto end; + } + + /* buffer audio source: the decoded frames from the decoder will be inserted here. */ + if (!dec_ctx->channel_layout) + dec_ctx->channel_layout = av_get_default_channel_layout(dec_ctx->channels); + snprintf(args, sizeof(args), + "time_base=%d/%d:sample_rate=%d:sample_fmt=%s:channel_layout=0x%"PRIx64, + time_base.num, time_base.den, dec_ctx->sample_rate, + av_get_sample_fmt_name(dec_ctx->sample_fmt), dec_ctx->channel_layout); + ret = avfilter_graph_create_filter(&buffersrc_ctx, abuffersrc, "in", + args, NULL, filter_graph); + if (ret < 0) { + av_log(NULL, AV_LOG_ERROR, "Cannot create audio buffer source\n"); + goto end; + } + + /* buffer audio sink: to terminate the filter chain. */ + ret = avfilter_graph_create_filter(&buffersink_ctx, abuffersink, "out", + NULL, NULL, filter_graph); + if (ret < 0) { + av_log(NULL, AV_LOG_ERROR, "Cannot create audio buffer sink\n"); + goto end; + } + + ret = av_opt_set_int_list(buffersink_ctx, "sample_fmts", out_sample_fmts, -1, + AV_OPT_SEARCH_CHILDREN); + if (ret < 0) { + av_log(NULL, AV_LOG_ERROR, "Cannot set output sample format\n"); + goto end; + } + + ret = av_opt_set_int_list(buffersink_ctx, "channel_layouts", out_channel_layouts, -1, + AV_OPT_SEARCH_CHILDREN); + if (ret < 0) { + av_log(NULL, AV_LOG_ERROR, "Cannot set output channel layout\n"); + goto end; + } + + ret = av_opt_set_int_list(buffersink_ctx, "sample_rates", out_sample_rates, -1, + AV_OPT_SEARCH_CHILDREN); + if (ret < 0) { + av_log(NULL, AV_LOG_ERROR, "Cannot set output sample rate\n"); + goto end; + } + + /* + * Set the endpoints for the filter graph. The filter_graph will + * be linked to the graph described by filters_descr. + */ + + /* + * The buffer source output must be connected to the input pad of + * the first filter described by filters_descr; since the first + * filter input label is not specified, it is set to "in" by + * default. + */ + outputs->name = av_strdup("in"); + outputs->filter_ctx = buffersrc_ctx; + outputs->pad_idx = 0; + outputs->next = NULL; + + /* + * The buffer sink input must be connected to the output pad of + * the last filter described by filters_descr; since the last + * filter output label is not specified, it is set to "out" by + * default. + */ + inputs->name = av_strdup("out"); + inputs->filter_ctx = buffersink_ctx; + inputs->pad_idx = 0; + inputs->next = NULL; + + if ((ret = avfilter_graph_parse_ptr(filter_graph, filters_descr, + &inputs, &outputs, NULL)) < 0) + goto end; + + if ((ret = avfilter_graph_config(filter_graph, NULL)) < 0) + goto end; + + /* Print summary of the sink buffer + * Note: args buffer is reused to store channel layout string */ + outlink = buffersink_ctx->inputs[0]; + av_get_channel_layout_string(args, sizeof(args), -1, outlink->channel_layout); + av_log(NULL, AV_LOG_INFO, "Output: srate:%dHz fmt:%s chlayout:%s\n", + (int)outlink->sample_rate, + (char *)av_x_if_null(av_get_sample_fmt_name(outlink->format), "?"), + args); + +end: + avfilter_inout_free(&inputs); + avfilter_inout_free(&outputs); + + return ret; +} + +static void print_frame(const AVFrame *frame) +{ + const int n = frame->nb_samples * av_get_channel_layout_nb_channels(av_frame_get_channel_layout(frame)); + const uint16_t *p = (uint16_t*)frame->data[0]; + const uint16_t *p_end = p + n; + + while (p < p_end) { + fputc(*p & 0xff, stdout); + fputc(*p>>8 & 0xff, stdout); + p++; + } + fflush(stdout); +} + +int main(int argc, char **argv) +{ + int ret; + AVPacket packet0, packet; + AVFrame *frame = av_frame_alloc(); + AVFrame *filt_frame = av_frame_alloc(); + int got_frame; + + if (!frame || !filt_frame) { + perror("Could not allocate frame"); + exit(1); + } + if (argc != 2) { + fprintf(stderr, "Usage: %s file | %s\n", argv[0], player); + exit(1); + } + + av_register_all(); + avfilter_register_all(); + + if ((ret = open_input_file(argv[1])) < 0) + goto end; + if ((ret = init_filters(filter_descr)) < 0) + goto end; + + /* read all packets */ + packet0.data = NULL; + packet.data = NULL; + while (1) { + if (!packet0.data) { + if ((ret = av_read_frame(fmt_ctx, &packet)) < 0) + break; + packet0 = packet; + } + + if (packet.stream_index == audio_stream_index) { + got_frame = 0; + ret = avcodec_decode_audio4(dec_ctx, frame, &got_frame, &packet); + if (ret < 0) { + av_log(NULL, AV_LOG_ERROR, "Error decoding audio\n"); + continue; + } + packet.size -= ret; + packet.data += ret; + + if (got_frame) { + /* push the audio data from decoded frame into the filtergraph */ + if (av_buffersrc_add_frame_flags(buffersrc_ctx, frame, 0) < 0) { + av_log(NULL, AV_LOG_ERROR, "Error while feeding the audio filtergraph\n"); + break; + } + + /* pull filtered audio from the filtergraph */ + while (1) { + ret = av_buffersink_get_frame(buffersink_ctx, filt_frame); + if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF) + break; + if (ret < 0) + goto end; + print_frame(filt_frame); + av_frame_unref(filt_frame); + } + } + + if (packet.size <= 0) + av_packet_unref(&packet0); + } else { + /* discard non-wanted packets */ + av_packet_unref(&packet0); + } + } +end: + avfilter_graph_free(&filter_graph); + avcodec_close(dec_ctx); + avformat_close_input(&fmt_ctx); + av_frame_free(&frame); + av_frame_free(&filt_frame); + + if (ret < 0 && ret != AVERROR_EOF) { + fprintf(stderr, "Error occurred: %s\n", av_err2str(ret)); + exit(1); + } + + exit(0); +} diff --git a/doc/examples/filtering_video.c b/doc/examples/filtering_video.c new file mode 100644 index 0000000000..3dabf13b10 --- /dev/null +++ b/doc/examples/filtering_video.c @@ -0,0 +1,280 @@ +/* + * Copyright (c) 2010 Nicolas George + * Copyright (c) 2011 Stefano Sabatini + * + * Permission is hereby granted, free of charge, to any person obtaining a copy + * of this software and associated documentation files (the "Software"), to deal + * in the Software without restriction, including without limitation the rights + * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell + * copies of the Software, and to permit persons to whom the Software is + * furnished to do so, subject to the following conditions: + * + * The above copyright notice and this permission notice shall be included in + * all copies or substantial portions of the Software. + * + * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR + * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, + * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL + * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER + * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, + * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN + * THE SOFTWARE. + */ + +/** + * @file + * API example for decoding and filtering + * @example filtering_video.c + */ + +#define _XOPEN_SOURCE 600 /* for usleep */ +#include <unistd.h> + +#include <libavcodec/avcodec.h> +#include <libavformat/avformat.h> +#include <libavfilter/avfiltergraph.h> +#include <libavfilter/buffersink.h> +#include <libavfilter/buffersrc.h> +#include <libavutil/opt.h> + +const char *filter_descr = "scale=78:24,transpose=cclock"; +/* other way: + scale=78:24 [scl]; [scl] transpose=cclock // assumes "[in]" and "[out]" to be input output pads respectively + */ + +static AVFormatContext *fmt_ctx; +static AVCodecContext *dec_ctx; +AVFilterContext *buffersink_ctx; +AVFilterContext *buffersrc_ctx; +AVFilterGraph *filter_graph; +static int video_stream_index = -1; +static int64_t last_pts = AV_NOPTS_VALUE; + +static int open_input_file(const char *filename) +{ + int ret; + AVCodec *dec; + + if ((ret = avformat_open_input(&fmt_ctx, filename, NULL, NULL)) < 0) { + av_log(NULL, AV_LOG_ERROR, "Cannot open input file\n"); + return ret; + } + + if ((ret = avformat_find_stream_info(fmt_ctx, NULL)) < 0) { + av_log(NULL, AV_LOG_ERROR, "Cannot find stream information\n"); + return ret; + } + + /* select the video stream */ + ret = av_find_best_stream(fmt_ctx, AVMEDIA_TYPE_VIDEO, -1, -1, &dec, 0); + if (ret < 0) { + av_log(NULL, AV_LOG_ERROR, "Cannot find a video stream in the input file\n"); + return ret; + } + video_stream_index = ret; + dec_ctx = fmt_ctx->streams[video_stream_index]->codec; + av_opt_set_int(dec_ctx, "refcounted_frames", 1, 0); + + /* init the video decoder */ + if ((ret = avcodec_open2(dec_ctx, dec, NULL)) < 0) { + av_log(NULL, AV_LOG_ERROR, "Cannot open video decoder\n"); + return ret; + } + + return 0; +} + +static int init_filters(const char *filters_descr) +{ + char args[512]; + int ret = 0; + AVFilter *buffersrc = avfilter_get_by_name("buffer"); + AVFilter *buffersink = avfilter_get_by_name("buffersink"); + AVFilterInOut *outputs = avfilter_inout_alloc(); + AVFilterInOut *inputs = avfilter_inout_alloc(); + AVRational time_base = fmt_ctx->streams[video_stream_index]->time_base; + enum AVPixelFormat pix_fmts[] = { AV_PIX_FMT_GRAY8, AV_PIX_FMT_NONE }; + + filter_graph = avfilter_graph_alloc(); + if (!outputs || !inputs || !filter_graph) { + ret = AVERROR(ENOMEM); + goto end; + } + + /* buffer video source: the decoded frames from the decoder will be inserted here. */ + snprintf(args, sizeof(args), + "video_size=%dx%d:pix_fmt=%d:time_base=%d/%d:pixel_aspect=%d/%d", + dec_ctx->width, dec_ctx->height, dec_ctx->pix_fmt, + time_base.num, time_base.den, + dec_ctx->sample_aspect_ratio.num, dec_ctx->sample_aspect_ratio.den); + + ret = avfilter_graph_create_filter(&buffersrc_ctx, buffersrc, "in", + args, NULL, filter_graph); + if (ret < 0) { + av_log(NULL, AV_LOG_ERROR, "Cannot create buffer source\n"); + goto end; + } + + /* buffer video sink: to terminate the filter chain. */ + ret = avfilter_graph_create_filter(&buffersink_ctx, buffersink, "out", + NULL, NULL, filter_graph); + if (ret < 0) { + av_log(NULL, AV_LOG_ERROR, "Cannot create buffer sink\n"); + goto end; + } + + ret = av_opt_set_int_list(buffersink_ctx, "pix_fmts", pix_fmts, + AV_PIX_FMT_NONE, AV_OPT_SEARCH_CHILDREN); + if (ret < 0) { + av_log(NULL, AV_LOG_ERROR, "Cannot set output pixel format\n"); + goto end; + } + + /* + * Set the endpoints for the filter graph. The filter_graph will + * be linked to the graph described by filters_descr. + */ + + /* + * The buffer source output must be connected to the input pad of + * the first filter described by filters_descr; since the first + * filter input label is not specified, it is set to "in" by + * default. + */ + outputs->name = av_strdup("in"); + outputs->filter_ctx = buffersrc_ctx; + outputs->pad_idx = 0; + outputs->next = NULL; + + /* + * The buffer sink input must be connected to the output pad of + * the last filter described by filters_descr; since the last + * filter output label is not specified, it is set to "out" by + * default. + */ + inputs->name = av_strdup("out"); + inputs->filter_ctx = buffersink_ctx; + inputs->pad_idx = 0; + inputs->next = NULL; + + if ((ret = avfilter_graph_parse_ptr(filter_graph, filters_descr, + &inputs, &outputs, NULL)) < 0) + goto end; + + if ((ret = avfilter_graph_config(filter_graph, NULL)) < 0) + goto end; + +end: + avfilter_inout_free(&inputs); + avfilter_inout_free(&outputs); + + return ret; +} + +static void display_frame(const AVFrame *frame, AVRational time_base) +{ + int x, y; + uint8_t *p0, *p; + int64_t delay; + + if (frame->pts != AV_NOPTS_VALUE) { + if (last_pts != AV_NOPTS_VALUE) { + /* sleep roughly the right amount of time; + * usleep is in microseconds, just like AV_TIME_BASE. */ + delay = av_rescale_q(frame->pts - last_pts, + time_base, AV_TIME_BASE_Q); + if (delay > 0 && delay < 1000000) + usleep(delay); + } + last_pts = frame->pts; + } + + /* Trivial ASCII grayscale display. */ + p0 = frame->data[0]; + puts("\033c"); + for (y = 0; y < frame->height; y++) { + p = p0; + for (x = 0; x < frame->width; x++) + putchar(" .-+#"[*(p++) / 52]); + putchar('\n'); + p0 += frame->linesize[0]; + } + fflush(stdout); +} + +int main(int argc, char **argv) +{ + int ret; + AVPacket packet; + AVFrame *frame = av_frame_alloc(); + AVFrame *filt_frame = av_frame_alloc(); + int got_frame; + + if (!frame || !filt_frame) { + perror("Could not allocate frame"); + exit(1); + } + if (argc != 2) { + fprintf(stderr, "Usage: %s file\n", argv[0]); + exit(1); + } + + av_register_all(); + avfilter_register_all(); + + if ((ret = open_input_file(argv[1])) < 0) + goto end; + if ((ret = init_filters(filter_descr)) < 0) + goto end; + + /* read all packets */ + while (1) { + if ((ret = av_read_frame(fmt_ctx, &packet)) < 0) + break; + + if (packet.stream_index == video_stream_index) { + got_frame = 0; + ret = avcodec_decode_video2(dec_ctx, frame, &got_frame, &packet); + if (ret < 0) { + av_log(NULL, AV_LOG_ERROR, "Error decoding video\n"); + break; + } + + if (got_frame) { + frame->pts = av_frame_get_best_effort_timestamp(frame); + + /* push the decoded frame into the filtergraph */ + if (av_buffersrc_add_frame_flags(buffersrc_ctx, frame, AV_BUFFERSRC_FLAG_KEEP_REF) < 0) { + av_log(NULL, AV_LOG_ERROR, "Error while feeding the filtergraph\n"); + break; + } + + /* pull filtered frames from the filtergraph */ + while (1) { + ret = av_buffersink_get_frame(buffersink_ctx, filt_frame); + if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF) + break; + if (ret < 0) + goto end; + display_frame(filt_frame, buffersink_ctx->inputs[0]->time_base); + av_frame_unref(filt_frame); + } + av_frame_unref(frame); + } + } + av_packet_unref(&packet); + } +end: + avfilter_graph_free(&filter_graph); + avcodec_close(dec_ctx); + avformat_close_input(&fmt_ctx); + av_frame_free(&frame); + av_frame_free(&filt_frame); + + if (ret < 0 && ret != AVERROR_EOF) { + fprintf(stderr, "Error occurred: %s\n", av_err2str(ret)); + exit(1); + } + + exit(0); +} diff --git a/doc/examples/http_multiclient.c b/doc/examples/http_multiclient.c new file mode 100644 index 0000000000..b9a306d835 --- /dev/null +++ b/doc/examples/http_multiclient.c @@ -0,0 +1,155 @@ +/* + * Copyright (c) 2015 Stephan Holljes + * + * Permission is hereby granted, free of charge, to any person obtaining a copy + * of this software and associated documentation files (the "Software"), to deal + * in the Software without restriction, including without limitation the rights + * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell + * copies of the Software, and to permit persons to whom the Software is + * furnished to do so, subject to the following conditions: + * + * The above copyright notice and this permission notice shall be included in + * all copies or substantial portions of the Software. + * + * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR + * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, + * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL + * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER + * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, + * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN + * THE SOFTWARE. + */ + +/** + * @file + * libavformat multi-client network API usage example. + * + * @example http_multiclient.c + * This example will serve a file without decoding or demuxing it over http. + * Multiple clients can connect and will receive the same file. + */ + +#include <libavformat/avformat.h> +#include <libavutil/opt.h> +#include <unistd.h> + +void process_client(AVIOContext *client, const char *in_uri) +{ + AVIOContext *input = NULL; + uint8_t buf[1024]; + int ret, n, reply_code; + char *resource = NULL; + while ((ret = avio_handshake(client)) > 0) { + av_opt_get(client, "resource", AV_OPT_SEARCH_CHILDREN, &resource); + // check for strlen(resource) is necessary, because av_opt_get() + // may return empty string. + if (resource && strlen(resource)) + break; + } + if (ret < 0) + goto end; + av_log(client, AV_LOG_TRACE, "resource=%p\n", resource); + if (resource && resource[0] == '/' && !strcmp((resource + 1), in_uri)) { + reply_code = 200; + } else { + reply_code = AVERROR_HTTP_NOT_FOUND; + } + if ((ret = av_opt_set_int(client, "reply_code", reply_code, AV_OPT_SEARCH_CHILDREN)) < 0) { + av_log(client, AV_LOG_ERROR, "Failed to set reply_code: %s.\n", av_err2str(ret)); + goto end; + } + av_log(client, AV_LOG_TRACE, "Set reply code to %d\n", reply_code); + + while ((ret = avio_handshake(client)) > 0); + + if (ret < 0) + goto end; + + fprintf(stderr, "Handshake performed.\n"); + if (reply_code != 200) + goto end; + fprintf(stderr, "Opening input file.\n"); + if ((ret = avio_open2(&input, in_uri, AVIO_FLAG_READ, NULL, NULL)) < 0) { + av_log(input, AV_LOG_ERROR, "Failed to open input: %s: %s.\n", in_uri, + av_err2str(ret)); + goto end; + } + for(;;) { + n = avio_read(input, buf, sizeof(buf)); + if (n < 0) { + if (n == AVERROR_EOF) + break; + av_log(input, AV_LOG_ERROR, "Error reading from input: %s.\n", + av_err2str(n)); + break; + } + avio_write(client, buf, n); + avio_flush(client); + } +end: + fprintf(stderr, "Flushing client\n"); + avio_flush(client); + fprintf(stderr, "Closing client\n"); + avio_close(client); + fprintf(stderr, "Closing input\n"); + avio_close(input); +} + +int main(int argc, char **argv) +{ + av_log_set_level(AV_LOG_TRACE); + AVDictionary *options = NULL; + AVIOContext *client = NULL, *server = NULL; + const char *in_uri, *out_uri; + int ret, pid; + if (argc < 3) { + printf("usage: %s input http://hostname[:port]\n" + "API example program to serve http to multiple clients.\n" + "\n", argv[0]); + return 1; + } + + in_uri = argv[1]; + out_uri = argv[2]; + + av_register_all(); + avformat_network_init(); + + if ((ret = av_dict_set(&options, "listen", "2", 0)) < 0) { + fprintf(stderr, "Failed to set listen mode for server: %s\n", av_err2str(ret)); + return ret; + } + if ((ret = avio_open2(&server, out_uri, AVIO_FLAG_WRITE, NULL, &options)) < 0) { + fprintf(stderr, "Failed to open server: %s\n", av_err2str(ret)); + return ret; + } + fprintf(stderr, "Entering main loop.\n"); + for(;;) { + if ((ret = avio_accept(server, &client)) < 0) + goto end; + fprintf(stderr, "Accepted client, forking process.\n"); + // XXX: Since we don't reap our children and don't ignore signals + // this produces zombie processes. + pid = fork(); + if (pid < 0) { + perror("Fork failed"); + ret = AVERROR(errno); + goto end; + } + if (pid == 0) { + fprintf(stderr, "In child.\n"); + process_client(client, in_uri); + avio_close(server); + exit(0); + } + if (pid > 0) + avio_close(client); + } +end: + avio_close(server); + if (ret < 0 && ret != AVERROR_EOF) { + fprintf(stderr, "Some errors occurred: %s\n", av_err2str(ret)); + return 1; + } + return 0; +} diff --git a/doc/examples/metadata.c b/doc/examples/metadata.c index f4c6eee9c3..f73c267369 100644 --- a/doc/examples/metadata.c +++ b/doc/examples/metadata.c @@ -22,8 +22,8 @@ /** * @file - * @example metadata.c * Shows how the metadata API can be used in application programs. + * @example metadata.c */ #include <stdio.h> @@ -51,6 +51,6 @@ int main (int argc, char **argv) while ((tag = av_dict_get(fmt_ctx->metadata, "", tag, AV_DICT_IGNORE_SUFFIX))) printf("%s=%s\n", tag->key, tag->value); - avformat_free_context(fmt_ctx); + avformat_close_input(&fmt_ctx); return 0; } diff --git a/doc/examples/output.c b/doc/examples/muxing.c index c883429cfc..d4dac5cd69 100644 --- a/doc/examples/output.c +++ b/doc/examples/muxing.c @@ -24,9 +24,9 @@ * @file * libavformat API example. * - * @example output.c * Output a media file in any supported libavformat format. The default * codecs are used. + * @example muxing.c */ #include <stdlib.h> @@ -34,17 +34,17 @@ #include <string.h> #include <math.h> -#include "libavutil/channel_layout.h" -#include "libavutil/mathematics.h" -#include "libavutil/opt.h" -#include "libavformat/avformat.h" -#include "libavresample/avresample.h" -#include "libswscale/swscale.h" +#include <libavutil/avassert.h> +#include <libavutil/channel_layout.h> +#include <libavutil/opt.h> +#include <libavutil/mathematics.h> +#include <libavutil/timestamp.h> +#include <libavformat/avformat.h> +#include <libswscale/swscale.h> +#include <libswresample/swresample.h> -/* 5 seconds stream duration */ -#define STREAM_DURATION 5.0 +#define STREAM_DURATION 10.0 #define STREAM_FRAME_RATE 25 /* 25 images/s */ -#define STREAM_NB_FRAMES ((int)(STREAM_DURATION * STREAM_FRAME_RATE)) #define STREAM_PIX_FMT AV_PIX_FMT_YUV420P /* default pix_fmt */ #define SCALE_FLAGS SWS_BICUBIC @@ -55,6 +55,7 @@ typedef struct OutputStream { /* pts of the next frame that will be generated */ int64_t next_pts; + int samples_count; AVFrame *frame; AVFrame *tmp_frame; @@ -62,75 +63,121 @@ typedef struct OutputStream { float t, tincr, tincr2; struct SwsContext *sws_ctx; - AVAudioResampleContext *avr; + struct SwrContext *swr_ctx; } OutputStream; -/**************************************************************/ -/* audio output */ +static void log_packet(const AVFormatContext *fmt_ctx, const AVPacket *pkt) +{ + AVRational *time_base = &fmt_ctx->streams[pkt->stream_index]->time_base; -/* - * add an audio output stream - */ -static void add_audio_stream(OutputStream *ost, AVFormatContext *oc, - enum AVCodecID codec_id) + printf("pts:%s pts_time:%s dts:%s dts_time:%s duration:%s duration_time:%s stream_index:%d\n", + av_ts2str(pkt->pts), av_ts2timestr(pkt->pts, time_base), + av_ts2str(pkt->dts), av_ts2timestr(pkt->dts, time_base), + av_ts2str(pkt->duration), av_ts2timestr(pkt->duration, time_base), + pkt->stream_index); +} + +static int write_frame(AVFormatContext *fmt_ctx, const AVRational *time_base, AVStream *st, AVPacket *pkt) +{ + /* rescale output packet timestamp values from codec to stream timebase */ + av_packet_rescale_ts(pkt, *time_base, st->time_base); + pkt->stream_index = st->index; + + /* Write the compressed frame to the media file. */ + log_packet(fmt_ctx, pkt); + return av_interleaved_write_frame(fmt_ctx, pkt); +} + +/* Add an output stream. */ +static void add_stream(OutputStream *ost, AVFormatContext *oc, + AVCodec **codec, + enum AVCodecID codec_id) { AVCodecContext *c; - AVCodec *codec; - int ret; + int i; - /* find the audio encoder */ - codec = avcodec_find_encoder(codec_id); - if (!codec) { - fprintf(stderr, "codec not found\n"); + /* find the encoder */ + *codec = avcodec_find_encoder(codec_id); + if (!(*codec)) { + fprintf(stderr, "Could not find encoder for '%s'\n", + avcodec_get_name(codec_id)); exit(1); } - ost->st = avformat_new_stream(oc, codec); + ost->st = avformat_new_stream(oc, *codec); if (!ost->st) { - fprintf(stderr, "Could not alloc stream\n"); + fprintf(stderr, "Could not allocate stream\n"); exit(1); } - + ost->st->id = oc->nb_streams-1; c = ost->st->codec; - /* put sample parameters */ - c->sample_fmt = codec->sample_fmts ? codec->sample_fmts[0] : AV_SAMPLE_FMT_S16; - c->sample_rate = codec->supported_samplerates ? codec->supported_samplerates[0] : 44100; - c->channel_layout = codec->channel_layouts ? codec->channel_layouts[0] : AV_CH_LAYOUT_STEREO; - c->channels = av_get_channel_layout_nb_channels(c->channel_layout); - c->bit_rate = 64000; + switch ((*codec)->type) { + case AVMEDIA_TYPE_AUDIO: + c->sample_fmt = (*codec)->sample_fmts ? + (*codec)->sample_fmts[0] : AV_SAMPLE_FMT_FLTP; + c->bit_rate = 64000; + c->sample_rate = 44100; + if ((*codec)->supported_samplerates) { + c->sample_rate = (*codec)->supported_samplerates[0]; + for (i = 0; (*codec)->supported_samplerates[i]; i++) { + if ((*codec)->supported_samplerates[i] == 44100) + c->sample_rate = 44100; + } + } + c->channels = av_get_channel_layout_nb_channels(c->channel_layout); + c->channel_layout = AV_CH_LAYOUT_STEREO; + if ((*codec)->channel_layouts) { + c->channel_layout = (*codec)->channel_layouts[0]; + for (i = 0; (*codec)->channel_layouts[i]; i++) { + if ((*codec)->channel_layouts[i] == AV_CH_LAYOUT_STEREO) + c->channel_layout = AV_CH_LAYOUT_STEREO; + } + } + c->channels = av_get_channel_layout_nb_channels(c->channel_layout); + ost->st->time_base = (AVRational){ 1, c->sample_rate }; + break; + + case AVMEDIA_TYPE_VIDEO: + c->codec_id = codec_id; + + c->bit_rate = 400000; + /* Resolution must be a multiple of two. */ + c->width = 352; + c->height = 288; + /* timebase: This is the fundamental unit of time (in seconds) in terms + * of which frame timestamps are represented. For fixed-fps content, + * timebase should be 1/framerate and timestamp increments should be + * identical to 1. */ + ost->st->time_base = (AVRational){ 1, STREAM_FRAME_RATE }; + c->time_base = ost->st->time_base; + + c->gop_size = 12; /* emit one intra frame every twelve frames at most */ + c->pix_fmt = STREAM_PIX_FMT; + if (c->codec_id == AV_CODEC_ID_MPEG2VIDEO) { + /* just for testing, we also add B frames */ + c->max_b_frames = 2; + } + if (c->codec_id == AV_CODEC_ID_MPEG1VIDEO) { + /* Needed to avoid using macroblocks in which some coeffs overflow. + * This does not happen with normal video, it just happens here as + * the motion of the chroma plane does not match the luma plane. */ + c->mb_decision = 2; + } + break; - ost->st->time_base = (AVRational){ 1, c->sample_rate }; + default: + break; + } - // some formats want stream headers to be separate + /* Some formats want stream headers to be separate. */ if (oc->oformat->flags & AVFMT_GLOBALHEADER) c->flags |= AV_CODEC_FLAG_GLOBAL_HEADER; - - /* initialize sample format conversion; - * to simplify the code, we always pass the data through lavr, even - * if the encoder supports the generated format directly -- the price is - * some extra data copying; - */ - ost->avr = avresample_alloc_context(); - if (!ost->avr) { - fprintf(stderr, "Error allocating the resampling context\n"); - exit(1); - } - - av_opt_set_int(ost->avr, "in_sample_fmt", AV_SAMPLE_FMT_S16, 0); - av_opt_set_int(ost->avr, "in_sample_rate", 44100, 0); - av_opt_set_int(ost->avr, "in_channel_layout", AV_CH_LAYOUT_STEREO, 0); - av_opt_set_int(ost->avr, "out_sample_fmt", c->sample_fmt, 0); - av_opt_set_int(ost->avr, "out_sample_rate", c->sample_rate, 0); - av_opt_set_int(ost->avr, "out_channel_layout", c->channel_layout, 0); - - ret = avresample_open(ost->avr); - if (ret < 0) { - fprintf(stderr, "Error opening the resampling context\n"); - exit(1); - } } +/**************************************************************/ +/* audio output */ + static AVFrame *alloc_audio_frame(enum AVSampleFormat sample_fmt, uint64_t channel_layout, int sample_rate, int nb_samples) @@ -159,16 +206,21 @@ static AVFrame *alloc_audio_frame(enum AVSampleFormat sample_fmt, return frame; } -static void open_audio(AVFormatContext *oc, OutputStream *ost) +static void open_audio(AVFormatContext *oc, AVCodec *codec, OutputStream *ost, AVDictionary *opt_arg) { AVCodecContext *c; int nb_samples; + int ret; + AVDictionary *opt = NULL; c = ost->st->codec; /* open it */ - if (avcodec_open2(c, NULL, NULL) < 0) { - fprintf(stderr, "could not open codec\n"); + av_dict_copy(&opt, opt_arg, 0); + ret = avcodec_open2(c, codec, &opt); + av_dict_free(&opt); + if (ret < 0) { + fprintf(stderr, "Could not open audio codec: %s\n", av_err2str(ret)); exit(1); } @@ -185,8 +237,29 @@ static void open_audio(AVFormatContext *oc, OutputStream *ost) ost->frame = alloc_audio_frame(c->sample_fmt, c->channel_layout, c->sample_rate, nb_samples); - ost->tmp_frame = alloc_audio_frame(AV_SAMPLE_FMT_S16, AV_CH_LAYOUT_STEREO, - 44100, nb_samples); + ost->tmp_frame = alloc_audio_frame(AV_SAMPLE_FMT_S16, c->channel_layout, + c->sample_rate, nb_samples); + + /* create resampler context */ + ost->swr_ctx = swr_alloc(); + if (!ost->swr_ctx) { + fprintf(stderr, "Could not allocate resampler context\n"); + exit(1); + } + + /* set options */ + av_opt_set_int (ost->swr_ctx, "in_channel_count", c->channels, 0); + av_opt_set_int (ost->swr_ctx, "in_sample_rate", c->sample_rate, 0); + av_opt_set_sample_fmt(ost->swr_ctx, "in_sample_fmt", AV_SAMPLE_FMT_S16, 0); + av_opt_set_int (ost->swr_ctx, "out_channel_count", c->channels, 0); + av_opt_set_int (ost->swr_ctx, "out_sample_rate", c->sample_rate, 0); + av_opt_set_sample_fmt(ost->swr_ctx, "out_sample_fmt", c->sample_fmt, 0); + + /* initialize the resampling context */ + if ((ret = swr_init(ost->swr_ctx)) < 0) { + fprintf(stderr, "Failed to initialize the resampling context\n"); + exit(1); + } } /* Prepare a 16 bit dummy audio frame of 'frame_size' samples and @@ -202,8 +275,7 @@ static AVFrame *get_audio_frame(OutputStream *ost) STREAM_DURATION, (AVRational){ 1, 1 }) >= 0) return NULL; - - for (j = 0; j < frame->nb_samples; j++) { + for (j = 0; j <frame->nb_samples; j++) { v = (int)(sin(ost->t) * 10000); for (i = 0; i < ost->st->codec->channels; i++) *q++ = v; @@ -211,62 +283,37 @@ static AVFrame *get_audio_frame(OutputStream *ost) ost->tincr += ost->tincr2; } - return frame; -} - -/* if a frame is provided, send it to the encoder, otherwise flush the encoder; - * return 1 when encoding is finished, 0 otherwise - */ -static int encode_audio_frame(AVFormatContext *oc, OutputStream *ost, - AVFrame *frame) -{ - AVPacket pkt = { 0 }; // data and size must be 0; - int got_packet; - - av_init_packet(&pkt); - avcodec_encode_audio2(ost->st->codec, &pkt, frame, &got_packet); - - if (got_packet) { - pkt.stream_index = ost->st->index; - - av_packet_rescale_ts(&pkt, ost->st->codec->time_base, ost->st->time_base); + frame->pts = ost->next_pts; + ost->next_pts += frame->nb_samples; - /* Write the compressed frame to the media file. */ - if (av_interleaved_write_frame(oc, &pkt) != 0) { - fprintf(stderr, "Error while writing audio frame\n"); - exit(1); - } - } - - return (frame || got_packet) ? 0 : 1; + return frame; } /* * encode one audio frame and send it to the muxer * return 1 when encoding is finished, 0 otherwise */ -static int process_audio_stream(AVFormatContext *oc, OutputStream *ost) +static int write_audio_frame(AVFormatContext *oc, OutputStream *ost) { + AVCodecContext *c; + AVPacket pkt = { 0 }; // data and size must be 0; AVFrame *frame; - int got_output = 0; int ret; + int got_packet; + int dst_nb_samples; + + av_init_packet(&pkt); + c = ost->st->codec; frame = get_audio_frame(ost); - got_output |= !!frame; - /* feed the data to lavr */ if (frame) { - ret = avresample_convert(ost->avr, NULL, 0, 0, - frame->extended_data, frame->linesize[0], - frame->nb_samples); - if (ret < 0) { - fprintf(stderr, "Error feeding audio data to the resampler\n"); - exit(1); - } - } + /* convert samples from native format to destination codec format, using the resampler */ + /* compute destination number of samples */ + dst_nb_samples = av_rescale_rnd(swr_get_delay(ost->swr_ctx, c->sample_rate) + frame->nb_samples, + c->sample_rate, c->sample_rate, AV_ROUND_UP); + av_assert0(dst_nb_samples == frame->nb_samples); - while ((frame && avresample_available(ost->avr) >= ost->frame->nb_samples) || - (!frame && avresample_get_out_samples(ost->avr, 0))) { /* when we pass a frame to the encoder, it may keep a reference to it * internally; * make sure we do not overwrite it here @@ -275,92 +322,41 @@ static int process_audio_stream(AVFormatContext *oc, OutputStream *ost) if (ret < 0) exit(1); - /* the difference between the two avresample calls here is that the - * first one just reads the already converted data that is buffered in - * the lavr output buffer, while the second one also flushes the - * resampler */ - if (frame) { - ret = avresample_read(ost->avr, ost->frame->extended_data, - ost->frame->nb_samples); - } else { - ret = avresample_convert(ost->avr, ost->frame->extended_data, - ost->frame->linesize[0], ost->frame->nb_samples, - NULL, 0, 0); - } + /* convert to destination format */ + ret = swr_convert(ost->swr_ctx, + ost->frame->data, dst_nb_samples, + (const uint8_t **)frame->data, frame->nb_samples); + if (ret < 0) { + fprintf(stderr, "Error while converting\n"); + exit(1); + } + frame = ost->frame; + + frame->pts = av_rescale_q(ost->samples_count, (AVRational){1, c->sample_rate}, c->time_base); + ost->samples_count += dst_nb_samples; + } + + ret = avcodec_encode_audio2(c, &pkt, frame, &got_packet); + if (ret < 0) { + fprintf(stderr, "Error encoding audio frame: %s\n", av_err2str(ret)); + exit(1); + } + if (got_packet) { + ret = write_frame(oc, &c->time_base, ost->st, &pkt); if (ret < 0) { - fprintf(stderr, "Error while resampling\n"); - exit(1); - } else if (frame && ret != ost->frame->nb_samples) { - fprintf(stderr, "Too few samples returned from lavr\n"); + fprintf(stderr, "Error while writing audio frame: %s\n", + av_err2str(ret)); exit(1); } - - ost->frame->nb_samples = ret; - - ost->frame->pts = ost->next_pts; - ost->next_pts += ost->frame->nb_samples; - - got_output |= encode_audio_frame(oc, ost, ret ? ost->frame : NULL); } - return !got_output; + return (frame || got_packet) ? 0 : 1; } /**************************************************************/ /* video output */ -/* Add a video output stream. */ -static void add_video_stream(OutputStream *ost, AVFormatContext *oc, - enum AVCodecID codec_id) -{ - AVCodecContext *c; - AVCodec *codec; - - /* find the video encoder */ - codec = avcodec_find_encoder(codec_id); - if (!codec) { - fprintf(stderr, "codec not found\n"); - exit(1); - } - - ost->st = avformat_new_stream(oc, codec); - if (!ost->st) { - fprintf(stderr, "Could not alloc stream\n"); - exit(1); - } - - c = ost->st->codec; - - /* Put sample parameters. */ - c->bit_rate = 400000; - /* Resolution must be a multiple of two. */ - c->width = 352; - c->height = 288; - /* timebase: This is the fundamental unit of time (in seconds) in terms - * of which frame timestamps are represented. For fixed-fps content, - * timebase should be 1/framerate and timestamp increments should be - * identical to 1. */ - ost->st->time_base = (AVRational){ 1, STREAM_FRAME_RATE }; - c->time_base = ost->st->time_base; - - c->gop_size = 12; /* emit one intra frame every twelve frames at most */ - c->pix_fmt = STREAM_PIX_FMT; - if (c->codec_id == AV_CODEC_ID_MPEG2VIDEO) { - /* just for testing, we also add B frames */ - c->max_b_frames = 2; - } - if (c->codec_id == AV_CODEC_ID_MPEG1VIDEO) { - /* Needed to avoid using macroblocks in which some coeffs overflow. - * This does not happen with normal video, it just happens here as - * the motion of the chroma plane does not match the luma plane. */ - c->mb_decision = 2; - } - /* Some formats want stream headers to be separate. */ - if (oc->oformat->flags & AVFMT_GLOBALHEADER) - c->flags |= AV_CODEC_FLAG_GLOBAL_HEADER; -} - static AVFrame *alloc_picture(enum AVPixelFormat pix_fmt, int width, int height) { AVFrame *picture; @@ -384,22 +380,26 @@ static AVFrame *alloc_picture(enum AVPixelFormat pix_fmt, int width, int height) return picture; } -static void open_video(AVFormatContext *oc, OutputStream *ost) +static void open_video(AVFormatContext *oc, AVCodec *codec, OutputStream *ost, AVDictionary *opt_arg) { - AVCodecContext *c; + int ret; + AVCodecContext *c = ost->st->codec; + AVDictionary *opt = NULL; - c = ost->st->codec; + av_dict_copy(&opt, opt_arg, 0); /* open the codec */ - if (avcodec_open2(c, NULL, NULL) < 0) { - fprintf(stderr, "could not open codec\n"); + ret = avcodec_open2(c, codec, &opt); + av_dict_free(&opt); + if (ret < 0) { + fprintf(stderr, "Could not open video codec: %s\n", av_err2str(ret)); exit(1); } - /* Allocate the encoded raw picture. */ + /* allocate and init a re-usable frame */ ost->frame = alloc_picture(c->pix_fmt, c->width, c->height); if (!ost->frame) { - fprintf(stderr, "Could not allocate picture\n"); + fprintf(stderr, "Could not allocate video frame\n"); exit(1); } @@ -466,12 +466,13 @@ static AVFrame *get_video_frame(OutputStream *ost) SCALE_FLAGS, NULL, NULL, NULL); if (!ost->sws_ctx) { fprintf(stderr, - "Cannot initialize the conversion context\n"); + "Could not initialize the conversion context\n"); exit(1); } } fill_yuv_image(ost->tmp_frame, ost->next_pts, c->width, c->height); - sws_scale(ost->sws_ctx, ost->tmp_frame->data, ost->tmp_frame->linesize, + sws_scale(ost->sws_ctx, + (const uint8_t * const *)ost->tmp_frame->data, ost->tmp_frame->linesize, 0, c->height, ost->frame->data, ost->frame->linesize); } else { fill_yuv_image(ost->frame, ost->next_pts, c->width, c->height); @@ -491,8 +492,8 @@ static int write_video_frame(AVFormatContext *oc, OutputStream *ost) int ret; AVCodecContext *c; AVFrame *frame; - AVPacket pkt = { 0 }; int got_packet = 0; + AVPacket pkt = { 0 }; c = ost->st->codec; @@ -503,20 +504,18 @@ static int write_video_frame(AVFormatContext *oc, OutputStream *ost) /* encode the image */ ret = avcodec_encode_video2(c, &pkt, frame, &got_packet); if (ret < 0) { - fprintf(stderr, "Error encoding a video frame\n"); + fprintf(stderr, "Error encoding video frame: %s\n", av_err2str(ret)); exit(1); } if (got_packet) { - av_packet_rescale_ts(&pkt, c->time_base, ost->st->time_base); - pkt.stream_index = ost->st->index; - - /* Write the compressed frame to the media file. */ - ret = av_interleaved_write_frame(oc, &pkt); + ret = write_frame(oc, &c->time_base, ost->st, &pkt); + } else { + ret = 0; } - if (ret != 0) { - fprintf(stderr, "Error while writing video frame\n"); + if (ret < 0) { + fprintf(stderr, "Error while writing video frame: %s\n", av_err2str(ret)); exit(1); } @@ -529,7 +528,7 @@ static void close_stream(AVFormatContext *oc, OutputStream *ost) av_frame_free(&ost->frame); av_frame_free(&ost->tmp_frame); sws_freeContext(ost->sws_ctx); - avresample_free(&ost->avr); + swr_free(&ost->swr_ctx); } /**************************************************************/ @@ -541,52 +540,51 @@ int main(int argc, char **argv) const char *filename; AVOutputFormat *fmt; AVFormatContext *oc; + AVCodec *audio_codec, *video_codec; + int ret; int have_video = 0, have_audio = 0; int encode_video = 0, encode_audio = 0; + AVDictionary *opt = NULL; /* Initialize libavcodec, and register all codecs and formats. */ av_register_all(); - if (argc != 2) { + if (argc < 2) { printf("usage: %s output_file\n" "API example program to output a media file with libavformat.\n" + "This program generates a synthetic audio and video stream, encodes and\n" + "muxes them into a file named output_file.\n" "The output format is automatically guessed according to the file extension.\n" - "Raw images can also be output by using '%%d' in the filename\n" + "Raw images can also be output by using '%%d' in the filename.\n" "\n", argv[0]); return 1; } filename = argv[1]; + if (argc > 3 && !strcmp(argv[2], "-flags")) { + av_dict_set(&opt, argv[2]+1, argv[3], 0); + } - /* Autodetect the output format from the name. default is MPEG. */ - fmt = av_guess_format(NULL, filename, NULL); - if (!fmt) { + /* allocate the output media context */ + avformat_alloc_output_context2(&oc, NULL, NULL, filename); + if (!oc) { printf("Could not deduce output format from file extension: using MPEG.\n"); - fmt = av_guess_format("mpeg", NULL, NULL); + avformat_alloc_output_context2(&oc, NULL, "mpeg", filename); } - if (!fmt) { - fprintf(stderr, "Could not find suitable output format\n"); + if (!oc) return 1; - } - /* Allocate the output media context. */ - oc = avformat_alloc_context(); - if (!oc) { - fprintf(stderr, "Memory error\n"); - return 1; - } - oc->oformat = fmt; - snprintf(oc->filename, sizeof(oc->filename), "%s", filename); + fmt = oc->oformat; /* Add the audio and video streams using the default format codecs * and initialize the codecs. */ if (fmt->video_codec != AV_CODEC_ID_NONE) { - add_video_stream(&video_st, oc, fmt->video_codec); + add_stream(&video_st, oc, &video_codec, fmt->video_codec); have_video = 1; encode_video = 1; } if (fmt->audio_codec != AV_CODEC_ID_NONE) { - add_audio_stream(&audio_st, oc, fmt->audio_codec); + add_stream(&audio_st, oc, &audio_codec, fmt->audio_codec); have_audio = 1; encode_audio = 1; } @@ -594,22 +592,30 @@ int main(int argc, char **argv) /* Now that all the parameters are set, we can open the audio and * video codecs and allocate the necessary encode buffers. */ if (have_video) - open_video(oc, &video_st); + open_video(oc, video_codec, &video_st, opt); + if (have_audio) - open_audio(oc, &audio_st); + open_audio(oc, audio_codec, &audio_st, opt); av_dump_format(oc, 0, filename, 1); /* open the output file, if needed */ if (!(fmt->flags & AVFMT_NOFILE)) { - if (avio_open(&oc->pb, filename, AVIO_FLAG_WRITE) < 0) { - fprintf(stderr, "Could not open '%s'\n", filename); + ret = avio_open(&oc->pb, filename, AVIO_FLAG_WRITE); + if (ret < 0) { + fprintf(stderr, "Could not open '%s': %s\n", filename, + av_err2str(ret)); return 1; } } /* Write the stream header, if any. */ - avformat_write_header(oc, NULL); + ret = avformat_write_header(oc, &opt); + if (ret < 0) { + fprintf(stderr, "Error occurred when opening output file: %s\n", + av_err2str(ret)); + return 1; + } while (encode_video || encode_audio) { /* select the stream to encode */ @@ -618,7 +624,7 @@ int main(int argc, char **argv) audio_st.next_pts, audio_st.st->codec->time_base) <= 0)) { encode_video = !write_video_frame(oc, &video_st); } else { - encode_audio = !process_audio_stream(oc, &audio_st); + encode_audio = !write_audio_frame(oc, &audio_st); } } @@ -636,7 +642,7 @@ int main(int argc, char **argv) if (!(fmt->flags & AVFMT_NOFILE)) /* Close the output file. */ - avio_close(oc->pb); + avio_closep(&oc->pb); /* free the stream */ avformat_free_context(oc); diff --git a/doc/examples/remuxing.c b/doc/examples/remuxing.c new file mode 100644 index 0000000000..65437d9abd --- /dev/null +++ b/doc/examples/remuxing.c @@ -0,0 +1,165 @@ +/* + * Copyright (c) 2013 Stefano Sabatini + * + * Permission is hereby granted, free of charge, to any person obtaining a copy + * of this software and associated documentation files (the "Software"), to deal + * in the Software without restriction, including without limitation the rights + * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell + * copies of the Software, and to permit persons to whom the Software is + * furnished to do so, subject to the following conditions: + * + * The above copyright notice and this permission notice shall be included in + * all copies or substantial portions of the Software. + * + * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR + * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, + * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL + * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER + * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, + * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN + * THE SOFTWARE. + */ + +/** + * @file + * libavformat/libavcodec demuxing and muxing API example. + * + * Remux streams from one container format to another. + * @example remuxing.c + */ + +#include <libavutil/timestamp.h> +#include <libavformat/avformat.h> + +static void log_packet(const AVFormatContext *fmt_ctx, const AVPacket *pkt, const char *tag) +{ + AVRational *time_base = &fmt_ctx->streams[pkt->stream_index]->time_base; + + printf("%s: pts:%s pts_time:%s dts:%s dts_time:%s duration:%s duration_time:%s stream_index:%d\n", + tag, + av_ts2str(pkt->pts), av_ts2timestr(pkt->pts, time_base), + av_ts2str(pkt->dts), av_ts2timestr(pkt->dts, time_base), + av_ts2str(pkt->duration), av_ts2timestr(pkt->duration, time_base), + pkt->stream_index); +} + +int main(int argc, char **argv) +{ + AVOutputFormat *ofmt = NULL; + AVFormatContext *ifmt_ctx = NULL, *ofmt_ctx = NULL; + AVPacket pkt; + const char *in_filename, *out_filename; + int ret, i; + + if (argc < 3) { + printf("usage: %s input output\n" + "API example program to remux a media file with libavformat and libavcodec.\n" + "The output format is guessed according to the file extension.\n" + "\n", argv[0]); + return 1; + } + + in_filename = argv[1]; + out_filename = argv[2]; + + av_register_all(); + + if ((ret = avformat_open_input(&ifmt_ctx, in_filename, 0, 0)) < 0) { + fprintf(stderr, "Could not open input file '%s'", in_filename); + goto end; + } + + if ((ret = avformat_find_stream_info(ifmt_ctx, 0)) < 0) { + fprintf(stderr, "Failed to retrieve input stream information"); + goto end; + } + + av_dump_format(ifmt_ctx, 0, in_filename, 0); + + avformat_alloc_output_context2(&ofmt_ctx, NULL, NULL, out_filename); + if (!ofmt_ctx) { + fprintf(stderr, "Could not create output context\n"); + ret = AVERROR_UNKNOWN; + goto end; + } + + ofmt = ofmt_ctx->oformat; + + for (i = 0; i < ifmt_ctx->nb_streams; i++) { + AVStream *in_stream = ifmt_ctx->streams[i]; + AVStream *out_stream = avformat_new_stream(ofmt_ctx, in_stream->codec->codec); + if (!out_stream) { + fprintf(stderr, "Failed allocating output stream\n"); + ret = AVERROR_UNKNOWN; + goto end; + } + + ret = avcodec_copy_context(out_stream->codec, in_stream->codec); + if (ret < 0) { + fprintf(stderr, "Failed to copy context from input to output stream codec context\n"); + goto end; + } + out_stream->codec->codec_tag = 0; + if (ofmt_ctx->oformat->flags & AVFMT_GLOBALHEADER) + out_stream->codec->flags |= AV_CODEC_FLAG_GLOBAL_HEADER; + } + av_dump_format(ofmt_ctx, 0, out_filename, 1); + + if (!(ofmt->flags & AVFMT_NOFILE)) { + ret = avio_open(&ofmt_ctx->pb, out_filename, AVIO_FLAG_WRITE); + if (ret < 0) { + fprintf(stderr, "Could not open output file '%s'", out_filename); + goto end; + } + } + + ret = avformat_write_header(ofmt_ctx, NULL); + if (ret < 0) { + fprintf(stderr, "Error occurred when opening output file\n"); + goto end; + } + + while (1) { + AVStream *in_stream, *out_stream; + + ret = av_read_frame(ifmt_ctx, &pkt); + if (ret < 0) + break; + + in_stream = ifmt_ctx->streams[pkt.stream_index]; + out_stream = ofmt_ctx->streams[pkt.stream_index]; + + log_packet(ifmt_ctx, &pkt, "in"); + + /* copy packet */ + pkt.pts = av_rescale_q_rnd(pkt.pts, in_stream->time_base, out_stream->time_base, AV_ROUND_NEAR_INF|AV_ROUND_PASS_MINMAX); + pkt.dts = av_rescale_q_rnd(pkt.dts, in_stream->time_base, out_stream->time_base, AV_ROUND_NEAR_INF|AV_ROUND_PASS_MINMAX); + pkt.duration = av_rescale_q(pkt.duration, in_stream->time_base, out_stream->time_base); + pkt.pos = -1; + log_packet(ofmt_ctx, &pkt, "out"); + + ret = av_interleaved_write_frame(ofmt_ctx, &pkt); + if (ret < 0) { + fprintf(stderr, "Error muxing packet\n"); + break; + } + av_packet_unref(&pkt); + } + + av_write_trailer(ofmt_ctx); +end: + + avformat_close_input(&ifmt_ctx); + + /* close output */ + if (ofmt_ctx && !(ofmt->flags & AVFMT_NOFILE)) + avio_closep(&ofmt_ctx->pb); + avformat_free_context(ofmt_ctx); + + if (ret < 0 && ret != AVERROR_EOF) { + fprintf(stderr, "Error occurred: %s\n", av_err2str(ret)); + return 1; + } + + return 0; +} diff --git a/doc/examples/resampling_audio.c b/doc/examples/resampling_audio.c new file mode 100644 index 0000000000..f35e7e1779 --- /dev/null +++ b/doc/examples/resampling_audio.c @@ -0,0 +1,214 @@ +/* + * Copyright (c) 2012 Stefano Sabatini + * + * Permission is hereby granted, free of charge, to any person obtaining a copy + * of this software and associated documentation files (the "Software"), to deal + * in the Software without restriction, including without limitation the rights + * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell + * copies of the Software, and to permit persons to whom the Software is + * furnished to do so, subject to the following conditions: + * + * The above copyright notice and this permission notice shall be included in + * all copies or substantial portions of the Software. + * + * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR + * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, + * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL + * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER + * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, + * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN + * THE SOFTWARE. + */ + +/** + * @example resampling_audio.c + * libswresample API use example. + */ + +#include <libavutil/opt.h> +#include <libavutil/channel_layout.h> +#include <libavutil/samplefmt.h> +#include <libswresample/swresample.h> + +static int get_format_from_sample_fmt(const char **fmt, + enum AVSampleFormat sample_fmt) +{ + int i; + struct sample_fmt_entry { + enum AVSampleFormat sample_fmt; const char *fmt_be, *fmt_le; + } sample_fmt_entries[] = { + { AV_SAMPLE_FMT_U8, "u8", "u8" }, + { AV_SAMPLE_FMT_S16, "s16be", "s16le" }, + { AV_SAMPLE_FMT_S32, "s32be", "s32le" }, + { AV_SAMPLE_FMT_FLT, "f32be", "f32le" }, + { AV_SAMPLE_FMT_DBL, "f64be", "f64le" }, + }; + *fmt = NULL; + + for (i = 0; i < FF_ARRAY_ELEMS(sample_fmt_entries); i++) { + struct sample_fmt_entry *entry = &sample_fmt_entries[i]; + if (sample_fmt == entry->sample_fmt) { + *fmt = AV_NE(entry->fmt_be, entry->fmt_le); + return 0; + } + } + + fprintf(stderr, + "Sample format %s not supported as output format\n", + av_get_sample_fmt_name(sample_fmt)); + return AVERROR(EINVAL); +} + +/** + * Fill dst buffer with nb_samples, generated starting from t. + */ +static void fill_samples(double *dst, int nb_samples, int nb_channels, int sample_rate, double *t) +{ + int i, j; + double tincr = 1.0 / sample_rate, *dstp = dst; + const double c = 2 * M_PI * 440.0; + + /* generate sin tone with 440Hz frequency and duplicated channels */ + for (i = 0; i < nb_samples; i++) { + *dstp = sin(c * *t); + for (j = 1; j < nb_channels; j++) + dstp[j] = dstp[0]; + dstp += nb_channels; + *t += tincr; + } +} + +int main(int argc, char **argv) +{ + int64_t src_ch_layout = AV_CH_LAYOUT_STEREO, dst_ch_layout = AV_CH_LAYOUT_SURROUND; + int src_rate = 48000, dst_rate = 44100; + uint8_t **src_data = NULL, **dst_data = NULL; + int src_nb_channels = 0, dst_nb_channels = 0; + int src_linesize, dst_linesize; + int src_nb_samples = 1024, dst_nb_samples, max_dst_nb_samples; + enum AVSampleFormat src_sample_fmt = AV_SAMPLE_FMT_DBL, dst_sample_fmt = AV_SAMPLE_FMT_S16; + const char *dst_filename = NULL; + FILE *dst_file; + int dst_bufsize; + const char *fmt; + struct SwrContext *swr_ctx; + double t; + int ret; + + if (argc != 2) { + fprintf(stderr, "Usage: %s output_file\n" + "API example program to show how to resample an audio stream with libswresample.\n" + "This program generates a series of audio frames, resamples them to a specified " + "output format and rate and saves them to an output file named output_file.\n", + argv[0]); + exit(1); + } + dst_filename = argv[1]; + + dst_file = fopen(dst_filename, "wb"); + if (!dst_file) { + fprintf(stderr, "Could not open destination file %s\n", dst_filename); + exit(1); + } + + /* create resampler context */ + swr_ctx = swr_alloc(); + if (!swr_ctx) { + fprintf(stderr, "Could not allocate resampler context\n"); + ret = AVERROR(ENOMEM); + goto end; + } + + /* set options */ + av_opt_set_int(swr_ctx, "in_channel_layout", src_ch_layout, 0); + av_opt_set_int(swr_ctx, "in_sample_rate", src_rate, 0); + av_opt_set_sample_fmt(swr_ctx, "in_sample_fmt", src_sample_fmt, 0); + + av_opt_set_int(swr_ctx, "out_channel_layout", dst_ch_layout, 0); + av_opt_set_int(swr_ctx, "out_sample_rate", dst_rate, 0); + av_opt_set_sample_fmt(swr_ctx, "out_sample_fmt", dst_sample_fmt, 0); + + /* initialize the resampling context */ + if ((ret = swr_init(swr_ctx)) < 0) { + fprintf(stderr, "Failed to initialize the resampling context\n"); + goto end; + } + + /* allocate source and destination samples buffers */ + + src_nb_channels = av_get_channel_layout_nb_channels(src_ch_layout); + ret = av_samples_alloc_array_and_samples(&src_data, &src_linesize, src_nb_channels, + src_nb_samples, src_sample_fmt, 0); + if (ret < 0) { + fprintf(stderr, "Could not allocate source samples\n"); + goto end; + } + + /* compute the number of converted samples: buffering is avoided + * ensuring that the output buffer will contain at least all the + * converted input samples */ + max_dst_nb_samples = dst_nb_samples = + av_rescale_rnd(src_nb_samples, dst_rate, src_rate, AV_ROUND_UP); + + /* buffer is going to be directly written to a rawaudio file, no alignment */ + dst_nb_channels = av_get_channel_layout_nb_channels(dst_ch_layout); + ret = av_samples_alloc_array_and_samples(&dst_data, &dst_linesize, dst_nb_channels, + dst_nb_samples, dst_sample_fmt, 0); + if (ret < 0) { + fprintf(stderr, "Could not allocate destination samples\n"); + goto end; + } + + t = 0; + do { + /* generate synthetic audio */ + fill_samples((double *)src_data[0], src_nb_samples, src_nb_channels, src_rate, &t); + + /* compute destination number of samples */ + dst_nb_samples = av_rescale_rnd(swr_get_delay(swr_ctx, src_rate) + + src_nb_samples, dst_rate, src_rate, AV_ROUND_UP); + if (dst_nb_samples > max_dst_nb_samples) { + av_freep(&dst_data[0]); + ret = av_samples_alloc(dst_data, &dst_linesize, dst_nb_channels, + dst_nb_samples, dst_sample_fmt, 1); + if (ret < 0) + break; + max_dst_nb_samples = dst_nb_samples; + } + + /* convert to destination format */ + ret = swr_convert(swr_ctx, dst_data, dst_nb_samples, (const uint8_t **)src_data, src_nb_samples); + if (ret < 0) { + fprintf(stderr, "Error while converting\n"); + goto end; + } + dst_bufsize = av_samples_get_buffer_size(&dst_linesize, dst_nb_channels, + ret, dst_sample_fmt, 1); + if (dst_bufsize < 0) { + fprintf(stderr, "Could not get sample buffer size\n"); + goto end; + } + printf("t:%f in:%d out:%d\n", t, src_nb_samples, ret); + fwrite(dst_data[0], 1, dst_bufsize, dst_file); + } while (t < 10); + + if ((ret = get_format_from_sample_fmt(&fmt, dst_sample_fmt)) < 0) + goto end; + fprintf(stderr, "Resampling succeeded. Play the output file with the command:\n" + "ffplay -f %s -channel_layout %"PRId64" -channels %d -ar %d %s\n", + fmt, dst_ch_layout, dst_nb_channels, dst_rate, dst_filename); + +end: + fclose(dst_file); + + if (src_data) + av_freep(&src_data[0]); + av_freep(&src_data); + + if (dst_data) + av_freep(&dst_data[0]); + av_freep(&dst_data); + + swr_free(&swr_ctx); + return ret < 0; +} diff --git a/doc/examples/scaling_video.c b/doc/examples/scaling_video.c new file mode 100644 index 0000000000..587f3abe4f --- /dev/null +++ b/doc/examples/scaling_video.c @@ -0,0 +1,140 @@ +/* + * Copyright (c) 2012 Stefano Sabatini + * + * Permission is hereby granted, free of charge, to any person obtaining a copy + * of this software and associated documentation files (the "Software"), to deal + * in the Software without restriction, including without limitation the rights + * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell + * copies of the Software, and to permit persons to whom the Software is + * furnished to do so, subject to the following conditions: + * + * The above copyright notice and this permission notice shall be included in + * all copies or substantial portions of the Software. + * + * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR + * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, + * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL + * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER + * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, + * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN + * THE SOFTWARE. + */ + +/** + * @file + * libswscale API use example. + * @example scaling_video.c + */ + +#include <libavutil/imgutils.h> +#include <libavutil/parseutils.h> +#include <libswscale/swscale.h> + +static void fill_yuv_image(uint8_t *data[4], int linesize[4], + int width, int height, int frame_index) +{ + int x, y; + + /* Y */ + for (y = 0; y < height; y++) + for (x = 0; x < width; x++) + data[0][y * linesize[0] + x] = x + y + frame_index * 3; + + /* Cb and Cr */ + for (y = 0; y < height / 2; y++) { + for (x = 0; x < width / 2; x++) { + data[1][y * linesize[1] + x] = 128 + y + frame_index * 2; + data[2][y * linesize[2] + x] = 64 + x + frame_index * 5; + } + } +} + +int main(int argc, char **argv) +{ + uint8_t *src_data[4], *dst_data[4]; + int src_linesize[4], dst_linesize[4]; + int src_w = 320, src_h = 240, dst_w, dst_h; + enum AVPixelFormat src_pix_fmt = AV_PIX_FMT_YUV420P, dst_pix_fmt = AV_PIX_FMT_RGB24; + const char *dst_size = NULL; + const char *dst_filename = NULL; + FILE *dst_file; + int dst_bufsize; + struct SwsContext *sws_ctx; + int i, ret; + + if (argc != 3) { + fprintf(stderr, "Usage: %s output_file output_size\n" + "API example program to show how to scale an image with libswscale.\n" + "This program generates a series of pictures, rescales them to the given " + "output_size and saves them to an output file named output_file\n." + "\n", argv[0]); + exit(1); + } + dst_filename = argv[1]; + dst_size = argv[2]; + + if (av_parse_video_size(&dst_w, &dst_h, dst_size) < 0) { + fprintf(stderr, + "Invalid size '%s', must be in the form WxH or a valid size abbreviation\n", + dst_size); + exit(1); + } + + dst_file = fopen(dst_filename, "wb"); + if (!dst_file) { + fprintf(stderr, "Could not open destination file %s\n", dst_filename); + exit(1); + } + + /* create scaling context */ + sws_ctx = sws_getContext(src_w, src_h, src_pix_fmt, + dst_w, dst_h, dst_pix_fmt, + SWS_BILINEAR, NULL, NULL, NULL); + if (!sws_ctx) { + fprintf(stderr, + "Impossible to create scale context for the conversion " + "fmt:%s s:%dx%d -> fmt:%s s:%dx%d\n", + av_get_pix_fmt_name(src_pix_fmt), src_w, src_h, + av_get_pix_fmt_name(dst_pix_fmt), dst_w, dst_h); + ret = AVERROR(EINVAL); + goto end; + } + + /* allocate source and destination image buffers */ + if ((ret = av_image_alloc(src_data, src_linesize, + src_w, src_h, src_pix_fmt, 16)) < 0) { + fprintf(stderr, "Could not allocate source image\n"); + goto end; + } + + /* buffer is going to be written to rawvideo file, no alignment */ + if ((ret = av_image_alloc(dst_data, dst_linesize, + dst_w, dst_h, dst_pix_fmt, 1)) < 0) { + fprintf(stderr, "Could not allocate destination image\n"); + goto end; + } + dst_bufsize = ret; + + for (i = 0; i < 100; i++) { + /* generate synthetic video */ + fill_yuv_image(src_data, src_linesize, src_w, src_h, i); + + /* convert to destination format */ + sws_scale(sws_ctx, (const uint8_t * const*)src_data, + src_linesize, 0, src_h, dst_data, dst_linesize); + + /* write scaled image to file */ + fwrite(dst_data[0], 1, dst_bufsize, dst_file); + } + + fprintf(stderr, "Scaling succeeded. Play the output file with the command:\n" + "ffplay -f rawvideo -pix_fmt %s -video_size %dx%d %s\n", + av_get_pix_fmt_name(dst_pix_fmt), dst_w, dst_h, dst_filename); + +end: + fclose(dst_file); + av_freep(&src_data[0]); + av_freep(&dst_data[0]); + sws_freeContext(sws_ctx); + return ret < 0; +} diff --git a/doc/examples/transcode_aac.c b/doc/examples/transcode_aac.c index 3eebfb9d02..486e54c281 100644 --- a/doc/examples/transcode_aac.c +++ b/doc/examples/transcode_aac.c @@ -1,18 +1,18 @@ /* - * This file is part of Libav. + * This file is part of FFmpeg. * - * Libav is free software; you can redistribute it and/or + * FFmpeg is free software; you can redistribute it and/or * modify it under the terms of the GNU Lesser General Public * License as published by the Free Software Foundation; either * version 2.1 of the License, or (at your option) any later version. * - * Libav is distributed in the hope that it will be useful, + * FFmpeg is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Lesser General Public License for more details. * * You should have received a copy of the GNU Lesser General Public - * License along with Libav; if not, write to the Free Software + * License along with FFmpeg; if not, write to the Free Software * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ @@ -21,7 +21,7 @@ * simple audio converter * * @example transcode_aac.c - * Convert an input audio file to AAC in an MP4 container using Libav. + * Convert an input audio file to AAC in an MP4 container using FFmpeg. * @author Andreas Unterweger (dustsigns@gmail.com) */ @@ -33,11 +33,12 @@ #include "libavcodec/avcodec.h" #include "libavutil/audio_fifo.h" +#include "libavutil/avassert.h" #include "libavutil/avstring.h" #include "libavutil/frame.h" #include "libavutil/opt.h" -#include "libavresample/avresample.h" +#include "libswresample/swresample.h" /** The output bit rate in kbit/s */ #define OUTPUT_BIT_RATE 96000 @@ -49,7 +50,7 @@ * @param error Error code to be converted * @return Corresponding error text (not thread-safe) */ -static char *const get_error_text(const int error) +static const char *get_error_text(const int error) { static char error_buffer[255]; av_strerror(error, error_buffer, sizeof(error_buffer)); @@ -203,7 +204,7 @@ static int open_output_file(const char *filename, return 0; cleanup: - avio_close((*output_format_context)->pb); + avio_closep(&(*output_format_context)->pb); avformat_free_context(*output_format_context); *output_format_context = NULL; return error < 0 ? error : AVERROR_EXIT; @@ -231,52 +232,46 @@ static int init_input_frame(AVFrame **frame) /** * Initialize the audio resampler based on the input and output codec settings. * If the input and output sample formats differ, a conversion is required - * libavresample takes care of this, but requires initialization. + * libswresample takes care of this, but requires initialization. */ static int init_resampler(AVCodecContext *input_codec_context, AVCodecContext *output_codec_context, - AVAudioResampleContext **resample_context) + SwrContext **resample_context) { - /** - * Only initialize the resampler if it is necessary, i.e., - * if and only if the sample formats differ. - */ - if (input_codec_context->sample_fmt != output_codec_context->sample_fmt || - input_codec_context->channels != output_codec_context->channels) { int error; - /** Create a resampler context for the conversion. */ - if (!(*resample_context = avresample_alloc_context())) { - fprintf(stderr, "Could not allocate resample context\n"); - return AVERROR(ENOMEM); - } - /** + * Create a resampler context for the conversion. * Set the conversion parameters. * Default channel layouts based on the number of channels * are assumed for simplicity (they are sometimes not detected * properly by the demuxer and/or decoder). */ - av_opt_set_int(*resample_context, "in_channel_layout", - av_get_default_channel_layout(input_codec_context->channels), 0); - av_opt_set_int(*resample_context, "out_channel_layout", - av_get_default_channel_layout(output_codec_context->channels), 0); - av_opt_set_int(*resample_context, "in_sample_rate", - input_codec_context->sample_rate, 0); - av_opt_set_int(*resample_context, "out_sample_rate", - output_codec_context->sample_rate, 0); - av_opt_set_int(*resample_context, "in_sample_fmt", - input_codec_context->sample_fmt, 0); - av_opt_set_int(*resample_context, "out_sample_fmt", - output_codec_context->sample_fmt, 0); + *resample_context = swr_alloc_set_opts(NULL, + av_get_default_channel_layout(output_codec_context->channels), + output_codec_context->sample_fmt, + output_codec_context->sample_rate, + av_get_default_channel_layout(input_codec_context->channels), + input_codec_context->sample_fmt, + input_codec_context->sample_rate, + 0, NULL); + if (!*resample_context) { + fprintf(stderr, "Could not allocate resample context\n"); + return AVERROR(ENOMEM); + } + /** + * Perform a sanity check so that the number of converted samples is + * not greater than the number of samples to be converted. + * If the sample rates differ, this case has to be handled differently + */ + av_assert0(output_codec_context->sample_rate == input_codec_context->sample_rate); /** Open the resampler with the specified parameters. */ - if ((error = avresample_open(*resample_context)) < 0) { + if ((error = swr_init(*resample_context)) < 0) { fprintf(stderr, "Could not open resample context\n"); - avresample_free(resample_context); + swr_free(resample_context); return error; } - } return 0; } @@ -317,7 +312,7 @@ static int decode_audio_frame(AVFrame *frame, /** Read one audio frame from the input file into a temporary packet. */ if ((error = av_read_frame(input_format_context, &input_packet)) < 0) { - /** If we are the the end of the file, flush the decoder below. */ + /** If we are at the end of the file, flush the decoder below. */ if (error == AVERROR_EOF) *finished = 1; else { @@ -396,30 +391,21 @@ static int init_converted_samples(uint8_t ***converted_input_samples, * The conversion happens on a per-frame basis, the size of which is specified * by frame_size. */ -static int convert_samples(uint8_t **input_data, +static int convert_samples(const uint8_t **input_data, uint8_t **converted_data, const int frame_size, - AVAudioResampleContext *resample_context) + SwrContext *resample_context) { int error; /** Convert the samples using the resampler. */ - if ((error = avresample_convert(resample_context, converted_data, 0, - frame_size, input_data, 0, frame_size)) < 0) { + if ((error = swr_convert(resample_context, + converted_data, frame_size, + input_data , frame_size)) < 0) { fprintf(stderr, "Could not convert input samples (error '%s')\n", get_error_text(error)); return error; } - /** - * Perform a sanity check so that the number of converted samples is - * not greater than the number of samples to be converted. - * If the sample rates differ, this case has to be handled differently - */ - if (avresample_available(resample_context)) { - fprintf(stderr, "Converted samples left over\n"); - return AVERROR_EXIT; - } - return 0; } @@ -456,7 +442,7 @@ static int read_decode_convert_and_store(AVAudioFifo *fifo, AVFormatContext *input_format_context, AVCodecContext *input_codec_context, AVCodecContext *output_codec_context, - AVAudioResampleContext *resampler_context, + SwrContext *resampler_context, int *finished) { /** Temporary storage of the input samples of the frame read from the file. */ @@ -493,7 +479,7 @@ static int read_decode_convert_and_store(AVAudioFifo *fifo, * Convert the input samples to the desired output sample format. * This requires a temporary storage provided by converted_input_samples. */ - if (convert_samples(input_frame->extended_data, converted_input_samples, + if (convert_samples((const uint8_t**)input_frame->extended_data, converted_input_samples, input_frame->nb_samples, resampler_context)) goto cleanup; @@ -664,7 +650,7 @@ int main(int argc, char **argv) { AVFormatContext *input_format_context = NULL, *output_format_context = NULL; AVCodecContext *input_codec_context = NULL, *output_codec_context = NULL; - AVAudioResampleContext *resample_context = NULL; + SwrContext *resample_context = NULL; AVAudioFifo *fifo = NULL; int ret = AVERROR_EXIT; @@ -768,14 +754,11 @@ int main(int argc, char **argv) cleanup: if (fifo) av_audio_fifo_free(fifo); - if (resample_context) { - avresample_close(resample_context); - avresample_free(&resample_context); - } + swr_free(&resample_context); if (output_codec_context) avcodec_close(output_codec_context); if (output_format_context) { - avio_close(output_format_context->pb); + avio_closep(&output_format_context->pb); avformat_free_context(output_format_context); } if (input_codec_context) diff --git a/doc/examples/transcoding.c b/doc/examples/transcoding.c new file mode 100644 index 0000000000..d5d410b168 --- /dev/null +++ b/doc/examples/transcoding.c @@ -0,0 +1,582 @@ +/* + * Copyright (c) 2010 Nicolas George + * Copyright (c) 2011 Stefano Sabatini + * Copyright (c) 2014 Andrey Utkin + * + * Permission is hereby granted, free of charge, to any person obtaining a copy + * of this software and associated documentation files (the "Software"), to deal + * in the Software without restriction, including without limitation the rights + * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell + * copies of the Software, and to permit persons to whom the Software is + * furnished to do so, subject to the following conditions: + * + * The above copyright notice and this permission notice shall be included in + * all copies or substantial portions of the Software. + * + * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR + * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, + * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL + * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER + * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, + * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN + * THE SOFTWARE. + */ + +/** + * @file + * API example for demuxing, decoding, filtering, encoding and muxing + * @example transcoding.c + */ + +#include <libavcodec/avcodec.h> +#include <libavformat/avformat.h> +#include <libavfilter/avfiltergraph.h> +#include <libavfilter/buffersink.h> +#include <libavfilter/buffersrc.h> +#include <libavutil/opt.h> +#include <libavutil/pixdesc.h> + +static AVFormatContext *ifmt_ctx; +static AVFormatContext *ofmt_ctx; +typedef struct FilteringContext { + AVFilterContext *buffersink_ctx; + AVFilterContext *buffersrc_ctx; + AVFilterGraph *filter_graph; +} FilteringContext; +static FilteringContext *filter_ctx; + +static int open_input_file(const char *filename) +{ + int ret; + unsigned int i; + + ifmt_ctx = NULL; + if ((ret = avformat_open_input(&ifmt_ctx, filename, NULL, NULL)) < 0) { + av_log(NULL, AV_LOG_ERROR, "Cannot open input file\n"); + return ret; + } + + if ((ret = avformat_find_stream_info(ifmt_ctx, NULL)) < 0) { + av_log(NULL, AV_LOG_ERROR, "Cannot find stream information\n"); + return ret; + } + + for (i = 0; i < ifmt_ctx->nb_streams; i++) { + AVStream *stream; + AVCodecContext *codec_ctx; + stream = ifmt_ctx->streams[i]; + codec_ctx = stream->codec; + /* Reencode video & audio and remux subtitles etc. */ + if (codec_ctx->codec_type == AVMEDIA_TYPE_VIDEO + || codec_ctx->codec_type == AVMEDIA_TYPE_AUDIO) { + /* Open decoder */ + ret = avcodec_open2(codec_ctx, + avcodec_find_decoder(codec_ctx->codec_id), NULL); + if (ret < 0) { + av_log(NULL, AV_LOG_ERROR, "Failed to open decoder for stream #%u\n", i); + return ret; + } + } + } + + av_dump_format(ifmt_ctx, 0, filename, 0); + return 0; +} + +static int open_output_file(const char *filename) +{ + AVStream *out_stream; + AVStream *in_stream; + AVCodecContext *dec_ctx, *enc_ctx; + AVCodec *encoder; + int ret; + unsigned int i; + + ofmt_ctx = NULL; + avformat_alloc_output_context2(&ofmt_ctx, NULL, NULL, filename); + if (!ofmt_ctx) { + av_log(NULL, AV_LOG_ERROR, "Could not create output context\n"); + return AVERROR_UNKNOWN; + } + + + for (i = 0; i < ifmt_ctx->nb_streams; i++) { + out_stream = avformat_new_stream(ofmt_ctx, NULL); + if (!out_stream) { + av_log(NULL, AV_LOG_ERROR, "Failed allocating output stream\n"); + return AVERROR_UNKNOWN; + } + + in_stream = ifmt_ctx->streams[i]; + dec_ctx = in_stream->codec; + enc_ctx = out_stream->codec; + + if (dec_ctx->codec_type == AVMEDIA_TYPE_VIDEO + || dec_ctx->codec_type == AVMEDIA_TYPE_AUDIO) { + /* in this example, we choose transcoding to same codec */ + encoder = avcodec_find_encoder(dec_ctx->codec_id); + if (!encoder) { + av_log(NULL, AV_LOG_FATAL, "Necessary encoder not found\n"); + return AVERROR_INVALIDDATA; + } + + /* In this example, we transcode to same properties (picture size, + * sample rate etc.). These properties can be changed for output + * streams easily using filters */ + if (dec_ctx->codec_type == AVMEDIA_TYPE_VIDEO) { + enc_ctx->height = dec_ctx->height; + enc_ctx->width = dec_ctx->width; + enc_ctx->sample_aspect_ratio = dec_ctx->sample_aspect_ratio; + /* take first format from list of supported formats */ + enc_ctx->pix_fmt = encoder->pix_fmts[0]; + /* video time_base can be set to whatever is handy and supported by encoder */ + enc_ctx->time_base = dec_ctx->time_base; + } else { + enc_ctx->sample_rate = dec_ctx->sample_rate; + enc_ctx->channel_layout = dec_ctx->channel_layout; + enc_ctx->channels = av_get_channel_layout_nb_channels(enc_ctx->channel_layout); + /* take first format from list of supported formats */ + enc_ctx->sample_fmt = encoder->sample_fmts[0]; + enc_ctx->time_base = (AVRational){1, enc_ctx->sample_rate}; + } + + /* Third parameter can be used to pass settings to encoder */ + ret = avcodec_open2(enc_ctx, encoder, NULL); + if (ret < 0) { + av_log(NULL, AV_LOG_ERROR, "Cannot open video encoder for stream #%u\n", i); + return ret; + } + } else if (dec_ctx->codec_type == AVMEDIA_TYPE_UNKNOWN) { + av_log(NULL, AV_LOG_FATAL, "Elementary stream #%d is of unknown type, cannot proceed\n", i); + return AVERROR_INVALIDDATA; + } else { + /* if this stream must be remuxed */ + ret = avcodec_copy_context(ofmt_ctx->streams[i]->codec, + ifmt_ctx->streams[i]->codec); + if (ret < 0) { + av_log(NULL, AV_LOG_ERROR, "Copying stream context failed\n"); + return ret; + } + } + + if (ofmt_ctx->oformat->flags & AVFMT_GLOBALHEADER) + enc_ctx->flags |= AV_CODEC_FLAG_GLOBAL_HEADER; + + } + av_dump_format(ofmt_ctx, 0, filename, 1); + + if (!(ofmt_ctx->oformat->flags & AVFMT_NOFILE)) { + ret = avio_open(&ofmt_ctx->pb, filename, AVIO_FLAG_WRITE); + if (ret < 0) { + av_log(NULL, AV_LOG_ERROR, "Could not open output file '%s'", filename); + return ret; + } + } + + /* init muxer, write output file header */ + ret = avformat_write_header(ofmt_ctx, NULL); + if (ret < 0) { + av_log(NULL, AV_LOG_ERROR, "Error occurred when opening output file\n"); + return ret; + } + + return 0; +} + +static int init_filter(FilteringContext* fctx, AVCodecContext *dec_ctx, + AVCodecContext *enc_ctx, const char *filter_spec) +{ + char args[512]; + int ret = 0; + AVFilter *buffersrc = NULL; + AVFilter *buffersink = NULL; + AVFilterContext *buffersrc_ctx = NULL; + AVFilterContext *buffersink_ctx = NULL; + AVFilterInOut *outputs = avfilter_inout_alloc(); + AVFilterInOut *inputs = avfilter_inout_alloc(); + AVFilterGraph *filter_graph = avfilter_graph_alloc(); + + if (!outputs || !inputs || !filter_graph) { + ret = AVERROR(ENOMEM); + goto end; + } + + if (dec_ctx->codec_type == AVMEDIA_TYPE_VIDEO) { + buffersrc = avfilter_get_by_name("buffer"); + buffersink = avfilter_get_by_name("buffersink"); + if (!buffersrc || !buffersink) { + av_log(NULL, AV_LOG_ERROR, "filtering source or sink element not found\n"); + ret = AVERROR_UNKNOWN; + goto end; + } + + snprintf(args, sizeof(args), + "video_size=%dx%d:pix_fmt=%d:time_base=%d/%d:pixel_aspect=%d/%d", + dec_ctx->width, dec_ctx->height, dec_ctx->pix_fmt, + dec_ctx->time_base.num, dec_ctx->time_base.den, + dec_ctx->sample_aspect_ratio.num, + dec_ctx->sample_aspect_ratio.den); + + ret = avfilter_graph_create_filter(&buffersrc_ctx, buffersrc, "in", + args, NULL, filter_graph); + if (ret < 0) { + av_log(NULL, AV_LOG_ERROR, "Cannot create buffer source\n"); + goto end; + } + + ret = avfilter_graph_create_filter(&buffersink_ctx, buffersink, "out", + NULL, NULL, filter_graph); + if (ret < 0) { + av_log(NULL, AV_LOG_ERROR, "Cannot create buffer sink\n"); + goto end; + } + + ret = av_opt_set_bin(buffersink_ctx, "pix_fmts", + (uint8_t*)&enc_ctx->pix_fmt, sizeof(enc_ctx->pix_fmt), + AV_OPT_SEARCH_CHILDREN); + if (ret < 0) { + av_log(NULL, AV_LOG_ERROR, "Cannot set output pixel format\n"); + goto end; + } + } else if (dec_ctx->codec_type == AVMEDIA_TYPE_AUDIO) { + buffersrc = avfilter_get_by_name("abuffer"); + buffersink = avfilter_get_by_name("abuffersink"); + if (!buffersrc || !buffersink) { + av_log(NULL, AV_LOG_ERROR, "filtering source or sink element not found\n"); + ret = AVERROR_UNKNOWN; + goto end; + } + + if (!dec_ctx->channel_layout) + dec_ctx->channel_layout = + av_get_default_channel_layout(dec_ctx->channels); + snprintf(args, sizeof(args), + "time_base=%d/%d:sample_rate=%d:sample_fmt=%s:channel_layout=0x%"PRIx64, + dec_ctx->time_base.num, dec_ctx->time_base.den, dec_ctx->sample_rate, + av_get_sample_fmt_name(dec_ctx->sample_fmt), + dec_ctx->channel_layout); + ret = avfilter_graph_create_filter(&buffersrc_ctx, buffersrc, "in", + args, NULL, filter_graph); + if (ret < 0) { + av_log(NULL, AV_LOG_ERROR, "Cannot create audio buffer source\n"); + goto end; + } + + ret = avfilter_graph_create_filter(&buffersink_ctx, buffersink, "out", + NULL, NULL, filter_graph); + if (ret < 0) { + av_log(NULL, AV_LOG_ERROR, "Cannot create audio buffer sink\n"); + goto end; + } + + ret = av_opt_set_bin(buffersink_ctx, "sample_fmts", + (uint8_t*)&enc_ctx->sample_fmt, sizeof(enc_ctx->sample_fmt), + AV_OPT_SEARCH_CHILDREN); + if (ret < 0) { + av_log(NULL, AV_LOG_ERROR, "Cannot set output sample format\n"); + goto end; + } + + ret = av_opt_set_bin(buffersink_ctx, "channel_layouts", + (uint8_t*)&enc_ctx->channel_layout, + sizeof(enc_ctx->channel_layout), AV_OPT_SEARCH_CHILDREN); + if (ret < 0) { + av_log(NULL, AV_LOG_ERROR, "Cannot set output channel layout\n"); + goto end; + } + + ret = av_opt_set_bin(buffersink_ctx, "sample_rates", + (uint8_t*)&enc_ctx->sample_rate, sizeof(enc_ctx->sample_rate), + AV_OPT_SEARCH_CHILDREN); + if (ret < 0) { + av_log(NULL, AV_LOG_ERROR, "Cannot set output sample rate\n"); + goto end; + } + } else { + ret = AVERROR_UNKNOWN; + goto end; + } + + /* Endpoints for the filter graph. */ + outputs->name = av_strdup("in"); + outputs->filter_ctx = buffersrc_ctx; + outputs->pad_idx = 0; + outputs->next = NULL; + + inputs->name = av_strdup("out"); + inputs->filter_ctx = buffersink_ctx; + inputs->pad_idx = 0; + inputs->next = NULL; + + if (!outputs->name || !inputs->name) { + ret = AVERROR(ENOMEM); + goto end; + } + + if ((ret = avfilter_graph_parse_ptr(filter_graph, filter_spec, + &inputs, &outputs, NULL)) < 0) + goto end; + + if ((ret = avfilter_graph_config(filter_graph, NULL)) < 0) + goto end; + + /* Fill FilteringContext */ + fctx->buffersrc_ctx = buffersrc_ctx; + fctx->buffersink_ctx = buffersink_ctx; + fctx->filter_graph = filter_graph; + +end: + avfilter_inout_free(&inputs); + avfilter_inout_free(&outputs); + + return ret; +} + +static int init_filters(void) +{ + const char *filter_spec; + unsigned int i; + int ret; + filter_ctx = av_malloc_array(ifmt_ctx->nb_streams, sizeof(*filter_ctx)); + if (!filter_ctx) + return AVERROR(ENOMEM); + + for (i = 0; i < ifmt_ctx->nb_streams; i++) { + filter_ctx[i].buffersrc_ctx = NULL; + filter_ctx[i].buffersink_ctx = NULL; + filter_ctx[i].filter_graph = NULL; + if (!(ifmt_ctx->streams[i]->codec->codec_type == AVMEDIA_TYPE_AUDIO + || ifmt_ctx->streams[i]->codec->codec_type == AVMEDIA_TYPE_VIDEO)) + continue; + + + if (ifmt_ctx->streams[i]->codec->codec_type == AVMEDIA_TYPE_VIDEO) + filter_spec = "null"; /* passthrough (dummy) filter for video */ + else + filter_spec = "anull"; /* passthrough (dummy) filter for audio */ + ret = init_filter(&filter_ctx[i], ifmt_ctx->streams[i]->codec, + ofmt_ctx->streams[i]->codec, filter_spec); + if (ret) + return ret; + } + return 0; +} + +static int encode_write_frame(AVFrame *filt_frame, unsigned int stream_index, int *got_frame) { + int ret; + int got_frame_local; + AVPacket enc_pkt; + int (*enc_func)(AVCodecContext *, AVPacket *, const AVFrame *, int *) = + (ifmt_ctx->streams[stream_index]->codec->codec_type == + AVMEDIA_TYPE_VIDEO) ? avcodec_encode_video2 : avcodec_encode_audio2; + + if (!got_frame) + got_frame = &got_frame_local; + + av_log(NULL, AV_LOG_INFO, "Encoding frame\n"); + /* encode filtered frame */ + enc_pkt.data = NULL; + enc_pkt.size = 0; + av_init_packet(&enc_pkt); + ret = enc_func(ofmt_ctx->streams[stream_index]->codec, &enc_pkt, + filt_frame, got_frame); + av_frame_free(&filt_frame); + if (ret < 0) + return ret; + if (!(*got_frame)) + return 0; + + /* prepare packet for muxing */ + enc_pkt.stream_index = stream_index; + av_packet_rescale_ts(&enc_pkt, + ofmt_ctx->streams[stream_index]->codec->time_base, + ofmt_ctx->streams[stream_index]->time_base); + + av_log(NULL, AV_LOG_DEBUG, "Muxing frame\n"); + /* mux encoded frame */ + ret = av_interleaved_write_frame(ofmt_ctx, &enc_pkt); + return ret; +} + +static int filter_encode_write_frame(AVFrame *frame, unsigned int stream_index) +{ + int ret; + AVFrame *filt_frame; + + av_log(NULL, AV_LOG_INFO, "Pushing decoded frame to filters\n"); + /* push the decoded frame into the filtergraph */ + ret = av_buffersrc_add_frame_flags(filter_ctx[stream_index].buffersrc_ctx, + frame, 0); + if (ret < 0) { + av_log(NULL, AV_LOG_ERROR, "Error while feeding the filtergraph\n"); + return ret; + } + + /* pull filtered frames from the filtergraph */ + while (1) { + filt_frame = av_frame_alloc(); + if (!filt_frame) { + ret = AVERROR(ENOMEM); + break; + } + av_log(NULL, AV_LOG_INFO, "Pulling filtered frame from filters\n"); + ret = av_buffersink_get_frame(filter_ctx[stream_index].buffersink_ctx, + filt_frame); + if (ret < 0) { + /* if no more frames for output - returns AVERROR(EAGAIN) + * if flushed and no more frames for output - returns AVERROR_EOF + * rewrite retcode to 0 to show it as normal procedure completion + */ + if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF) + ret = 0; + av_frame_free(&filt_frame); + break; + } + + filt_frame->pict_type = AV_PICTURE_TYPE_NONE; + ret = encode_write_frame(filt_frame, stream_index, NULL); + if (ret < 0) + break; + } + + return ret; +} + +static int flush_encoder(unsigned int stream_index) +{ + int ret; + int got_frame; + + if (!(ofmt_ctx->streams[stream_index]->codec->codec->capabilities & + AV_CODEC_CAP_DELAY)) + return 0; + + while (1) { + av_log(NULL, AV_LOG_INFO, "Flushing stream #%u encoder\n", stream_index); + ret = encode_write_frame(NULL, stream_index, &got_frame); + if (ret < 0) + break; + if (!got_frame) + return 0; + } + return ret; +} + +int main(int argc, char **argv) +{ + int ret; + AVPacket packet = { .data = NULL, .size = 0 }; + AVFrame *frame = NULL; + enum AVMediaType type; + unsigned int stream_index; + unsigned int i; + int got_frame; + int (*dec_func)(AVCodecContext *, AVFrame *, int *, const AVPacket *); + + if (argc != 3) { + av_log(NULL, AV_LOG_ERROR, "Usage: %s <input file> <output file>\n", argv[0]); + return 1; + } + + av_register_all(); + avfilter_register_all(); + + if ((ret = open_input_file(argv[1])) < 0) + goto end; + if ((ret = open_output_file(argv[2])) < 0) + goto end; + if ((ret = init_filters()) < 0) + goto end; + + /* read all packets */ + while (1) { + if ((ret = av_read_frame(ifmt_ctx, &packet)) < 0) + break; + stream_index = packet.stream_index; + type = ifmt_ctx->streams[packet.stream_index]->codec->codec_type; + av_log(NULL, AV_LOG_DEBUG, "Demuxer gave frame of stream_index %u\n", + stream_index); + + if (filter_ctx[stream_index].filter_graph) { + av_log(NULL, AV_LOG_DEBUG, "Going to reencode&filter the frame\n"); + frame = av_frame_alloc(); + if (!frame) { + ret = AVERROR(ENOMEM); + break; + } + av_packet_rescale_ts(&packet, + ifmt_ctx->streams[stream_index]->time_base, + ifmt_ctx->streams[stream_index]->codec->time_base); + dec_func = (type == AVMEDIA_TYPE_VIDEO) ? avcodec_decode_video2 : + avcodec_decode_audio4; + ret = dec_func(ifmt_ctx->streams[stream_index]->codec, frame, + &got_frame, &packet); + if (ret < 0) { + av_frame_free(&frame); + av_log(NULL, AV_LOG_ERROR, "Decoding failed\n"); + break; + } + + if (got_frame) { + frame->pts = av_frame_get_best_effort_timestamp(frame); + ret = filter_encode_write_frame(frame, stream_index); + av_frame_free(&frame); + if (ret < 0) + goto end; + } else { + av_frame_free(&frame); + } + } else { + /* remux this frame without reencoding */ + av_packet_rescale_ts(&packet, + ifmt_ctx->streams[stream_index]->time_base, + ofmt_ctx->streams[stream_index]->time_base); + + ret = av_interleaved_write_frame(ofmt_ctx, &packet); + if (ret < 0) + goto end; + } + av_packet_unref(&packet); + } + + /* flush filters and encoders */ + for (i = 0; i < ifmt_ctx->nb_streams; i++) { + /* flush filter */ + if (!filter_ctx[i].filter_graph) + continue; + ret = filter_encode_write_frame(NULL, i); + if (ret < 0) { + av_log(NULL, AV_LOG_ERROR, "Flushing filter failed\n"); + goto end; + } + + /* flush encoder */ + ret = flush_encoder(i); + if (ret < 0) { + av_log(NULL, AV_LOG_ERROR, "Flushing encoder failed\n"); + goto end; + } + } + + av_write_trailer(ofmt_ctx); +end: + av_packet_unref(&packet); + av_frame_free(&frame); + for (i = 0; i < ifmt_ctx->nb_streams; i++) { + avcodec_close(ifmt_ctx->streams[i]->codec); + if (ofmt_ctx && ofmt_ctx->nb_streams > i && ofmt_ctx->streams[i] && ofmt_ctx->streams[i]->codec) + avcodec_close(ofmt_ctx->streams[i]->codec); + if (filter_ctx && filter_ctx[i].filter_graph) + avfilter_graph_free(&filter_ctx[i].filter_graph); + } + av_free(filter_ctx); + avformat_close_input(&ifmt_ctx); + if (ofmt_ctx && !(ofmt_ctx->oformat->flags & AVFMT_NOFILE)) + avio_closep(&ofmt_ctx->pb); + avformat_free_context(ofmt_ctx); + + if (ret < 0) + av_log(NULL, AV_LOG_ERROR, "Error occurred: %s\n", av_err2str(ret)); + + return ret ? 1 : 0; +} diff --git a/doc/faq.texi b/doc/faq.texi index b400124f69..ef111c70e5 100644 --- a/doc/faq.texi +++ b/doc/faq.texi @@ -1,8 +1,9 @@ \input texinfo @c -*- texinfo -*- +@documentencoding UTF-8 -@settitle Libav FAQ +@settitle FFmpeg FAQ @titlepage -@center @titlefont{Libav FAQ} +@center @titlefont{FFmpeg FAQ} @end titlepage @top @@ -11,23 +12,23 @@ @chapter General Questions -@section Why doesn't Libav support feature [xyz]? +@section Why doesn't FFmpeg support feature [xyz]? -Because no one has taken on that task yet. Libav development is +Because no one has taken on that task yet. FFmpeg development is driven by the tasks that are important to the individual developers. If there is a feature that is important to you, the best way to get it implemented is to undertake the task yourself or sponsor a developer. -@section Libav does not support codec XXX. Can you include a Windows DLL loader to support it? +@section FFmpeg does not support codec XXX. Can you include a Windows DLL loader to support it? No. Windows DLLs are not portable, bloated and often slow. -Moreover Libav strives to support all codecs natively. +Moreover FFmpeg strives to support all codecs natively. A DLL loader is not conducive to that goal. -@section I cannot read this file although this format seems to be supported by avconv. +@section I cannot read this file although this format seems to be supported by ffmpeg. -Even if avconv can read the container format, it may not support all its -codecs. Please consult the supported codec list in the avconv +Even if ffmpeg can read the container format, it may not support all its +codecs. Please consult the supported codec list in the ffmpeg documentation. @section Which codecs are supported by Windows? @@ -79,8 +80,75 @@ not a bug they should fix: Then again, some of them do not know the difference between an undecidable problem and an NP-hard problem... +@section I have installed this library with my distro's package manager. Why does @command{configure} not see it? + +Distributions usually split libraries in several packages. The main package +contains the files necessary to run programs using the library. The +development package contains the files necessary to build programs using the +library. Sometimes, docs and/or data are in a separate package too. + +To build FFmpeg, you need to install the development package. It is usually +called @file{libfoo-dev} or @file{libfoo-devel}. You can remove it after the +build is finished, but be sure to keep the main package. + +@section How do I make @command{pkg-config} find my libraries? + +Somewhere along with your libraries, there is a @file{.pc} file (or several) +in a @file{pkgconfig} directory. You need to set environment variables to +point @command{pkg-config} to these files. + +If you need to @emph{add} directories to @command{pkg-config}'s search list +(typical use case: library installed separately), add it to +@code{$PKG_CONFIG_PATH}: + +@example +export PKG_CONFIG_PATH=/opt/x264/lib/pkgconfig:/opt/opus/lib/pkgconfig +@end example + +If you need to @emph{replace} @command{pkg-config}'s search list +(typical use case: cross-compiling), set it in +@code{$PKG_CONFIG_LIBDIR}: + +@example +export PKG_CONFIG_LIBDIR=/home/me/cross/usr/lib/pkgconfig:/home/me/cross/usr/local/lib/pkgconfig +@end example + +If you need to know the library's internal dependencies (typical use: static +linking), add the @code{--static} option to @command{pkg-config}: + +@example +./configure --pkg-config-flags=--static +@end example + +@section How do I use @command{pkg-config} when cross-compiling? + +The best way is to install @command{pkg-config} in your cross-compilation +environment. It will automatically use the cross-compilation libraries. + +You can also use @command{pkg-config} from the host environment by +specifying explicitly @code{--pkg-config=pkg-config} to @command{configure}. +In that case, you must point @command{pkg-config} to the correct directories +using the @code{PKG_CONFIG_LIBDIR}, as explained in the previous entry. + +As an intermediate solution, you can place in your cross-compilation +environment a script that calls the host @command{pkg-config} with +@code{PKG_CONFIG_LIBDIR} set. That script can look like that: + +@example +#!/bin/sh +PKG_CONFIG_LIBDIR=/path/to/cross/lib/pkgconfig +export PKG_CONFIG_LIBDIR +exec /usr/bin/pkg-config "$@@" +@end example + @chapter Usage +@section ffmpeg does not work; what is wrong? + +Try a @code{make distclean} in the ffmpeg source directory before the build. +If this does not help see +(@url{https://ffmpeg.org/bugreports.html}). + @section How do I encode single pictures into movies? First, rename your pictures to follow a numerical sequence. @@ -88,12 +156,21 @@ For example, img1.jpg, img2.jpg, img3.jpg,... Then you may run: @example - avconv -f image2 -i img%d.jpg /tmp/a.mpg +ffmpeg -f image2 -i img%d.jpg /tmp/a.mpg @end example Notice that @samp{%d} is replaced by the image number. -@file{img%03d.jpg} means the sequence @file{img001.jpg}, @file{img002.jpg}, etc... +@file{img%03d.jpg} means the sequence @file{img001.jpg}, @file{img002.jpg}, etc. + +Use the @option{-start_number} option to declare a starting number for +the sequence. This is useful if your sequence does not start with +@file{img001.jpg} but is still in a numerical order. The following +example will start with @file{img100.jpg}: + +@example +ffmpeg -f image2 -start_number 100 -i img%d.jpg /tmp/a.mpg +@end example If you have large number of pictures to rename, you can use the following command to ease the burden. The command, using the bourne @@ -102,7 +179,7 @@ that match @code{*jpg} to the @file{/tmp} directory in the sequence of @file{img001.jpg}, @file{img002.jpg} and so on. @example - x=1; for i in *jpg; do counter=$(printf %03d $x); ln -s "$i" /tmp/img"$counter".jpg; x=$(($x+1)); done +x=1; for i in *jpg; do counter=$(printf %03d $x); ln -s "$i" /tmp/img"$counter".jpg; x=$(($x+1)); done @end example If you want to sequence them by oldest modified first, substitute @@ -111,17 +188,23 @@ If you want to sequence them by oldest modified first, substitute Then run: @example - avconv -f image2 -i /tmp/img%03d.jpg /tmp/a.mpg +ffmpeg -f image2 -i /tmp/img%03d.jpg /tmp/a.mpg @end example -The same logic is used for any image format that avconv reads. +The same logic is used for any image format that ffmpeg reads. + +You can also use @command{cat} to pipe images to ffmpeg: + +@example +cat *.jpg | ffmpeg -f image2pipe -c:v mjpeg -i - output.mpg +@end example @section How do I encode movie to single pictures? Use: @example - avconv -i movie.mpg movie%d.jpg +ffmpeg -i movie.mpg movie%d.jpg @end example The @file{movie.mpg} used as input will be converted to @@ -137,7 +220,7 @@ to force the encoding. Applying that to the previous example: @example - avconv -i movie.mpg -f image2 -c:v mjpeg menu%d.jpg +ffmpeg -i movie.mpg -f image2 -c:v mjpeg menu%d.jpg @end example Beware that there is no "jpeg" codec. Use "mjpeg" instead. @@ -156,12 +239,12 @@ Use @file{-} as file name. Try '-f image2 test%d.jpg'. -@section Why can I not change the framerate? +@section Why can I not change the frame rate? -Some codecs, like MPEG-1/2, only allow a small number of fixed framerates. +Some codecs, like MPEG-1/2, only allow a small number of fixed frame rates. Choose a different codec with the -c:v command line option. -@section How do I encode Xvid or DivX video with avconv? +@section How do I encode Xvid or DivX video with ffmpeg? Both Xvid and DivX (version 4+) are implementations of the ISO MPEG-4 standard (note that there are many other coding formats that use this @@ -182,24 +265,24 @@ things to try: '-bf 2', '-flags qprd', '-flags mv0', '-flags skiprd'. but beware the '-g 100' might cause problems with some decoders. Things to try: '-bf 2', '-flags qprd', '-flags mv0', '-flags skiprd. -@section Interlaced video looks very bad when encoded with avconv, what is wrong? +@section Interlaced video looks very bad when encoded with ffmpeg, what is wrong? You should use '-flags +ilme+ildct' and maybe '-flags +alt' for interlaced material, and try '-top 0/1' if the result looks really messed-up. @section How can I read DirectShow files? -If you have built Libav with @code{./configure --enable-avisynth} +If you have built FFmpeg with @code{./configure --enable-avisynth} (only possible on MinGW/Cygwin platforms), then you may use any file that DirectShow can read as input. Just create an "input.avs" text file with this single line ... @example - DirectShowSource("C:\path to your file\yourfile.asf") +DirectShowSource("C:\path to your file\yourfile.asf") @end example -... and then feed that text file to avconv: +... and then feed that text file to ffmpeg: @example - avconv -i input.avs +ffmpeg -i input.avs @end example For ANY other help on AviSynth, please visit the @@ -207,8 +290,67 @@ For ANY other help on AviSynth, please visit the @section How can I join video files? -A few multimedia containers (MPEG-1, MPEG-2 PS, DV) allow to join video files by -merely concatenating them. +To "join" video files is quite ambiguous. The following list explains the +different kinds of "joining" and points out how those are addressed in +FFmpeg. To join video files may mean: + +@itemize + +@item +To put them one after the other: this is called to @emph{concatenate} them +(in short: concat) and is addressed +@ref{How can I concatenate video files, in this very faq}. + +@item +To put them together in the same file, to let the user choose between the +different versions (example: different audio languages): this is called to +@emph{multiplex} them together (in short: mux), and is done by simply +invoking ffmpeg with several @option{-i} options. + +@item +For audio, to put all channels together in a single stream (example: two +mono streams into one stereo stream): this is sometimes called to +@emph{merge} them, and can be done using the +@url{https://ffmpeg.org/ffmpeg-filters.html#amerge, @code{amerge}} filter. + +@item +For audio, to play one on top of the other: this is called to @emph{mix} +them, and can be done by first merging them into a single stream and then +using the @url{https://ffmpeg.org/ffmpeg-filters.html#pan, @code{pan}} filter to mix +the channels at will. + +@item +For video, to display both together, side by side or one on top of a part of +the other; it can be done using the +@url{https://ffmpeg.org/ffmpeg-filters.html#overlay, @code{overlay}} video filter. + +@end itemize + +@anchor{How can I concatenate video files} +@section How can I concatenate video files? + +There are several solutions, depending on the exact circumstances. + +@subsection Concatenating using the concat @emph{filter} + +FFmpeg has a @url{https://ffmpeg.org/ffmpeg-filters.html#concat, +@code{concat}} filter designed specifically for that, with examples in the +documentation. This operation is recommended if you need to re-encode. + +@subsection Concatenating using the concat @emph{demuxer} + +FFmpeg has a @url{https://www.ffmpeg.org/ffmpeg-formats.html#concat, +@code{concat}} demuxer which you can use when you want to avoid a re-encode and +your format doesn't support file level concatenation. + +@subsection Concatenating using the concat @emph{protocol} (file level) + +FFmpeg has a @url{https://ffmpeg.org/ffmpeg-protocols.html#concat, +@code{concat}} protocol designed specifically for that, with examples in the +documentation. + +A few multimedia containers (MPEG-1, MPEG-2 PS, DV) allow one to concatenate +video by merely concatenating the files containing them. Hence you may concatenate your multimedia files by first transcoding them to these privileged formats, then using the humble @code{cat} command (or the @@ -216,27 +358,38 @@ equally humble @code{copy} under Windows), and finally transcoding back to your format of choice. @example -avconv -i input1.avi intermediate1.mpg -avconv -i input2.avi intermediate2.mpg +ffmpeg -i input1.avi -qscale:v 1 intermediate1.mpg +ffmpeg -i input2.avi -qscale:v 1 intermediate2.mpg cat intermediate1.mpg intermediate2.mpg > intermediate_all.mpg -avconv -i intermediate_all.mpg output.avi +ffmpeg -i intermediate_all.mpg -qscale:v 2 output.avi @end example -Notice that you should set a reasonably high bitrate for your intermediate and -output files, if you want to preserve video quality. +Additionally, you can use the @code{concat} protocol instead of @code{cat} or +@code{copy} which will avoid creation of a potentially huge intermediate file. -Also notice that you may avoid the huge intermediate files by taking advantage -of named pipes, should your platform support it: +@example +ffmpeg -i input1.avi -qscale:v 1 intermediate1.mpg +ffmpeg -i input2.avi -qscale:v 1 intermediate2.mpg +ffmpeg -i concat:"intermediate1.mpg|intermediate2.mpg" -c copy intermediate_all.mpg +ffmpeg -i intermediate_all.mpg -qscale:v 2 output.avi +@end example + +Note that you may need to escape the character "|" which is special for many +shells. + +Another option is usage of named pipes, should your platform support it: @example mkfifo intermediate1.mpg mkfifo intermediate2.mpg -avconv -i input1.avi -y intermediate1.mpg < /dev/null & -avconv -i input2.avi -y intermediate2.mpg < /dev/null & +ffmpeg -i input1.avi -qscale:v 1 -y intermediate1.mpg < /dev/null & +ffmpeg -i input2.avi -qscale:v 1 -y intermediate2.mpg < /dev/null & cat intermediate1.mpg intermediate2.mpg |\ -avconv -f mpeg -i - -c:v mpeg4 -acodec libmp3lame output.avi +ffmpeg -f mpeg -i - -c:v mpeg4 -acodec libmp3lame output.avi @end example +@subsection Concatenating using raw audio and video + Similarly, the yuv4mpegpipe format, and the raw video, raw audio codecs also allow concatenation, and the transcoding step is almost lossless. When using multiple yuv4mpegpipe(s), the first line needs to be discarded @@ -244,7 +397,8 @@ from all but the first stream. This can be accomplished by piping through @code{tail} as seen below. Note that when piping through @code{tail} you must use command grouping, @code{@{ ;@}}, to background properly. -For example, let's say we want to join two FLV files into an output.flv file: +For example, let's say we want to concatenate two FLV files into an +output.flv file: @example mkfifo temp1.a @@ -253,45 +407,114 @@ mkfifo temp2.a mkfifo temp2.v mkfifo all.a mkfifo all.v -avconv -i input1.flv -vn -f u16le -acodec pcm_s16le -ac 2 -ar 44100 - > temp1.a < /dev/null & -avconv -i input2.flv -vn -f u16le -acodec pcm_s16le -ac 2 -ar 44100 - > temp2.a < /dev/null & -avconv -i input1.flv -an -f yuv4mpegpipe - > temp1.v < /dev/null & -@{ avconv -i input2.flv -an -f yuv4mpegpipe - < /dev/null | tail -n +2 > temp2.v ; @} & +ffmpeg -i input1.flv -vn -f u16le -acodec pcm_s16le -ac 2 -ar 44100 - > temp1.a < /dev/null & +ffmpeg -i input2.flv -vn -f u16le -acodec pcm_s16le -ac 2 -ar 44100 - > temp2.a < /dev/null & +ffmpeg -i input1.flv -an -f yuv4mpegpipe - > temp1.v < /dev/null & +@{ ffmpeg -i input2.flv -an -f yuv4mpegpipe - < /dev/null | tail -n +2 > temp2.v ; @} & cat temp1.a temp2.a > all.a & cat temp1.v temp2.v > all.v & -avconv -f u16le -acodec pcm_s16le -ac 2 -ar 44100 -i all.a \ +ffmpeg -f u16le -acodec pcm_s16le -ac 2 -ar 44100 -i all.a \ -f yuv4mpegpipe -i all.v \ -y output.flv rm temp[12].[av] all.[av] @end example -@section -profile option fails when encoding H.264 video with AAC audio +@section Using @option{-f lavfi}, audio becomes mono for no apparent reason. + +Use @option{-dumpgraph -} to find out exactly where the channel layout is +lost. -@command{avconv} prints an error like +Most likely, it is through @code{auto-inserted aresample}. Try to understand +why the converting filter was needed at that place. + +Just before the output is a likely place, as @option{-f lavfi} currently +only support packed S16. + +Then insert the correct @code{aformat} explicitly in the filtergraph, +specifying the exact format. @example -Undefined constant or missing '(' in 'baseline' -Unable to parse option value "baseline" -Error setting option profile to value baseline. +aformat=sample_fmts=s16:channel_layouts=stereo @end example -Short answer: write @option{-profile:v} instead of @option{-profile}. +@section Why does FFmpeg not see the subtitles in my VOB file? + +VOB and a few other formats do not have a global header that describes +everything present in the file. Instead, applications are supposed to scan +the file to see what it contains. Since VOB files are frequently large, only +the beginning is scanned. If the subtitles happen only later in the file, +they will not be initially detected. + +Some applications, including the @code{ffmpeg} command-line tool, can only +work with streams that were detected during the initial scan; streams that +are detected later are ignored. + +The size of the initial scan is controlled by two options: @code{probesize} +(default ~5 Mo) and @code{analyzeduration} (default 5,000,000 µs = 5 s). For +the subtitle stream to be detected, both values must be large enough. + +@section Why was the @command{ffmpeg} @option{-sameq} option removed? What to use instead? + +The @option{-sameq} option meant "same quantizer", and made sense only in a +very limited set of cases. Unfortunately, a lot of people mistook it for +"same quality" and used it in places where it did not make sense: it had +roughly the expected visible effect, but achieved it in a very inefficient +way. + +Each encoder has its own set of options to set the quality-vs-size balance, +use the options for the encoder you are using to set the quality level to a +point acceptable for your tastes. The most common options to do that are +@option{-qscale} and @option{-qmax}, but you should peruse the documentation +of the encoder you chose. + +@section I have a stretched video, why does scaling does not fix it? + +A lot of video codecs and formats can store the @emph{aspect ratio} of the +video: this is the ratio between the width and the height of either the full +image (DAR, display aspect ratio) or individual pixels (SAR, sample aspect +ratio). For example, EGA screens at resolution 640×350 had 4:3 DAR and 35:48 +SAR. + +Most still image processing work with square pixels, i.e. 1:1 SAR, but a lot +of video standards, especially from the analogic-numeric transition era, use +non-square pixels. -Long answer: this happens because the @option{-profile} option can apply to both -video and audio. Specifically the AAC encoder also defines some profiles, none -of which are named @var{baseline}. +Most processing filters in FFmpeg handle the aspect ratio to avoid +stretching the image: cropping adjusts the DAR to keep the SAR constant, +scaling adjusts the SAR to keep the DAR constant. -The solution is to apply the @option{-profile} option to the video stream only -by using @url{http://libav.org/avconv.html#Stream-specifiers-1, Stream specifiers}. -Appending @code{:v} to it will do exactly that. +If you want to stretch, or “unstretch”, the image, you need to override the +information with the +@url{https://ffmpeg.org/ffmpeg-filters.html#setdar_002c-setsar, @code{setdar or setsar filters}}. + +Do not forget to examine carefully the original video to check whether the +stretching comes from the image or from the aspect ratio information. + +For example, to fix a badly encoded EGA capture, use the following commands, +either the first one to upscale to square pixels or the second one to set +the correct aspect ratio or the third one to avoid transcoding (may not work +depending on the format / codec / player / phase of the moon): + +@example +ffmpeg -i ega_screen.nut -vf scale=640:480,setsar=1 ega_screen_scaled.nut +ffmpeg -i ega_screen.nut -vf setdar=4/3 ega_screen_anamorphic.nut +ffmpeg -i ega_screen.nut -aspect 4/3 -c copy ega_screen_overridden.nut +@end example @chapter Development -@section Are there examples illustrating how to use the Libav libraries, particularly libavcodec and libavformat? +@section Are there examples illustrating how to use the FFmpeg libraries, particularly libavcodec and libavformat? + +Yes. Check the @file{doc/examples} directory in the source +repository, also available online at: +@url{https://github.com/FFmpeg/FFmpeg/tree/master/doc/examples}. -Yes. Read the Developers Guide of the Libav documentation. Alternatively, +Examples are also installed by default, usually in +@code{$PREFIX/share/ffmpeg/examples}. + +Also you may read the Developers Guide of the FFmpeg documentation. Alternatively, examine the source code for one of the many open source projects that -already incorporate Libav at (@url{projects.html}). +already incorporate FFmpeg at (@url{projects.html}). @section Can you support my C compiler XXX? @@ -302,42 +525,86 @@ with @code{#ifdef}s related to the compiler. @section Is Microsoft Visual C++ supported? Yes. Please see the @uref{platform.html, Microsoft Visual C++} -section in the Libav documentation. +section in the FFmpeg documentation. @section Can you add automake, libtool or autoconf support? No. These tools are too bloated and they complicate the build. -@section Why not rewrite Libav in object-oriented C++? +@section Why not rewrite FFmpeg in object-oriented C++? -Libav is already organized in a highly modular manner and does not need to +FFmpeg is already organized in a highly modular manner and does not need to be rewritten in a formal object language. Further, many of the developers favor straight C; it works for them. For more arguments on this matter, read @uref{http://www.tux.org/lkml/#s15, "Programming Religion"}. +@section Why are the ffmpeg programs devoid of debugging symbols? + +The build process creates @command{ffmpeg_g}, @command{ffplay_g}, etc. which +contain full debug information. Those binaries are stripped to create +@command{ffmpeg}, @command{ffplay}, etc. If you need the debug information, use +the *_g versions. + @section I do not like the LGPL, can I contribute code under the GPL instead? Yes, as long as the code is optional and can easily and cleanly be placed -under #if CONFIG_GPL without breaking anything. So for example a new codec +under #if CONFIG_GPL without breaking anything. So, for example, a new codec or filter would be OK under GPL while a bug fix to LGPL code would not. -@section I'm using Libav from within my C++ application but the linker complains about missing symbols which seem to be available. +@section I'm using FFmpeg from within my C application but the linker complains about missing symbols from the libraries themselves. + +FFmpeg builds static libraries by default. In static libraries, dependencies +are not handled. That has two consequences. First, you must specify the +libraries in dependency order: @code{-lavdevice} must come before +@code{-lavformat}, @code{-lavutil} must come after everything else, etc. +Second, external libraries that are used in FFmpeg have to be specified too. + +An easy way to get the full list of required libraries in dependency order +is to use @code{pkg-config}. -Libav is a pure C project, so to use the libraries within your C++ application +@example +c99 -o program program.c $(pkg-config --cflags --libs libavformat libavcodec) +@end example + +See @file{doc/example/Makefile} and @file{doc/example/pc-uninstalled} for +more details. + +@section I'm using FFmpeg from within my C++ application but the linker complains about missing symbols which seem to be available. + +FFmpeg is a pure C project, so to use the libraries within your C++ application you need to explicitly state that you are using a C library. You can do this by -encompassing your Libav includes using @code{extern "C"}. +encompassing your FFmpeg includes using @code{extern "C"}. See @url{http://www.parashift.com/c++-faq-lite/mixing-c-and-cpp.html#faq-32.3} @section I'm using libavutil from within my C++ application but the compiler complains about 'UINT64_C' was not declared in this scope -Libav is a pure C project using C99 math features, in order to enable C++ +FFmpeg is a pure C project using C99 math features, in order to enable C++ to use them you have to append -D__STDC_CONSTANT_MACROS to your CXXFLAGS @section I have a file in memory / a API different from *open/*read/ libc how do I use it with libavformat? You have to create a custom AVIOContext using @code{avio_alloc_context}, -see @file{libavformat/aviobuf.c} in Libav and @file{libmpdemux/demux_lavf.c} in MPlayer2 sources. +see @file{libavformat/aviobuf.c} in FFmpeg and @file{libmpdemux/demux_lavf.c} in MPlayer or MPlayer2 sources. + +@section Where is the documentation about ffv1, msmpeg4, asv1, 4xm? + +see @url{https://www.ffmpeg.org/~michael/} + +@section How do I feed H.263-RTP (and other codecs in RTP) to libavcodec? + +Even if peculiar since it is network oriented, RTP is a container like any +other. You have to @emph{demux} RTP before feeding the payload to libavcodec. +In this specific case please look at RFC 4629 to see how it should be done. + +@section AVStream.r_frame_rate is wrong, it is much larger than the frame rate. + +@code{r_frame_rate} is NOT the average frame rate, it is the smallest frame rate +that can accurately represent all timestamps. So no, it is not +wrong if it is larger than the average! +For example, if you have mixed 25 and 30 fps content, then @code{r_frame_rate} +will be 150 (it is the least common multiple). +If you are looking for the average frame rate, see @code{AVStream.avg_frame_rate}. @section Why is @code{make fate} not running all tests? diff --git a/doc/fate.texi b/doc/fate.texi index d6beaa5c24..353443a17b 100644 --- a/doc/fate.texi +++ b/doc/fate.texi @@ -1,84 +1,179 @@ \input texinfo @c -*- texinfo -*- +@documentencoding UTF-8 -@settitle FATE Automated Testing Environment +@settitle FFmpeg Automated Testing Environment @titlepage -@center @titlefont{FATE Automated Testing Environment} +@center @titlefont{FFmpeg Automated Testing Environment} @end titlepage +@node Top @top @contents @chapter Introduction -FATE provides a regression testsuite embedded within the Libav build system. -It can be run locally and optionally configured to send reports to a web -aggregator and viewer @url{http://fate.libav.org}. +FATE is an extended regression suite on the client-side and a means +for results aggregation and presentation on the server-side. -It is advised to run FATE before submitting patches to the current codebase -and provide new tests when submitting patches to add additional features. +The first part of this document explains how you can use FATE from +your FFmpeg source directory to test your ffmpeg binary. The second +part describes how you can run FATE to submit the results to FFmpeg's +FATE server. -@chapter Running FATE +In any way you can have a look at the publicly viewable FATE results +by visiting this website: -@section Samples and References -In order to run, FATE needs a large amount of data (samples and references) -that is provided separately from the actual source distribution. +@url{http://fate.ffmpeg.org/} -To inform the build system about the testsuite location, pass -@option{--samples=<path to the samples>} to @command{configure} or set the -@var{SAMPLES} Make variable or the @var{LIBAV_SAMPLES} environment variable -to a suitable value. +This is especially recommended for all people contributing source +code to FFmpeg, as it can be seen if some test on some platform broke +with their recent contribution. This usually happens on the platforms +the developers could not test on. -To use a custom wrapper to run the test, pass @option{--target-exec} to -@command{configure} or set the @var{TARGET_EXEC} Make variable. +The second part of this document describes how you can run FATE to +submit your results to FFmpeg's FATE server. If you want to submit your +results be sure to check that your combination of CPU, OS and compiler +is not already listed on the above mentioned website. + +In the third part you can find a comprehensive listing of FATE makefile +targets and variables. -The dataset is available through @command{rsync}, is possible to fetch -the current sample using the straight rsync command or through a specific -@ref{Makefile target}. + +@chapter Using FATE from your FFmpeg source directory + +If you want to run FATE on your machine you need to have the samples +in place. You can get the samples via the build target fate-rsync. +Use this command from the top-level source directory: @example -# rsync -aL rsync://fate-suite.libav.org/fate-suite/ fate-suite +make fate-rsync SAMPLES=fate-suite/ +make fate SAMPLES=fate-suite/ @end example +The above commands set the samples location by passing a makefile +variable via command line. It is also possible to set the samples +location at source configuration time by invoking configure with +@option{--samples=<path to the samples directory>}. Afterwards you can +invoke the makefile targets without setting the @var{SAMPLES} makefile +variable. This is illustrated by the following commands: + @example -# make fate-rsync SAMPLES=fate-suite +./configure --samples=fate-suite/ +make fate-rsync +make fate @end example +Yet another way to tell FATE about the location of the sample +directory is by making sure the environment variable FATE_SAMPLES +contains the path to your samples directory. This can be achieved +by e.g. putting that variable in your shell profile or by setting +it in your interactive session. -@chapter Manual Run -FATE regression test can be run through @command{make}. -Specific Makefile targets and Makefile variables are available: +@example +FATE_SAMPLES=fate-suite/ make fate +@end example -@anchor{Makefile target} -@section FATE Makefile targets +@float NOTE +Do not put a '~' character in the samples path to indicate a home +directory. Because of shell nuances, this will cause FATE to fail. +@end float -@table @option -@item fate-list -List all fate/regression test targets. +To use a custom wrapper to run the test, pass @option{--target-exec} to +@command{configure} or set the @var{TARGET_EXEC} Make variable. -@item fate-rsync -Shortcut to download the fate test samples to the specified testsuite location. -@item fate -Run the FATE test suite (requires the fate-suite dataset). +@chapter Submitting the results to the FFmpeg result aggregation server + +To submit your results to the server you should run fate through the +shell script @file{tests/fate.sh} from the FFmpeg sources. This script needs +to be invoked with a configuration file as its first argument. + +@example +tests/fate.sh /path/to/fate_config +@end example + +A configuration file template with comments describing the individual +configuration variables can be found at @file{doc/fate_config.sh.template}. + +@ifhtml +The mentioned configuration template is also available here: +@verbatiminclude fate_config.sh.template +@end ifhtml + +Create a configuration that suits your needs, based on the configuration +template. The @env{slot} configuration variable can be any string that is not +yet used, but it is suggested that you name it adhering to the following +pattern @samp{@var{arch}-@var{os}-@var{compiler}-@var{compiler version}}. The +configuration file itself will be sourced in a shell script, therefore all +shell features may be used. This enables you to setup the environment as you +need it for your build. + +For your first test runs the @env{fate_recv} variable should be empty or +commented out. This will run everything as normal except that it will omit +the submission of the results to the server. The following files should be +present in $workdir as specified in the configuration file: + +@itemize + @item configure.log + @item compile.log + @item test.log + @item report + @item version +@end itemize + +When you have everything working properly you can create an SSH key pair +and send the public key to the FATE server administrator who can be contacted +at the email address @email{fate-admin@@ffmpeg.org}. + +Configure your SSH client to use public key authentication with that key +when connecting to the FATE server. Also do not forget to check the identity +of the server and to accept its host key. This can usually be achieved by +running your SSH client manually and killing it after you accepted the key. +The FATE server's fingerprint is: + +@table @samp +@item RSA + d3:f1:83:97:a4:75:2b:a6:fb:d6:e8:aa:81:93:97:51 +@item ECDSA + 76:9f:68:32:04:1e:d5:d4:ec:47:3f:dc:fc:18:17:86 @end table -@section FATE Makefile variables -@table @option -@item V -Verbosity level, can be set to 0, 1 or 2. +If you have problems connecting to the FATE server, it may help to try out +the @command{ssh} command with one or more @option{-v} options. You should +get detailed output concerning your SSH configuration and the authentication +process. + +The only thing left is to automate the execution of the fate.sh script and +the synchronisation of the samples directory. + + +@chapter FATE makefile targets and variables + +@section Makefile targets @table @option -@item 0 -show just the test arguments +@item fate-rsync +Download/synchronize sample files to the configured samples directory. -@item 1 -show just the command used in the test +@item fate-list +Will list all fate/regression test targets. -@item 2 -show everything +@item fate +Run the FATE test suite (requires the fate-suite dataset). @end table +@section Makefile variables + +@table @env +@item V +Verbosity level, can be set to 0, 1 or 2. + @itemize + @item 0: show just the test arguments + @item 1: show just the command used in the test + @item 2: show everything + @end itemize + @item SAMPLES Specify or override the path to the FATE samples at make time, it has a meaning only while running the regression tests. @@ -88,81 +183,24 @@ Specify how many threads to use while running regression tests, it is quite useful to detect thread-related regressions. @item THREAD_TYPE -Specify which threading strategy test, either @var{slice} or @var{frame}, -by default @var{slice+frame} +Specify which threading strategy test, either @samp{slice} or @samp{frame}, +by default @samp{slice+frame} @item CPUFLAGS -Specify a mask to be applied to autodetected CPU flags. +Specify CPU flags. @item TARGET_EXEC Specify or override the wrapper used to run the tests. +The @env{TARGET_EXEC} option provides a way to run FATE wrapped in +@command{valgrind}, @command{qemu-user} or @command{wine} or on remote targets +through @command{ssh}. @item GEN -Set to @var{1} to generate the missing or mismatched references. +Set to @samp{1} to generate the missing or mismatched references. @end table -@example - make V=1 SAMPLES=/var/fate/samples THREADS=2 CPUFLAGS=mmx fate -@end example - -@chapter Automated Tests -In order to automatically testing specific configurations, e.g. multiple -compilers, @command{tests/fate.sh} is provided. - -This shell script builds Libav, runs the regression tests and prepares -a report that can be sent to @url{http://fate.libav.org/} or directly -examined locally. - -@section Testing Profiles -The configuration file passed to @command{fate.sh} is shell scripts as well. - -It must provide at least a @var{slot} identifier, the @var{repo} from -which fetch the sources, the @var{samples} directory, a @var{workdir} with -enough space to build and run all the tests. -Optional submit command @var{fate_recv} and a @var{comment} to describe -the testing profile are available. - -Additional optional parameter to tune the Libav building and reporting process -can be passed. +@section Examples @example -slot= # some unique identifier -repo=git://git.libav.org/libav.git # the source repository -#branch=release/10 # the branch to test -samples=/path/to/fate/samples -workdir= # directory in which to do all the work -fate_recv="ssh -T fate@@fate.libav.org" # command to submit report -comment= # optional description -build_only= # set to "yes" for a compile-only instance that skips tests - -# the following are optional and map to configure options -arch= -cpu= -cross_prefix= -as= -cc= -ld= -target_os= -sysroot= -target_exec= -target_path= -target_samples= -extra_cflags= -extra_ldflags= -extra_libs= -extra_conf= # extra configure options not covered above - -#make= # name of GNU make if not 'make' -makeopts= # extra options passed to 'make' -#tar= # command to create a tar archive from its arguments on - # stdout, defaults to 'tar c' +make V=1 SAMPLES=/var/fate/samples THREADS=2 CPUFLAGS=mmx fate @end example - -@section Special Instances -The @var{TARGET_EXEC} option provides a way to run FATE wrapped in -@command{valgrind}, @command{qemu-user} or @command{wine} or on remote targets -through @command{ssh}. - -@section Submitting Reports -In order to send reports you need to create an @command{ssh} key and send it -to @email{root@@libav.org}. diff --git a/doc/fate_config.sh.template b/doc/fate_config.sh.template new file mode 100644 index 0000000000..059a1f862f --- /dev/null +++ b/doc/fate_config.sh.template @@ -0,0 +1,30 @@ +slot= # some unique identifier +repo=git://source.ffmpeg.org/ffmpeg.git # the source repository +#branch=release/2.6 # the branch to test +samples= # path to samples directory +workdir= # directory in which to do all the work +#fate_recv="ssh -T fate@fate.ffmpeg.org" # command to submit report +comment= # optional description +build_only= # set to "yes" for a compile-only instance that skips tests + +# the following are optional and map to configure options +arch= +cpu= +cross_prefix= +as= +cc= +ld= +target_os= +sysroot= +target_exec= +target_path= +target_samples= +extra_cflags= +extra_ldflags= +extra_libs= +extra_conf= # extra configure options not covered above + +#make= # name of GNU make if not 'make' +makeopts= # extra options passed to 'make' +#tar= # command to create a tar archive from its arguments on stdout, + # defaults to 'tar c' diff --git a/doc/ffmpeg-bitstream-filters.texi b/doc/ffmpeg-bitstream-filters.texi new file mode 100644 index 0000000000..bbde25708f --- /dev/null +++ b/doc/ffmpeg-bitstream-filters.texi @@ -0,0 +1,46 @@ +\input texinfo @c -*- texinfo -*- +@documentencoding UTF-8 + +@settitle FFmpeg Bitstream Filters Documentation +@titlepage +@center @titlefont{FFmpeg Bitstream Filters Documentation} +@end titlepage + +@top + +@contents + +@chapter Description +@c man begin DESCRIPTION + +This document describes the bitstream filters provided by the +libavcodec library. + +A bitstream filter operates on the encoded stream data, and performs +bitstream level modifications without performing decoding. + +@c man end DESCRIPTION + +@include bitstream_filters.texi + +@chapter See Also + +@ifhtml +@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver}, +@url{libavcodec.html,libavcodec} +@end ifhtml + +@ifnothtml +ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), libavcodec(3) +@end ifnothtml + +@include authors.texi + +@ignore + +@setfilename ffmpeg-bitstream-filters +@settitle FFmpeg bitstream filters + +@end ignore + +@bye diff --git a/doc/ffmpeg-codecs.texi b/doc/ffmpeg-codecs.texi new file mode 100644 index 0000000000..7df4391ae7 --- /dev/null +++ b/doc/ffmpeg-codecs.texi @@ -0,0 +1,43 @@ +\input texinfo @c -*- texinfo -*- +@documentencoding UTF-8 + +@settitle FFmpeg Codecs Documentation +@titlepage +@center @titlefont{FFmpeg Codecs Documentation} +@end titlepage + +@top + +@contents + +@chapter Description +@c man begin DESCRIPTION + +This document describes the codecs (decoders and encoders) provided by +the libavcodec library. + +@c man end DESCRIPTION + +@include codecs.texi + +@chapter See Also + +@ifhtml +@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver}, +@url{libavcodec.html,libavcodec} +@end ifhtml + +@ifnothtml +ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), libavcodec(3) +@end ifnothtml + +@include authors.texi + +@ignore + +@setfilename ffmpeg-codecs +@settitle FFmpeg codecs + +@end ignore + +@bye diff --git a/doc/ffmpeg-devices.texi b/doc/ffmpeg-devices.texi new file mode 100644 index 0000000000..721c0df800 --- /dev/null +++ b/doc/ffmpeg-devices.texi @@ -0,0 +1,43 @@ +\input texinfo @c -*- texinfo -*- +@documentencoding UTF-8 + +@settitle FFmpeg Devices Documentation +@titlepage +@center @titlefont{FFmpeg Devices Documentation} +@end titlepage + +@top + +@contents + +@chapter Description +@c man begin DESCRIPTION + +This document describes the input and output devices provided by the +libavdevice library. + +@c man end DESCRIPTION + +@include devices.texi + +@chapter See Also + +@ifhtml +@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver}, +@url{libavdevice.html,libavdevice} +@end ifhtml + +@ifnothtml +ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), libavdevice(3) +@end ifnothtml + +@include authors.texi + +@ignore + +@setfilename ffmpeg-devices +@settitle FFmpeg devices + +@end ignore + +@bye diff --git a/doc/ffmpeg-filters.texi b/doc/ffmpeg-filters.texi new file mode 100644 index 0000000000..b643f2c027 --- /dev/null +++ b/doc/ffmpeg-filters.texi @@ -0,0 +1,43 @@ +\input texinfo @c -*- texinfo -*- +@documentencoding UTF-8 + +@settitle FFmpeg Filters Documentation +@titlepage +@center @titlefont{FFmpeg Filters Documentation} +@end titlepage + +@top + +@contents + +@chapter Description +@c man begin DESCRIPTION + +This document describes filters, sources, and sinks provided by the +libavfilter library. + +@c man end DESCRIPTION + +@include filters.texi + +@chapter See Also + +@ifhtml +@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver}, +@url{libavfilter.html,libavfilter} +@end ifhtml + +@ifnothtml +ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), libavfilter(3) +@end ifnothtml + +@include authors.texi + +@ignore + +@setfilename ffmpeg-filters +@settitle FFmpeg filters + +@end ignore + +@bye diff --git a/doc/ffmpeg-formats.texi b/doc/ffmpeg-formats.texi new file mode 100644 index 0000000000..d916ee84b7 --- /dev/null +++ b/doc/ffmpeg-formats.texi @@ -0,0 +1,43 @@ +\input texinfo @c -*- texinfo -*- +@documentencoding UTF-8 + +@settitle FFmpeg Formats Documentation +@titlepage +@center @titlefont{FFmpeg Formats Documentation} +@end titlepage + +@top + +@contents + +@chapter Description +@c man begin DESCRIPTION + +This document describes the supported formats (muxers and demuxers) +provided by the libavformat library. + +@c man end DESCRIPTION + +@include formats.texi + +@chapter See Also + +@ifhtml +@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver}, +@url{libavformat.html,libavformat} +@end ifhtml + +@ifnothtml +ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), libavformat(3) +@end ifnothtml + +@include authors.texi + +@ignore + +@setfilename ffmpeg-formats +@settitle FFmpeg formats + +@end ignore + +@bye diff --git a/doc/ffmpeg-protocols.texi b/doc/ffmpeg-protocols.texi new file mode 100644 index 0000000000..f3a09f6a69 --- /dev/null +++ b/doc/ffmpeg-protocols.texi @@ -0,0 +1,43 @@ +\input texinfo @c -*- texinfo -*- +@documentencoding UTF-8 + +@settitle FFmpeg Protocols Documentation +@titlepage +@center @titlefont{FFmpeg Protocols Documentation} +@end titlepage + +@top + +@contents + +@chapter Description +@c man begin DESCRIPTION + +This document describes the input and output protocols provided by the +libavformat library. + +@c man end DESCRIPTION + +@include protocols.texi + +@chapter See Also + +@ifhtml +@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver}, +@url{libavformat.html,libavformat} +@end ifhtml + +@ifnothtml +ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), libavformat(3) +@end ifnothtml + +@include authors.texi + +@ignore + +@setfilename ffmpeg-protocols +@settitle FFmpeg protocols + +@end ignore + +@bye diff --git a/doc/ffmpeg-resampler.texi b/doc/ffmpeg-resampler.texi new file mode 100644 index 0000000000..be3784f3ed --- /dev/null +++ b/doc/ffmpeg-resampler.texi @@ -0,0 +1,45 @@ +\input texinfo @c -*- texinfo -*- +@documentencoding UTF-8 + +@settitle FFmpeg Resampler Documentation +@titlepage +@center @titlefont{FFmpeg Resampler Documentation} +@end titlepage + +@top + +@contents + +@chapter Description +@c man begin DESCRIPTION + +The FFmpeg resampler provides a high-level interface to the +libswresample library audio resampling utilities. In particular it +allows one to perform audio resampling, audio channel layout rematrixing, +and convert audio format and packing layout. + +@c man end DESCRIPTION + +@include resampler.texi + +@chapter See Also + +@ifhtml +@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver}, +@url{libswresample.html,libswresample} +@end ifhtml + +@ifnothtml +ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), libswresample(3) +@end ifnothtml + +@include authors.texi + +@ignore + +@setfilename ffmpeg-resampler +@settitle FFmpeg Resampler + +@end ignore + +@bye diff --git a/doc/ffmpeg-scaler.texi b/doc/ffmpeg-scaler.texi new file mode 100644 index 0000000000..9ab12a1f95 --- /dev/null +++ b/doc/ffmpeg-scaler.texi @@ -0,0 +1,44 @@ +\input texinfo @c -*- texinfo -*- +@documentencoding UTF-8 + +@settitle FFmpeg Scaler Documentation +@titlepage +@center @titlefont{FFmpeg Scaler Documentation} +@end titlepage + +@top + +@contents + +@chapter Description +@c man begin DESCRIPTION + +The FFmpeg rescaler provides a high-level interface to the libswscale +library image conversion utilities. In particular it allows one to perform +image rescaling and pixel format conversion. + +@c man end DESCRIPTION + +@include scaler.texi + +@chapter See Also + +@ifhtml +@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver}, +@url{libswscale.html,libswscale} +@end ifhtml + +@ifnothtml +ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), libswscale(3) +@end ifnothtml + +@include authors.texi + +@ignore + +@setfilename ffmpeg-scaler +@settitle FFmpeg video scaling and pixel format converter + +@end ignore + +@bye diff --git a/doc/ffmpeg-utils.texi b/doc/ffmpeg-utils.texi new file mode 100644 index 0000000000..e39cfa85ec --- /dev/null +++ b/doc/ffmpeg-utils.texi @@ -0,0 +1,43 @@ +\input texinfo @c -*- texinfo -*- +@documentencoding UTF-8 + +@settitle FFmpeg Utilities Documentation +@titlepage +@center @titlefont{FFmpeg Utilities Documentation} +@end titlepage + +@top + +@contents + +@chapter Description +@c man begin DESCRIPTION + +This document describes some generic features and utilities provided +by the libavutil library. + +@c man end DESCRIPTION + +@include utils.texi + +@chapter See Also + +@ifhtml +@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver}, +@url{libavutil.html,libavutil} +@end ifhtml + +@ifnothtml +ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), libavutil(3) +@end ifnothtml + +@include authors.texi + +@ignore + +@setfilename ffmpeg-utils +@settitle FFmpeg utilities + +@end ignore + +@bye diff --git a/doc/avconv.texi b/doc/ffmpeg.texi index 6aaf445d55..e02807cb47 100644 --- a/doc/avconv.texi +++ b/doc/ffmpeg.texi @@ -1,8 +1,9 @@ \input texinfo @c -*- texinfo -*- +@documentencoding UTF-8 -@settitle avconv Documentation +@settitle ffmpeg Documentation @titlepage -@center @titlefont{avconv Documentation} +@center @titlefont{ffmpeg Documentation} @end titlepage @top @@ -11,37 +12,31 @@ @chapter Synopsis -The generic syntax is: - -@example -@c man begin SYNOPSIS -avconv [global options] [[infile options][@option{-i} @var{infile}]]... @{[outfile options] @var{outfile}@}... -@c man end -@end example +ffmpeg [@var{global_options}] @{[@var{input_file_options}] -i @file{input_file}@} ... @{[@var{output_file_options}] @file{output_file}@} ... @chapter Description @c man begin DESCRIPTION -avconv is a very fast video and audio converter that can also grab from +@command{ffmpeg} is a very fast video and audio converter that can also grab from a live audio/video source. It can also convert between arbitrary sample rates and resize video on the fly with a high quality polyphase filter. -avconv reads from an arbitrary number of input "files" (which can be regular +@command{ffmpeg} reads from an arbitrary number of input "files" (which can be regular files, pipes, network streams, grabbing devices, etc.), specified by the @code{-i} option, and writes to an arbitrary number of output "files", which are specified by a plain output filename. Anything found on the command line which cannot be interpreted as an option is considered to be an output filename. -Each input or output file can in principle contain any number of streams of -different types (video/audio/subtitle/attachment/data). Allowed number and/or -types of streams can be limited by the container format. Selecting, which -streams from which inputs go into output, is done either automatically or with -the @code{-map} option (see the Stream selection chapter). +Each input or output file can, in principle, contain any number of streams of +different types (video/audio/subtitle/attachment/data). The allowed number and/or +types of streams may be limited by the container format. Selecting which +streams from which inputs will go into which output is either done automatically +or with the @code{-map} option (see the Stream selection chapter). To refer to input files in options, you must use their indices (0-based). E.g. -the first input file is @code{0}, the second is @code{1} etc. Similarly, streams +the first input file is @code{0}, the second is @code{1}, etc. Similarly, streams within a file are referred to by their indices. E.g. @code{2:3} refers to the -fourth stream in the third input file. See also the Stream specifiers chapter. +fourth stream in the third input file. Also see the Stream specifiers chapter. As a general rule, options are applied to the next specified file. Therefore, order is important, and you can have the same @@ -56,22 +51,22 @@ options apply ONLY to the next input or output file and are reset between files. @itemize @item -To set the video bitrate of the output file to 64kbit/s: +To set the video bitrate of the output file to 64 kbit/s: @example -avconv -i input.avi -b 64k output.avi +ffmpeg -i input.avi -b:v 64k -bufsize 64k output.avi @end example @item To force the frame rate of the output file to 24 fps: @example -avconv -i input.avi -r 24 output.avi +ffmpeg -i input.avi -r 24 output.avi @end example @item To force the frame rate of the input file (valid for raw formats only) to 1 fps and the frame rate of the output file to 24 fps: @example -avconv -r 1 -i input.m2v -r 24 output.avi +ffmpeg -r 1 -i input.m2v -r 24 output.avi @end example @end itemize @@ -82,10 +77,10 @@ The format option may be needed for raw input files. @chapter Detailed description @c man begin DETAILED DESCRIPTION -The transcoding process in @command{avconv} for each output can be described by +The transcoding process in @command{ffmpeg} for each output can be described by the following diagram: -@example +@verbatim _______ ______________ | | | | | input | demuxer | encoded data | decoder @@ -104,24 +99,24 @@ the following diagram: |________| |______________| -@end example +@end verbatim -@command{avconv} calls the libavformat library (containing demuxers) to read +@command{ffmpeg} calls the libavformat library (containing demuxers) to read input files and get packets containing encoded data from them. When there are -multiple input files, @command{avconv} tries to keep them synchronized by +multiple input files, @command{ffmpeg} tries to keep them synchronized by tracking lowest timestamp on any active input stream. Encoded packets are then passed to the decoder (unless streamcopy is selected for the stream, see further for a description). The decoder produces uncompressed frames (raw video/PCM audio/...) which can be processed further by -filtering (see next section). After filtering the frames are passed to the -encoder, which encodes them and outputs encoded packets again. Finally those are +filtering (see next section). After filtering, the frames are passed to the +encoder, which encodes them and outputs encoded packets. Finally those are passed to the muxer, which writes the encoded packets to the output file. @section Filtering -Before encoding, @command{avconv} can process raw audio and video frames using +Before encoding, @command{ffmpeg} can process raw audio and video frames using filters from the libavfilter library. Several chained filters form a filter -graph. @command{avconv} distinguishes between two types of filtergraphs - +graph. @command{ffmpeg} distinguishes between two types of filtergraphs: simple and complex. @subsection Simple filtergraphs @@ -129,31 +124,31 @@ Simple filtergraphs are those that have exactly one input and output, both of the same type. In the above diagram they can be represented by simply inserting an additional step between decoding and encoding: -@example +@verbatim _________ ______________ | | | | | decoded | | encoded data | -| frames |\ /| packets | -|_________| \ / |______________| +| frames |\ _ | packets | +|_________| \ /||______________| \ __________ / - simple \ | | / encoder - filtergraph \| filtered |/ + simple _\|| | / encoder + filtergraph | filtered |/ | frames | |__________| -@end example +@end verbatim Simple filtergraphs are configured with the per-stream @option{-filter} option (with @option{-vf} and @option{-af} aliases for video and audio respectively). A simple filtergraph for video can look for example like this: -@example +@verbatim _______ _____________ _______ ________ | | | | | | | | | input | ---> | deinterlace | ---> | scale | ---> | output | |_______| |_____________| |_______| |________| -@end example +@end verbatim Note that some filters change frame properties but not frame contents. E.g. the @code{fps} filter in the example above changes number of frames, but does not @@ -162,11 +157,11 @@ only sets timestamps and otherwise passes the frames unchanged. @subsection Complex filtergraphs Complex filtergraphs are those which cannot be described as simply a linear -processing chain applied to one stream. This is the case e.g. when the graph has +processing chain applied to one stream. This is the case, for example, when the graph has more than one input and/or output, or when output stream type is different from input. They can be represented with the following diagram: -@example +@verbatim _________ | | | input 0 |\ __________ @@ -184,34 +179,36 @@ input. They can be represented with the following diagram: | input 2 |/ |_________| -@end example +@end verbatim Complex filtergraphs are configured with the @option{-filter_complex} option. -Note that this option is global, since a complex filtergraph by its nature +Note that this option is global, since a complex filtergraph, by its nature, cannot be unambiguously associated with a single stream or file. +The @option{-lavfi} option is equivalent to @option{-filter_complex}. + A trivial example of a complex filtergraph is the @code{overlay} filter, which has two video inputs and one video output, containing one video overlaid on top of the other. Its audio counterpart is the @code{amix} filter. @section Stream copy Stream copy is a mode selected by supplying the @code{copy} parameter to the -@option{-codec} option. It makes @command{avconv} omit the decoding and encoding +@option{-codec} option. It makes @command{ffmpeg} omit the decoding and encoding step for the specified stream, so it does only demuxing and muxing. It is useful for changing the container format or modifying container-level metadata. The -diagram above will in this case simplify to this: +diagram above will, in this case, simplify to this: -@example +@verbatim _______ ______________ ________ | | | | | | | input | demuxer | encoded data | muxer | output | | file | ---------> | packets | -------> | file | |_______| |______________| |________| -@end example +@end verbatim Since there is no decoding or encoding, it is very fast and there is no quality -loss. However it might not work in some cases because of many factors. Applying +loss. However, it might not work in some cases because of many factors. Applying filters is obviously also impossible, since filters work on uncompressed data. @c man end DETAILED DESCRIPTION @@ -219,12 +216,14 @@ filters is obviously also impossible, since filters work on uncompressed data. @chapter Stream selection @c man begin STREAM SELECTION -By default avconv tries to pick the "best" stream of each type present in input -files and add them to each output file. For video, this means the highest -resolution, for audio the highest channel count. For subtitle it's simply the -first subtitle stream. +By default, @command{ffmpeg} includes only one stream of each type (video, audio, subtitle) +present in the input files and adds them to each output file. It picks the +"best" of each based upon the following criteria: for video, it is the stream +with the highest resolution, for audio, it is the stream with the most channels, for +subtitles, it is the first subtitle stream. In the case where several streams of +the same type rate equally, the stream with the lowest index is chosen. -You can disable some of those defaults by using @code{-vn/-an/-sn} options. For +You can disable some of those defaults by using the @code{-vn/-an/-sn} options. For full manual control, use the @code{-map} option, which disables the defaults just described. @@ -233,15 +232,15 @@ described. @chapter Options @c man begin OPTIONS -@include avtools-common-opts.texi +@include fftools-common-opts.texi @section Main options @table @option @item -f @var{fmt} (@emph{input/output}) -Force input or output file format. The format is normally autodetected for input -files and guessed from file extension for output files, so this option is not +Force input or output file format. The format is normally auto detected for input +files and guessed from the file extension for output files, so this option is not needed in most cases. @item -i @var{filename} (@emph{input}) @@ -251,9 +250,10 @@ input file name Overwrite output files without asking. @item -n (@emph{global}) -Immediately exit when output files already exist. +Do not overwrite output files, and exit immediately if a specified +output file already exists. -@item -loop @var{number} (@emph{input}) +@item -stream_loop @var{number} (@emph{input}) Set number of times input stream shall be looped. Loop 0 means no loop, loop -1 means infinite loop. @@ -262,32 +262,49 @@ loop -1 means infinite loop. Select an encoder (when used before an output file) or a decoder (when used before an input file) for one or more streams. @var{codec} is the name of a decoder/encoder or a special value @code{copy} (output only) to indicate that -the stream is not to be reencoded. +the stream is not to be re-encoded. For example @example -avconv -i INPUT -map 0 -c:v libx264 -c:a copy OUTPUT +ffmpeg -i INPUT -map 0 -c:v libx264 -c:a copy OUTPUT @end example encodes all video streams with libx264 and copies all audio streams. For each stream, the last matching @code{c} option is applied, so @example -avconv -i INPUT -map 0 -c copy -c:v:1 libx264 -c:a:137 libvorbis OUTPUT +ffmpeg -i INPUT -map 0 -c copy -c:v:1 libx264 -c:a:137 libvorbis OUTPUT @end example will copy all the streams except the second video, which will be encoded with libx264, and the 138th audio, which will be encoded with libvorbis. -@item -t @var{duration} (@emph{output}) -Stop writing the output after its duration reaches @var{duration}. -@var{duration} may be a number in seconds, or in @code{hh:mm:ss[.xxx]} form. +@item -t @var{duration} (@emph{input/output}) +When used as an input option (before @code{-i}), limit the @var{duration} of +data read from the input file. + +When used as an output option (before an output filename), stop writing the +output after its duration reaches @var{duration}. + +@var{duration} must be a time duration specification, +see @ref{time duration syntax,,the Time duration section in the ffmpeg-utils(1) manual,ffmpeg-utils}. + +-to and -t are mutually exclusive and -t has priority. + +@item -to @var{position} (@emph{output}) +Stop writing the output at @var{position}. +@var{position} must be a time duration specification, +see @ref{time duration syntax,,the Time duration section in the ffmpeg-utils(1) manual,ffmpeg-utils}. + +-to and -t are mutually exclusive and -t has priority. @item -fs @var{limit_size} (@emph{output}) -Set the file size limit. +Set the file size limit, expressed in bytes. No further chunk of bytes is written +after the limit is exceeded. The size of the output file is slightly more than the +requested file size. @item -ss @var{position} (@emph{input/output}) When used as an input option (before @code{-i}), seeks in this input file to -@var{position}. Note the in most formats it is not possible to seek exactly, so -@command{avconv} will seek to the closest seek point before @var{position}. +@var{position}. Note that in most formats it is not possible to seek exactly, +so @command{ffmpeg} will seek to the closest seek point before @var{position}. When transcoding and @option{-accurate_seek} is enabled (the default), this extra segment between the seek point and @var{position} will be decoded and discarded. When doing stream copy or when @option{-noaccurate_seek} is used, it @@ -296,35 +313,55 @@ will be preserved. When used as an output option (before an output filename), decodes but discards input until the timestamps reach @var{position}. -@var{position} may be either in seconds or in @code{hh:mm:ss[.xxx]} form. +@var{position} must be a time duration specification, +see @ref{time duration syntax,,the Time duration section in the ffmpeg-utils(1) manual,ffmpeg-utils}. + +@item -sseof @var{position} (@emph{input/output}) + +Like the @code{-ss} option but relative to the "end of file". That is negative +values are earlier in the file, 0 is at EOF. @item -itsoffset @var{offset} (@emph{input}) -Set the input time offset in seconds. -@code{[-]hh:mm:ss[.xxx]} syntax is also supported. -The offset is added to the timestamps of the input files. -Specifying a positive offset means that the corresponding -streams are delayed by @var{offset} seconds. +Set the input time offset. + +@var{offset} must be a time duration specification, +see @ref{time duration syntax,,the Time duration section in the ffmpeg-utils(1) manual,ffmpeg-utils}. + +The offset is added to the timestamps of the input files. Specifying +a positive offset means that the corresponding streams are delayed by +the time duration specified in @var{offset}. + +@item -timestamp @var{date} (@emph{output}) +Set the recording timestamp in the container. + +@var{date} must be a date specification, +see @ref{date syntax,,the Date section in the ffmpeg-utils(1) manual,ffmpeg-utils}. @item -metadata[:metadata_specifier] @var{key}=@var{value} (@emph{output,per-metadata}) Set a metadata key/value pair. An optional @var{metadata_specifier} may be given to set metadata -on streams or chapters. See @code{-map_metadata} documentation for -details. +on streams, chapters or programs. See @code{-map_metadata} +documentation for details. This option overrides metadata set with @code{-map_metadata}. It is also possible to delete metadata by using an empty value. For example, for setting the title in the output file: @example -avconv -i in.avi -metadata title="my title" out.flv +ffmpeg -i in.avi -metadata title="my title" out.flv @end example To set the language of the first audio stream: @example -avconv -i INPUT -metadata:s:a:0 language=eng OUTPUT +ffmpeg -i INPUT -metadata:s:a:0 language=eng OUTPUT @end example +@item -program [title=@var{title}:][program_num=@var{program_num}:]st=@var{stream}[:st=@var{stream}...] (@emph{output}) + +Creates a program with the specified @var{title}, @var{program_num} and adds the specified +@var{stream}(s) to it. + @item -target @var{type} (@emph{output}) Specify target file type (@code{vcd}, @code{svcd}, @code{dvd}, @code{dv}, @code{dv50}). @var{type} may be prefixed with @code{pal-}, @code{ntsc-} or @@ -332,34 +369,46 @@ Specify target file type (@code{vcd}, @code{svcd}, @code{dvd}, @code{dv}, (bitrate, codecs, buffer sizes) are then set automatically. You can just type: @example -avconv -i myfile.avi -target vcd /tmp/vcd.mpg +ffmpeg -i myfile.avi -target vcd /tmp/vcd.mpg @end example Nevertheless you can specify additional options as long as you know they do not conflict with the standard, as in: @example -avconv -i myfile.avi -target vcd -bf 2 /tmp/vcd.mpg +ffmpeg -i myfile.avi -target vcd -bf 2 /tmp/vcd.mpg @end example @item -dframes @var{number} (@emph{output}) -Set the number of data frames to record. This is an alias for @code{-frames:d}. +Set the number of data frames to output. This is an alias for @code{-frames:d}. @item -frames[:@var{stream_specifier}] @var{framecount} (@emph{output,per-stream}) Stop writing to the stream after @var{framecount} frames. @item -q[:@var{stream_specifier}] @var{q} (@emph{output,per-stream}) @itemx -qscale[:@var{stream_specifier}] @var{q} (@emph{output,per-stream}) -Use fixed quality scale (VBR). The meaning of @var{q} is +Use fixed quality scale (VBR). The meaning of @var{q}/@var{qscale} is codec-dependent. - -@item -filter[:@var{stream_specifier}] @var{filter_graph} (@emph{output,per-stream}) -@var{filter_graph} is a description of the filter graph to apply to -the stream. Use @code{-filters} to show all the available filters -(including also sources and sinks). - -See also the @option{-filter_complex} option if you want to create filter graphs -with multiple inputs and/or outputs. +If @var{qscale} is used without a @var{stream_specifier} then it applies only +to the video stream, this is to maintain compatibility with previous behavior +and as specifying the same codec specific value to 2 different codecs that is +audio and video generally is not what is intended when no stream_specifier is +used. + +@anchor{filter_option} +@item -filter[:@var{stream_specifier}] @var{filtergraph} (@emph{output,per-stream}) +Create the filtergraph specified by @var{filtergraph} and use it to +filter the stream. + +@var{filtergraph} is a description of the filtergraph to apply to +the stream, and must have a single input and a single output of the +same type of the stream. In the filtergraph, the input is associated +to the label @code{in}, and the output to the label @code{out}. See +the ffmpeg-filters manual for more information about the filtergraph +syntax. + +See the @ref{filter_complex_option,,-filter_complex option} if you +want to create filtergraphs with multiple inputs and/or outputs. @item -filter_script[:@var{stream_specifier}] @var{filename} (@emph{output,per-stream}) This option is similar to @option{-filter}, the only difference is that its @@ -370,7 +419,34 @@ read. Specify the preset for matching stream(s). @item -stats (@emph{global}) -Print encoding progress/statistics. On by default. +Print encoding progress/statistics. It is on by default, to explicitly +disable it you need to specify @code{-nostats}. + +@item -progress @var{url} (@emph{global}) +Send program-friendly progress information to @var{url}. + +Progress information is written approximately every second and at the end of +the encoding process. It is made of "@var{key}=@var{value}" lines. @var{key} +consists of only alphanumeric characters. The last key of a sequence of +progress information is always "progress". + +@item -stdin +Enable interaction on standard input. On by default unless standard input is +used as an input. To explicitly disable interaction you need to specify +@code{-nostdin}. + +Disabling interaction on standard input is useful, for example, if +ffmpeg is in the background process group. Roughly the same result can +be achieved with @code{ffmpeg ... < /dev/null} but it requires a +shell. + +@item -debug_ts (@emph{global}) +Print timestamp information. It is off by default. This option is +mostly useful for testing and debugging purposes, and the output +format may change from one version to another, so it should not be +employed by portable scripts. + +See also the option @code{-fdebug ts}. @item -attach @var{filename} (@emph{output}) Add an attachment to the output file. This is supported by a few formats @@ -383,7 +459,7 @@ with @code{-map} or automatic mappings). Note that for Matroska you also have to set the mimetype metadata tag: @example -avconv -i INPUT -attach DejaVuSans.ttf -metadata:s:2 mimetype=application/x-truetype-font out.mkv +ffmpeg -i INPUT -attach DejaVuSans.ttf -metadata:s:2 mimetype=application/x-truetype-font out.mkv @end example (assuming that the attachment stream will be third in the output file). @@ -394,11 +470,11 @@ will be used. E.g. to extract the first attachment to a file named 'out.ttf': @example -avconv -dump_attachment:t:0 out.ttf INPUT +ffmpeg -dump_attachment:t:0 out.ttf -i INPUT @end example To extract all attachments to files determined by the @code{filename} tag: @example -avconv -dump_attachment:t "" INPUT +ffmpeg -dump_attachment:t "" -i INPUT @end example Technical note -- attachments are implemented as codec extradata, so this @@ -414,16 +490,18 @@ Disable automatically rotating video based on file metadata. @table @option @item -vframes @var{number} (@emph{output}) -Set the number of video frames to record. This is an alias for @code{-frames:v}. +Set the number of video frames to output. This is an alias for @code{-frames:v}. @item -r[:@var{stream_specifier}] @var{fps} (@emph{input/output,per-stream}) Set frame rate (Hz value, fraction or abbreviation). As an input option, ignore any timestamps stored in the file and instead generate timestamps assuming constant frame rate @var{fps}. +This is not the same as the @option{-framerate} option used for some input formats +like image2 or v4l2 (it used to be the same in older versions of FFmpeg). +If in doubt use @option{-framerate} instead of the input option @option{-r}. As an output option, duplicate or drop input frames to achieve constant output -frame rate @var{fps} (note that this actually causes the @code{fps} filter to be -inserted to the end of the corresponding filtergraph). +frame rate @var{fps}. @item -s[:@var{stream_specifier}] @var{size} (@emph{input/output,per-stream}) Set frame size. @@ -436,76 +514,7 @@ As an output option, this inserts the @code{scale} video filter to the @emph{end} of the corresponding filtergraph. Please use the @code{scale} filter directly to insert it at the beginning or some other place. -The format is @samp{wxh} (default - same as source). The following -abbreviations are recognized: -@table @samp -@item sqcif -128x96 -@item qcif -176x144 -@item cif -352x288 -@item 4cif -704x576 -@item 16cif -1408x1152 -@item qqvga -160x120 -@item qvga -320x240 -@item vga -640x480 -@item svga -800x600 -@item xga -1024x768 -@item uxga -1600x1200 -@item qxga -2048x1536 -@item sxga -1280x1024 -@item qsxga -2560x2048 -@item hsxga -5120x4096 -@item wvga -852x480 -@item wxga -1366x768 -@item wsxga -1600x1024 -@item wuxga -1920x1200 -@item woxga -2560x1600 -@item wqsxga -3200x2048 -@item wquxga -3840x2400 -@item whsxga -6400x4096 -@item whuxga -7680x4800 -@item cga -320x200 -@item ega -640x350 -@item hd480 -852x480 -@item hd720 -1280x720 -@item hd1080 -1920x1080 -@item 2kdci -2048x1080 -@item 4kdci -4096x2160 -@item uhd2160 -3840x2160 -@item uhd4320 -7680x4320 -@end table +The format is @samp{wxh} (default - same as source). @item -aspect[:@var{stream_specifier}] @var{aspect} (@emph{output,per-stream}) Set the video display aspect ratio specified by @var{aspect}. @@ -515,6 +524,10 @@ form @var{num}:@var{den}, where @var{num} and @var{den} are the numerator and denominator of the aspect ratio. For example "4:3", "16:9", "1.3333", and "1.7777" are valid argument values. +If used together with @option{-vcodec copy}, it will affect the aspect ratio +stored at container level, but not the aspect ratio stored in encoded +frames, if it exists. + @item -vn (@emph{output}) Disable video recording. @@ -530,38 +543,56 @@ at the exact requested bitrate. On pass 1, you may just deactivate audio and set output to null, examples for Windows and Unix: @example -avconv -i foo.mov -c:v libxvid -pass 1 -an -f rawvideo -y NUL -avconv -i foo.mov -c:v libxvid -pass 1 -an -f rawvideo -y /dev/null +ffmpeg -i foo.mov -c:v libxvid -pass 1 -an -f rawvideo -y NUL +ffmpeg -i foo.mov -c:v libxvid -pass 1 -an -f rawvideo -y /dev/null @end example @item -passlogfile[:@var{stream_specifier}] @var{prefix} (@emph{output,per-stream}) Set two-pass log file name prefix to @var{prefix}, the default file name -prefix is ``av2pass''. The complete file name will be +prefix is ``ffmpeg2pass''. The complete file name will be @file{PREFIX-N.log}, where N is a number specific to the output -stream. +stream -@item -vf @var{filter_graph} (@emph{output}) -@var{filter_graph} is a description of the filter graph to apply to -the input video. -Use the option "-filters" to show all the available filters (including -also sources and sinks). This is an alias for @code{-filter:v}. +@item -vf @var{filtergraph} (@emph{output}) +Create the filtergraph specified by @var{filtergraph} and use it to +filter the stream. +This is an alias for @code{-filter:v}, see the @ref{filter_option,,-filter option}. @end table -@section Advanced Video Options +@section Advanced Video options @table @option @item -pix_fmt[:@var{stream_specifier}] @var{format} (@emph{input/output,per-stream}) Set pixel format. Use @code{-pix_fmts} to show all the supported pixel formats. +If the selected pixel format can not be selected, ffmpeg will print a +warning and select the best pixel format supported by the encoder. +If @var{pix_fmt} is prefixed by a @code{+}, ffmpeg will exit with an error +if the requested pixel format can not be selected, and automatic conversions +inside filtergraphs are disabled. +If @var{pix_fmt} is a single @code{+}, ffmpeg selects the same pixel format +as the input (or graph output) and automatic conversions are disabled. + @item -sws_flags @var{flags} (@emph{input/output}) Set SwScaler flags. @item -vdt @var{n} Discard threshold. @item -rc_override[:@var{stream_specifier}] @var{override} (@emph{output,per-stream}) -rate control override for specific intervals - +Rate control override for specific intervals, formatted as "int,int,int" +list separated with slashes. Two first values are the beginning and +end frame numbers, last one is quantizer to use if positive, or quality +factor if negative. + +@item -ilme +Force interlacing support in encoder (MPEG-2 and MPEG-4 only). +Use this option if your input file is interlaced and you want +to keep the interlaced format for minimum losses. +The alternative is to deinterlace the input stream with +@option{-deinterlace}, but deinterlacing introduces losses. +@item -psnr +Calculate PSNR of compressed frames. @item -vstats Dump video coding statistics to @file{vstats_HHMMSS.log}. @item -vstats_file @var{file} @@ -573,13 +604,61 @@ Intra_dc_precision. @item -vtag @var{fourcc/tag} (@emph{output}) Force video tag/fourcc. This is an alias for @code{-tag:v}. @item -qphist (@emph{global}) -Show QP histogram. +Show QP histogram +@item -vbsf @var{bitstream_filter} +Deprecated see -bsf + @item -force_key_frames[:@var{stream_specifier}] @var{time}[,@var{time}...] (@emph{output,per-stream}) +@item -force_key_frames[:@var{stream_specifier}] expr:@var{expr} (@emph{output,per-stream}) Force key frames at the specified timestamps, more precisely at the first frames after each specified time. + +If the argument is prefixed with @code{expr:}, the string @var{expr} +is interpreted like an expression and is evaluated for each frame. A +key frame is forced in case the evaluation is non-zero. + +If one of the times is "@code{chapters}[@var{delta}]", it is expanded into +the time of the beginning of all chapters in the file, shifted by +@var{delta}, expressed as a time in seconds. This option can be useful to ensure that a seek point is present at a chapter mark or any other designated place in the output file. -The timestamps must be specified in ascending order. + +For example, to insert a key frame at 5 minutes, plus key frames 0.1 second +before the beginning of every chapter: +@example +-force_key_frames 0:05:00,chapters-0.1 +@end example + +The expression in @var{expr} can contain the following constants: +@table @option +@item n +the number of current processed frame, starting from 0 +@item n_forced +the number of forced frames +@item prev_forced_n +the number of the previous forced frame, it is @code{NAN} when no +keyframe was forced yet +@item prev_forced_t +the time of the previous forced frame, it is @code{NAN} when no +keyframe was forced yet +@item t +the time of the current processed frame +@end table + +For example to force a key frame every 5 seconds, you can specify: +@example +-force_key_frames expr:gte(t,n_forced*5) +@end example + +To force a key frame 5 seconds after the time of the last forced one, +starting from second 13: +@example +-force_key_frames expr:if(isnan(prev_forced_t),gte(t,13),gte(t,prev_forced_t+5)) +@end example + +Note that forcing too many keyframes is very harmful for the lookahead +algorithms of certain encoders: using fixed-GOP options or similar +would be more efficient. @item -copyinkf[:@var{stream_specifier}] (@emph{output,per-stream}) When doing stream copy, copy also non-key frames found at the @@ -619,7 +698,7 @@ This option has no effect if the selected hwaccel is not available or not supported by the chosen decoder. Note that most acceleration methods are intended for playback and will not be -faster than software decoding on modern CPUs. Additionally, @command{avconv} +faster than software decoding on modern CPUs. Additionally, @command{ffmpeg} will usually need to copy the decoded frames from the GPU memory into the system memory, resulting in further performance loss. This option is thus mainly useful for testing. @@ -656,7 +735,7 @@ are: @end table @item -hwaccels -List all hardware acceleration methods supported in this build of avconv. +List all hardware acceleration methods supported in this build of ffmpeg. @end table @@ -664,7 +743,7 @@ List all hardware acceleration methods supported in this build of avconv. @table @option @item -aframes @var{number} (@emph{output}) -Set the number of audio frames to record. This is an alias for @code{-frames:a}. +Set the number of audio frames to output. This is an alias for @code{-frames:a}. @item -ar[:@var{stream_specifier}] @var{freq} (@emph{input/output,per-stream}) Set the audio sampling frequency. For output streams it is set by default to the frequency of the corresponding input stream. For input @@ -684,27 +763,60 @@ Set the audio codec. This is an alias for @code{-codec:a}. @item -sample_fmt[:@var{stream_specifier}] @var{sample_fmt} (@emph{output,per-stream}) Set the audio sample format. Use @code{-sample_fmts} to get a list of supported sample formats. -@item -af @var{filter_graph} (@emph{output}) -@var{filter_graph} is a description of the filter graph to apply to -the input audio. -Use the option "-filters" to show all the available filters (including -also sources and sinks). This is an alias for @code{-filter:a}. + +@item -af @var{filtergraph} (@emph{output}) +Create the filtergraph specified by @var{filtergraph} and use it to +filter the stream. + +This is an alias for @code{-filter:a}, see the @ref{filter_option,,-filter option}. @end table -@section Advanced Audio options: +@section Advanced Audio options @table @option @item -atag @var{fourcc/tag} (@emph{output}) Force audio tag/fourcc. This is an alias for @code{-tag:a}. +@item -absf @var{bitstream_filter} +Deprecated, see -bsf +@item -guess_layout_max @var{channels} (@emph{input,per-stream}) +If some input channel layout is not known, try to guess only if it +corresponds to at most the specified number of channels. For example, 2 +tells to @command{ffmpeg} to recognize 1 channel as mono and 2 channels as +stereo but not 6 channels as 5.1. The default is to always try to guess. Use +0 to disable all guessing. @end table -@section Subtitle options: +@section Subtitle options @table @option @item -scodec @var{codec} (@emph{input/output}) Set the subtitle codec. This is an alias for @code{-codec:s}. @item -sn (@emph{output}) Disable subtitle recording. +@item -sbsf @var{bitstream_filter} +Deprecated, see -bsf +@end table + +@section Advanced Subtitle options + +@table @option + +@item -fix_sub_duration +Fix subtitles durations. For each subtitle, wait for the next packet in the +same stream and adjust the duration of the first to avoid overlap. This is +necessary with some subtitles codecs, especially DVB subtitles, because the +duration in the original packet is only a rough estimate and the end is +actually marked by an empty subtitle frame. Failing to use this option when +necessary can result in exaggerated durations or muxing failures due to +non-monotonic timestamps. + +Note that this option will delay the output of all data until the next +subtitle packet is decoded: it may increase memory consumption and latency a +lot. + +@item -canvas_size @var{size} +Set the size of the canvas used to render subtitles. + @end table @section Advanced options @@ -732,7 +844,7 @@ graphs (see the @option{-filter_complex} option) to the output file. For example, to map ALL streams from the first input file to output @example -avconv -i INPUT -map 0 output +ffmpeg -i INPUT -map 0 output @end example For example, if you have two audio streams in the first input file, @@ -740,7 +852,7 @@ these streams are identified by "0:0" and "0:1". You can use @code{-map} to select which streams to place in an output file. For example: @example -avconv -i INPUT -map 0:1 out.wav +ffmpeg -i INPUT -map 0:1 out.wav @end example will map the input stream in @file{INPUT} identified by "0:1" to the (single) output stream in @file{out.wav}. @@ -750,26 +862,90 @@ For example, to select the stream with index 2 from input file index 6 from input @file{b.mov} (specified by the identifier "1:6"), and copy them to the output file @file{out.mov}: @example -avconv -i a.mov -i b.mov -c copy -map 0:2 -map 1:6 out.mov +ffmpeg -i a.mov -i b.mov -c copy -map 0:2 -map 1:6 out.mov @end example To select all video and the third audio stream from an input file: @example -avconv -i INPUT -map 0:v -map 0:a:2 OUTPUT +ffmpeg -i INPUT -map 0:v -map 0:a:2 OUTPUT @end example To map all the streams except the second audio, use negative mappings @example -avconv -i INPUT -map 0 -map -0:a:1 OUTPUT +ffmpeg -i INPUT -map 0 -map -0:a:1 OUTPUT @end example To pick the English audio stream: @example -avconv -i INPUT -map 0:m:language:eng OUTPUT +ffmpeg -i INPUT -map 0:m:language:eng OUTPUT @end example Note that using this option disables the default mappings for this output file. +@item -ignore_unknown +Ignore input streams with unknown type instead of failing if copying +such streams is attempted. + +@item -copy_unknown +Allow input streams with unknown type to be copied instead of failing if copying +such streams is attempted. + +@item -map_channel [@var{input_file_id}.@var{stream_specifier}.@var{channel_id}|-1][:@var{output_file_id}.@var{stream_specifier}] +Map an audio channel from a given input to an output. If +@var{output_file_id}.@var{stream_specifier} is not set, the audio channel will +be mapped on all the audio streams. + +Using "-1" instead of +@var{input_file_id}.@var{stream_specifier}.@var{channel_id} will map a muted +channel. + +For example, assuming @var{INPUT} is a stereo audio file, you can switch the +two audio channels with the following command: +@example +ffmpeg -i INPUT -map_channel 0.0.1 -map_channel 0.0.0 OUTPUT +@end example + +If you want to mute the first channel and keep the second: +@example +ffmpeg -i INPUT -map_channel -1 -map_channel 0.0.1 OUTPUT +@end example + +The order of the "-map_channel" option specifies the order of the channels in +the output stream. The output channel layout is guessed from the number of +channels mapped (mono if one "-map_channel", stereo if two, etc.). Using "-ac" +in combination of "-map_channel" makes the channel gain levels to be updated if +input and output channel layouts don't match (for instance two "-map_channel" +options and "-ac 6"). + +You can also extract each channel of an input to specific outputs; the following +command extracts two channels of the @var{INPUT} audio stream (file 0, stream 0) +to the respective @var{OUTPUT_CH0} and @var{OUTPUT_CH1} outputs: +@example +ffmpeg -i INPUT -map_channel 0.0.0 OUTPUT_CH0 -map_channel 0.0.1 OUTPUT_CH1 +@end example + +The following example splits the channels of a stereo input into two separate +streams, which are put into the same output file: +@example +ffmpeg -i stereo.wav -map 0:0 -map 0:0 -map_channel 0.0.0:0.0 -map_channel 0.0.1:0.1 -y out.ogg +@end example + +Note that currently each output stream can only contain channels from a single +input stream; you can't for example use "-map_channel" to pick multiple input +audio channels contained in different streams (from the same or different files) +and merge them into a single output stream. It is therefore not currently +possible, for example, to turn two separate mono streams into a single stereo +stream. However splitting a stereo stream into two single channel mono streams +is possible. + +If you need this feature, a possible workaround is to use the @emph{amerge} +filter. For example, if you need to merge a media (here @file{input.mkv}) with 2 +mono audio streams into one single stereo channel audio stream (and keep the +video stream), you can use the following command: +@example +ffmpeg -i input.mkv -filter_complex "[0:1] [0:2] amerge" -c:a pcm_s16le -c:v copy output.mkv +@end example + @item -map_metadata[:@var{metadata_spec_out}] @var{infile}[:@var{metadata_spec_in}] (@emph{output,per-metadata}) Set metadata information of the next output file from @var{infile}. Note that those are file indices (zero-based), not filenames. @@ -801,12 +977,12 @@ file index can be used to create a dummy mapping that just disables automatic co For example to copy metadata from the first stream of the input file to global metadata of the output file: @example -avconv -i in.ogg -map_metadata 0:s:0 out.mp3 +ffmpeg -i in.ogg -map_metadata 0:s:0 out.mp3 @end example To do the reverse, i.e. copy global metadata to all audio streams: @example -avconv -i in.mkv -map_metadata:s:a 0:g out.mkv +ffmpeg -i in.mkv -map_metadata:s:a 0:g out.mkv @end example Note that simple @code{0} would work as well in this example, since global metadata is assumed by default. @@ -816,55 +992,126 @@ Copy chapters from input file with index @var{input_file_index} to the next output file. If no chapter mapping is specified, then chapters are copied from the first input file with at least one chapter. Use a negative file index to disable any chapter copying. -@item -debug -Print specific debug info. + @item -benchmark (@emph{global}) Show benchmarking information at the end of an encode. Shows CPU time used and maximum memory consumption. Maximum memory consumption is not supported on all systems, it will usually display as 0 if not supported. +@item -benchmark_all (@emph{global}) +Show benchmarking information during the encode. +Shows CPU time used in various steps (audio/video encode/decode). @item -timelimit @var{duration} (@emph{global}) -Exit after avconv has been running for @var{duration} seconds. +Exit after ffmpeg has been running for @var{duration} seconds. @item -dump (@emph{global}) Dump each input packet to stderr. @item -hex (@emph{global}) When dumping packets, also dump the payload. @item -re (@emph{input}) -Read input at native frame rate. Mainly used to simulate a grab device +Read input at native frame rate. Mainly used to simulate a grab device. or live input stream (e.g. when reading from a file). Should not be used with actual grab devices or live input streams (where it can cause packet loss). +By default @command{ffmpeg} attempts to read the input(s) as fast as possible. +This option will slow down the reading of the input(s) to the native frame rate +of the input(s). It is useful for real-time output (e.g. live streaming). +@item -loop_input +Loop over the input stream. Currently it works only for image +streams. This option is used for automatic FFserver testing. +This option is deprecated, use -loop 1. +@item -loop_output @var{number_of_times} +Repeatedly loop output for formats that support looping such as animated GIF +(0 will loop the output infinitely). +This option is deprecated, use -loop. @item -vsync @var{parameter} Video sync method. +For compatibility reasons old values can be specified as numbers. +Newly added values will have to be specified as strings always. @table @option -@item passthrough +@item 0, passthrough Each frame is passed with its timestamp from the demuxer to the muxer. -@item cfr +@item 1, cfr Frames will be duplicated and dropped to achieve exactly the requested -constant framerate. -@item vfr +constant frame rate. +@item 2, vfr Frames are passed through with their timestamp or dropped so as to prevent 2 frames from having the same timestamp. -@item auto +@item drop +As passthrough but destroys all timestamps, making the muxer generate +fresh timestamps based on frame-rate. +@item -1, auto Chooses between 1 and 2 depending on muxer capabilities. This is the default method. @end table +Note that the timestamps may be further modified by the muxer, after this. +For example, in the case that the format option @option{avoid_negative_ts} +is enabled. + With -map you can select from which stream the timestamps should be taken. You can leave either video or audio unchanged and sync the remaining stream(s) to the unchanged one. +@item -frame_drop_threshold @var{parameter} +Frame drop threshold, which specifies how much behind video frames can +be before they are dropped. In frame rate units, so 1.0 is one frame. +The default is -1.1. One possible usecase is to avoid framedrops in case +of noisy timestamps or to increase frame drop precision in case of exact +timestamps. + @item -async @var{samples_per_second} Audio sync method. "Stretches/squeezes" the audio stream to match the timestamps, the parameter is the maximum samples per second by which the audio is changed. -async 1 is a special case where only the start of the audio stream is corrected without any later correction. -This option has been deprecated. Use the @code{asyncts} audio filter instead. + +Note that the timestamps may be further modified by the muxer, after this. +For example, in the case that the format option @option{avoid_negative_ts} +is enabled. + +This option has been deprecated. Use the @code{aresample} audio filter instead. + @item -copyts -Copy timestamps from input to output. -@item -copytb -Copy input stream time base from input to output when stream copying. +Do not process input timestamps, but keep their values without trying +to sanitize them. In particular, do not remove the initial start time +offset value. + +Note that, depending on the @option{vsync} option or on specific muxer +processing (e.g. in case the format option @option{avoid_negative_ts} +is enabled) the output timestamps may mismatch with the input +timestamps even when this option is selected. + +@item -start_at_zero +When used with @option{copyts}, shift input timestamps so they start at zero. + +This means that using e.g. @code{-ss 50} will make output timestamps start at +50 seconds, regardless of what timestamp the input file started at. + +@item -copytb @var{mode} +Specify how to set the encoder timebase when stream copying. @var{mode} is an +integer numeric value, and can assume one of the following values: + +@table @option +@item 1 +Use the demuxer timebase. + +The time base is copied to the output encoder from the corresponding input +demuxer. This is sometimes required to avoid non monotonically increasing +timestamps when copying video streams with variable frame rate. + +@item 0 +Use the decoder timebase. + +The time base is copied to the output encoder from the corresponding input +decoder. + +@item -1 +Try to make the choice automatically, in order to generate a sane output. +@end table + +Default value is -1. + @item -shortest (@emph{output}) Finish encoding when the shortest input stream ends. @item -dts_delta_threshold @@ -882,28 +1129,37 @@ may be reassigned to a different value. For example, to set the stream 0 PID to 33 and the stream 1 PID to 36 for an output mpegts file: @example -avconv -i infile -streamid 0:33 -streamid 1:36 out.ts +ffmpeg -i infile -streamid 0:33 -streamid 1:36 out.ts @end example @item -bsf[:@var{stream_specifier}] @var{bitstream_filters} (@emph{output,per-stream}) -Set bitstream filters for matching streams. @var{bistream_filters} is +Set bitstream filters for matching streams. @var{bitstream_filters} is a comma-separated list of bitstream filters. Use the @code{-bsfs} option to get the list of bitstream filters. @example -avconv -i h264.mp4 -c:v copy -bsf:v h264_mp4toannexb -an out.h264 +ffmpeg -i h264.mp4 -c:v copy -bsf:v h264_mp4toannexb -an out.h264 @end example @example -avconv -i file.mov -an -vn -bsf:s mov2textsub -c:s copy -f rawvideo sub.txt +ffmpeg -i file.mov -an -vn -bsf:s mov2textsub -c:s copy -f rawvideo sub.txt @end example @item -tag[:@var{stream_specifier}] @var{codec_tag} (@emph{input/output,per-stream}) Force a tag/fourcc for matching streams. +@item -timecode @var{hh}:@var{mm}:@var{ss}SEP@var{ff} +Specify Timecode for writing. @var{SEP} is ':' for non drop timecode and ';' +(or '.') for drop. +@example +ffmpeg -i input.mpg -timecode 01:02:03.04 -r 30000/1001 -s ntsc output.mpg +@end example + +@anchor{filter_complex_option} @item -filter_complex @var{filtergraph} (@emph{global}) -Define a complex filter graph, i.e. one with arbitrary number of inputs and/or +Define a complex filtergraph, i.e. one with arbitrary number of inputs and/or outputs. For simple graphs -- those with one input and one output of the same type -- see the @option{-filter} options. @var{filtergraph} is a description of -the filter graph, as described in @ref{Filtergraph syntax}. +the filtergraph, as described in the ``Filtergraph syntax'' section of the +ffmpeg-filters manual. Input link labels must refer to input streams using the @code{[file_index:stream_specifier]} syntax (i.e. the same as @option{-map} @@ -919,7 +1175,7 @@ normal input files. For example, to overlay an image over video @example -avconv -i video.mkv -i image.png -filter_complex '[0:v][1:v]overlay[out]' -map +ffmpeg -i video.mkv -i image.png -filter_complex '[0:v][1:v]overlay[out]' -map '[out]' out.mkv @end example Here @code{[0:v]} refers to the first video stream in the first input file, @@ -930,21 +1186,25 @@ of overlay. Assuming there is only one video stream in each input file, we can omit input labels, so the above is equivalent to @example -avconv -i video.mkv -i image.png -filter_complex 'overlay[out]' -map +ffmpeg -i video.mkv -i image.png -filter_complex 'overlay[out]' -map '[out]' out.mkv @end example Furthermore we can omit the output label and the single output from the filter graph will be added to the output file automatically, so we can simply write @example -avconv -i video.mkv -i image.png -filter_complex 'overlay' out.mkv +ffmpeg -i video.mkv -i image.png -filter_complex 'overlay' out.mkv @end example To generate 5 seconds of pure red video using lavfi @code{color} source: @example -avconv -filter_complex 'color=red' -t 5 out.mkv +ffmpeg -filter_complex 'color=c=red' -t 5 out.mkv @end example +@item -lavfi @var{filtergraph} (@emph{global}) +Define a complex filtergraph, i.e. one with arbitrary number of inputs and/or +outputs. Equivalent to @option{-filter_complex}. + @item -filter_complex_script @var{filename} (@emph{global}) This option is similar to @option{-filter_complex}, the only difference is that its argument is the name of the file from which a complex filtergraph @@ -956,96 +1216,180 @@ This option enables or disables accurate seeking in input files with the transcoding. Use @option{-noaccurate_seek} to disable it, which may be useful e.g. when copying some streams and transcoding the others. -@end table -@c man end OPTIONS +@item -seek_timestamp (@emph{input}) +This option enables or disables seeking by timestamp in input files with the +@option{-ss} option. It is disabled by default. If enabled, the argument +to the @option{-ss} option is considered an actual timestamp, and is not +offset by the start time of the file. This matters only for files which do +not start from timestamp 0, such as transport streams. + +@item -thread_queue_size @var{size} (@emph{input}) +This option sets the maximum number of queued packets when reading from the +file or device. With low latency / high rate live streams, packets may be +discarded if they are not read in a timely manner; raising this value can +avoid it. + +@item -override_ffserver (@emph{global}) +Overrides the input specifications from @command{ffserver}. Using this +option you can map any input stream to @command{ffserver} and control +many aspects of the encoding from @command{ffmpeg}. Without this +option @command{ffmpeg} will transmit to @command{ffserver} what is +requested by @command{ffserver}. + +The option is intended for cases where features are needed that cannot be +specified to @command{ffserver} but can be to @command{ffmpeg}. + +@item -sdp_file @var{file} (@emph{global}) +Print sdp information for an output stream to @var{file}. +This allows dumping sdp information when at least one output isn't an +rtp stream. (Requires at least one of the output formats to be rtp). + +@item -discard (@emph{input}) +Allows discarding specific streams or frames of streams at the demuxer. +Not all demuxers support this. -@chapter Tips -@c man begin TIPS +@table @option +@item none +Discard no frame. -@itemize -@item -For streaming at very low bitrate application, use a low frame rate -and a small GOP size. This is especially true for RealVideo where -the Linux player does not seem to be very fast, so it can miss -frames. An example is: +@item default +Default, which discards no frames. -@example -avconv -g 3 -r 3 -t 10 -b 50k -s qcif -f rv10 /tmp/b.rm -@end example +@item noref +Discard all non-reference frames. -@item -The parameter 'q' which is displayed while encoding is the current -quantizer. The value 1 indicates that a very good quality could -be achieved. The value 31 indicates the worst quality. If q=31 appears -too often, it means that the encoder cannot compress enough to meet -your bitrate. You must either increase the bitrate, decrease the -frame rate or decrease the frame size. +@item bidir +Discard all bidirectional frames. -@item -If your computer is not fast enough, you can speed up the -compression at the expense of the compression ratio. You can use -'-me zero' to speed up motion estimation, and '-g 0' to disable -motion estimation completely (you have only I-frames, which means it -is about as good as JPEG compression). +@item nokey +Discard all frames excepts keyframes. -@item -To have very low audio bitrates, reduce the sampling frequency -(down to 22050 Hz for MPEG audio, 22050 or 11025 for AC-3). +@item all +Discard all frames. +@end table -@item -To have a constant quality (but a variable bitrate), use the option -'-qscale n' when 'n' is between 1 (excellent quality) and 31 (worst -quality). +@item -abort_on @var{flags} (@emph{global}) +Stop and abort on various conditions. The following flags are available: -@end itemize -@c man end TIPS +@table @option +@item empty_output +No packets were passed to the muxer, the output is empty. +@end table -@chapter Examples -@c man begin EXAMPLES +@item -xerror (@emph{global}) +Stop and exit on error + +@end table + +As a special exception, you can use a bitmap subtitle stream as input: it +will be converted into a video with the same size as the largest video in +the file, or 720x576 if no video is present. Note that this is an +experimental and temporary solution. It will be removed once libavfilter has +proper support for subtitles. + +For example, to hardcode subtitles on top of a DVB-T recording stored in +MPEG-TS format, delaying the subtitles by 1 second: +@example +ffmpeg -i input.ts -filter_complex \ + '[#0x2ef] setpts=PTS+1/TB [sub] ; [#0x2d0] [sub] overlay' \ + -sn -map '#0x2dc' output.mkv +@end example +(0x2d0, 0x2dc and 0x2ef are the MPEG-TS PIDs of respectively the video, +audio and subtitles streams; 0:0, 0:3 and 0:7 would have worked too) @section Preset files +A preset file contains a sequence of @var{option}=@var{value} pairs, +one for each line, specifying a sequence of options which would be +awkward to specify on the command line. Lines starting with the hash +('#') character are ignored and are used to provide comments. Check +the @file{presets} directory in the FFmpeg source tree for examples. + +There are two types of preset files: ffpreset and avpreset files. + +@subsection ffpreset files +ffpreset files are specified with the @code{vpre}, @code{apre}, +@code{spre}, and @code{fpre} options. The @code{fpre} option takes the +filename of the preset instead of a preset name as input and can be +used for any kind of codec. For the @code{vpre}, @code{apre}, and +@code{spre} options, the options specified in a preset file are +applied to the currently selected codec of the same type as the preset +option. + +The argument passed to the @code{vpre}, @code{apre}, and @code{spre} +preset options identifies the preset file to use according to the +following rules: + +First ffmpeg searches for a file named @var{arg}.ffpreset in the +directories @file{$FFMPEG_DATADIR} (if set), and @file{$HOME/.ffmpeg}, and in +the datadir defined at configuration time (usually @file{PREFIX/share/ffmpeg}) +or in a @file{ffpresets} folder along the executable on win32, +in that order. For example, if the argument is @code{libvpx-1080p}, it will +search for the file @file{libvpx-1080p.ffpreset}. + +If no such file is found, then ffmpeg will search for a file named +@var{codec_name}-@var{arg}.ffpreset in the above-mentioned +directories, where @var{codec_name} is the name of the codec to which +the preset file options will be applied. For example, if you select +the video codec with @code{-vcodec libvpx} and use @code{-vpre 1080p}, +then it will search for the file @file{libvpx-1080p.ffpreset}. + +@subsection avpreset files +avpreset files are specified with the @code{pre} option. They work similar to +ffpreset files, but they only allow encoder- specific options. Therefore, an +@var{option}=@var{value} pair specifying an encoder cannot be used. + +When the @code{pre} option is specified, ffmpeg will look for files with the +suffix .avpreset in the directories @file{$AVCONV_DATADIR} (if set), and +@file{$HOME/.avconv}, and in the datadir defined at configuration time (usually +@file{PREFIX/share/ffmpeg}), in that order. + +First ffmpeg searches for a file named @var{codec_name}-@var{arg}.avpreset in +the above-mentioned directories, where @var{codec_name} is the name of the codec +to which the preset file options will be applied. For example, if you select the +video codec with @code{-vcodec libvpx} and use @code{-pre 1080p}, then it will +search for the file @file{libvpx-1080p.avpreset}. + +If no such file is found, then ffmpeg will search for a file named +@var{arg}.avpreset in the same directories. -A preset file contains a sequence of @var{option=value} pairs, one for -each line, specifying a sequence of options which can be specified also on -the command line. Lines starting with the hash ('#') character are ignored and -are used to provide comments. Empty lines are also ignored. Check the -@file{presets} directory in the Libav source tree for examples. +@c man end OPTIONS -Preset files are specified with the @code{pre} option, this option takes a -preset name as input. Avconv searches for a file named @var{preset_name}.avpreset in -the directories @file{$AVCONV_DATADIR} (if set), and @file{$HOME/.avconv}, and in -the data directory defined at configuration time (usually @file{$PREFIX/share/avconv}) -in that order. For example, if the argument is @code{libx264-max}, it will -search for the file @file{libx264-max.avpreset}. +@chapter Examples +@c man begin EXAMPLES @section Video and Audio grabbing -If you specify the input format and device then avconv can grab video +If you specify the input format and device then ffmpeg can grab video and audio directly. @example -avconv -f oss -i /dev/dsp -f video4linux2 -i /dev/video0 /tmp/out.mpg +ffmpeg -f oss -i /dev/dsp -f video4linux2 -i /dev/video0 /tmp/out.mpg +@end example + +Or with an ALSA audio source (mono input, card id 1) instead of OSS: +@example +ffmpeg -f alsa -ac 1 -i hw:1 -f video4linux2 -i /dev/video0 /tmp/out.mpg @end example Note that you must activate the right video source and channel before -launching avconv with any TV viewer such as +launching ffmpeg with any TV viewer such as @uref{http://linux.bytesex.org/xawtv/, xawtv} by Gerd Knorr. You also have to set the audio recording levels correctly with a standard mixer. @section X11 grabbing -Grab the X11 display with avconv via +Grab the X11 display with ffmpeg via @example -avconv -f x11grab -s cif -r 25 -i :0.0 /tmp/out.mpg +ffmpeg -f x11grab -video_size cif -framerate 25 -i :0.0 /tmp/out.mpg @end example 0.0 is display.screen number of your X11 server, same as the DISPLAY environment variable. @example -avconv -f x11grab -s cif -r 25 -i :0.0+10,20 /tmp/out.mpg +ffmpeg -f x11grab -video_size cif -framerate 25 -i :0.0+10,20 /tmp/out.mpg @end example 0.0 is display.screen number of your X11 server, same as the DISPLAY environment @@ -1053,7 +1397,7 @@ variable. 10 is the x-offset and 20 the y-offset for the grabbing. @section Video and Audio file format conversion -Any supported file format and protocol can serve as input to avconv: +Any supported file format and protocol can serve as input to ffmpeg: Examples: @itemize @@ -1061,7 +1405,7 @@ Examples: You can use YUV files as input: @example -avconv -i /tmp/test%d.Y /tmp/out.mpg +ffmpeg -i /tmp/test%d.Y /tmp/out.mpg @end example It will use the files: @@ -1073,13 +1417,13 @@ It will use the files: The Y files use twice the resolution of the U and V files. They are raw files, without header. They can be generated by all decent video decoders. You must specify the size of the image with the @option{-s} option -if avconv cannot guess it. +if ffmpeg cannot guess it. @item You can input from a raw YUV420P file: @example -avconv -i /tmp/test.yuv /tmp/out.avi +ffmpeg -i /tmp/test.yuv /tmp/out.avi @end example test.yuv is a file containing raw YUV planar data. Each frame is composed @@ -1090,14 +1434,14 @@ horizontal resolution. You can output to a raw YUV420P file: @example -avconv -i mydivx.avi hugefile.yuv +ffmpeg -i mydivx.avi hugefile.yuv @end example @item You can set several input files and output files: @example -avconv -i /tmp/a.wav -s 640x480 -i /tmp/a.yuv /tmp/a.mpg +ffmpeg -i /tmp/a.wav -s 640x480 -i /tmp/a.yuv /tmp/a.mpg @end example Converts the audio file a.wav and the raw YUV video file a.yuv @@ -1107,7 +1451,7 @@ to MPEG file a.mpg. You can also do audio and video conversions at the same time: @example -avconv -i /tmp/a.wav -ar 22050 /tmp/a.mp2 +ffmpeg -i /tmp/a.wav -ar 22050 /tmp/a.mp2 @end example Converts a.wav to MPEG audio at 22050 Hz sample rate. @@ -1117,7 +1461,7 @@ You can encode to several formats at the same time and define a mapping from input stream to output streams: @example -avconv -i /tmp/a.wav -map 0:a -b 64k /tmp/a.mp2 -map 0:a -b 128k /tmp/b.mp2 +ffmpeg -i /tmp/a.wav -map 0:a -b:a 64k /tmp/a.mp2 -map 0:a -b:a 128k /tmp/b.mp2 @end example Converts a.wav to a.mp2 at 64 kbits and to b.mp2 at 128 kbits. '-map @@ -1128,7 +1472,7 @@ stream, in the order of the definition of output streams. You can transcode decrypted VOBs: @example -avconv -i snatch_1.vob -f avi -c:v mpeg4 -b:v 800k -g 300 -bf 2 -c:a libmp3lame -b:a 128k snatch.avi +ffmpeg -i snatch_1.vob -f avi -c:v mpeg4 -b:v 800k -g 300 -bf 2 -c:a libmp3lame -b:a 128k snatch.avi @end example This is a typical DVD ripping example; the input is a VOB file, the @@ -1140,14 +1484,14 @@ to enable LAME support by passing @code{--enable-libmp3lame} to configure. The mapping is particularly useful for DVD transcoding to get the desired audio language. -NOTE: To see the supported input formats, use @code{avconv -formats}. +NOTE: To see the supported input formats, use @code{ffmpeg -formats}. @item You can extract images from a video, or create a video from many images: For extracting images from a video: @example -avconv -i foo.avi -r 1 -s WxH -f image2 foo-%03d.jpeg +ffmpeg -i foo.avi -r 1 -s WxH -f image2 foo-%03d.jpeg @end example This will extract one video frame per second from the video and will @@ -1160,7 +1504,7 @@ combination with -ss to start extracting from a certain point in time. For creating a video from many images: @example -avconv -f image2 -i foo-%03d.jpeg -r 12 -s WxH foo.avi +ffmpeg -f image2 -framerate 12 -i foo-%03d.jpeg -s WxH foo.avi @end example The syntax @code{foo-%03d.jpeg} specifies to use a decimal number @@ -1168,11 +1512,21 @@ composed of three digits padded with zeroes to express the sequence number. It is the same syntax supported by the C printf function, but only formats accepting a normal integer are suitable. +When importing an image sequence, -i also supports expanding +shell-like wildcard patterns (globbing) internally, by selecting the +image2-specific @code{-pattern_type glob} option. + +For example, for creating a video from filenames matching the glob pattern +@code{foo-*.jpeg}: +@example +ffmpeg -f image2 -pattern_type glob -framerate 12 -i 'foo-*.jpeg' -s WxH foo.avi +@end example + @item You can put many streams of the same type in the output: @example -avconv -i test1.avi -i test2.avi -map 1:1 -map 1:0 -map 0:1 -map 0:0 -c copy -y test12.nut +ffmpeg -i test1.avi -i test2.avi -map 1:1 -map 1:0 -map 0:1 -map 0:0 -c copy -y test12.nut @end example The resulting output file @file{test12.nut} will contain the first four streams @@ -1181,43 +1535,86 @@ from the input files in reverse order. @item To force CBR video output: @example -avconv -i myfile.avi -b 4000k -minrate 4000k -maxrate 4000k -bufsize 1835k out.m2v +ffmpeg -i myfile.avi -b 4000k -minrate 4000k -maxrate 4000k -bufsize 1835k out.m2v @end example @item The four options lmin, lmax, mblmin and mblmax use 'lambda' units, but you may use the QP2LAMBDA constant to easily convert from 'q' units: @example -avconv -i src.ext -lmax 21*QP2LAMBDA dst.ext +ffmpeg -i src.ext -lmax 21*QP2LAMBDA dst.ext @end example @end itemize @c man end EXAMPLES -@include eval.texi -@include decoders.texi -@include encoders.texi -@include demuxers.texi -@include muxers.texi -@include indevs.texi -@include outdevs.texi -@include protocols.texi +@include config.texi +@ifset config-all +@ifset config-avutil +@include utils.texi +@end ifset +@ifset config-avcodec +@include codecs.texi @include bitstream_filters.texi +@end ifset +@ifset config-avformat +@include formats.texi +@include protocols.texi +@end ifset +@ifset config-avdevice +@include devices.texi +@end ifset +@ifset config-swresample +@include resampler.texi +@end ifset +@ifset config-swscale +@include scaler.texi +@end ifset +@ifset config-avfilter @include filters.texi -@include metadata.texi +@end ifset +@end ifset + +@chapter See Also + +@ifhtml +@ifset config-all +@url{ffmpeg.html,ffmpeg} +@end ifset +@ifset config-not-all +@url{ffmpeg-all.html,ffmpeg-all}, +@end ifset +@url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver}, +@url{ffmpeg-utils.html,ffmpeg-utils}, +@url{ffmpeg-scaler.html,ffmpeg-scaler}, +@url{ffmpeg-resampler.html,ffmpeg-resampler}, +@url{ffmpeg-codecs.html,ffmpeg-codecs}, +@url{ffmpeg-bitstream-filters.html,ffmpeg-bitstream-filters}, +@url{ffmpeg-formats.html,ffmpeg-formats}, +@url{ffmpeg-devices.html,ffmpeg-devices}, +@url{ffmpeg-protocols.html,ffmpeg-protocols}, +@url{ffmpeg-filters.html,ffmpeg-filters} +@end ifhtml + +@ifnothtml +@ifset config-all +ffmpeg(1), +@end ifset +@ifset config-not-all +ffmpeg-all(1), +@end ifset +ffplay(1), ffprobe(1), ffserver(1), +ffmpeg-utils(1), ffmpeg-scaler(1), ffmpeg-resampler(1), +ffmpeg-codecs(1), ffmpeg-bitstream-filters(1), ffmpeg-formats(1), +ffmpeg-devices(1), ffmpeg-protocols(1), ffmpeg-filters(1) +@end ifnothtml + +@include authors.texi @ignore -@setfilename avconv -@settitle avconv video converter - -@c man begin SEEALSO -avplay(1), avprobe(1) and the Libav HTML documentation -@c man end - -@c man begin AUTHORS -The Libav developers -@c man end +@setfilename ffmpeg +@settitle ffmpeg video converter @end ignore diff --git a/doc/ffmpeg.txt b/doc/ffmpeg.txt new file mode 100644 index 0000000000..a028ca23d2 --- /dev/null +++ b/doc/ffmpeg.txt @@ -0,0 +1,47 @@ + : + ffmpeg.c : libav* + ======== : ====== + : + : + --------------------------------:---> AVStream... + InputStream input_streams[] / : + / : + InputFile input_files[] +==========================+ / ^ : + ------> 0 | : st ---:-----------:--/ : : + ^ +------+-----------+-----+ / +--------------------------+ : : + : | :ist_index--:-----:---------/ 1 | : st : | : : + : +------+-----------+-----+ +==========================+ : : + nb_input_files : | :ist_index--:-----:------------------> 2 | : st : | : : + : +------+-----------+-----+ +--------------------------+ : nb_input_streams : + : | :ist_index : | 3 | ... | : : + v +------+-----------+-----+ +--------------------------+ : : + --> 4 | | : : + | +--------------------------+ : : + | 5 | | : : + | +==========================+ v : + | : + | : + | : + | : + --------- --------------------------------:---> AVStream... + \ / : + OutputStream output_streams[] / : + \ / : + +======\======================/======+ ^ : + ------> 0 | : source_index : st-:--- | : : + OutputFile output_files[] / +------------------------------------+ : : + / 1 | : : : | : : + ^ +------+------------+-----+ / +------------------------------------+ : : + : | : ost_index -:-----:------/ 2 | : : : | : : + nb_output_files : +------+------------+-----+ +====================================+ : : + : | : ost_index -:-----|-----------------> 3 | : : : | : : + : +------+------------+-----+ +------------------------------------+ : nb_output_streams : + : | : : | 4 | | : : + : +------+------------+-----+ +------------------------------------+ : : + : | : : | 5 | | : : + v +------+------------+-----+ +------------------------------------+ : : + 6 | | : : + +------------------------------------+ : : + 7 | | : : + +====================================+ v : + : diff --git a/doc/ffplay.texi b/doc/ffplay.texi new file mode 100644 index 0000000000..4bc3ced39a --- /dev/null +++ b/doc/ffplay.texi @@ -0,0 +1,322 @@ +\input texinfo @c -*- texinfo -*- +@documentencoding UTF-8 + +@settitle ffplay Documentation +@titlepage +@center @titlefont{ffplay Documentation} +@end titlepage + +@top + +@contents + +@chapter Synopsis + +ffplay [@var{options}] [@file{input_file}] + +@chapter Description +@c man begin DESCRIPTION + +FFplay is a very simple and portable media player using the FFmpeg +libraries and the SDL library. It is mostly used as a testbed for the +various FFmpeg APIs. +@c man end + +@chapter Options +@c man begin OPTIONS + +@include fftools-common-opts.texi + +@section Main options + +@table @option +@item -x @var{width} +Force displayed width. +@item -y @var{height} +Force displayed height. +@item -s @var{size} +Set frame size (WxH or abbreviation), needed for videos which do +not contain a header with the frame size like raw YUV. This option +has been deprecated in favor of private options, try -video_size. +@item -fs +Start in fullscreen mode. +@item -an +Disable audio. +@item -vn +Disable video. +@item -sn +Disable subtitles. +@item -ss @var{pos} +Seek to @var{pos}. Note that in most formats it is not possible to seek +exactly, so @command{ffplay} will seek to the nearest seek point to +@var{pos}. + +@var{pos} must be a time duration specification, +see @ref{time duration syntax,,the Time duration section in the ffmpeg-utils(1) manual,ffmpeg-utils}. +@item -t @var{duration} +Play @var{duration} seconds of audio/video. + +@var{duration} must be a time duration specification, +see @ref{time duration syntax,,the Time duration section in the ffmpeg-utils(1) manual,ffmpeg-utils}. +@item -bytes +Seek by bytes. +@item -nodisp +Disable graphical display. +@item -f @var{fmt} +Force format. +@item -window_title @var{title} +Set window title (default is the input filename). +@item -loop @var{number} +Loops movie playback <number> times. 0 means forever. +@item -showmode @var{mode} +Set the show mode to use. +Available values for @var{mode} are: +@table @samp +@item 0, video +show video +@item 1, waves +show audio waves +@item 2, rdft +show audio frequency band using RDFT ((Inverse) Real Discrete Fourier Transform) +@end table + +Default value is "video", if video is not present or cannot be played +"rdft" is automatically selected. + +You can interactively cycle through the available show modes by +pressing the key @key{w}. + +@item -vf @var{filtergraph} +Create the filtergraph specified by @var{filtergraph} and use it to +filter the video stream. + +@var{filtergraph} is a description of the filtergraph to apply to +the stream, and must have a single video input and a single video +output. In the filtergraph, the input is associated to the label +@code{in}, and the output to the label @code{out}. See the +ffmpeg-filters manual for more information about the filtergraph +syntax. + +You can specify this parameter multiple times and cycle through the specified +filtergraphs along with the show modes by pressing the key @key{w}. + +@item -af @var{filtergraph} +@var{filtergraph} is a description of the filtergraph to apply to +the input audio. +Use the option "-filters" to show all the available filters (including +sources and sinks). + +@item -i @var{input_file} +Read @var{input_file}. +@end table + +@section Advanced options +@table @option +@item -pix_fmt @var{format} +Set pixel format. +This option has been deprecated in favor of private options, try -pixel_format. + +@item -stats +Print several playback statistics, in particular show the stream +duration, the codec parameters, the current position in the stream and +the audio/video synchronisation drift. It is on by default, to +explicitly disable it you need to specify @code{-nostats}. + +@item -fast +Non-spec-compliant optimizations. +@item -genpts +Generate pts. +@item -sync @var{type} +Set the master clock to audio (@code{type=audio}), video +(@code{type=video}) or external (@code{type=ext}). Default is audio. The +master clock is used to control audio-video synchronization. Most media +players use audio as master clock, but in some cases (streaming or high +quality broadcast) it is necessary to change that. This option is mainly +used for debugging purposes. +@item -ast @var{audio_stream_specifier} +Select the desired audio stream using the given stream specifier. The stream +specifiers are described in the @ref{Stream specifiers} chapter. If this option +is not specified, the "best" audio stream is selected in the program of the +already selected video stream. +@item -vst @var{video_stream_specifier} +Select the desired video stream using the given stream specifier. The stream +specifiers are described in the @ref{Stream specifiers} chapter. If this option +is not specified, the "best" video stream is selected. +@item -sst @var{subtitle_stream_specifier} +Select the desired subtitle stream using the given stream specifier. The stream +specifiers are described in the @ref{Stream specifiers} chapter. If this option +is not specified, the "best" subtitle stream is selected in the program of the +already selected video or audio stream. +@item -autoexit +Exit when video is done playing. +@item -exitonkeydown +Exit if any key is pressed. +@item -exitonmousedown +Exit if any mouse button is pressed. + +@item -codec:@var{media_specifier} @var{codec_name} +Force a specific decoder implementation for the stream identified by +@var{media_specifier}, which can assume the values @code{a} (audio), +@code{v} (video), and @code{s} subtitle. + +@item -acodec @var{codec_name} +Force a specific audio decoder. + +@item -vcodec @var{codec_name} +Force a specific video decoder. + +@item -scodec @var{codec_name} +Force a specific subtitle decoder. + +@item -autorotate +Automatically rotate the video according to file metadata. Enabled by +default, use @option{-noautorotate} to disable it. + +@item -framedrop +Drop video frames if video is out of sync. Enabled by default if the master +clock is not set to video. Use this option to enable frame dropping for all +master clock sources, use @option{-noframedrop} to disable it. + +@item -infbuf +Do not limit the input buffer size, read as much data as possible from the +input as soon as possible. Enabled by default for realtime streams, where data +may be dropped if not read in time. Use this option to enable infinite buffers +for all inputs, use @option{-noinfbuf} to disable it. + +@end table + +@section While playing + +@table @key +@item q, ESC +Quit. + +@item f +Toggle full screen. + +@item p, SPC +Pause. + +@item m +Toggle mute. + +@item 9, 0 +Decrease and increase volume respectively. + +@item /, * +Decrease and increase volume respectively. + +@item a +Cycle audio channel in the current program. + +@item v +Cycle video channel. + +@item t +Cycle subtitle channel in the current program. + +@item c +Cycle program. + +@item w +Cycle video filters or show modes. + +@item s +Step to the next frame. + +Pause if the stream is not already paused, step to the next video +frame, and pause. + +@item left/right +Seek backward/forward 10 seconds. + +@item down/up +Seek backward/forward 1 minute. + +@item page down/page up +Seek to the previous/next chapter. +or if there are no chapters +Seek backward/forward 10 minutes. + +@item right mouse click +Seek to percentage in file corresponding to fraction of width. + +@item left mouse double-click +Toggle full screen. + +@end table + +@c man end + +@include config.texi +@ifset config-all +@set config-readonly +@ifset config-avutil +@include utils.texi +@end ifset +@ifset config-avcodec +@include codecs.texi +@include bitstream_filters.texi +@end ifset +@ifset config-avformat +@include formats.texi +@include protocols.texi +@end ifset +@ifset config-avdevice +@include devices.texi +@end ifset +@ifset config-swresample +@include resampler.texi +@end ifset +@ifset config-swscale +@include scaler.texi +@end ifset +@ifset config-avfilter +@include filters.texi +@end ifset +@end ifset + +@chapter See Also + +@ifhtml +@ifset config-all +@url{ffplay.html,ffplay}, +@end ifset +@ifset config-not-all +@url{ffplay-all.html,ffmpeg-all}, +@end ifset +@url{ffmpeg.html,ffmpeg}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver}, +@url{ffmpeg-utils.html,ffmpeg-utils}, +@url{ffmpeg-scaler.html,ffmpeg-scaler}, +@url{ffmpeg-resampler.html,ffmpeg-resampler}, +@url{ffmpeg-codecs.html,ffmpeg-codecs}, +@url{ffmpeg-bitstream-filters.html,ffmpeg-bitstream-filters}, +@url{ffmpeg-formats.html,ffmpeg-formats}, +@url{ffmpeg-devices.html,ffmpeg-devices}, +@url{ffmpeg-protocols.html,ffmpeg-protocols}, +@url{ffmpeg-filters.html,ffmpeg-filters} +@end ifhtml + +@ifnothtml +@ifset config-all +ffplay(1), +@end ifset +@ifset config-not-all +ffplay-all(1), +@end ifset +ffmpeg(1), ffprobe(1), ffserver(1), +ffmpeg-utils(1), ffmpeg-scaler(1), ffmpeg-resampler(1), +ffmpeg-codecs(1), ffmpeg-bitstream-filters(1), ffmpeg-formats(1), +ffmpeg-devices(1), ffmpeg-protocols(1), ffmpeg-filters(1) +@end ifnothtml + +@include authors.texi + +@ignore + +@setfilename ffplay +@settitle FFplay media player + +@end ignore + +@bye diff --git a/doc/ffprobe.texi b/doc/ffprobe.texi new file mode 100644 index 0000000000..2024eed4e5 --- /dev/null +++ b/doc/ffprobe.texi @@ -0,0 +1,683 @@ +\input texinfo @c -*- texinfo -*- +@documentencoding UTF-8 + +@settitle ffprobe Documentation +@titlepage +@center @titlefont{ffprobe Documentation} +@end titlepage + +@top + +@contents + +@chapter Synopsis + +ffprobe [@var{options}] [@file{input_file}] + +@chapter Description +@c man begin DESCRIPTION + +ffprobe gathers information from multimedia streams and prints it in +human- and machine-readable fashion. + +For example it can be used to check the format of the container used +by a multimedia stream and the format and type of each media stream +contained in it. + +If a filename is specified in input, ffprobe will try to open and +probe the file content. If the file cannot be opened or recognized as +a multimedia file, a positive exit code is returned. + +ffprobe may be employed both as a standalone application or in +combination with a textual filter, which may perform more +sophisticated processing, e.g. statistical processing or plotting. + +Options are used to list some of the formats supported by ffprobe or +for specifying which information to display, and for setting how +ffprobe will show it. + +ffprobe output is designed to be easily parsable by a textual filter, +and consists of one or more sections of a form defined by the selected +writer, which is specified by the @option{print_format} option. + +Sections may contain other nested sections, and are identified by a +name (which may be shared by other sections), and an unique +name. See the output of @option{sections}. + +Metadata tags stored in the container or in the streams are recognized +and printed in the corresponding "FORMAT", "STREAM" or "PROGRAM_STREAM" +section. + +@c man end + +@chapter Options +@c man begin OPTIONS + +@include fftools-common-opts.texi + +@section Main options + +@table @option + +@item -f @var{format} +Force format to use. + +@item -unit +Show the unit of the displayed values. + +@item -prefix +Use SI prefixes for the displayed values. +Unless the "-byte_binary_prefix" option is used all the prefixes +are decimal. + +@item -byte_binary_prefix +Force the use of binary prefixes for byte values. + +@item -sexagesimal +Use sexagesimal format HH:MM:SS.MICROSECONDS for time values. + +@item -pretty +Prettify the format of the displayed values, it corresponds to the +options "-unit -prefix -byte_binary_prefix -sexagesimal". + +@item -of, -print_format @var{writer_name}[=@var{writer_options}] +Set the output printing format. + +@var{writer_name} specifies the name of the writer, and +@var{writer_options} specifies the options to be passed to the writer. + +For example for printing the output in JSON format, specify: +@example +-print_format json +@end example + +For more details on the available output printing formats, see the +Writers section below. + +@item -sections +Print sections structure and section information, and exit. The output +is not meant to be parsed by a machine. + +@item -select_streams @var{stream_specifier} +Select only the streams specified by @var{stream_specifier}. This +option affects only the options related to streams +(e.g. @code{show_streams}, @code{show_packets}, etc.). + +For example to show only audio streams, you can use the command: +@example +ffprobe -show_streams -select_streams a INPUT +@end example + +To show only video packets belonging to the video stream with index 1: +@example +ffprobe -show_packets -select_streams v:1 INPUT +@end example + +@item -show_data +Show payload data, as a hexadecimal and ASCII dump. Coupled with +@option{-show_packets}, it will dump the packets' data. Coupled with +@option{-show_streams}, it will dump the codec extradata. + +The dump is printed as the "data" field. It may contain newlines. + +@item -show_data_hash @var{algorithm} +Show a hash of payload data, for packets with @option{-show_packets} and for +codec extradata with @option{-show_streams}. + +@item -show_error +Show information about the error found when trying to probe the input. + +The error information is printed within a section with name "ERROR". + +@item -show_format +Show information about the container format of the input multimedia +stream. + +All the container format information is printed within a section with +name "FORMAT". + +@item -show_format_entry @var{name} +Like @option{-show_format}, but only prints the specified entry of the +container format information, rather than all. This option may be given more +than once, then all specified entries will be shown. + +This option is deprecated, use @code{show_entries} instead. + +@item -show_entries @var{section_entries} +Set list of entries to show. + +Entries are specified according to the following +syntax. @var{section_entries} contains a list of section entries +separated by @code{:}. Each section entry is composed by a section +name (or unique name), optionally followed by a list of entries local +to that section, separated by @code{,}. + +If section name is specified but is followed by no @code{=}, all +entries are printed to output, together with all the contained +sections. Otherwise only the entries specified in the local section +entries list are printed. In particular, if @code{=} is specified but +the list of local entries is empty, then no entries will be shown for +that section. + +Note that the order of specification of the local section entries is +not honored in the output, and the usual display order will be +retained. + +The formal syntax is given by: +@example +@var{LOCAL_SECTION_ENTRIES} ::= @var{SECTION_ENTRY_NAME}[,@var{LOCAL_SECTION_ENTRIES}] +@var{SECTION_ENTRY} ::= @var{SECTION_NAME}[=[@var{LOCAL_SECTION_ENTRIES}]] +@var{SECTION_ENTRIES} ::= @var{SECTION_ENTRY}[:@var{SECTION_ENTRIES}] +@end example + +For example, to show only the index and type of each stream, and the PTS +time, duration time, and stream index of the packets, you can specify +the argument: +@example +packet=pts_time,duration_time,stream_index : stream=index,codec_type +@end example + +To show all the entries in the section "format", but only the codec +type in the section "stream", specify the argument: +@example +format : stream=codec_type +@end example + +To show all the tags in the stream and format sections: +@example +stream_tags : format_tags +@end example + +To show only the @code{title} tag (if available) in the stream +sections: +@example +stream_tags=title +@end example + +@item -show_packets +Show information about each packet contained in the input multimedia +stream. + +The information for each single packet is printed within a dedicated +section with name "PACKET". + +@item -show_frames +Show information about each frame and subtitle contained in the input +multimedia stream. + +The information for each single frame is printed within a dedicated +section with name "FRAME" or "SUBTITLE". + +@item -show_streams +Show information about each media stream contained in the input +multimedia stream. + +Each media stream information is printed within a dedicated section +with name "STREAM". + +@item -show_programs +Show information about programs and their streams contained in the input +multimedia stream. + +Each media stream information is printed within a dedicated section +with name "PROGRAM_STREAM". + +@item -show_chapters +Show information about chapters stored in the format. + +Each chapter is printed within a dedicated section with name "CHAPTER". + +@item -count_frames +Count the number of frames per stream and report it in the +corresponding stream section. + +@item -count_packets +Count the number of packets per stream and report it in the +corresponding stream section. + +@item -read_intervals @var{read_intervals} + +Read only the specified intervals. @var{read_intervals} must be a +sequence of interval specifications separated by ",". +@command{ffprobe} will seek to the interval starting point, and will +continue reading from that. + +Each interval is specified by two optional parts, separated by "%". + +The first part specifies the interval start position. It is +interpreted as an abolute position, or as a relative offset from the +current position if it is preceded by the "+" character. If this first +part is not specified, no seeking will be performed when reading this +interval. + +The second part specifies the interval end position. It is interpreted +as an absolute position, or as a relative offset from the current +position if it is preceded by the "+" character. If the offset +specification starts with "#", it is interpreted as the number of +packets to read (not including the flushing packets) from the interval +start. If no second part is specified, the program will read until the +end of the input. + +Note that seeking is not accurate, thus the actual interval start +point may be different from the specified position. Also, when an +interval duration is specified, the absolute end time will be computed +by adding the duration to the interval start point found by seeking +the file, rather than to the specified start value. + +The formal syntax is given by: +@example +@var{INTERVAL} ::= [@var{START}|+@var{START_OFFSET}][%[@var{END}|+@var{END_OFFSET}]] +@var{INTERVALS} ::= @var{INTERVAL}[,@var{INTERVALS}] +@end example + +A few examples follow. +@itemize +@item +Seek to time 10, read packets until 20 seconds after the found seek +point, then seek to position @code{01:30} (1 minute and thirty +seconds) and read packets until position @code{01:45}. +@example +10%+20,01:30%01:45 +@end example + +@item +Read only 42 packets after seeking to position @code{01:23}: +@example +01:23%+#42 +@end example + +@item +Read only the first 20 seconds from the start: +@example +%+20 +@end example + +@item +Read from the start until position @code{02:30}: +@example +%02:30 +@end example +@end itemize + +@item -show_private_data, -private +Show private data, that is data depending on the format of the +particular shown element. +This option is enabled by default, but you may need to disable it +for specific uses, for example when creating XSD-compliant XML output. + +@item -show_program_version +Show information related to program version. + +Version information is printed within a section with name +"PROGRAM_VERSION". + +@item -show_library_versions +Show information related to library versions. + +Version information for each library is printed within a section with +name "LIBRARY_VERSION". + +@item -show_versions +Show information related to program and library versions. This is the +equivalent of setting both @option{-show_program_version} and +@option{-show_library_versions} options. + +@item -show_pixel_formats +Show information about all pixel formats supported by FFmpeg. + +Pixel format information for each format is printed within a section +with name "PIXEL_FORMAT". + +@item -bitexact +Force bitexact output, useful to produce output which is not dependent +on the specific build. + +@item -i @var{input_file} +Read @var{input_file}. + +@end table +@c man end + +@chapter Writers +@c man begin WRITERS + +A writer defines the output format adopted by @command{ffprobe}, and will be +used for printing all the parts of the output. + +A writer may accept one or more arguments, which specify the options +to adopt. The options are specified as a list of @var{key}=@var{value} +pairs, separated by ":". + +All writers support the following options: + +@table @option +@item string_validation, sv +Set string validation mode. + +The following values are accepted. +@table @samp +@item fail +The writer will fail immediately in case an invalid string (UTF-8) +sequence or code point is found in the input. This is especially +useful to validate input metadata. + +@item ignore +Any validation error will be ignored. This will result in possibly +broken output, especially with the json or xml writer. + +@item replace +The writer will substitute invalid UTF-8 sequences or code points with +the string specified with the @option{string_validation_replacement}. +@end table + +Default value is @samp{replace}. + +@item string_validation_replacement, svr +Set replacement string to use in case @option{string_validation} is +set to @samp{replace}. + +In case the option is not specified, the writer will assume the empty +string, that is it will remove the invalid sequences from the input +strings. +@end table + +A description of the currently available writers follows. + +@section default +Default format. + +Print each section in the form: +@example +[SECTION] +key1=val1 +... +keyN=valN +[/SECTION] +@end example + +Metadata tags are printed as a line in the corresponding FORMAT, STREAM or +PROGRAM_STREAM section, and are prefixed by the string "TAG:". + +A description of the accepted options follows. + +@table @option + +@item nokey, nk +If set to 1 specify not to print the key of each field. Default value +is 0. + +@item noprint_wrappers, nw +If set to 1 specify not to print the section header and footer. +Default value is 0. +@end table + +@section compact, csv +Compact and CSV format. + +The @code{csv} writer is equivalent to @code{compact}, but supports +different defaults. + +Each section is printed on a single line. +If no option is specifid, the output has the form: +@example +section|key1=val1| ... |keyN=valN +@end example + +Metadata tags are printed in the corresponding "format" or "stream" +section. A metadata tag key, if printed, is prefixed by the string +"tag:". + +The description of the accepted options follows. + +@table @option + +@item item_sep, s +Specify the character to use for separating fields in the output line. +It must be a single printable character, it is "|" by default ("," for +the @code{csv} writer). + +@item nokey, nk +If set to 1 specify not to print the key of each field. Its default +value is 0 (1 for the @code{csv} writer). + +@item escape, e +Set the escape mode to use, default to "c" ("csv" for the @code{csv} +writer). + +It can assume one of the following values: +@table @option +@item c +Perform C-like escaping. Strings containing a newline (@samp{\n}), carriage +return (@samp{\r}), a tab (@samp{\t}), a form feed (@samp{\f}), the escaping +character (@samp{\}) or the item separator character @var{SEP} are escaped +using C-like fashioned escaping, so that a newline is converted to the +sequence @samp{\n}, a carriage return to @samp{\r}, @samp{\} to @samp{\\} and +the separator @var{SEP} is converted to @samp{\@var{SEP}}. + +@item csv +Perform CSV-like escaping, as described in RFC4180. Strings +containing a newline (@samp{\n}), a carriage return (@samp{\r}), a double quote +(@samp{"}), or @var{SEP} are enclosed in double-quotes. + +@item none +Perform no escaping. +@end table + +@item print_section, p +Print the section name at the begin of each line if the value is +@code{1}, disable it with value set to @code{0}. Default value is +@code{1}. + +@end table + +@section flat +Flat format. + +A free-form output where each line contains an explicit key=value, such as +"streams.stream.3.tags.foo=bar". The output is shell escaped, so it can be +directly embedded in sh scripts as long as the separator character is an +alphanumeric character or an underscore (see @var{sep_char} option). + +The description of the accepted options follows. + +@table @option +@item sep_char, s +Separator character used to separate the chapter, the section name, IDs and +potential tags in the printed field key. + +Default value is @samp{.}. + +@item hierarchical, h +Specify if the section name specification should be hierarchical. If +set to 1, and if there is more than one section in the current +chapter, the section name will be prefixed by the name of the +chapter. A value of 0 will disable this behavior. + +Default value is 1. +@end table + +@section ini +INI format output. + +Print output in an INI based format. + +The following conventions are adopted: + +@itemize +@item +all key and values are UTF-8 +@item +@samp{.} is the subgroup separator +@item +newline, @samp{\t}, @samp{\f}, @samp{\b} and the following characters are +escaped +@item +@samp{\} is the escape character +@item +@samp{#} is the comment indicator +@item +@samp{=} is the key/value separator +@item +@samp{:} is not used but usually parsed as key/value separator +@end itemize + +This writer accepts options as a list of @var{key}=@var{value} pairs, +separated by @samp{:}. + +The description of the accepted options follows. + +@table @option +@item hierarchical, h +Specify if the section name specification should be hierarchical. If +set to 1, and if there is more than one section in the current +chapter, the section name will be prefixed by the name of the +chapter. A value of 0 will disable this behavior. + +Default value is 1. +@end table + +@section json +JSON based format. + +Each section is printed using JSON notation. + +The description of the accepted options follows. + +@table @option + +@item compact, c +If set to 1 enable compact output, that is each section will be +printed on a single line. Default value is 0. +@end table + +For more information about JSON, see @url{http://www.json.org/}. + +@section xml +XML based format. + +The XML output is described in the XML schema description file +@file{ffprobe.xsd} installed in the FFmpeg datadir. + +An updated version of the schema can be retrieved at the url +@url{http://www.ffmpeg.org/schema/ffprobe.xsd}, which redirects to the +latest schema committed into the FFmpeg development source code tree. + +Note that the output issued will be compliant to the +@file{ffprobe.xsd} schema only when no special global output options +(@option{unit}, @option{prefix}, @option{byte_binary_prefix}, +@option{sexagesimal} etc.) are specified. + +The description of the accepted options follows. + +@table @option + +@item fully_qualified, q +If set to 1 specify if the output should be fully qualified. Default +value is 0. +This is required for generating an XML file which can be validated +through an XSD file. + +@item xsd_compliant, x +If set to 1 perform more checks for ensuring that the output is XSD +compliant. Default value is 0. +This option automatically sets @option{fully_qualified} to 1. +@end table + +For more information about the XML format, see +@url{http://www.w3.org/XML/}. +@c man end WRITERS + +@chapter Timecode +@c man begin TIMECODE + +@command{ffprobe} supports Timecode extraction: + +@itemize + +@item +MPEG1/2 timecode is extracted from the GOP, and is available in the video +stream details (@option{-show_streams}, see @var{timecode}). + +@item +MOV timecode is extracted from tmcd track, so is available in the tmcd +stream metadata (@option{-show_streams}, see @var{TAG:timecode}). + +@item +DV, GXF and AVI timecodes are available in format metadata +(@option{-show_format}, see @var{TAG:timecode}). + +@end itemize +@c man end TIMECODE + +@include config.texi +@ifset config-all +@set config-readonly +@ifset config-avutil +@include utils.texi +@end ifset +@ifset config-avcodec +@include codecs.texi +@include bitstream_filters.texi +@end ifset +@ifset config-avformat +@include formats.texi +@include protocols.texi +@end ifset +@ifset config-avdevice +@include devices.texi +@end ifset +@ifset config-swresample +@include resampler.texi +@end ifset +@ifset config-swscale +@include scaler.texi +@end ifset +@ifset config-avfilter +@include filters.texi +@end ifset +@end ifset + +@chapter See Also + +@ifhtml +@ifset config-all +@url{ffprobe.html,ffprobe}, +@end ifset +@ifset config-not-all +@url{ffprobe-all.html,ffprobe-all}, +@end ifset +@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffserver.html,ffserver}, +@url{ffmpeg-utils.html,ffmpeg-utils}, +@url{ffmpeg-scaler.html,ffmpeg-scaler}, +@url{ffmpeg-resampler.html,ffmpeg-resampler}, +@url{ffmpeg-codecs.html,ffmpeg-codecs}, +@url{ffmpeg-bitstream-filters.html,ffmpeg-bitstream-filters}, +@url{ffmpeg-formats.html,ffmpeg-formats}, +@url{ffmpeg-devices.html,ffmpeg-devices}, +@url{ffmpeg-protocols.html,ffmpeg-protocols}, +@url{ffmpeg-filters.html,ffmpeg-filters} +@end ifhtml + +@ifnothtml +@ifset config-all +ffprobe(1), +@end ifset +@ifset config-not-all +ffprobe-all(1), +@end ifset +ffmpeg(1), ffplay(1), ffserver(1), +ffmpeg-utils(1), ffmpeg-scaler(1), ffmpeg-resampler(1), +ffmpeg-codecs(1), ffmpeg-bitstream-filters(1), ffmpeg-formats(1), +ffmpeg-devices(1), ffmpeg-protocols(1), ffmpeg-filters(1) +@end ifnothtml + +@include authors.texi + +@ignore + +@setfilename ffprobe +@settitle ffprobe media prober + +@end ignore + +@bye diff --git a/doc/ffprobe.xsd b/doc/ffprobe.xsd new file mode 100644 index 0000000000..c7d5101965 --- /dev/null +++ b/doc/ffprobe.xsd @@ -0,0 +1,356 @@ +<?xml version="1.0" encoding="UTF-8"?> + +<xsd:schema xmlns:xsd="http://www.w3.org/2001/XMLSchema" + targetNamespace="http://www.ffmpeg.org/schema/ffprobe" + xmlns:ffprobe="http://www.ffmpeg.org/schema/ffprobe"> + + <xsd:element name="ffprobe" type="ffprobe:ffprobeType"/> + + <xsd:complexType name="ffprobeType"> + <xsd:sequence> + <xsd:element name="program_version" type="ffprobe:programVersionType" minOccurs="0" maxOccurs="1" /> + <xsd:element name="library_versions" type="ffprobe:libraryVersionsType" minOccurs="0" maxOccurs="1" /> + <xsd:element name="pixel_formats" type="ffprobe:pixelFormatsType" minOccurs="0" maxOccurs="1" /> + <xsd:element name="packets" type="ffprobe:packetsType" minOccurs="0" maxOccurs="1" /> + <xsd:element name="frames" type="ffprobe:framesType" minOccurs="0" maxOccurs="1" /> + <xsd:element name="packets_and_frames" type="ffprobe:packetsAndFramesType" minOccurs="0" maxOccurs="1" /> + <xsd:element name="programs" type="ffprobe:programsType" minOccurs="0" maxOccurs="1" /> + <xsd:element name="streams" type="ffprobe:streamsType" minOccurs="0" maxOccurs="1" /> + <xsd:element name="chapters" type="ffprobe:chaptersType" minOccurs="0" maxOccurs="1" /> + <xsd:element name="format" type="ffprobe:formatType" minOccurs="0" maxOccurs="1" /> + <xsd:element name="error" type="ffprobe:errorType" minOccurs="0" maxOccurs="1" /> + </xsd:sequence> + </xsd:complexType> + + <xsd:complexType name="packetsType"> + <xsd:sequence> + <xsd:element name="packet" type="ffprobe:packetType" minOccurs="0" maxOccurs="unbounded"/> + </xsd:sequence> + </xsd:complexType> + + <xsd:complexType name="framesType"> + <xsd:sequence> + <xsd:choice minOccurs="0" maxOccurs="unbounded"> + <xsd:element name="frame" type="ffprobe:frameType" minOccurs="0" maxOccurs="unbounded"/> + <xsd:element name="subtitle" type="ffprobe:subtitleType" minOccurs="0" maxOccurs="unbounded"/> + </xsd:choice> + </xsd:sequence> + </xsd:complexType> + + <xsd:complexType name="packetsAndFramesType"> + <xsd:sequence> + <xsd:choice minOccurs="0" maxOccurs="unbounded"> + <xsd:element name="packet" type="ffprobe:packetType" minOccurs="0" maxOccurs="unbounded"/> + <xsd:element name="frame" type="ffprobe:frameType" minOccurs="0" maxOccurs="unbounded"/> + <xsd:element name="subtitle" type="ffprobe:subtitleType" minOccurs="0" maxOccurs="unbounded"/> + </xsd:choice> + </xsd:sequence> + </xsd:complexType> + + <xsd:complexType name="packetType"> + <xsd:sequence> + <xsd:element name="tag" type="ffprobe:tagType" minOccurs="0" maxOccurs="unbounded"/> + <xsd:element name="side_data_list" type="ffprobe:packetSideDataListType" minOccurs="0" maxOccurs="1" /> + </xsd:sequence> + + <xsd:attribute name="codec_type" type="xsd:string" use="required" /> + <xsd:attribute name="stream_index" type="xsd:int" use="required" /> + <xsd:attribute name="pts" type="xsd:long" /> + <xsd:attribute name="pts_time" type="xsd:float" /> + <xsd:attribute name="dts" type="xsd:long" /> + <xsd:attribute name="dts_time" type="xsd:float" /> + <xsd:attribute name="duration" type="xsd:long" /> + <xsd:attribute name="duration_time" type="xsd:float" /> + <xsd:attribute name="convergence_duration" type="xsd:long" /> + <xsd:attribute name="convergence_duration_time" type="xsd:float" /> + <xsd:attribute name="size" type="xsd:long" use="required" /> + <xsd:attribute name="pos" type="xsd:long" /> + <xsd:attribute name="flags" type="xsd:string" use="required" /> + <xsd:attribute name="data" type="xsd:string" /> + <xsd:attribute name="data_hash" type="xsd:string" /> + </xsd:complexType> + + <xsd:complexType name="packetSideDataListType"> + <xsd:sequence> + <xsd:element name="side_data" type="ffprobe:packetSideDataType" minOccurs="1" maxOccurs="unbounded"/> + </xsd:sequence> + </xsd:complexType> + <xsd:complexType name="packetSideDataType"> + <xsd:attribute name="side_data_type" type="xsd:string"/> + <xsd:attribute name="side_data_size" type="xsd:int" /> + </xsd:complexType> + + <xsd:complexType name="frameType"> + <xsd:sequence> + <xsd:element name="tag" type="ffprobe:tagType" minOccurs="0" maxOccurs="unbounded"/> + <xsd:element name="side_data_list" type="ffprobe:frameSideDataListType" minOccurs="0" maxOccurs="1" /> + </xsd:sequence> + + <xsd:attribute name="media_type" type="xsd:string" use="required"/> + <xsd:attribute name="stream_index" type="xsd:int" /> + <xsd:attribute name="key_frame" type="xsd:int" use="required"/> + <xsd:attribute name="pts" type="xsd:long" /> + <xsd:attribute name="pts_time" type="xsd:float"/> + <xsd:attribute name="pkt_pts" type="xsd:long" /> + <xsd:attribute name="pkt_pts_time" type="xsd:float"/> + <xsd:attribute name="pkt_dts" type="xsd:long" /> + <xsd:attribute name="pkt_dts_time" type="xsd:float"/> + <xsd:attribute name="best_effort_timestamp" type="xsd:long" /> + <xsd:attribute name="best_effort_timestamp_time" type="xsd:float" /> + <xsd:attribute name="pkt_duration" type="xsd:long" /> + <xsd:attribute name="pkt_duration_time" type="xsd:float"/> + <xsd:attribute name="pkt_pos" type="xsd:long" /> + <xsd:attribute name="pkt_size" type="xsd:int" /> + + <!-- audio attributes --> + <xsd:attribute name="sample_fmt" type="xsd:string"/> + <xsd:attribute name="nb_samples" type="xsd:long" /> + <xsd:attribute name="channels" type="xsd:int" /> + <xsd:attribute name="channel_layout" type="xsd:string"/> + + <!-- video attributes --> + <xsd:attribute name="width" type="xsd:long" /> + <xsd:attribute name="height" type="xsd:long" /> + <xsd:attribute name="pix_fmt" type="xsd:string"/> + <xsd:attribute name="sample_aspect_ratio" type="xsd:string"/> + <xsd:attribute name="pict_type" type="xsd:string"/> + <xsd:attribute name="coded_picture_number" type="xsd:long" /> + <xsd:attribute name="display_picture_number" type="xsd:long" /> + <xsd:attribute name="interlaced_frame" type="xsd:int" /> + <xsd:attribute name="top_field_first" type="xsd:int" /> + <xsd:attribute name="repeat_pict" type="xsd:int" /> + </xsd:complexType> + + <xsd:complexType name="frameSideDataListType"> + <xsd:sequence> + <xsd:element name="side_data" type="ffprobe:frameSideDataType" minOccurs="1" maxOccurs="unbounded"/> + </xsd:sequence> + </xsd:complexType> + <xsd:complexType name="frameSideDataType"> + <xsd:attribute name="side_data_type" type="xsd:string"/> + <xsd:attribute name="side_data_size" type="xsd:int" /> + </xsd:complexType> + + <xsd:complexType name="subtitleType"> + <xsd:attribute name="media_type" type="xsd:string" fixed="subtitle" use="required"/> + <xsd:attribute name="pts" type="xsd:long" /> + <xsd:attribute name="pts_time" type="xsd:float"/> + <xsd:attribute name="format" type="xsd:int" /> + <xsd:attribute name="start_display_time" type="xsd:int" /> + <xsd:attribute name="end_display_time" type="xsd:int" /> + <xsd:attribute name="num_rects" type="xsd:int" /> + </xsd:complexType> + + <xsd:complexType name="streamsType"> + <xsd:sequence> + <xsd:element name="stream" type="ffprobe:streamType" minOccurs="0" maxOccurs="unbounded"/> + </xsd:sequence> + </xsd:complexType> + + <xsd:complexType name="programsType"> + <xsd:sequence> + <xsd:element name="program" type="ffprobe:programType" minOccurs="0" maxOccurs="unbounded"/> + </xsd:sequence> + </xsd:complexType> + + <xsd:complexType name="streamDispositionType"> + <xsd:attribute name="default" type="xsd:int" use="required" /> + <xsd:attribute name="dub" type="xsd:int" use="required" /> + <xsd:attribute name="original" type="xsd:int" use="required" /> + <xsd:attribute name="comment" type="xsd:int" use="required" /> + <xsd:attribute name="lyrics" type="xsd:int" use="required" /> + <xsd:attribute name="karaoke" type="xsd:int" use="required" /> + <xsd:attribute name="forced" type="xsd:int" use="required" /> + <xsd:attribute name="hearing_impaired" type="xsd:int" use="required" /> + <xsd:attribute name="visual_impaired" type="xsd:int" use="required" /> + <xsd:attribute name="clean_effects" type="xsd:int" use="required" /> + <xsd:attribute name="attached_pic" type="xsd:int" use="required" /> + </xsd:complexType> + + <xsd:complexType name="streamType"> + <xsd:sequence> + <xsd:element name="disposition" type="ffprobe:streamDispositionType" minOccurs="0" maxOccurs="1"/> + <xsd:element name="tag" type="ffprobe:tagType" minOccurs="0" maxOccurs="unbounded"/> + <xsd:element name="side_data_list" type="ffprobe:packetSideDataListType" minOccurs="0" maxOccurs="1" /> + </xsd:sequence> + + <xsd:attribute name="index" type="xsd:int" use="required"/> + <xsd:attribute name="codec_name" type="xsd:string" /> + <xsd:attribute name="codec_long_name" type="xsd:string" /> + <xsd:attribute name="profile" type="xsd:string" /> + <xsd:attribute name="codec_type" type="xsd:string" /> + <xsd:attribute name="codec_time_base" type="xsd:string" use="required"/> + <xsd:attribute name="codec_tag" type="xsd:string" use="required"/> + <xsd:attribute name="codec_tag_string" type="xsd:string" use="required"/> + <xsd:attribute name="extradata" type="xsd:string" /> + <xsd:attribute name="extradata_hash" type="xsd:string" /> + + <!-- video attributes --> + <xsd:attribute name="width" type="xsd:int"/> + <xsd:attribute name="height" type="xsd:int"/> + <xsd:attribute name="coded_width" type="xsd:int"/> + <xsd:attribute name="coded_height" type="xsd:int"/> + <xsd:attribute name="has_b_frames" type="xsd:int"/> + <xsd:attribute name="sample_aspect_ratio" type="xsd:string"/> + <xsd:attribute name="display_aspect_ratio" type="xsd:string"/> + <xsd:attribute name="pix_fmt" type="xsd:string"/> + <xsd:attribute name="level" type="xsd:int"/> + <xsd:attribute name="color_range" type="xsd:string"/> + <xsd:attribute name="color_space" type="xsd:string"/> + <xsd:attribute name="color_transfer" type="xsd:string"/> + <xsd:attribute name="color_primaries" type="xsd:string"/> + <xsd:attribute name="chroma_location" type="xsd:string"/> + <xsd:attribute name="timecode" type="xsd:string"/> + <xsd:attribute name="refs" type="xsd:int"/> + + <!-- audio attributes --> + <xsd:attribute name="sample_fmt" type="xsd:string"/> + <xsd:attribute name="sample_rate" type="xsd:int"/> + <xsd:attribute name="channels" type="xsd:int"/> + <xsd:attribute name="channel_layout" type="xsd:string"/> + <xsd:attribute name="bits_per_sample" type="xsd:int"/> + + <xsd:attribute name="id" type="xsd:string"/> + <xsd:attribute name="r_frame_rate" type="xsd:string" use="required"/> + <xsd:attribute name="avg_frame_rate" type="xsd:string" use="required"/> + <xsd:attribute name="time_base" type="xsd:string" use="required"/> + <xsd:attribute name="start_pts" type="xsd:long"/> + <xsd:attribute name="start_time" type="xsd:float"/> + <xsd:attribute name="duration_ts" type="xsd:long"/> + <xsd:attribute name="duration" type="xsd:float"/> + <xsd:attribute name="bit_rate" type="xsd:int"/> + <xsd:attribute name="max_bit_rate" type="xsd:int"/> + <xsd:attribute name="bits_per_raw_sample" type="xsd:int"/> + <xsd:attribute name="nb_frames" type="xsd:int"/> + <xsd:attribute name="nb_read_frames" type="xsd:int"/> + <xsd:attribute name="nb_read_packets" type="xsd:int"/> + </xsd:complexType> + + <xsd:complexType name="programType"> + <xsd:sequence> + <xsd:element name="tag" type="ffprobe:tagType" minOccurs="0" maxOccurs="unbounded"/> + <xsd:element name="streams" type="ffprobe:streamsType" minOccurs="0" maxOccurs="1"/> + </xsd:sequence> + + <xsd:attribute name="program_id" type="xsd:int" use="required"/> + <xsd:attribute name="program_num" type="xsd:int" use="required"/> + <xsd:attribute name="nb_streams" type="xsd:int" use="required"/> + <xsd:attribute name="start_time" type="xsd:float"/> + <xsd:attribute name="start_pts" type="xsd:long"/> + <xsd:attribute name="end_time" type="xsd:float"/> + <xsd:attribute name="end_pts" type="xsd:long"/> + <xsd:attribute name="pmt_pid" type="xsd:int" use="required"/> + <xsd:attribute name="pcr_pid" type="xsd:int" use="required"/> + </xsd:complexType> + + <xsd:complexType name="formatType"> + <xsd:sequence> + <xsd:element name="tag" type="ffprobe:tagType" minOccurs="0" maxOccurs="unbounded"/> + </xsd:sequence> + + <xsd:attribute name="filename" type="xsd:string" use="required"/> + <xsd:attribute name="nb_streams" type="xsd:int" use="required"/> + <xsd:attribute name="nb_programs" type="xsd:int" use="required"/> + <xsd:attribute name="format_name" type="xsd:string" use="required"/> + <xsd:attribute name="format_long_name" type="xsd:string"/> + <xsd:attribute name="start_time" type="xsd:float"/> + <xsd:attribute name="duration" type="xsd:float"/> + <xsd:attribute name="size" type="xsd:long"/> + <xsd:attribute name="bit_rate" type="xsd:long"/> + <xsd:attribute name="probe_score" type="xsd:int"/> + </xsd:complexType> + + <xsd:complexType name="tagType"> + <xsd:attribute name="key" type="xsd:string" use="required"/> + <xsd:attribute name="value" type="xsd:string" use="required"/> + </xsd:complexType> + + <xsd:complexType name="errorType"> + <xsd:attribute name="code" type="xsd:int" use="required"/> + <xsd:attribute name="string" type="xsd:string" use="required"/> + </xsd:complexType> + + <xsd:complexType name="programVersionType"> + <xsd:attribute name="version" type="xsd:string" use="required"/> + <xsd:attribute name="copyright" type="xsd:string" use="required"/> + <xsd:attribute name="build_date" type="xsd:string"/> + <xsd:attribute name="build_time" type="xsd:string"/> + <xsd:attribute name="compiler_ident" type="xsd:string" use="required"/> + <xsd:attribute name="configuration" type="xsd:string" use="required"/> + </xsd:complexType> + + <xsd:complexType name="chaptersType"> + <xsd:sequence> + <xsd:element name="chapter" type="ffprobe:chapterType" minOccurs="0" maxOccurs="unbounded"/> + </xsd:sequence> + </xsd:complexType> + + <xsd:complexType name="chapterType"> + <xsd:sequence> + <xsd:element name="tag" type="ffprobe:tagType" minOccurs="0" maxOccurs="unbounded"/> + </xsd:sequence> + + <xsd:attribute name="id" type="xsd:int" use="required"/> + <xsd:attribute name="time_base" type="xsd:string" use="required"/> + <xsd:attribute name="start" type="xsd:int" use="required"/> + <xsd:attribute name="start_time" type="xsd:float"/> + <xsd:attribute name="end" type="xsd:int" use="required"/> + <xsd:attribute name="end_time" type="xsd:float" use="required"/> + </xsd:complexType> + + <xsd:complexType name="libraryVersionType"> + <xsd:attribute name="name" type="xsd:string" use="required"/> + <xsd:attribute name="major" type="xsd:int" use="required"/> + <xsd:attribute name="minor" type="xsd:int" use="required"/> + <xsd:attribute name="micro" type="xsd:int" use="required"/> + <xsd:attribute name="version" type="xsd:int" use="required"/> + <xsd:attribute name="ident" type="xsd:string" use="required"/> + </xsd:complexType> + + <xsd:complexType name="libraryVersionsType"> + <xsd:sequence> + <xsd:element name="library_version" type="ffprobe:libraryVersionType" minOccurs="0" maxOccurs="unbounded"/> + </xsd:sequence> + </xsd:complexType> + + <xsd:complexType name="pixelFormatFlagsType"> + <xsd:attribute name="big_endian" type="xsd:int" use="required"/> + <xsd:attribute name="palette" type="xsd:int" use="required"/> + <xsd:attribute name="bitstream" type="xsd:int" use="required"/> + <xsd:attribute name="hwaccel" type="xsd:int" use="required"/> + <xsd:attribute name="planar" type="xsd:int" use="required"/> + <xsd:attribute name="rgb" type="xsd:int" use="required"/> + <xsd:attribute name="pseudopal" type="xsd:int" use="required"/> + <xsd:attribute name="alpha" type="xsd:int" use="required"/> + </xsd:complexType> + + <xsd:complexType name="pixelFormatComponentType"> + <xsd:attribute name="index" type="xsd:int" use="required"/> + <xsd:attribute name="bit_depth" type="xsd:int" use="required"/> + </xsd:complexType> + + <xsd:complexType name="pixelFormatComponentsType"> + <xsd:sequence> + <xsd:element name="component" type="ffprobe:pixelFormatComponentType" minOccurs="0" maxOccurs="unbounded"/> + </xsd:sequence> + </xsd:complexType> + + <xsd:complexType name="pixelFormatType"> + <xsd:sequence> + <xsd:element name="flags" type="ffprobe:pixelFormatFlagsType" minOccurs="0" maxOccurs="1"/> + <xsd:element name="components" type="ffprobe:pixelFormatComponentsType" minOccurs="0" maxOccurs="1"/> + </xsd:sequence> + + <xsd:attribute name="name" type="xsd:string" use="required"/> + <xsd:attribute name="nb_components" type="xsd:int" use="required"/> + <xsd:attribute name="log2_chroma_w" type="xsd:int"/> + <xsd:attribute name="log2_chroma_h" type="xsd:int"/> + <xsd:attribute name="bits_per_pixel" type="xsd:int"/> + </xsd:complexType> + + <xsd:complexType name="pixelFormatsType"> + <xsd:sequence> + <xsd:element name="pixel_format" type="ffprobe:pixelFormatType" minOccurs="0" maxOccurs="unbounded"/> + </xsd:sequence> + </xsd:complexType> +</xsd:schema> diff --git a/doc/ffserver.conf b/doc/ffserver.conf new file mode 100644 index 0000000000..7a30fb6c3b --- /dev/null +++ b/doc/ffserver.conf @@ -0,0 +1,372 @@ +# Port on which the server is listening. You must select a different +# port from your standard HTTP web server if it is running on the same +# computer. +HTTPPort 8090 + +# Address on which the server is bound. Only useful if you have +# several network interfaces. +HTTPBindAddress 0.0.0.0 + +# Number of simultaneous HTTP connections that can be handled. It has +# to be defined *before* the MaxClients parameter, since it defines the +# MaxClients maximum limit. +MaxHTTPConnections 2000 + +# Number of simultaneous requests that can be handled. Since FFServer +# is very fast, it is more likely that you will want to leave this high +# and use MaxBandwidth, below. +MaxClients 1000 + +# This the maximum amount of kbit/sec that you are prepared to +# consume when streaming to clients. +MaxBandwidth 1000 + +# Access log file (uses standard Apache log file format) +# '-' is the standard output. +CustomLog - + +################################################################## +# Definition of the live feeds. Each live feed contains one video +# and/or audio sequence coming from an ffmpeg encoder or another +# ffserver. This sequence may be encoded simultaneously with several +# codecs at several resolutions. + +<Feed feed1.ffm> + +# You must use 'ffmpeg' to send a live feed to ffserver. In this +# example, you can type: +# +# ffmpeg http://localhost:8090/feed1.ffm + +# ffserver can also do time shifting. It means that it can stream any +# previously recorded live stream. The request should contain: +# "http://xxxx?date=[YYYY-MM-DDT][[HH:]MM:]SS[.m...]".You must specify +# a path where the feed is stored on disk. You also specify the +# maximum size of the feed, where zero means unlimited. Default: +# File=/tmp/feed_name.ffm FileMaxSize=5M +File /tmp/feed1.ffm +FileMaxSize 200K + +# You could specify +# ReadOnlyFile /saved/specialvideo.ffm +# This marks the file as readonly and it will not be deleted or updated. + +# Specify launch in order to start ffmpeg automatically. +# First ffmpeg must be defined with an appropriate path if needed, +# after that options can follow, but avoid adding the http:// field +#Launch ffmpeg + +# Only allow connections from localhost to the feed. +ACL allow 127.0.0.1 + +</Feed> + + +################################################################## +# Now you can define each stream which will be generated from the +# original audio and video stream. Each format has a filename (here +# 'test1.mpg'). FFServer will send this stream when answering a +# request containing this filename. + +<Stream test1.mpg> + +# coming from live feed 'feed1' +Feed feed1.ffm + +# Format of the stream : you can choose among: +# mpeg : MPEG-1 multiplexed video and audio +# mpegvideo : only MPEG-1 video +# mp2 : MPEG-2 audio (use AudioCodec to select layer 2 and 3 codec) +# ogg : Ogg format (Vorbis audio codec) +# rm : RealNetworks-compatible stream. Multiplexed audio and video. +# ra : RealNetworks-compatible stream. Audio only. +# mpjpeg : Multipart JPEG (works with Netscape without any plugin) +# jpeg : Generate a single JPEG image. +# mjpeg : Generate a M-JPEG stream. +# asf : ASF compatible streaming (Windows Media Player format). +# swf : Macromedia Flash compatible stream +# avi : AVI format (MPEG-4 video, MPEG audio sound) +Format mpeg + +# Bitrate for the audio stream. Codecs usually support only a few +# different bitrates. +AudioBitRate 32 + +# Number of audio channels: 1 = mono, 2 = stereo +AudioChannels 1 + +# Sampling frequency for audio. When using low bitrates, you should +# lower this frequency to 22050 or 11025. The supported frequencies +# depend on the selected audio codec. +AudioSampleRate 44100 + +# Bitrate for the video stream +VideoBitRate 64 + +# Ratecontrol buffer size +VideoBufferSize 40 + +# Number of frames per second +VideoFrameRate 3 + +# Size of the video frame: WxH (default: 160x128) +# The following abbreviations are defined: sqcif, qcif, cif, 4cif, qqvga, +# qvga, vga, svga, xga, uxga, qxga, sxga, qsxga, hsxga, wvga, wxga, wsxga, +# wuxga, woxga, wqsxga, wquxga, whsxga, whuxga, cga, ega, hd480, hd720, +# hd1080 +VideoSize 160x128 + +# Transmit only intra frames (useful for low bitrates, but kills frame rate). +#VideoIntraOnly + +# If non-intra only, an intra frame is transmitted every VideoGopSize +# frames. Video synchronization can only begin at an intra frame. +VideoGopSize 12 + +# More MPEG-4 parameters +# VideoHighQuality +# Video4MotionVector + +# Choose your codecs: +#AudioCodec mp2 +#VideoCodec mpeg1video + +# Suppress audio +#NoAudio + +# Suppress video +#NoVideo + +#VideoQMin 3 +#VideoQMax 31 + +# Set this to the number of seconds backwards in time to start. Note that +# most players will buffer 5-10 seconds of video, and also you need to allow +# for a keyframe to appear in the data stream. +#Preroll 15 + +# ACL: + +# You can allow ranges of addresses (or single addresses) +#ACL ALLOW <first address> <last address> + +# You can deny ranges of addresses (or single addresses) +#ACL DENY <first address> <last address> + +# You can repeat the ACL allow/deny as often as you like. It is on a per +# stream basis. The first match defines the action. If there are no matches, +# then the default is the inverse of the last ACL statement. +# +# Thus 'ACL allow localhost' only allows access from localhost. +# 'ACL deny 1.0.0.0 1.255.255.255' would deny the whole of network 1 and +# allow everybody else. + +</Stream> + + +################################################################## +# Example streams + + +# Multipart JPEG + +#<Stream test.mjpg> +#Feed feed1.ffm +#Format mpjpeg +#VideoFrameRate 2 +#VideoIntraOnly +#NoAudio +#Strict -1 +#</Stream> + + +# Single JPEG + +#<Stream test.jpg> +#Feed feed1.ffm +#Format jpeg +#VideoFrameRate 2 +#VideoIntraOnly +##VideoSize 352x240 +#NoAudio +#Strict -1 +#</Stream> + + +# Flash + +#<Stream test.swf> +#Feed feed1.ffm +#Format swf +#VideoFrameRate 2 +#VideoIntraOnly +#NoAudio +#</Stream> + + +# ASF compatible + +<Stream test.asf> +Feed feed1.ffm +Format asf +VideoFrameRate 15 +VideoSize 352x240 +VideoBitRate 256 +VideoBufferSize 40 +VideoGopSize 30 +AudioBitRate 64 +StartSendOnKey +</Stream> + + +# MP3 audio + +#<Stream test.mp3> +#Feed feed1.ffm +#Format mp2 +#AudioCodec mp3 +#AudioBitRate 64 +#AudioChannels 1 +#AudioSampleRate 44100 +#NoVideo +#</Stream> + + +# Ogg Vorbis audio + +#<Stream test.ogg> +#Feed feed1.ffm +#Metadata title "Stream title" +#AudioBitRate 64 +#AudioChannels 2 +#AudioSampleRate 44100 +#NoVideo +#</Stream> + + +# Real with audio only at 32 kbits + +#<Stream test.ra> +#Feed feed1.ffm +#Format rm +#AudioBitRate 32 +#NoVideo +#NoAudio +#</Stream> + + +# Real with audio and video at 64 kbits + +#<Stream test.rm> +#Feed feed1.ffm +#Format rm +#AudioBitRate 32 +#VideoBitRate 128 +#VideoFrameRate 25 +#VideoGopSize 25 +#NoAudio +#</Stream> + + +################################################################## +# A stream coming from a file: you only need to set the input +# filename and optionally a new format. Supported conversions: +# AVI -> ASF + +#<Stream file.rm> +#File "/usr/local/httpd/htdocs/tlive.rm" +#NoAudio +#</Stream> + +#<Stream file.asf> +#File "/usr/local/httpd/htdocs/test.asf" +#NoAudio +#Metadata author "Me" +#Metadata copyright "Super MegaCorp" +#Metadata title "Test stream from disk" +#Metadata comment "Test comment" +#</Stream> + + +################################################################## +# RTSP examples +# +# You can access this stream with the RTSP URL: +# rtsp://localhost:5454/test1-rtsp.mpg +# +# A non-standard RTSP redirector is also created. Its URL is: +# http://localhost:8090/test1-rtsp.rtsp + +#<Stream test1-rtsp.mpg> +#Format rtp +#File "/usr/local/httpd/htdocs/test1.mpg" +#</Stream> + + +# Transcode an incoming live feed to another live feed, +# using libx264 and video presets + +#<Stream live.h264> +#Format rtp +#Feed feed1.ffm +#VideoCodec libx264 +#VideoFrameRate 24 +#VideoBitRate 100 +#VideoSize 480x272 +#AVPresetVideo default +#AVPresetVideo baseline +#AVOptionVideo flags +global_header +# +#AudioCodec libfaac +#AudioBitRate 32 +#AudioChannels 2 +#AudioSampleRate 22050 +#AVOptionAudio flags +global_header +#</Stream> + +################################################################## +# SDP/multicast examples +# +# If you want to send your stream in multicast, you must set the +# multicast address with MulticastAddress. The port and the TTL can +# also be set. +# +# An SDP file is automatically generated by ffserver by adding the +# 'sdp' extension to the stream name (here +# http://localhost:8090/test1-sdp.sdp). You should usually give this +# file to your player to play the stream. +# +# The 'NoLoop' option can be used to avoid looping when the stream is +# terminated. + +#<Stream test1-sdp.mpg> +#Format rtp +#File "/usr/local/httpd/htdocs/test1.mpg" +#MulticastAddress 224.124.0.1 +#MulticastPort 5000 +#MulticastTTL 16 +#NoLoop +#</Stream> + + +################################################################## +# Special streams + +# Server status + +<Stream stat.html> +Format status + +# Only allow local people to get the status +ACL allow localhost +ACL allow 192.168.0.0 192.168.255.255 + +#FaviconURL http://pond1.gladstonefamily.net:8080/favicon.ico +</Stream> + + +# Redirect index.html to the appropriate site + +<Redirect index.html> +URL http://www.ffmpeg.org/ +</Redirect> diff --git a/doc/ffserver.texi b/doc/ffserver.texi new file mode 100644 index 0000000000..ad48f47a8f --- /dev/null +++ b/doc/ffserver.texi @@ -0,0 +1,923 @@ +\input texinfo @c -*- texinfo -*- +@documentencoding UTF-8 + +@settitle ffserver Documentation +@titlepage +@center @titlefont{ffserver Documentation} +@end titlepage + +@top + +@contents + +@chapter Synopsis + +ffserver [@var{options}] + +@chapter Description +@c man begin DESCRIPTION + +@command{ffserver} is a streaming server for both audio and video. +It supports several live feeds, streaming from files and time shifting +on live feeds. You can seek to positions in the past on each live +feed, provided you specify a big enough feed storage. + +@command{ffserver} is configured through a configuration file, which +is read at startup. If not explicitly specified, it will read from +@file{/etc/ffserver.conf}. + +@command{ffserver} receives prerecorded files or FFM streams from some +@command{ffmpeg} instance as input, then streams them over +RTP/RTSP/HTTP. + +An @command{ffserver} instance will listen on some port as specified +in the configuration file. You can launch one or more instances of +@command{ffmpeg} and send one or more FFM streams to the port where +ffserver is expecting to receive them. Alternately, you can make +@command{ffserver} launch such @command{ffmpeg} instances at startup. + +Input streams are called feeds, and each one is specified by a +@code{<Feed>} section in the configuration file. + +For each feed you can have different output streams in various +formats, each one specified by a @code{<Stream>} section in the +configuration file. + +@chapter Detailed description + +@command{ffserver} works by forwarding streams encoded by +@command{ffmpeg}, or pre-recorded streams which are read from disk. + +Precisely, @command{ffserver} acts as an HTTP server, accepting POST +requests from @command{ffmpeg} to acquire the stream to publish, and +serving RTSP clients or HTTP clients GET requests with the stream +media content. + +A feed is an @ref{FFM} stream created by @command{ffmpeg}, and sent to +a port where @command{ffserver} is listening. + +Each feed is identified by a unique name, corresponding to the name +of the resource published on @command{ffserver}, and is configured by +a dedicated @code{Feed} section in the configuration file. + +The feed publish URL is given by: +@example +http://@var{ffserver_ip_address}:@var{http_port}/@var{feed_name} +@end example + +where @var{ffserver_ip_address} is the IP address of the machine where +@command{ffserver} is installed, @var{http_port} is the port number of +the HTTP server (configured through the @option{HTTPPort} option), and +@var{feed_name} is the name of the corresponding feed defined in the +configuration file. + +Each feed is associated to a file which is stored on disk. This stored +file is used to send pre-recorded data to a player as fast as +possible when new content is added in real-time to the stream. + +A "live-stream" or "stream" is a resource published by +@command{ffserver}, and made accessible through the HTTP protocol to +clients. + +A stream can be connected to a feed, or to a file. In the first case, +the published stream is forwarded from the corresponding feed +generated by a running instance of @command{ffmpeg}, in the second +case the stream is read from a pre-recorded file. + +Each stream is identified by a unique name, corresponding to the name +of the resource served by @command{ffserver}, and is configured by +a dedicated @code{Stream} section in the configuration file. + +The stream access HTTP URL is given by: +@example +http://@var{ffserver_ip_address}:@var{http_port}/@var{stream_name}[@var{options}] +@end example + +The stream access RTSP URL is given by: +@example +http://@var{ffserver_ip_address}:@var{rtsp_port}/@var{stream_name}[@var{options}] +@end example + +@var{stream_name} is the name of the corresponding stream defined in +the configuration file. @var{options} is a list of options specified +after the URL which affects how the stream is served by +@command{ffserver}. @var{http_port} and @var{rtsp_port} are the HTTP +and RTSP ports configured with the options @var{HTTPPort} and +@var{RTSPPort} respectively. + +In case the stream is associated to a feed, the encoding parameters +must be configured in the stream configuration. They are sent to +@command{ffmpeg} when setting up the encoding. This allows +@command{ffserver} to define the encoding parameters used by +the @command{ffmpeg} encoders. + +The @command{ffmpeg} @option{override_ffserver} commandline option +allows one to override the encoding parameters set by the server. + +Multiple streams can be connected to the same feed. + +For example, you can have a situation described by the following +graph: + +@verbatim + _________ __________ + | | | | +ffmpeg 1 -----| feed 1 |-----| stream 1 | + \ |_________|\ |__________| + \ \ + \ \ __________ + \ \ | | + \ \| stream 2 | + \ |__________| + \ + \ _________ __________ + \ | | | | + \| feed 2 |-----| stream 3 | + |_________| |__________| + + _________ __________ + | | | | +ffmpeg 2 -----| feed 3 |-----| stream 4 | + |_________| |__________| + + _________ __________ + | | | | + | file 1 |-----| stream 5 | + |_________| |__________| + +@end verbatim + +@anchor{FFM} +@section FFM, FFM2 formats + +FFM and FFM2 are formats used by ffserver. They allow storing a wide variety of +video and audio streams and encoding options, and can store a moving time segment +of an infinite movie or a whole movie. + +FFM is version specific, and there is limited compatibility of FFM files +generated by one version of ffmpeg/ffserver and another version of +ffmpeg/ffserver. It may work but it is not guaranteed to work. + +FFM2 is extensible while maintaining compatibility and should work between +differing versions of tools. FFM2 is the default. + +@section Status stream + +@command{ffserver} supports an HTTP interface which exposes the +current status of the server. + +Simply point your browser to the address of the special status stream +specified in the configuration file. + +For example if you have: +@example +<Stream status.html> +Format status + +# Only allow local people to get the status +ACL allow localhost +ACL allow 192.168.0.0 192.168.255.255 +</Stream> +@end example + +then the server will post a page with the status information when +the special stream @file{status.html} is requested. + +@section How do I make it work? + +As a simple test, just run the following two command lines where INPUTFILE +is some file which you can decode with ffmpeg: + +@example +ffserver -f doc/ffserver.conf & +ffmpeg -i INPUTFILE http://localhost:8090/feed1.ffm +@end example + +At this point you should be able to go to your Windows machine and fire up +Windows Media Player (WMP). Go to Open URL and enter + +@example + http://<linuxbox>:8090/test.asf +@end example + +You should (after a short delay) see video and hear audio. + +WARNING: trying to stream test1.mpg doesn't work with WMP as it tries to +transfer the entire file before starting to play. +The same is true of AVI files. + +You should edit the @file{ffserver.conf} file to suit your needs (in +terms of frame rates etc). Then install @command{ffserver} and +@command{ffmpeg}, write a script to start them up, and off you go. + +@section What else can it do? + +You can replay video from .ffm files that was recorded earlier. +However, there are a number of caveats, including the fact that the +ffserver parameters must match the original parameters used to record the +file. If they do not, then ffserver deletes the file before recording into it. +(Now that I write this, it seems broken). + +You can fiddle with many of the codec choices and encoding parameters, and +there are a bunch more parameters that you cannot control. Post a message +to the mailing list if there are some 'must have' parameters. Look in +ffserver.conf for a list of the currently available controls. + +It will automatically generate the ASX or RAM files that are often used +in browsers. These files are actually redirections to the underlying ASF +or RM file. The reason for this is that the browser often fetches the +entire file before starting up the external viewer. The redirection files +are very small and can be transferred quickly. [The stream itself is +often 'infinite' and thus the browser tries to download it and never +finishes.] + +@section Tips + +* When you connect to a live stream, most players (WMP, RA, etc) want to +buffer a certain number of seconds of material so that they can display the +signal continuously. However, ffserver (by default) starts sending data +in realtime. This means that there is a pause of a few seconds while the +buffering is being done by the player. The good news is that this can be +cured by adding a '?buffer=5' to the end of the URL. This means that the +stream should start 5 seconds in the past -- and so the first 5 seconds +of the stream are sent as fast as the network will allow. It will then +slow down to real time. This noticeably improves the startup experience. + +You can also add a 'Preroll 15' statement into the ffserver.conf that will +add the 15 second prebuffering on all requests that do not otherwise +specify a time. In addition, ffserver will skip frames until a key_frame +is found. This further reduces the startup delay by not transferring data +that will be discarded. + +@section Why does the ?buffer / Preroll stop working after a time? + +It turns out that (on my machine at least) the number of frames successfully +grabbed is marginally less than the number that ought to be grabbed. This +means that the timestamp in the encoded data stream gets behind realtime. +This means that if you say 'Preroll 10', then when the stream gets 10 +or more seconds behind, there is no Preroll left. + +Fixing this requires a change in the internals of how timestamps are +handled. + +@section Does the @code{?date=} stuff work. + +Yes (subject to the limitation outlined above). Also note that whenever you +start ffserver, it deletes the ffm file (if any parameters have changed), +thus wiping out what you had recorded before. + +The format of the @code{?date=xxxxxx} is fairly flexible. You should use one +of the following formats (the 'T' is literal): + +@example +* YYYY-MM-DDTHH:MM:SS (localtime) +* YYYY-MM-DDTHH:MM:SSZ (UTC) +@end example + +You can omit the YYYY-MM-DD, and then it refers to the current day. However +note that @samp{?date=16:00:00} refers to 16:00 on the current day -- this +may be in the future and so is unlikely to be useful. + +You use this by adding the ?date= to the end of the URL for the stream. +For example: @samp{http://localhost:8080/test.asf?date=2002-07-26T23:05:00}. +@c man end + +@chapter Options +@c man begin OPTIONS + +@include fftools-common-opts.texi + +@section Main options + +@table @option +@item -f @var{configfile} +Read configuration file @file{configfile}. If not specified it will +read by default from @file{/etc/ffserver.conf}. + +@item -n +Enable no-launch mode. This option disables all the @code{Launch} +directives within the various @code{<Feed>} sections. Since +@command{ffserver} will not launch any @command{ffmpeg} instances, you +will have to launch them manually. + +@item -d +Enable debug mode. This option increases log verbosity, and directs +log messages to stdout. When specified, the @option{CustomLog} option +is ignored. +@end table + +@chapter Configuration file syntax + +@command{ffserver} reads a configuration file containing global +options and settings for each stream and feed. + +The configuration file consists of global options and dedicated +sections, which must be introduced by "<@var{SECTION_NAME} +@var{ARGS}>" on a separate line and must be terminated by a line in +the form "</@var{SECTION_NAME}>". @var{ARGS} is optional. + +Currently the following sections are recognized: @samp{Feed}, +@samp{Stream}, @samp{Redirect}. + +A line starting with @code{#} is ignored and treated as a comment. + +Name of options and sections are case-insensitive. + +@section ACL syntax +An ACL (Access Control List) specifies the address which are allowed +to access a given stream, or to write a given feed. + +It accepts the folling forms +@itemize +@item +Allow/deny access to @var{address}. +@example +ACL ALLOW <address> +ACL DENY <address> +@end example + +@item +Allow/deny access to ranges of addresses from @var{first_address} to +@var{last_address}. +@example +ACL ALLOW <first_address> <last_address> +ACL DENY <first_address> <last_address> +@end example +@end itemize + +You can repeat the ACL allow/deny as often as you like. It is on a per +stream basis. The first match defines the action. If there are no matches, +then the default is the inverse of the last ACL statement. + +Thus 'ACL allow localhost' only allows access from localhost. +'ACL deny 1.0.0.0 1.255.255.255' would deny the whole of network 1 and +allow everybody else. + +@section Global options +@table @option +@item HTTPPort @var{port_number} +@item Port @var{port_number} +@item RTSPPort @var{port_number} + +@var{HTTPPort} sets the HTTP server listening TCP port number, +@var{RTSPPort} sets the RTSP server listening TCP port number. + +@var{Port} is the equivalent of @var{HTTPPort} and is deprecated. + +You must select a different port from your standard HTTP web server if +it is running on the same computer. + +If not specified, no corresponding server will be created. + +@item HTTPBindAddress @var{ip_address} +@item BindAddress @var{ip_address} +@item RTSPBindAddress @var{ip_address} +Set address on which the HTTP/RTSP server is bound. Only useful if you +have several network interfaces. + +@var{BindAddress} is the equivalent of @var{HTTPBindAddress} and is +deprecated. + +@item MaxHTTPConnections @var{n} +Set number of simultaneous HTTP connections that can be handled. It +has to be defined @emph{before} the @option{MaxClients} parameter, +since it defines the @option{MaxClients} maximum limit. + +Default value is 2000. + +@item MaxClients @var{n} +Set number of simultaneous requests that can be handled. Since +@command{ffserver} is very fast, it is more likely that you will want +to leave this high and use @option{MaxBandwidth}. + +Default value is 5. + +@item MaxBandwidth @var{kbps} +Set the maximum amount of kbit/sec that you are prepared to consume +when streaming to clients. + +Default value is 1000. + +@item CustomLog @var{filename} +Set access log file (uses standard Apache log file format). '-' is the +standard output. + +If not specified @command{ffserver} will produce no log. + +In case the commandline option @option{-d} is specified this option is +ignored, and the log is written to standard output. + +@item NoDaemon +Set no-daemon mode. This option is currently ignored since now +@command{ffserver} will always work in no-daemon mode, and is +deprecated. + +@item UseDefaults +@item NoDefaults +Control whether default codec options are used for the all streams or not. +Each stream may overwrite this setting for its own. Default is @var{UseDefaults}. +The lastest occurrence overrides previous if multiple definitions. +@end table + +@section Feed section + +A Feed section defines a feed provided to @command{ffserver}. + +Each live feed contains one video and/or audio sequence coming from an +@command{ffmpeg} encoder or another @command{ffserver}. This sequence +may be encoded simultaneously with several codecs at several +resolutions. + +A feed instance specification is introduced by a line in the form: +@example +<Feed FEED_FILENAME> +@end example + +where @var{FEED_FILENAME} specifies the unique name of the FFM stream. + +The following options are recognized within a Feed section. + +@table @option +@item File @var{filename} +@item ReadOnlyFile @var{filename} +Set the path where the feed file is stored on disk. + +If not specified, the @file{/tmp/FEED.ffm} is assumed, where +@var{FEED} is the feed name. + +If @option{ReadOnlyFile} is used the file is marked as read-only and +it will not be deleted or updated. + +@item Truncate +Truncate the feed file, rather than appending to it. By default +@command{ffserver} will append data to the file, until the maximum +file size value is reached (see @option{FileMaxSize} option). + +@item FileMaxSize @var{size} +Set maximum size of the feed file in bytes. 0 means unlimited. The +postfixes @code{K} (2^10), @code{M} (2^20), and @code{G} (2^30) are +recognized. + +Default value is 5M. + +@item Launch @var{args} +Launch an @command{ffmpeg} command when creating @command{ffserver}. + +@var{args} must be a sequence of arguments to be provided to an +@command{ffmpeg} instance. The first provided argument is ignored, and +it is replaced by a path with the same dirname of the @command{ffserver} +instance, followed by the remaining argument and terminated with a +path corresponding to the feed. + +When the launched process exits, @command{ffserver} will launch +another program instance. + +In case you need a more complex @command{ffmpeg} configuration, +e.g. if you need to generate multiple FFM feeds with a single +@command{ffmpeg} instance, you should launch @command{ffmpeg} by hand. + +This option is ignored in case the commandline option @option{-n} is +specified. + +@item ACL @var{spec} +Specify the list of IP address which are allowed or denied to write +the feed. Multiple ACL options can be specified. +@end table + +@section Stream section + +A Stream section defines a stream provided by @command{ffserver}, and +identified by a single name. + +The stream is sent when answering a request containing the stream +name. + +A stream section must be introduced by the line: +@example +<Stream STREAM_NAME> +@end example + +where @var{STREAM_NAME} specifies the unique name of the stream. + +The following options are recognized within a Stream section. + +Encoding options are marked with the @emph{encoding} tag, and they are +used to set the encoding parameters, and are mapped to libavcodec +encoding options. Not all encoding options are supported, in +particular it is not possible to set encoder private options. In order +to override the encoding options specified by @command{ffserver}, you +can use the @command{ffmpeg} @option{override_ffserver} commandline +option. + +Only one of the @option{Feed} and @option{File} options should be set. + +@table @option +@item Feed @var{feed_name} +Set the input feed. @var{feed_name} must correspond to an existing +feed defined in a @code{Feed} section. + +When this option is set, encoding options are used to setup the +encoding operated by the remote @command{ffmpeg} process. + +@item File @var{filename} +Set the filename of the pre-recorded input file to stream. + +When this option is set, encoding options are ignored and the input +file content is re-streamed as is. + +@item Format @var{format_name} +Set the format of the output stream. + +Must be the name of a format recognized by FFmpeg. If set to +@samp{status}, it is treated as a status stream. + +@item InputFormat @var{format_name} +Set input format. If not specified, it is automatically guessed. + +@item Preroll @var{n} +Set this to the number of seconds backwards in time to start. Note that +most players will buffer 5-10 seconds of video, and also you need to allow +for a keyframe to appear in the data stream. + +Default value is 0. + +@item StartSendOnKey +Do not send stream until it gets the first key frame. By default +@command{ffserver} will send data immediately. + +@item MaxTime @var{n} +Set the number of seconds to run. This value set the maximum duration +of the stream a client will be able to receive. + +A value of 0 means that no limit is set on the stream duration. + +@item ACL @var{spec} +Set ACL for the stream. + +@item DynamicACL @var{spec} + +@item RTSPOption @var{option} + +@item MulticastAddress @var{address} + +@item MulticastPort @var{port} + +@item MulticastTTL @var{integer} + +@item NoLoop + +@item FaviconURL @var{url} +Set favicon (favourite icon) for the server status page. It is ignored +for regular streams. + +@item Author @var{value} +@item Comment @var{value} +@item Copyright @var{value} +@item Title @var{value} +Set metadata corresponding to the option. All these options are +deprecated in favor of @option{Metadata}. + +@item Metadata @var{key} @var{value} +Set metadata value on the output stream. + +@item UseDefaults +@item NoDefaults +Control whether default codec options are used for the stream or not. +Default is @var{UseDefaults} unless disabled globally. + +@item NoAudio +@item NoVideo +Suppress audio/video. + +@item AudioCodec @var{codec_name} (@emph{encoding,audio}) +Set audio codec. + +@item AudioBitRate @var{rate} (@emph{encoding,audio}) +Set bitrate for the audio stream in kbits per second. + +@item AudioChannels @var{n} (@emph{encoding,audio}) +Set number of audio channels. + +@item AudioSampleRate @var{n} (@emph{encoding,audio}) +Set sampling frequency for audio. When using low bitrates, you should +lower this frequency to 22050 or 11025. The supported frequencies +depend on the selected audio codec. + +@item AVOptionAudio [@var{codec}:]@var{option} @var{value} (@emph{encoding,audio}) +Set generic or private option for audio stream. +Private option must be prefixed with codec name or codec must be defined before. + +@item AVPresetAudio @var{preset} (@emph{encoding,audio}) +Set preset for audio stream. + +@item VideoCodec @var{codec_name} (@emph{encoding,video}) +Set video codec. + +@item VideoBitRate @var{n} (@emph{encoding,video}) +Set bitrate for the video stream in kbits per second. + +@item VideoBitRateRange @var{range} (@emph{encoding,video}) +Set video bitrate range. + +A range must be specified in the form @var{minrate}-@var{maxrate}, and +specifies the @option{minrate} and @option{maxrate} encoding options +expressed in kbits per second. + +@item VideoBitRateRangeTolerance @var{n} (@emph{encoding,video}) +Set video bitrate tolerance in kbits per second. + +@item PixelFormat @var{pixel_format} (@emph{encoding,video}) +Set video pixel format. + +@item Debug @var{integer} (@emph{encoding,video}) +Set video @option{debug} encoding option. + +@item Strict @var{integer} (@emph{encoding,video}) +Set video @option{strict} encoding option. + +@item VideoBufferSize @var{n} (@emph{encoding,video}) +Set ratecontrol buffer size, expressed in KB. + +@item VideoFrameRate @var{n} (@emph{encoding,video}) +Set number of video frames per second. + +@item VideoSize (@emph{encoding,video}) +Set size of the video frame, must be an abbreviation or in the form +@var{W}x@var{H}. See @ref{video size syntax,,the Video size section +in the ffmpeg-utils(1) manual,ffmpeg-utils}. + +Default value is @code{160x128}. + +@item VideoIntraOnly (@emph{encoding,video}) +Transmit only intra frames (useful for low bitrates, but kills frame rate). + +@item VideoGopSize @var{n} (@emph{encoding,video}) +If non-intra only, an intra frame is transmitted every VideoGopSize +frames. Video synchronization can only begin at an intra frame. + +@item VideoTag @var{tag} (@emph{encoding,video}) +Set video tag. + +@item VideoHighQuality (@emph{encoding,video}) +@item Video4MotionVector (@emph{encoding,video}) + +@item BitExact (@emph{encoding,video}) +Set bitexact encoding flag. + +@item IdctSimple (@emph{encoding,video}) +Set simple IDCT algorithm. + +@item Qscale @var{n} (@emph{encoding,video}) +Enable constant quality encoding, and set video qscale (quantization +scale) value, expressed in @var{n} QP units. + +@item VideoQMin @var{n} (@emph{encoding,video}) +@item VideoQMax @var{n} (@emph{encoding,video}) +Set video qmin/qmax. + +@item VideoQDiff @var{integer} (@emph{encoding,video}) +Set video @option{qdiff} encoding option. + +@item LumiMask @var{float} (@emph{encoding,video}) +@item DarkMask @var{float} (@emph{encoding,video}) +Set @option{lumi_mask}/@option{dark_mask} encoding options. + +@item AVOptionVideo [@var{codec}:]@var{option} @var{value} (@emph{encoding,video}) +Set generic or private option for video stream. +Private option must be prefixed with codec name or codec must be defined before. + +@item AVPresetVideo @var{preset} (@emph{encoding,video}) +Set preset for video stream. + +@var{preset} must be the path of a preset file. +@end table + +@subsection Server status stream + +A server status stream is a special stream which is used to show +statistics about the @command{ffserver} operations. + +It must be specified setting the option @option{Format} to +@samp{status}. + +@section Redirect section + +A redirect section specifies where to redirect the requested URL to +another page. + +A redirect section must be introduced by the line: +@example +<Redirect NAME> +@end example + +where @var{NAME} is the name of the page which should be redirected. + +It only accepts the option @option{URL}, which specify the redirection +URL. + +@chapter Stream examples + +@itemize +@item +Multipart JPEG +@example +<Stream test.mjpg> +Feed feed1.ffm +Format mpjpeg +VideoFrameRate 2 +VideoIntraOnly +NoAudio +Strict -1 +</Stream> +@end example + +@item +Single JPEG +@example +<Stream test.jpg> +Feed feed1.ffm +Format jpeg +VideoFrameRate 2 +VideoIntraOnly +VideoSize 352x240 +NoAudio +Strict -1 +</Stream> +@end example + +@item +Flash +@example +<Stream test.swf> +Feed feed1.ffm +Format swf +VideoFrameRate 2 +VideoIntraOnly +NoAudio +</Stream> +@end example + +@item +ASF compatible +@example +<Stream test.asf> +Feed feed1.ffm +Format asf +VideoFrameRate 15 +VideoSize 352x240 +VideoBitRate 256 +VideoBufferSize 40 +VideoGopSize 30 +AudioBitRate 64 +StartSendOnKey +</Stream> +@end example + +@item +MP3 audio +@example +<Stream test.mp3> +Feed feed1.ffm +Format mp2 +AudioCodec mp3 +AudioBitRate 64 +AudioChannels 1 +AudioSampleRate 44100 +NoVideo +</Stream> +@end example + +@item +Ogg Vorbis audio +@example +<Stream test.ogg> +Feed feed1.ffm +Metadata title "Stream title" +AudioBitRate 64 +AudioChannels 2 +AudioSampleRate 44100 +NoVideo +</Stream> +@end example + +@item +Real with audio only at 32 kbits +@example +<Stream test.ra> +Feed feed1.ffm +Format rm +AudioBitRate 32 +NoVideo +</Stream> +@end example + +@item +Real with audio and video at 64 kbits +@example +<Stream test.rm> +Feed feed1.ffm +Format rm +AudioBitRate 32 +VideoBitRate 128 +VideoFrameRate 25 +VideoGopSize 25 +</Stream> +@end example + +@item +For stream coming from a file: you only need to set the input filename +and optionally a new format. + +@example +<Stream file.rm> +File "/usr/local/httpd/htdocs/tlive.rm" +NoAudio +</Stream> +@end example + +@example +<Stream file.asf> +File "/usr/local/httpd/htdocs/test.asf" +NoAudio +Metadata author "Me" +Metadata copyright "Super MegaCorp" +Metadata title "Test stream from disk" +Metadata comment "Test comment" +</Stream> +@end example +@end itemize + +@c man end + +@include config.texi +@ifset config-all +@ifset config-avutil +@include utils.texi +@end ifset +@ifset config-avcodec +@include codecs.texi +@include bitstream_filters.texi +@end ifset +@ifset config-avformat +@include formats.texi +@include protocols.texi +@end ifset +@ifset config-avdevice +@include devices.texi +@end ifset +@ifset config-swresample +@include resampler.texi +@end ifset +@ifset config-swscale +@include scaler.texi +@end ifset +@ifset config-avfilter +@include filters.texi +@end ifset +@end ifset + +@chapter See Also + +@ifhtml +@ifset config-all +@url{ffserver.html,ffserver}, +@end ifset +@ifset config-not-all +@url{ffserver-all.html,ffserver-all}, +@end ifset +the @file{doc/ffserver.conf} example, +@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, +@url{ffmpeg-utils.html,ffmpeg-utils}, +@url{ffmpeg-scaler.html,ffmpeg-scaler}, +@url{ffmpeg-resampler.html,ffmpeg-resampler}, +@url{ffmpeg-codecs.html,ffmpeg-codecs}, +@url{ffmpeg-bitstream-filters.html,ffmpeg-bitstream-filters}, +@url{ffmpeg-formats.html,ffmpeg-formats}, +@url{ffmpeg-devices.html,ffmpeg-devices}, +@url{ffmpeg-protocols.html,ffmpeg-protocols}, +@url{ffmpeg-filters.html,ffmpeg-filters} +@end ifhtml + +@ifnothtml +@ifset config-all +ffserver(1), +@end ifset +@ifset config-not-all +ffserver-all(1), +@end ifset +the @file{doc/ffserver.conf} example, ffmpeg(1), ffplay(1), ffprobe(1), +ffmpeg-utils(1), ffmpeg-scaler(1), ffmpeg-resampler(1), +ffmpeg-codecs(1), ffmpeg-bitstream-filters(1), ffmpeg-formats(1), +ffmpeg-devices(1), ffmpeg-protocols(1), ffmpeg-filters(1) +@end ifnothtml + +@include authors.texi + +@ignore + +@setfilename ffserver +@settitle ffserver video server + +@end ignore + +@bye diff --git a/doc/fftools-common-opts.texi b/doc/fftools-common-opts.texi new file mode 100644 index 0000000000..509c8bca7c --- /dev/null +++ b/doc/fftools-common-opts.texi @@ -0,0 +1,389 @@ +All the numerical options, if not specified otherwise, accept a string +representing a number as input, which may be followed by one of the SI +unit prefixes, for example: 'K', 'M', or 'G'. + +If 'i' is appended to the SI unit prefix, the complete prefix will be +interpreted as a unit prefix for binary multiples, which are based on +powers of 1024 instead of powers of 1000. Appending 'B' to the SI unit +prefix multiplies the value by 8. This allows using, for example: +'KB', 'MiB', 'G' and 'B' as number suffixes. + +Options which do not take arguments are boolean options, and set the +corresponding value to true. They can be set to false by prefixing +the option name with "no". For example using "-nofoo" +will set the boolean option with name "foo" to false. + +@anchor{Stream specifiers} +@section Stream specifiers +Some options are applied per-stream, e.g. bitrate or codec. Stream specifiers +are used to precisely specify which stream(s) a given option belongs to. + +A stream specifier is a string generally appended to the option name and +separated from it by a colon. E.g. @code{-codec:a:1 ac3} contains the +@code{a:1} stream specifier, which matches the second audio stream. Therefore, it +would select the ac3 codec for the second audio stream. + +A stream specifier can match several streams, so that the option is applied to all +of them. E.g. the stream specifier in @code{-b:a 128k} matches all audio +streams. + +An empty stream specifier matches all streams. For example, @code{-codec copy} +or @code{-codec: copy} would copy all the streams without reencoding. + +Possible forms of stream specifiers are: +@table @option +@item @var{stream_index} +Matches the stream with this index. E.g. @code{-threads:1 4} would set the +thread count for the second stream to 4. +@item @var{stream_type}[:@var{stream_index}] +@var{stream_type} is one of following: 'v' or 'V' for video, 'a' for audio, 's' +for subtitle, 'd' for data, and 't' for attachments. 'v' matches all video +streams, 'V' only matches video streams which are not attached pictures, video +thumbnails or cover arts. If @var{stream_index} is given, then it matches +stream number @var{stream_index} of this type. Otherwise, it matches all +streams of this type. +@item p:@var{program_id}[:@var{stream_index}] +If @var{stream_index} is given, then it matches the stream with number @var{stream_index} +in the program with the id @var{program_id}. Otherwise, it matches all streams in the +program. +@item #@var{stream_id} or i:@var{stream_id} +Match the stream by stream id (e.g. PID in MPEG-TS container). +@item m:@var{key}[:@var{value}] +Matches streams with the metadata tag @var{key} having the specified value. If +@var{value} is not given, matches streams that contain the given tag with any +value. +@item u +Matches streams with usable configuration, the codec must be defined and the +essential information such as video dimension or audio sample rate must be present. + +Note that in @command{ffmpeg}, matching by metadata will only work properly for +input files. +@end table + +@section Generic options + +These options are shared amongst the ff* tools. + +@table @option + +@item -L +Show license. + +@item -h, -?, -help, --help [@var{arg}] +Show help. An optional parameter may be specified to print help about a specific +item. If no argument is specified, only basic (non advanced) tool +options are shown. + +Possible values of @var{arg} are: +@table @option +@item long +Print advanced tool options in addition to the basic tool options. + +@item full +Print complete list of options, including shared and private options +for encoders, decoders, demuxers, muxers, filters, etc. + +@item decoder=@var{decoder_name} +Print detailed information about the decoder named @var{decoder_name}. Use the +@option{-decoders} option to get a list of all decoders. + +@item encoder=@var{encoder_name} +Print detailed information about the encoder named @var{encoder_name}. Use the +@option{-encoders} option to get a list of all encoders. + +@item demuxer=@var{demuxer_name} +Print detailed information about the demuxer named @var{demuxer_name}. Use the +@option{-formats} option to get a list of all demuxers and muxers. + +@item muxer=@var{muxer_name} +Print detailed information about the muxer named @var{muxer_name}. Use the +@option{-formats} option to get a list of all muxers and demuxers. + +@item filter=@var{filter_name} +Print detailed information about the filter name @var{filter_name}. Use the +@option{-filters} option to get a list of all filters. +@end table + +@item -version +Show version. + +@item -formats +Show available formats (including devices). + +@item -devices +Show available devices. + +@item -codecs +Show all codecs known to libavcodec. + +Note that the term 'codec' is used throughout this documentation as a shortcut +for what is more correctly called a media bitstream format. + +@item -decoders +Show available decoders. + +@item -encoders +Show all available encoders. + +@item -bsfs +Show available bitstream filters. + +@item -protocols +Show available protocols. + +@item -filters +Show available libavfilter filters. + +@item -pix_fmts +Show available pixel formats. + +@item -sample_fmts +Show available sample formats. + +@item -layouts +Show channel names and standard channel layouts. + +@item -colors +Show recognized color names. + +@item -sources @var{device}[,@var{opt1}=@var{val1}[,@var{opt2}=@var{val2}]...] +Show autodetected sources of the intput device. +Some devices may provide system-dependent source names that cannot be autodetected. +The returned list cannot be assumed to be always complete. +@example +ffmpeg -sources pulse,server=192.168.0.4 +@end example + +@item -sinks @var{device}[,@var{opt1}=@var{val1}[,@var{opt2}=@var{val2}]...] +Show autodetected sinks of the output device. +Some devices may provide system-dependent sink names that cannot be autodetected. +The returned list cannot be assumed to be always complete. +@example +ffmpeg -sinks pulse,server=192.168.0.4 +@end example + +@item -loglevel [repeat+]@var{loglevel} | -v [repeat+]@var{loglevel} +Set the logging level used by the library. +Adding "repeat+" indicates that repeated log output should not be compressed +to the first line and the "Last message repeated n times" line will be +omitted. "repeat" can also be used alone. +If "repeat" is used alone, and with no prior loglevel set, the default +loglevel will be used. If multiple loglevel parameters are given, using +'repeat' will not change the loglevel. +@var{loglevel} is a string or a number containing one of the following values: +@table @samp +@item quiet, -8 +Show nothing at all; be silent. +@item panic, 0 +Only show fatal errors which could lead the process to crash, such as +and assert failure. This is not currently used for anything. +@item fatal, 8 +Only show fatal errors. These are errors after which the process absolutely +cannot continue after. +@item error, 16 +Show all errors, including ones which can be recovered from. +@item warning, 24 +Show all warnings and errors. Any message related to possibly +incorrect or unexpected events will be shown. +@item info, 32 +Show informative messages during processing. This is in addition to +warnings and errors. This is the default value. +@item verbose, 40 +Same as @code{info}, except more verbose. +@item debug, 48 +Show everything, including debugging information. +@item trace, 56 +@end table + +By default the program logs to stderr, if coloring is supported by the +terminal, colors are used to mark errors and warnings. Log coloring +can be disabled setting the environment variable +@env{AV_LOG_FORCE_NOCOLOR} or @env{NO_COLOR}, or can be forced setting +the environment variable @env{AV_LOG_FORCE_COLOR}. +The use of the environment variable @env{NO_COLOR} is deprecated and +will be dropped in a following FFmpeg version. + +@item -report +Dump full command line and console output to a file named +@code{@var{program}-@var{YYYYMMDD}-@var{HHMMSS}.log} in the current +directory. +This file can be useful for bug reports. +It also implies @code{-loglevel verbose}. + +Setting the environment variable @env{FFREPORT} to any value has the +same effect. If the value is a ':'-separated key=value sequence, these +options will affect the report; option values must be escaped if they +contain special characters or the options delimiter ':' (see the +``Quoting and escaping'' section in the ffmpeg-utils manual). + +The following options are recognized: +@table @option +@item file +set the file name to use for the report; @code{%p} is expanded to the name +of the program, @code{%t} is expanded to a timestamp, @code{%%} is expanded +to a plain @code{%} +@item level +set the log verbosity level using a numerical value (see @code{-loglevel}). +@end table + +For example, to output a report to a file named @file{ffreport.log} +using a log level of @code{32} (alias for log level @code{info}): + +@example +FFREPORT=file=ffreport.log:level=32 ffmpeg -i input output +@end example + +Errors in parsing the environment variable are not fatal, and will not +appear in the report. + +@item -hide_banner +Suppress printing banner. + +All FFmpeg tools will normally show a copyright notice, build options +and library versions. This option can be used to suppress printing +this information. + +@item -cpuflags flags (@emph{global}) +Allows setting and clearing cpu flags. This option is intended +for testing. Do not use it unless you know what you're doing. +@example +ffmpeg -cpuflags -sse+mmx ... +ffmpeg -cpuflags mmx ... +ffmpeg -cpuflags 0 ... +@end example +Possible flags for this option are: +@table @samp +@item x86 +@table @samp +@item mmx +@item mmxext +@item sse +@item sse2 +@item sse2slow +@item sse3 +@item sse3slow +@item ssse3 +@item atom +@item sse4.1 +@item sse4.2 +@item avx +@item avx2 +@item xop +@item fma3 +@item fma4 +@item 3dnow +@item 3dnowext +@item bmi1 +@item bmi2 +@item cmov +@end table +@item ARM +@table @samp +@item armv5te +@item armv6 +@item armv6t2 +@item vfp +@item vfpv3 +@item neon +@item setend +@end table +@item AArch64 +@table @samp +@item armv8 +@item vfp +@item neon +@end table +@item PowerPC +@table @samp +@item altivec +@end table +@item Specific Processors +@table @samp +@item pentium2 +@item pentium3 +@item pentium4 +@item k6 +@item k62 +@item athlon +@item athlonxp +@item k8 +@end table +@end table + +@item -opencl_bench +This option is used to benchmark all available OpenCL devices and print the +results. This option is only available when FFmpeg has been compiled with +@code{--enable-opencl}. + +When FFmpeg is configured with @code{--enable-opencl}, the options for the +global OpenCL context are set via @option{-opencl_options}. See the +"OpenCL Options" section in the ffmpeg-utils manual for the complete list of +supported options. Amongst others, these options include the ability to select +a specific platform and device to run the OpenCL code on. By default, FFmpeg +will run on the first device of the first platform. While the options for the +global OpenCL context provide flexibility to the user in selecting the OpenCL +device of their choice, most users would probably want to select the fastest +OpenCL device for their system. + +This option assists the selection of the most efficient configuration by +identifying the appropriate device for the user's system. The built-in +benchmark is run on all the OpenCL devices and the performance is measured for +each device. The devices in the results list are sorted based on their +performance with the fastest device listed first. The user can subsequently +invoke @command{ffmpeg} using the device deemed most appropriate via +@option{-opencl_options} to obtain the best performance for the OpenCL +accelerated code. + +Typical usage to use the fastest OpenCL device involve the following steps. + +Run the command: +@example +ffmpeg -opencl_bench +@end example +Note down the platform ID (@var{pidx}) and device ID (@var{didx}) of the first +i.e. fastest device in the list. +Select the platform and device using the command: +@example +ffmpeg -opencl_options platform_idx=@var{pidx}:device_idx=@var{didx} ... +@end example + +@item -opencl_options options (@emph{global}) +Set OpenCL environment options. This option is only available when +FFmpeg has been compiled with @code{--enable-opencl}. + +@var{options} must be a list of @var{key}=@var{value} option pairs +separated by ':'. See the ``OpenCL Options'' section in the +ffmpeg-utils manual for the list of supported options. +@end table + +@section AVOptions + +These options are provided directly by the libavformat, libavdevice and +libavcodec libraries. To see the list of available AVOptions, use the +@option{-help} option. They are separated into two categories: +@table @option +@item generic +These options can be set for any container, codec or device. Generic options +are listed under AVFormatContext options for containers/devices and under +AVCodecContext options for codecs. +@item private +These options are specific to the given container, device or codec. Private +options are listed under their corresponding containers/devices/codecs. +@end table + +For example to write an ID3v2.3 header instead of a default ID3v2.4 to +an MP3 file, use the @option{id3v2_version} private option of the MP3 +muxer: +@example +ffmpeg -i input.flac -id3v2_version 3 out.mp3 +@end example + +All codec AVOptions are per-stream, and thus a stream specifier +should be attached to them. + +Note: the @option{-nooption} syntax cannot be used for boolean +AVOptions, use @option{-option 0}/@option{-option 1}. + +Note: the old undocumented way of specifying per-stream AVOptions by +prepending v/a/s to the options name is now obsolete and will be +removed soon. diff --git a/doc/filter_design.txt b/doc/filter_design.txt new file mode 100644 index 0000000000..e8a7c53ee9 --- /dev/null +++ b/doc/filter_design.txt @@ -0,0 +1,269 @@ +Filter design +============= + +This document explains guidelines that should be observed (or ignored with +good reason) when writing filters for libavfilter. + +In this document, the word “frame” indicates either a video frame or a group +of audio samples, as stored in an AVFilterBuffer structure. + + +Format negotiation +================== + + The query_formats method should set, for each input and each output links, + the list of supported formats. + + For video links, that means pixel format. For audio links, that means + channel layout, sample format (the sample packing is implied by the sample + format) and sample rate. + + The lists are not just lists, they are references to shared objects. When + the negotiation mechanism computes the intersection of the formats + supported at each end of a link, all references to both lists are replaced + with a reference to the intersection. And when a single format is + eventually chosen for a link amongst the remaining list, again, all + references to the list are updated. + + That means that if a filter requires that its input and output have the + same format amongst a supported list, all it has to do is use a reference + to the same list of formats. + + query_formats can leave some formats unset and return AVERROR(EAGAIN) to + cause the negotiation mechanism to try again later. That can be used by + filters with complex requirements to use the format negotiated on one link + to set the formats supported on another. + + +Buffer references ownership and permissions +=========================================== + + Principle + --------- + + Audio and video data are voluminous; the buffer and buffer reference + mechanism is intended to avoid, as much as possible, expensive copies of + that data while still allowing the filters to produce correct results. + + The data is stored in buffers represented by AVFilterBuffer structures. + They must not be accessed directly, but through references stored in + AVFilterBufferRef structures. Several references can point to the + same buffer; the buffer is automatically deallocated once all + corresponding references have been destroyed. + + The characteristics of the data (resolution, sample rate, etc.) are + stored in the reference; different references for the same buffer can + show different characteristics. In particular, a video reference can + point to only a part of a video buffer. + + A reference is usually obtained as input to the start_frame or + filter_frame method or requested using the ff_get_video_buffer or + ff_get_audio_buffer functions. A new reference on an existing buffer can + be created with the avfilter_ref_buffer. A reference is destroyed using + the avfilter_unref_bufferp function. + + Reference ownership + ------------------- + + At any time, a reference “belongs” to a particular piece of code, + usually a filter. With a few caveats that will be explained below, only + that piece of code is allowed to access it. It is also responsible for + destroying it, although this is sometimes done automatically (see the + section on link reference fields). + + Here are the (fairly obvious) rules for reference ownership: + + * A reference received by the filter_frame method (or its start_frame + deprecated version) belongs to the corresponding filter. + + Special exception: for video references: the reference may be used + internally for automatic copying and must not be destroyed before + end_frame; it can be given away to ff_start_frame. + + * A reference passed to ff_filter_frame (or the deprecated + ff_start_frame) is given away and must no longer be used. + + * A reference created with avfilter_ref_buffer belongs to the code that + created it. + + * A reference obtained with ff_get_video_buffer or ff_get_audio_buffer + belongs to the code that requested it. + + * A reference given as return value by the get_video_buffer or + get_audio_buffer method is given away and must no longer be used. + + Link reference fields + --------------------- + + The AVFilterLink structure has a few AVFilterBufferRef fields. The + cur_buf and out_buf were used with the deprecated + start_frame/draw_slice/end_frame API and should no longer be used. + src_buf and partial_buf are used by libavfilter internally + and must not be accessed by filters. + + Reference permissions + --------------------- + + The AVFilterBufferRef structure has a perms field that describes what + the code that owns the reference is allowed to do to the buffer data. + Different references for the same buffer can have different permissions. + + For video filters that implement the deprecated + start_frame/draw_slice/end_frame API, the permissions only apply to the + parts of the buffer that have already been covered by the draw_slice + method. + + The value is a binary OR of the following constants: + + * AV_PERM_READ: the owner can read the buffer data; this is essentially + always true and is there for self-documentation. + + * AV_PERM_WRITE: the owner can modify the buffer data. + + * AV_PERM_PRESERVE: the owner can rely on the fact that the buffer data + will not be modified by previous filters. + + * AV_PERM_REUSE: the owner can output the buffer several times, without + modifying the data in between. + + * AV_PERM_REUSE2: the owner can output the buffer several times and + modify the data in between (useless without the WRITE permissions). + + * AV_PERM_ALIGN: the owner can access the data using fast operations + that require data alignment. + + The READ, WRITE and PRESERVE permissions are about sharing the same + buffer between several filters to avoid expensive copies without them + doing conflicting changes on the data. + + The REUSE and REUSE2 permissions are about special memory for direct + rendering. For example a buffer directly allocated in video memory must + not modified once it is displayed on screen, or it will cause tearing; + it will therefore not have the REUSE2 permission. + + The ALIGN permission is about extracting part of the buffer, for + copy-less padding or cropping for example. + + + References received on input pads are guaranteed to have all the + permissions stated in the min_perms field and none of the permissions + stated in the rej_perms. + + References obtained by ff_get_video_buffer and ff_get_audio_buffer are + guaranteed to have at least all the permissions requested as argument. + + References created by avfilter_ref_buffer have the same permissions as + the original reference minus the ones explicitly masked; the mask is + usually ~0 to keep the same permissions. + + Filters should remove permissions on reference they give to output + whenever necessary. It can be automatically done by setting the + rej_perms field on the output pad. + + Here are a few guidelines corresponding to common situations: + + * Filters that modify and forward their frame (like drawtext) need the + WRITE permission. + + * Filters that read their input to produce a new frame on output (like + scale) need the READ permission on input and must request a buffer + with the WRITE permission. + + * Filters that intend to keep a reference after the filtering process + is finished (after filter_frame returns) must have the PRESERVE + permission on it and remove the WRITE permission if they create a new + reference to give it away. + + * Filters that intend to modify a reference they have kept after the end + of the filtering process need the REUSE2 permission and must remove + the PRESERVE permission if they create a new reference to give it + away. + + +Frame scheduling +================ + + The purpose of these rules is to ensure that frames flow in the filter + graph without getting stuck and accumulating somewhere. + + Simple filters that output one frame for each input frame should not have + to worry about it. + + filter_frame + ------------ + + This method is called when a frame is pushed to the filter's input. It + can be called at any time except in a reentrant way. + + If the input frame is enough to produce output, then the filter should + push the output frames on the output link immediately. + + As an exception to the previous rule, if the input frame is enough to + produce several output frames, then the filter needs output only at + least one per link. The additional frames can be left buffered in the + filter; these buffered frames must be flushed immediately if a new input + produces new output. + + (Example: frame rate-doubling filter: filter_frame must (1) flush the + second copy of the previous frame, if it is still there, (2) push the + first copy of the incoming frame, (3) keep the second copy for later.) + + If the input frame is not enough to produce output, the filter must not + call request_frame to get more. It must just process the frame or queue + it. The task of requesting more frames is left to the filter's + request_frame method or the application. + + If a filter has several inputs, the filter must be ready for frames + arriving randomly on any input. Therefore, any filter with several inputs + will most likely require some kind of queuing mechanism. It is perfectly + acceptable to have a limited queue and to drop frames when the inputs + are too unbalanced. + + request_frame + ------------- + + This method is called when a frame is wanted on an output. + + For an input, it should directly call filter_frame on the corresponding + output. + + For a filter, if there are queued frames already ready, one of these + frames should be pushed. If not, the filter should request a frame on + one of its inputs, repeatedly until at least one frame has been pushed. + + Return values: + if request_frame could produce a frame, or at least make progress + towards producing a frame, it should return 0; + if it could not for temporary reasons, it should return AVERROR(EAGAIN); + if it could not because there are no more frames, it should return + AVERROR_EOF. + + The typical implementation of request_frame for a filter with several + inputs will look like that: + + if (frames_queued) { + push_one_frame(); + return 0; + } + input = input_where_a_frame_is_most_needed(); + ret = ff_request_frame(input); + if (ret == AVERROR_EOF) { + process_eof_on_input(); + } else if (ret < 0) { + return ret; + } + return 0; + + Note that, except for filters that can have queued frames, request_frame + does not push frames: it requests them to its input, and as a reaction, + the filter_frame method possibly will be called and do the work. + +Legacy API +========== + + Until libavfilter 3.23, the filter_frame method was split: + + - for video filters, it was made of start_frame, draw_slice (that could be + called several times on distinct parts of the frame) and end_frame; + + - for audio filters, it was called filter_samples. diff --git a/doc/filters.texi b/doc/filters.texi index 9804c0e29d..6c5003f207 100644 --- a/doc/filters.texi +++ b/doc/filters.texi @@ -1,3 +1,100 @@ +@chapter Filtering Introduction +@c man begin FILTERING INTRODUCTION + +Filtering in FFmpeg is enabled through the libavfilter library. + +In libavfilter, a filter can have multiple inputs and multiple +outputs. +To illustrate the sorts of things that are possible, we consider the +following filtergraph. + +@verbatim + [main] +input --> split ---------------------> overlay --> output + | ^ + |[tmp] [flip]| + +-----> crop --> vflip -------+ +@end verbatim + +This filtergraph splits the input stream in two streams, then sends one +stream through the crop filter and the vflip filter, before merging it +back with the other stream by overlaying it on top. You can use the +following command to achieve this: + +@example +ffmpeg -i INPUT -vf "split [main][tmp]; [tmp] crop=iw:ih/2:0:0, vflip [flip]; [main][flip] overlay=0:H/2" OUTPUT +@end example + +The result will be that the top half of the video is mirrored +onto the bottom half of the output video. + +Filters in the same linear chain are separated by commas, and distinct +linear chains of filters are separated by semicolons. In our example, +@var{crop,vflip} are in one linear chain, @var{split} and +@var{overlay} are separately in another. The points where the linear +chains join are labelled by names enclosed in square brackets. In the +example, the split filter generates two outputs that are associated to +the labels @var{[main]} and @var{[tmp]}. + +The stream sent to the second output of @var{split}, labelled as +@var{[tmp]}, is processed through the @var{crop} filter, which crops +away the lower half part of the video, and then vertically flipped. The +@var{overlay} filter takes in input the first unchanged output of the +split filter (which was labelled as @var{[main]}), and overlay on its +lower half the output generated by the @var{crop,vflip} filterchain. + +Some filters take in input a list of parameters: they are specified +after the filter name and an equal sign, and are separated from each other +by a colon. + +There exist so-called @var{source filters} that do not have an +audio/video input, and @var{sink filters} that will not have audio/video +output. + +@c man end FILTERING INTRODUCTION + +@chapter graph2dot +@c man begin GRAPH2DOT + +The @file{graph2dot} program included in the FFmpeg @file{tools} +directory can be used to parse a filtergraph description and issue a +corresponding textual representation in the dot language. + +Invoke the command: +@example +graph2dot -h +@end example + +to see how to use @file{graph2dot}. + +You can then pass the dot description to the @file{dot} program (from +the graphviz suite of programs) and obtain a graphical representation +of the filtergraph. + +For example the sequence of commands: +@example +echo @var{GRAPH_DESCRIPTION} | \ +tools/graph2dot -o graph.tmp && \ +dot -Tpng graph.tmp -o graph.png && \ +display graph.png +@end example + +can be used to create and display an image representing the graph +described by the @var{GRAPH_DESCRIPTION} string. Note that this string must be +a complete self-contained graph, with its inputs and outputs explicitly defined. +For example if your command line is of the form: +@example +ffmpeg -i infile -vf scale=640:360 outfile +@end example +your @var{GRAPH_DESCRIPTION} string will need to be of the form: +@example +nullsrc,scale=640:360,nullsink +@end example +you may also need to set the @var{nullsrc} parameters and add a @var{format} +filter in order to simulate a specific input file. + +@c man end GRAPH2DOT + @chapter Filtergraph description @c man begin FILTERGRAPH DESCRIPTION @@ -17,10 +114,11 @@ output pads is called a "sink". @anchor{Filtergraph syntax} @section Filtergraph syntax -A filtergraph has a textual representation, which is -recognized by the @option{-filter}/@option{-vf} and @option{-filter_complex} -options in @command{avconv} and @option{-vf} in @command{avplay}, and by the -@code{avfilter_graph_parse()}/@code{avfilter_graph_parse2()} functions defined in +A filtergraph has a textual representation, which is recognized by the +@option{-filter}/@option{-vf}/@option{-af} and +@option{-filter_complex} options in @command{ffmpeg} and +@option{-vf}/@option{-af} in @command{ffplay}, and by the +@code{avfilter_graph_parse_ptr()} function defined in @file{libavfilter/avfilter.h}. A filterchain consists of a sequence of connected filters, each one @@ -55,21 +153,27 @@ declares three options in this order -- @option{type}, @option{start_frame} and @var{in} is assigned to the option @option{type}, @var{0} to @option{start_frame} and @var{30} to @option{nb_frames}. +@item +A ':'-separated list of mixed direct @var{value} and long @var{key=value} +pairs. The direct @var{value} must precede the @var{key=value} pairs, and +follow the same constraints order of the previous point. The following +@var{key=value} pairs can be set in any preferred order. + @end itemize If the option value itself is a list of items (e.g. the @code{format} filter takes a list of pixel formats), the items in the list are usually separated by -'|'. +@samp{|}. -The list of arguments can be quoted using the character "'" as initial -and ending mark, and the character '\' for escaping the characters +The list of arguments can be quoted using the character @samp{'} as initial +and ending mark, and the character @samp{\} for escaping the characters within the quoted text; otherwise the argument string is considered terminated when the next special character (belonging to the set -"[]=;,") is encountered. +@samp{[]=;,}) is encountered. The name and arguments of the filter are optionally preceded and followed by a list of link labels. -A link label allows to name a link and associate it to a filter output +A link label allows one to name a link and associate it to a filter output or input pad. The preceding labels @var{in_link_1} ... @var{in_link_N}, are associated to the filter input pads, the following labels @var{out_link_1} ... @var{out_link_M}, are @@ -91,6 +195,10 @@ instance two input pads. The first output pad of split is labelled output pad of split is linked to the second input pad of overlay, which are both unlabelled. +In a filter description, if the input label of the first filter is not +specified, "in" is assumed; if the output label of the last filter is not +specified, "out" is assumed. + In a complete filterchain all the unlabelled filter input and output pads must be connected. A filtergraph is considered valid if all the filter input and output pads of all the filterchains are connected. @@ -112,21 +220,600 @@ Here is a BNF description of the filtergraph syntax: @var{FILTERGRAPH} ::= [sws_flags=@var{flags};] @var{FILTERCHAIN} [;@var{FILTERGRAPH}] @end example +@section Notes on filtergraph escaping + +Filtergraph description composition entails several levels of +escaping. See @ref{quoting_and_escaping,,the "Quoting and escaping" +section in the ffmpeg-utils(1) manual,ffmpeg-utils} for more +information about the employed escaping procedure. + +A first level escaping affects the content of each filter option +value, which may contain the special character @code{:} used to +separate values, or one of the escaping characters @code{\'}. + +A second level escaping affects the whole filter description, which +may contain the escaping characters @code{\'} or the special +characters @code{[],;} used by the filtergraph description. + +Finally, when you specify a filtergraph on a shell commandline, you +need to perform a third level escaping for the shell special +characters contained within it. + +For example, consider the following string to be embedded in +the @ref{drawtext} filter description @option{text} value: +@example +this is a 'string': may contain one, or more, special characters +@end example + +This string contains the @code{'} special escaping character, and the +@code{:} special character, so it needs to be escaped in this way: +@example +text=this is a \'string\'\: may contain one, or more, special characters +@end example + +A second level of escaping is required when embedding the filter +description in a filtergraph description, in order to escape all the +filtergraph special characters. Thus the example above becomes: +@example +drawtext=text=this is a \\\'string\\\'\\: may contain one\, or more\, special characters +@end example +(note that in addition to the @code{\'} escaping special characters, +also @code{,} needs to be escaped). + +Finally an additional level of escaping is needed when writing the +filtergraph description in a shell command, which depends on the +escaping rules of the adopted shell. For example, assuming that +@code{\} is special and needs to be escaped with another @code{\}, the +previous string will finally result in: +@example +-vf "drawtext=text=this is a \\\\\\'string\\\\\\'\\\\: may contain one\\, or more\\, special characters" +@end example + +@chapter Timeline editing + +Some filters support a generic @option{enable} option. For the filters +supporting timeline editing, this option can be set to an expression which is +evaluated before sending a frame to the filter. If the evaluation is non-zero, +the filter will be enabled, otherwise the frame will be sent unchanged to the +next filter in the filtergraph. + +The expression accepts the following values: +@table @samp +@item t +timestamp expressed in seconds, NAN if the input timestamp is unknown + +@item n +sequential number of the input frame, starting from 0 + +@item pos +the position in the file of the input frame, NAN if unknown + +@item w +@item h +width and height of the input frame if video +@end table + +Additionally, these filters support an @option{enable} command that can be used +to re-define the expression. + +Like any other filtering option, the @option{enable} option follows the same +rules. + +For example, to enable a blur filter (@ref{smartblur}) from 10 seconds to 3 +minutes, and a @ref{curves} filter starting at 3 seconds: +@example +smartblur = enable='between(t,10,3*60)', +curves = enable='gte(t,3)' : preset=cross_process +@end example + @c man end FILTERGRAPH DESCRIPTION @chapter Audio Filters @c man begin AUDIO FILTERS -When you configure your Libav build, you can disable any of the -existing filters using --disable-filters. +When you configure your FFmpeg build, you can disable any of the +existing filters using @code{--disable-filters}. The configure output will show the audio filters included in your build. Below is a description of the currently available audio filters. +@section acompressor + +A compressor is mainly used to reduce the dynamic range of a signal. +Especially modern music is mostly compressed at a high ratio to +improve the overall loudness. It's done to get the highest attention +of a listener, "fatten" the sound and bring more "power" to the track. +If a signal is compressed too much it may sound dull or "dead" +afterwards or it may start to "pump" (which could be a powerful effect +but can also destroy a track completely). +The right compression is the key to reach a professional sound and is +the high art of mixing and mastering. Because of its complex settings +it may take a long time to get the right feeling for this kind of effect. + +Compression is done by detecting the volume above a chosen level +@code{threshold} and dividing it by the factor set with @code{ratio}. +So if you set the threshold to -12dB and your signal reaches -6dB a ratio +of 2:1 will result in a signal at -9dB. Because an exact manipulation of +the signal would cause distortion of the waveform the reduction can be +levelled over the time. This is done by setting "Attack" and "Release". +@code{attack} determines how long the signal has to rise above the threshold +before any reduction will occur and @code{release} sets the time the signal +has to fall below the threshold to reduce the reduction again. Shorter signals +than the chosen attack time will be left untouched. +The overall reduction of the signal can be made up afterwards with the +@code{makeup} setting. So compressing the peaks of a signal about 6dB and +raising the makeup to this level results in a signal twice as loud than the +source. To gain a softer entry in the compression the @code{knee} flattens the +hard edge at the threshold in the range of the chosen decibels. + +The filter accepts the following options: + +@table @option +@item level_in +Set input gain. Default is 1. Range is between 0.015625 and 64. + +@item threshold +If a signal of second stream rises above this level it will affect the gain +reduction of the first stream. +By default it is 0.125. Range is between 0.00097563 and 1. + +@item ratio +Set a ratio by which the signal is reduced. 1:2 means that if the level +rose 4dB above the threshold, it will be only 2dB above after the reduction. +Default is 2. Range is between 1 and 20. + +@item attack +Amount of milliseconds the signal has to rise above the threshold before gain +reduction starts. Default is 20. Range is between 0.01 and 2000. + +@item release +Amount of milliseconds the signal has to fall below the threshold before +reduction is decreased again. Default is 250. Range is between 0.01 and 9000. + +@item makeup +Set the amount by how much signal will be amplified after processing. +Default is 2. Range is from 1 and 64. + +@item knee +Curve the sharp knee around the threshold to enter gain reduction more softly. +Default is 2.82843. Range is between 1 and 8. + +@item link +Choose if the @code{average} level between all channels of input stream +or the louder(@code{maximum}) channel of input stream affects the +reduction. Default is @code{average}. + +@item detection +Should the exact signal be taken in case of @code{peak} or an RMS one in case +of @code{rms}. Default is @code{rms} which is mostly smoother. + +@item mix +How much to use compressed signal in output. Default is 1. +Range is between 0 and 1. +@end table + +@section acrossfade + +Apply cross fade from one input audio stream to another input audio stream. +The cross fade is applied for specified duration near the end of first stream. + +The filter accepts the following options: + +@table @option +@item nb_samples, ns +Specify the number of samples for which the cross fade effect has to last. +At the end of the cross fade effect the first input audio will be completely +silent. Default is 44100. + +@item duration, d +Specify the duration of the cross fade effect. See +@ref{time duration syntax,,the Time duration section in the ffmpeg-utils(1) manual,ffmpeg-utils} +for the accepted syntax. +By default the duration is determined by @var{nb_samples}. +If set this option is used instead of @var{nb_samples}. + +@item overlap, o +Should first stream end overlap with second stream start. Default is enabled. + +@item curve1 +Set curve for cross fade transition for first stream. + +@item curve2 +Set curve for cross fade transition for second stream. + +For description of available curve types see @ref{afade} filter description. +@end table + +@subsection Examples + +@itemize +@item +Cross fade from one input to another: +@example +ffmpeg -i first.flac -i second.flac -filter_complex acrossfade=d=10:c1=exp:c2=exp output.flac +@end example + +@item +Cross fade from one input to another but without overlapping: +@example +ffmpeg -i first.flac -i second.flac -filter_complex acrossfade=d=10:o=0:c1=exp:c2=exp output.flac +@end example +@end itemize + +@section adelay + +Delay one or more audio channels. + +Samples in delayed channel are filled with silence. + +The filter accepts the following option: + +@table @option +@item delays +Set list of delays in milliseconds for each channel separated by '|'. +At least one delay greater than 0 should be provided. +Unused delays will be silently ignored. If number of given delays is +smaller than number of channels all remaining channels will not be delayed. +@end table + +@subsection Examples + +@itemize +@item +Delay first channel by 1.5 seconds, the third channel by 0.5 seconds and leave +the second channel (and any other channels that may be present) unchanged. +@example +adelay=1500|0|500 +@end example +@end itemize + +@section aecho + +Apply echoing to the input audio. + +Echoes are reflected sound and can occur naturally amongst mountains +(and sometimes large buildings) when talking or shouting; digital echo +effects emulate this behaviour and are often used to help fill out the +sound of a single instrument or vocal. The time difference between the +original signal and the reflection is the @code{delay}, and the +loudness of the reflected signal is the @code{decay}. +Multiple echoes can have different delays and decays. + +A description of the accepted parameters follows. + +@table @option +@item in_gain +Set input gain of reflected signal. Default is @code{0.6}. + +@item out_gain +Set output gain of reflected signal. Default is @code{0.3}. + +@item delays +Set list of time intervals in milliseconds between original signal and reflections +separated by '|'. Allowed range for each @code{delay} is @code{(0 - 90000.0]}. +Default is @code{1000}. + +@item decays +Set list of loudnesses of reflected signals separated by '|'. +Allowed range for each @code{decay} is @code{(0 - 1.0]}. +Default is @code{0.5}. +@end table + +@subsection Examples + +@itemize +@item +Make it sound as if there are twice as many instruments as are actually playing: +@example +aecho=0.8:0.88:60:0.4 +@end example + +@item +If delay is very short, then it sound like a (metallic) robot playing music: +@example +aecho=0.8:0.88:6:0.4 +@end example + +@item +A longer delay will sound like an open air concert in the mountains: +@example +aecho=0.8:0.9:1000:0.3 +@end example + +@item +Same as above but with one more mountain: +@example +aecho=0.8:0.9:1000|1800:0.3|0.25 +@end example +@end itemize + +@section aemphasis +Audio emphasis filter creates or restores material directly taken from LPs or +emphased CDs with different filter curves. E.g. to store music on vinyl the +signal has to be altered by a filter first to even out the disadvantages of +this recording medium. +Once the material is played back the inverse filter has to be applied to +restore the distortion of the frequency response. + +The filter accepts the following options: + +@table @option +@item level_in +Set input gain. + +@item level_out +Set output gain. + +@item mode +Set filter mode. For restoring material use @code{reproduction} mode, otherwise +use @code{production} mode. Default is @code{reproduction} mode. + +@item type +Set filter type. Selects medium. Can be one of the following: + +@table @option +@item col +select Columbia. +@item emi +select EMI. +@item bsi +select BSI (78RPM). +@item riaa +select RIAA. +@item cd +select Compact Disc (CD). +@item 50fm +select 50µs (FM). +@item 75fm +select 75µs (FM). +@item 50kf +select 50µs (FM-KF). +@item 75kf +select 75µs (FM-KF). +@end table +@end table + +@section aeval + +Modify an audio signal according to the specified expressions. + +This filter accepts one or more expressions (one for each channel), +which are evaluated and used to modify a corresponding audio signal. + +It accepts the following parameters: + +@table @option +@item exprs +Set the '|'-separated expressions list for each separate channel. If +the number of input channels is greater than the number of +expressions, the last specified expression is used for the remaining +output channels. + +@item channel_layout, c +Set output channel layout. If not specified, the channel layout is +specified by the number of expressions. If set to @samp{same}, it will +use by default the same input channel layout. +@end table + +Each expression in @var{exprs} can contain the following constants and functions: + +@table @option +@item ch +channel number of the current expression + +@item n +number of the evaluated sample, starting from 0 + +@item s +sample rate + +@item t +time of the evaluated sample expressed in seconds + +@item nb_in_channels +@item nb_out_channels +input and output number of channels + +@item val(CH) +the value of input channel with number @var{CH} +@end table + +Note: this filter is slow. For faster processing you should use a +dedicated filter. + +@subsection Examples + +@itemize +@item +Half volume: +@example +aeval=val(ch)/2:c=same +@end example + +@item +Invert phase of the second channel: +@example +aeval=val(0)|-val(1) +@end example +@end itemize + +@anchor{afade} +@section afade + +Apply fade-in/out effect to input audio. + +A description of the accepted parameters follows. + +@table @option +@item type, t +Specify the effect type, can be either @code{in} for fade-in, or +@code{out} for a fade-out effect. Default is @code{in}. + +@item start_sample, ss +Specify the number of the start sample for starting to apply the fade +effect. Default is 0. + +@item nb_samples, ns +Specify the number of samples for which the fade effect has to last. At +the end of the fade-in effect the output audio will have the same +volume as the input audio, at the end of the fade-out transition +the output audio will be silence. Default is 44100. + +@item start_time, st +Specify the start time of the fade effect. Default is 0. +The value must be specified as a time duration; see +@ref{time duration syntax,,the Time duration section in the ffmpeg-utils(1) manual,ffmpeg-utils} +for the accepted syntax. +If set this option is used instead of @var{start_sample}. + +@item duration, d +Specify the duration of the fade effect. See +@ref{time duration syntax,,the Time duration section in the ffmpeg-utils(1) manual,ffmpeg-utils} +for the accepted syntax. +At the end of the fade-in effect the output audio will have the same +volume as the input audio, at the end of the fade-out transition +the output audio will be silence. +By default the duration is determined by @var{nb_samples}. +If set this option is used instead of @var{nb_samples}. + +@item curve +Set curve for fade transition. + +It accepts the following values: +@table @option +@item tri +select triangular, linear slope (default) +@item qsin +select quarter of sine wave +@item hsin +select half of sine wave +@item esin +select exponential sine wave +@item log +select logarithmic +@item ipar +select inverted parabola +@item qua +select quadratic +@item cub +select cubic +@item squ +select square root +@item cbr +select cubic root +@item par +select parabola +@item exp +select exponential +@item iqsin +select inverted quarter of sine wave +@item ihsin +select inverted half of sine wave +@item dese +select double-exponential seat +@item desi +select double-exponential sigmoid +@end table +@end table + +@subsection Examples + +@itemize +@item +Fade in first 15 seconds of audio: +@example +afade=t=in:ss=0:d=15 +@end example + +@item +Fade out last 25 seconds of a 900 seconds audio: +@example +afade=t=out:st=875:d=25 +@end example +@end itemize + +@section afftfilt +Apply arbitrary expressions to samples in frequency domain. + +@table @option +@item real +Set frequency domain real expression for each separate channel separated +by '|'. Default is "1". +If the number of input channels is greater than the number of +expressions, the last specified expression is used for the remaining +output channels. + +@item imag +Set frequency domain imaginary expression for each separate channel +separated by '|'. If not set, @var{real} option is used. + +Each expression in @var{real} and @var{imag} can contain the following +constants: + +@table @option +@item sr +sample rate + +@item b +current frequency bin number + +@item nb +number of available bins + +@item ch +channel number of the current expression + +@item chs +number of channels + +@item pts +current frame pts +@end table + +@item win_size +Set window size. + +It accepts the following values: +@table @samp +@item w16 +@item w32 +@item w64 +@item w128 +@item w256 +@item w512 +@item w1024 +@item w2048 +@item w4096 +@item w8192 +@item w16384 +@item w32768 +@item w65536 +@end table +Default is @code{w4096} + +@item win_func +Set window function. Default is @code{hann}. + +@item overlap +Set window overlap. If set to 1, the recommended overlap for selected +window function will be picked. Default is @code{0.75}. +@end table + +@subsection Examples + +@itemize +@item +Leave almost only low frequencies in audio: +@example +afftfilt="1-clip((b/nb)*b,0,1)" +@end example +@end itemize + +@anchor{aformat} @section aformat -Convert the input audio to one of the specified formats. The framework will +Set output format constraints for the input audio. The framework will negotiate the most appropriate format to minimize conversions. It accepts the following parameters: @@ -141,6 +828,8 @@ A '|'-separated list of requested sample rates. @item channel_layouts A '|'-separated list of requested channel layouts. +See @ref{channel layout syntax,,the Channel Layout section in the ffmpeg-utils(1) manual,ffmpeg-utils} +for the required syntax. @end table If a parameter is omitted, all values are allowed. @@ -150,13 +839,205 @@ Force the output to either unsigned 8-bit or signed 16-bit stereo aformat=sample_fmts=u8|s16:channel_layouts=stereo @end example +@section agate + +A gate is mainly used to reduce lower parts of a signal. This kind of signal +processing reduces disturbing noise between useful signals. + +Gating is done by detecting the volume below a chosen level @var{threshold} +and divide it by the factor set with @var{ratio}. The bottom of the noise +floor is set via @var{range}. Because an exact manipulation of the signal +would cause distortion of the waveform the reduction can be levelled over +time. This is done by setting @var{attack} and @var{release}. + +@var{attack} determines how long the signal has to fall below the threshold +before any reduction will occur and @var{release} sets the time the signal +has to raise above the threshold to reduce the reduction again. +Shorter signals than the chosen attack time will be left untouched. + +@table @option +@item level_in +Set input level before filtering. +Default is 1. Allowed range is from 0.015625 to 64. + +@item range +Set the level of gain reduction when the signal is below the threshold. +Default is 0.06125. Allowed range is from 0 to 1. + +@item threshold +If a signal rises above this level the gain reduction is released. +Default is 0.125. Allowed range is from 0 to 1. + +@item ratio +Set a ratio about which the signal is reduced. +Default is 2. Allowed range is from 1 to 9000. + +@item attack +Amount of milliseconds the signal has to rise above the threshold before gain +reduction stops. +Default is 20 milliseconds. Allowed range is from 0.01 to 9000. + +@item release +Amount of milliseconds the signal has to fall below the threshold before the +reduction is increased again. Default is 250 milliseconds. +Allowed range is from 0.01 to 9000. + +@item makeup +Set amount of amplification of signal after processing. +Default is 1. Allowed range is from 1 to 64. + +@item knee +Curve the sharp knee around the threshold to enter gain reduction more softly. +Default is 2.828427125. Allowed range is from 1 to 8. + +@item detection +Choose if exact signal should be taken for detection or an RMS like one. +Default is rms. Can be peak or rms. + +@item link +Choose if the average level between all channels or the louder channel affects +the reduction. +Default is average. Can be average or maximum. +@end table + +@section alimiter + +The limiter prevents input signal from raising over a desired threshold. +This limiter uses lookahead technology to prevent your signal from distorting. +It means that there is a small delay after signal is processed. Keep in mind +that the delay it produces is the attack time you set. + +The filter accepts the following options: + +@table @option +@item level_in +Set input gain. Default is 1. + +@item level_out +Set output gain. Default is 1. + +@item limit +Don't let signals above this level pass the limiter. Default is 1. + +@item attack +The limiter will reach its attenuation level in this amount of time in +milliseconds. Default is 5 milliseconds. + +@item release +Come back from limiting to attenuation 1.0 in this amount of milliseconds. +Default is 50 milliseconds. + +@item asc +When gain reduction is always needed ASC takes care of releasing to an +average reduction level rather than reaching a reduction of 0 in the release +time. + +@item asc_level +Select how much the release time is affected by ASC, 0 means nearly no changes +in release time while 1 produces higher release times. + +@item level +Auto level output signal. Default is enabled. +This normalizes audio back to 0dB if enabled. +@end table + +Depending on picked setting it is recommended to upsample input 2x or 4x times +with @ref{aresample} before applying this filter. + +@section allpass + +Apply a two-pole all-pass filter with central frequency (in Hz) +@var{frequency}, and filter-width @var{width}. +An all-pass filter changes the audio's frequency to phase relationship +without changing its frequency to amplitude relationship. + +The filter accepts the following options: + +@table @option +@item frequency, f +Set frequency in Hz. + +@item width_type +Set method to specify band-width of filter. +@table @option +@item h +Hz +@item q +Q-Factor +@item o +octave +@item s +slope +@end table + +@item width, w +Specify the band-width of a filter in width_type units. +@end table + +@anchor{amerge} +@section amerge + +Merge two or more audio streams into a single multi-channel stream. + +The filter accepts the following options: + +@table @option + +@item inputs +Set the number of inputs. Default is 2. + +@end table + +If the channel layouts of the inputs are disjoint, and therefore compatible, +the channel layout of the output will be set accordingly and the channels +will be reordered as necessary. If the channel layouts of the inputs are not +disjoint, the output will have all the channels of the first input then all +the channels of the second input, in that order, and the channel layout of +the output will be the default value corresponding to the total number of +channels. + +For example, if the first input is in 2.1 (FL+FR+LF) and the second input +is FC+BL+BR, then the output will be in 5.1, with the channels in the +following order: a1, a2, b1, a3, b2, b3 (a1 is the first channel of the +first input, b1 is the first channel of the second input). + +On the other hand, if both input are in stereo, the output channels will be +in the default order: a1, a2, b1, b2, and the channel layout will be +arbitrarily set to 4.0, which may or may not be the expected value. + +All inputs must have the same sample rate, and format. + +If inputs do not have the same duration, the output will stop with the +shortest. + +@subsection Examples + +@itemize +@item +Merge two mono files into a stereo stream: +@example +amovie=left.wav [l] ; amovie=right.mp3 [r] ; [l] [r] amerge +@end example + +@item +Multiple merges assuming 1 video stream and 6 audio streams in @file{input.mkv}: +@example +ffmpeg -i input.mkv -filter_complex "[0:1][0:2][0:3][0:4][0:5][0:6] amerge=inputs=6" -c:a pcm_s16le output.mkv +@end example +@end itemize + @section amix Mixes multiple audio inputs into a single output. +Note that this filter only supports float samples (the @var{amerge} +and @var{pan} audio filters support many formats). If the @var{amix} +input has integer samples then @ref{aresample} will be automatically +inserted to perform the conversion to float samples. + For example @example -avconv -i INPUT1 -i INPUT2 -i INPUT3 -filter_complex amix=inputs=3:duration=first:dropout_transition=3 OUTPUT +ffmpeg -i INPUT1 -i INPUT2 -i INPUT3 -filter_complex amix=inputs=3:duration=first:dropout_transition=3 OUTPUT @end example will mix 3 input audio streams to a single output with the same duration as the first input and a dropout transition time of 3 seconds. @@ -188,118 +1069,325 @@ stream ends. The default value is 2 seconds. @end table -@section anull - -Pass the audio source unchanged to the output. - -@section asetpts +@section anequalizer -Change the PTS (presentation timestamp) of the input audio frames. +High-order parametric multiband equalizer for each channel. It accepts the following parameters: +@table @option +@item params + +This option string is in format: +"c@var{chn} f=@var{cf} w=@var{w} g=@var{g} t=@var{f} | ..." +Each equalizer band is separated by '|'. @table @option +@item chn +Set channel number to which equalization will be applied. +If input doesn't have that channel the entry is ignored. -@item expr -The expression which is evaluated for each frame to construct its timestamp. +@item cf +Set central frequency for band. +If input doesn't have that frequency the entry is ignored. +@item w +Set band width in hertz. + +@item g +Set band gain in dB. + +@item f +Set filter type for band, optional, can be: + +@table @samp +@item 0 +Butterworth, this is default. + +@item 1 +Chebyshev type 1. + +@item 2 +Chebyshev type 2. +@end table @end table -The expression is evaluated through the eval API and can contain the following -constants: +@item curves +With this option activated frequency response of anequalizer is displayed +in video stream. -@table @option -@item FRAME_RATE -frame rate, only defined for constant frame-rate video +@item size +Set video stream size. Only useful if curves option is activated. + +@item mgain +Set max gain that will be displayed. Only useful if curves option is activated. +Setting this to reasonable value allows to display gain which is derived from +neighbour bands which are too close to each other and thus produce higher gain +when both are activated. + +@item fscale +Set frequency scale used to draw frequency response in video output. +Can be linear or logarithmic. Default is logarithmic. + +@item colors +Set color for each channel curve which is going to be displayed in video stream. +This is list of color names separated by space or by '|'. +Unrecognised or missing colors will be replaced by white color. +@end table -@item PTS -the presentation timestamp in input +@subsection Examples -@item E, PI, PHI -These are approximated values for the mathematical constants e -(Euler's number), pi (Greek pi), and phi (the golden ratio). +@itemize +@item +Lower gain by 10 of central frequency 200Hz and width 100 Hz +for first 2 channels using Chebyshev type 1 filter: +@example +anequalizer=c0 f=200 w=100 g=-10 t=1|c1 f=200 w=100 g=-10 t=1 +@end example +@end itemize -@item N -The number of audio samples passed through the filter so far, starting at 0. +@subsection Commands -@item S -The number of audio samples in the current frame. +This filter supports the following commands: +@table @option +@item change +Alter existing filter parameters. +Syntax for the commands is : "@var{fN}|f=@var{freq}|w=@var{width}|g=@var{gain}" + +@var{fN} is existing filter number, starting from 0, if no such filter is available +error is returned. +@var{freq} set new frequency parameter. +@var{width} set new width parameter in herz. +@var{gain} set new gain parameter in dB. + +Full filter invocation with asendcmd may look like this: +asendcmd=c='4.0 anequalizer change 0|f=200|w=50|g=1',anequalizer=... +@end table -@item SR -The audio sample rate. +@section anull -@item STARTPTS -The PTS of the first frame. +Pass the audio source unchanged to the output. -@item PREV_INPTS -The previous input PTS. +@section apad -@item PREV_OUTPTS -The previous output PTS. +Pad the end of an audio stream with silence. -@item RTCTIME -The wallclock (RTC) time in microseconds. +This can be used together with @command{ffmpeg} @option{-shortest} to +extend audio streams to the same length as the video stream. -@item RTCSTART -The wallclock (RTC) time at the start of the movie in microseconds. +A description of the accepted options follows. +@table @option +@item packet_size +Set silence packet size. Default value is 4096. + +@item pad_len +Set the number of samples of silence to add to the end. After the +value is reached, the stream is terminated. This option is mutually +exclusive with @option{whole_len}. + +@item whole_len +Set the minimum total number of samples in the output audio stream. If +the value is longer than the input audio length, silence is added to +the end, until the value is reached. This option is mutually exclusive +with @option{pad_len}. @end table -Some examples: +If neither the @option{pad_len} nor the @option{whole_len} option is +set, the filter will add silence to the end of the input stream +indefinitely. + +@subsection Examples +@itemize +@item +Add 1024 samples of silence to the end of the input: @example -# Start counting PTS from zero -asetpts=expr=PTS-STARTPTS +apad=pad_len=1024 +@end example -# Generate timestamps by counting samples -asetpts=expr=N/SR/TB +@item +Make sure the audio output will contain at least 10000 samples, pad +the input with silence if required: +@example +apad=whole_len=10000 +@end example -# Generate timestamps from a "live source" and rebase onto the current timebase -asetpts='(RTCTIME - RTCSTART) / (TB * 1000000)" +@item +Use @command{ffmpeg} to pad the audio input with silence, so that the +video stream will always result the shortest and will be converted +until the end in the output file when using the @option{shortest} +option: +@example +ffmpeg -i VIDEO -i AUDIO -filter_complex "[1:0]apad" -shortest OUTPUT @end example +@end itemize -@section asettb +@section aphaser +Add a phasing effect to the input audio. -Set the timebase to use for the output frames timestamps. -It is mainly useful for testing timebase configuration. +A phaser filter creates series of peaks and troughs in the frequency spectrum. +The position of the peaks and troughs are modulated so that they vary over time, creating a sweeping effect. -This filter accepts the following parameters: +A description of the accepted parameters follows. @table @option +@item in_gain +Set input gain. Default is 0.4. -@item expr -The expression which is evaluated into the output timebase. +@item out_gain +Set output gain. Default is 0.74 +@item delay +Set delay in milliseconds. Default is 3.0. + +@item decay +Set decay. Default is 0.4. + +@item speed +Set modulation speed in Hz. Default is 0.5. + +@item type +Set modulation type. Default is triangular. + +It accepts the following values: +@table @samp +@item triangular, t +@item sinusoidal, s +@end table @end table -The expression can contain the constants @var{PI}, @var{E}, @var{PHI}, @var{AVTB} (the -default timebase), @var{intb} (the input timebase), and @var{sr} (the sample rate, -audio only). +@section apulsator -The default value for the input is @var{intb}. +Audio pulsator is something between an autopanner and a tremolo. +But it can produce funny stereo effects as well. Pulsator changes the volume +of the left and right channel based on a LFO (low frequency oscillator) with +different waveforms and shifted phases. +This filter have the ability to define an offset between left and right +channel. An offset of 0 means that both LFO shapes match each other. +The left and right channel are altered equally - a conventional tremolo. +An offset of 50% means that the shape of the right channel is exactly shifted +in phase (or moved backwards about half of the frequency) - pulsator acts as +an autopanner. At 1 both curves match again. Every setting in between moves the +phase shift gapless between all stages and produces some "bypassing" sounds with +sine and triangle waveforms. The more you set the offset near 1 (starting from +the 0.5) the faster the signal passes from the left to the right speaker. -Some examples: +The filter accepts the following options: + +@table @option +@item level_in +Set input gain. By default it is 1. Range is [0.015625 - 64]. + +@item level_out +Set output gain. By default it is 1. Range is [0.015625 - 64]. + +@item mode +Set waveform shape the LFO will use. Can be one of: sine, triangle, square, +sawup or sawdown. Default is sine. + +@item amount +Set modulation. Define how much of original signal is affected by the LFO. + +@item offset_l +Set left channel offset. Default is 0. Allowed range is [0 - 1]. + +@item offset_r +Set right channel offset. Default is 0.5. Allowed range is [0 - 1]. + +@item width +Set pulse width. Default is 1. Allowed range is [0 - 2]. + +@item timing +Set possible timing mode. Can be one of: bpm, ms or hz. Default is hz. + +@item bpm +Set bpm. Default is 120. Allowed range is [30 - 300]. Only used if timing +is set to bpm. + +@item ms +Set ms. Default is 500. Allowed range is [10 - 2000]. Only used if timing +is set to ms. + +@item hz +Set frequency in Hz. Default is 2. Allowed range is [0.01 - 100]. Only used +if timing is set to hz. +@end table + +@anchor{aresample} +@section aresample + +Resample the input audio to the specified parameters, using the +libswresample library. If none are specified then the filter will +automatically convert between its input and output. + +This filter is also able to stretch/squeeze the audio data to make it match +the timestamps or to inject silence / cut out audio to make it match the +timestamps, do a combination of both or do neither. + +The filter accepts the syntax +[@var{sample_rate}:]@var{resampler_options}, where @var{sample_rate} +expresses a sample rate and @var{resampler_options} is a list of +@var{key}=@var{value} pairs, separated by ":". See the +ffmpeg-resampler manual for the complete list of supported options. + +@subsection Examples + +@itemize +@item +Resample the input audio to 44100Hz: +@example +aresample=44100 +@end example +@item +Stretch/squeeze samples to the given timestamps, with a maximum of 1000 +samples per second compensation: @example -# Set the timebase to 1/25: -settb=1/25 +aresample=async=1000 +@end example +@end itemize -# Set the timebase to 1/10: -settb=0.1 +@section asetnsamples -# Set the timebase to 1001/1000: -settb=1+0.001 +Set the number of samples per each output audio frame. -# Set the timebase to 2*intb: -settb=2*intb +The last output packet may contain a different number of samples, as +the filter will flush all the remaining samples when the input audio +signal its end. -# Set the default timebase value: -settb=AVTB +The filter accepts the following options: + +@table @option + +@item nb_out_samples, n +Set the number of frames per each output audio frame. The number is +intended as the number of samples @emph{per each channel}. +Default value is 1024. -# Set the timebase to twice the sample rate: -asettb=sr*2 +@item pad, p +If set to 1, the filter will pad the last audio frame with zeroes, so +that the last frame will contain the same number of samples as the +previous ones. Default value is 1. +@end table + +For example, to set the number of per-frame samples to 1234 and +disable padding for the last frame, use: +@example +asetnsamples=n=1234:p=0 @end example +@section asetrate + +Set the sample rate without altering the PCM data. +This will result in a change of speed and pitch. + +The filter accepts the following options: + +@table @option +@item sample_rate, r +Set the output sample rate. Default is 44100 Hz. +@end table + @section ashowinfo Show a line containing various information for each input audio frame. @@ -308,7 +1396,7 @@ The input audio is not modified. The shown line contains a sequence of key/value pairs of the form @var{key}:@var{value}. -It accepts the following parameters: +The following values are shown in the output: @table @option @item n @@ -321,6 +1409,10 @@ depends on the filter input pad, and is usually 1/@var{sample_rate}. @item pts_time The presentation timestamp of the input frame in seconds. +@item pos +position of the frame in the input stream, -1 if this information in +unavailable and/or meaningless (for example in case of synthetic audio) + @item fmt The sample format. @@ -341,23 +1433,118 @@ audio, the data is treated as if all the planes were concatenated. A list of Adler-32 checksums for each data plane. @end table -@section asplit +@anchor{astats} +@section astats -Split input audio into several identical outputs. +Display time domain statistical information about the audio channels. +Statistics are calculated and displayed for each audio channel and, +where applicable, an overall figure is also given. -It accepts a single parameter, which specifies the number of outputs. If -unspecified, it defaults to 2. +It accepts the following option: +@table @option +@item length +Short window length in seconds, used for peak and trough RMS measurement. +Default is @code{0.05} (50 milliseconds). Allowed range is @code{[0.1 - 10]}. + +@item metadata + +Set metadata injection. All the metadata keys are prefixed with @code{lavfi.astats.X}, +where @code{X} is channel number starting from 1 or string @code{Overall}. Default is +disabled. + +Available keys for each channel are: +DC_offset +Min_level +Max_level +Min_difference +Max_difference +Mean_difference +Peak_level +RMS_peak +RMS_trough +Crest_factor +Flat_factor +Peak_count +Bit_depth + +and for Overall: +DC_offset +Min_level +Max_level +Min_difference +Max_difference +Mean_difference +Peak_level +RMS_level +RMS_peak +RMS_trough +Flat_factor +Peak_count +Bit_depth +Number_of_samples + +For example full key look like this @code{lavfi.astats.1.DC_offset} or +this @code{lavfi.astats.Overall.Peak_count}. + +For description what each key means read below. -For example, -@example -avconv -i INPUT -filter_complex asplit=5 OUTPUT -@end example -will create 5 copies of the input audio. +@item reset +Set number of frame after which stats are going to be recalculated. +Default is disabled. +@end table + +A description of each shown parameter follows: + +@table @option +@item DC offset +Mean amplitude displacement from zero. + +@item Min level +Minimal sample level. + +@item Max level +Maximal sample level. + +@item Min difference +Minimal difference between two consecutive samples. + +@item Max difference +Maximal difference between two consecutive samples. + +@item Mean difference +Mean difference between two consecutive samples. +The average of each difference between two consecutive samples. + +@item Peak level dB +@item RMS level dB +Standard peak and RMS level measured in dBFS. + +@item RMS peak dB +@item RMS trough dB +Peak and trough values for RMS level measured over a short window. + +@item Crest factor +Standard ratio of peak to RMS level (note: not in dB). + +@item Flat factor +Flatness (i.e. consecutive samples with the same value) of the signal at its peak levels +(i.e. either @var{Min level} or @var{Max level}). + +@item Peak count +Number of occasions (not the number of samples) that the signal attained either +@var{Min level} or @var{Max level}. + +@item Bit depth +Overall bit depth of audio. Number of bits used for each sample. +@end table @section asyncts + Synchronize audio data with timestamps by squeezing/stretching it and/or dropping samples/adding silence when needed. +This filter is not built by default, please use @ref{aresample} to do squeezing/stretching. + It accepts the following parameters: @table @option @@ -384,7 +1571,32 @@ with a negative PTS due to encoder delay. @end table +@section atempo + +Adjust audio tempo. + +The filter accepts exactly one parameter, the audio tempo. If not +specified then the filter will assume nominal 1.0 tempo. Tempo must +be in the [0.5, 2.0] range. + +@subsection Examples + +@itemize +@item +Slow down audio to 80% tempo: +@example +atempo=0.8 +@end example + +@item +To speed up audio to 125% tempo: +@example +atempo=1.25 +@end example +@end itemize + @section atrim + Trim the input so that the output contains one continuous subpart of the input. It accepts the following parameters: @@ -394,7 +1606,7 @@ Timestamp (in seconds) of the start of the section to keep. I.e. the audio sample with the timestamp @var{start} will be the first sample in the output. @item end -Timestamp (in seconds) of the first audio sample that will be dropped. I.e. the +Specify time of the first audio sample that will be dropped, i.e. the audio sample immediately preceding the one with the timestamp @var{end} will be the last sample in the output. @@ -416,6 +1628,10 @@ The number of the first sample that should be output. The number of the first sample that should be dropped. @end table +@option{start}, @option{end}, and @option{duration} are expressed as time +duration specifications; see +@ref{time duration syntax,,the Time duration section in the ffmpeg-utils(1) manual,ffmpeg-utils}. + Note that the first two sets of the start/end options and the @option{duration} option look at the frame timestamp, while the _sample options simply count the samples that pass through the filter. So start/end_pts and start/end_sample will @@ -437,17 +1653,122 @@ Examples: @item Drop everything except the second minute of input: @example -avconv -i INPUT -af atrim=60:120 +ffmpeg -i INPUT -af atrim=60:120 @end example @item Keep only the first 1000 samples: @example -avconv -i INPUT -af atrim=end_sample=1000 +ffmpeg -i INPUT -af atrim=end_sample=1000 @end example @end itemize +@section bandpass + +Apply a two-pole Butterworth band-pass filter with central +frequency @var{frequency}, and (3dB-point) band-width width. +The @var{csg} option selects a constant skirt gain (peak gain = Q) +instead of the default: constant 0dB peak gain. +The filter roll off at 6dB per octave (20dB per decade). + +The filter accepts the following options: + +@table @option +@item frequency, f +Set the filter's central frequency. Default is @code{3000}. + +@item csg +Constant skirt gain if set to 1. Defaults to 0. + +@item width_type +Set method to specify band-width of filter. +@table @option +@item h +Hz +@item q +Q-Factor +@item o +octave +@item s +slope +@end table + +@item width, w +Specify the band-width of a filter in width_type units. +@end table + +@section bandreject + +Apply a two-pole Butterworth band-reject filter with central +frequency @var{frequency}, and (3dB-point) band-width @var{width}. +The filter roll off at 6dB per octave (20dB per decade). + +The filter accepts the following options: + +@table @option +@item frequency, f +Set the filter's central frequency. Default is @code{3000}. + +@item width_type +Set method to specify band-width of filter. +@table @option +@item h +Hz +@item q +Q-Factor +@item o +octave +@item s +slope +@end table + +@item width, w +Specify the band-width of a filter in width_type units. +@end table + +@section bass + +Boost or cut the bass (lower) frequencies of the audio using a two-pole +shelving filter with a response similar to that of a standard +hi-fi's tone-controls. This is also known as shelving equalisation (EQ). + +The filter accepts the following options: + +@table @option +@item gain, g +Give the gain at 0 Hz. Its useful range is about -20 +(for a large cut) to +20 (for a large boost). +Beware of clipping when using a positive gain. + +@item frequency, f +Set the filter's central frequency and so can be used +to extend or reduce the frequency range to be boosted or cut. +The default value is @code{100} Hz. + +@item width_type +Set method to specify band-width of filter. +@table @option +@item h +Hz +@item q +Q-Factor +@item o +octave +@item s +slope +@end table + +@item width, w +Determine how steep is the filter's shelf transition. +@end table + +@section biquad + +Apply a biquad IIR filter with the given coefficients. +Where @var{b0}, @var{b1}, @var{b2} and @var{a0}, @var{a1}, @var{a2} +are the numerator and denominator coefficients respectively. + @section bs2b Bauer stereo to binaural transformation, which improves headphone listening of stereo audio records. @@ -478,7 +1799,42 @@ Feed level (in Hz). @end table +@section channelmap + +Remap input channels to new locations. + +It accepts the following parameters: +@table @option +@item channel_layout +The channel layout of the output stream. + +@item map +Map channels from input to output. The argument is a '|'-separated list of +mappings, each in the @code{@var{in_channel}-@var{out_channel}} or +@var{in_channel} form. @var{in_channel} can be either the name of the input +channel (e.g. FL for front left) or its index in the input channel layout. +@var{out_channel} is the name of the output channel or its index in the output +channel layout. If @var{out_channel} is not given then it is implicitly an +index, starting with zero and increasing by one for each mapping. +@end table + +If no mapping is present, the filter will implicitly map input channels to +output channels, preserving indices. + +For example, assuming a 5.1+downmix input MOV file, +@example +ffmpeg -i in.mov -filter 'channelmap=map=DL-FL|DR-FR' out.wav +@end example +will create an output WAV file tagged as stereo from the downmix channels of +the input. + +To fix a 5.1 WAV improperly encoded in AAC's native channel order +@example +ffmpeg -i in.wav -filter 'channelmap=1|2|0|5|3|4:5.1' out.wav +@end example + @section channelsplit + Split each channel from an input audio stream into a separate output stream. It accepts the following parameters: @@ -489,53 +1845,75 @@ The channel layout of the input stream. The default is "stereo". For example, assuming a stereo input MP3 file, @example -avconv -i in.mp3 -filter_complex channelsplit out.mkv +ffmpeg -i in.mp3 -filter_complex channelsplit out.mkv @end example will create an output Matroska file with two audio streams, one containing only the left channel and the other the right channel. Split a 5.1 WAV file into per-channel files: @example -avconv -i in.wav -filter_complex +ffmpeg -i in.wav -filter_complex 'channelsplit=channel_layout=5.1[FL][FR][FC][LFE][SL][SR]' -map '[FL]' front_left.wav -map '[FR]' front_right.wav -map '[FC]' front_center.wav -map '[LFE]' lfe.wav -map '[SL]' side_left.wav -map '[SR]' side_right.wav @end example -@section channelmap -Remap input channels to new locations. +@section chorus +Add a chorus effect to the audio. + +Can make a single vocal sound like a chorus, but can also be applied to instrumentation. + +Chorus resembles an echo effect with a short delay, but whereas with echo the delay is +constant, with chorus, it is varied using using sinusoidal or triangular modulation. +The modulation depth defines the range the modulated delay is played before or after +the delay. Hence the delayed sound will sound slower or faster, that is the delayed +sound tuned around the original one, like in a chorus where some vocals are slightly +off key. It accepts the following parameters: @table @option -@item channel_layout -The channel layout of the output stream. +@item in_gain +Set input gain. Default is 0.4. -@item map -Map channels from input to output. The argument is a '|'-separated list of -mappings, each in the @code{@var{in_channel}-@var{out_channel}} or -@var{in_channel} form. @var{in_channel} can be either the name of the input -channel (e.g. FL for front left) or its index in the input channel layout. -@var{out_channel} is the name of the output channel or its index in the output -channel layout. If @var{out_channel} is not given then it is implicitly an -index, starting with zero and increasing by one for each mapping. +@item out_gain +Set output gain. Default is 0.4. + +@item delays +Set delays. A typical delay is around 40ms to 60ms. + +@item decays +Set decays. + +@item speeds +Set speeds. + +@item depths +Set depths. @end table -If no mapping is present, the filter will implicitly map input channels to -output channels, preserving indices. +@subsection Examples -For example, assuming a 5.1+downmix input MOV file, +@itemize +@item +A single delay: @example -avconv -i in.mov -filter 'channelmap=map=DL-FL|DR-FR' out.wav +chorus=0.7:0.9:55:0.4:0.25:2 @end example -will create an output WAV file tagged as stereo from the downmix channels of -the input. -To fix a 5.1 WAV improperly encoded in AAC's native channel order +@item +Two delays: @example -avconv -i in.wav -filter 'channelmap=1|2|0|5|3|4:5.1' out.wav +chorus=0.6:0.9:50|60:0.4|0.32:0.25|0.4:2|1.3 @end example +@item +Fuller sounding chorus with three delays: +@example +chorus=0.5:0.9:50|60|40:0.4|0.32|0.3:0.25|0.4|0.3:2|2.3|1.3 +@end example +@end itemize + @section compand Compress or expand the audio's dynamic range. @@ -552,11 +1930,14 @@ situations, the attack time (response to the audio getting louder) should be shorter than the decay time, because the human ear is more sensitive to sudden loud audio than sudden soft audio. A typical value for attack is 0.3 seconds and a typical value for decay is 0.8 seconds. +If specified number of attacks & decays is lower than number of channels, the last +set attack/decay will be used for all remaining channels. @item points A list of points for the transfer function, specified in dB relative to the maximum possible signal amplitude. Each key points list must be defined using -the following syntax: @code{x0/y0|x1/y1|x2/y2|....} +the following syntax: @code{x0/y0|x1/y1|x2/y2|....} or +@code{x0/y0 x1/y1 x2/y2 ....} The input values must be in strictly increasing order but the transfer function does not have to be monotonically rising. The point @code{0/0} is assumed but @@ -596,6 +1977,11 @@ noisy environment: compand=.3|.3:1|1:-90/-60|-60/-40|-40/-30|-20/-20:6:0:-90:0.2 @end example +Another example for audio with whisper and explosion parts: +@example +compand=0|0:1|1:-90/-900|-70/-70|-30/-9|0/-3:6:0:0:0 +@end example + @item A noise gate for when the noise is at a lower level than the signal: @example @@ -608,9 +1994,450 @@ than the signal (making it, in some ways, similar to squelch): @example compand=.1|.1:.1|.1:-45.1/-45.1|-45/-900|0/-900:.01:45:-90:.1 @end example + +@item +2:1 compression starting at -6dB: +@example +compand=points=-80/-80|-6/-6|0/-3.8|20/3.5 +@end example + +@item +2:1 compression starting at -9dB: +@example +compand=points=-80/-80|-9/-9|0/-5.3|20/2.9 +@end example + +@item +2:1 compression starting at -12dB: +@example +compand=points=-80/-80|-12/-12|0/-6.8|20/1.9 +@end example + +@item +2:1 compression starting at -18dB: +@example +compand=points=-80/-80|-18/-18|0/-9.8|20/0.7 +@end example + +@item +3:1 compression starting at -15dB: +@example +compand=points=-80/-80|-15/-15|0/-10.8|20/-5.2 +@end example + +@item +Compressor/Gate: +@example +compand=points=-80/-105|-62/-80|-15.4/-15.4|0/-12|20/-7.6 +@end example + +@item +Expander: +@example +compand=attacks=0:points=-80/-169|-54/-80|-49.5/-64.6|-41.1/-41.1|-25.8/-15|-10.8/-4.5|0/0|20/8.3 +@end example + +@item +Hard limiter at -6dB: +@example +compand=attacks=0:points=-80/-80|-6/-6|20/-6 +@end example + +@item +Hard limiter at -12dB: +@example +compand=attacks=0:points=-80/-80|-12/-12|20/-12 +@end example + +@item +Hard noise gate at -35 dB: +@example +compand=attacks=0:points=-80/-115|-35.1/-80|-35/-35|20/20 +@end example + +@item +Soft limiter: +@example +compand=attacks=0:points=-80/-80|-12.4/-12.4|-6/-8|0/-6.8|20/-2.8 +@end example @end itemize +@section compensationdelay + +Compensation Delay Line is a metric based delay to compensate differing +positions of microphones or speakers. + +For example, you have recorded guitar with two microphones placed in +different location. Because the front of sound wave has fixed speed in +normal conditions, the phasing of microphones can vary and depends on +their location and interposition. The best sound mix can be achieved when +these microphones are in phase (synchronized). Note that distance of +~30 cm between microphones makes one microphone to capture signal in +antiphase to another microphone. That makes the final mix sounding moody. +This filter helps to solve phasing problems by adding different delays +to each microphone track and make them synchronized. + +The best result can be reached when you take one track as base and +synchronize other tracks one by one with it. +Remember that synchronization/delay tolerance depends on sample rate, too. +Higher sample rates will give more tolerance. + +It accepts the following parameters: + +@table @option +@item mm +Set millimeters distance. This is compensation distance for fine tuning. +Default is 0. + +@item cm +Set cm distance. This is compensation distance for tightening distance setup. +Default is 0. + +@item m +Set meters distance. This is compensation distance for hard distance setup. +Default is 0. + +@item dry +Set dry amount. Amount of unprocessed (dry) signal. +Default is 0. + +@item wet +Set wet amount. Amount of processed (wet) signal. +Default is 1. + +@item temp +Set temperature degree in Celsius. This is the temperature of the environment. +Default is 20. +@end table + +@section dcshift +Apply a DC shift to the audio. + +This can be useful to remove a DC offset (caused perhaps by a hardware problem +in the recording chain) from the audio. The effect of a DC offset is reduced +headroom and hence volume. The @ref{astats} filter can be used to determine if +a signal has a DC offset. + +@table @option +@item shift +Set the DC shift, allowed range is [-1, 1]. It indicates the amount to shift +the audio. + +@item limitergain +Optional. It should have a value much less than 1 (e.g. 0.05 or 0.02) and is +used to prevent clipping. +@end table + +@section dynaudnorm +Dynamic Audio Normalizer. + +This filter applies a certain amount of gain to the input audio in order +to bring its peak magnitude to a target level (e.g. 0 dBFS). However, in +contrast to more "simple" normalization algorithms, the Dynamic Audio +Normalizer *dynamically* re-adjusts the gain factor to the input audio. +This allows for applying extra gain to the "quiet" sections of the audio +while avoiding distortions or clipping the "loud" sections. In other words: +The Dynamic Audio Normalizer will "even out" the volume of quiet and loud +sections, in the sense that the volume of each section is brought to the +same target level. Note, however, that the Dynamic Audio Normalizer achieves +this goal *without* applying "dynamic range compressing". It will retain 100% +of the dynamic range *within* each section of the audio file. + +@table @option +@item f +Set the frame length in milliseconds. In range from 10 to 8000 milliseconds. +Default is 500 milliseconds. +The Dynamic Audio Normalizer processes the input audio in small chunks, +referred to as frames. This is required, because a peak magnitude has no +meaning for just a single sample value. Instead, we need to determine the +peak magnitude for a contiguous sequence of sample values. While a "standard" +normalizer would simply use the peak magnitude of the complete file, the +Dynamic Audio Normalizer determines the peak magnitude individually for each +frame. The length of a frame is specified in milliseconds. By default, the +Dynamic Audio Normalizer uses a frame length of 500 milliseconds, which has +been found to give good results with most files. +Note that the exact frame length, in number of samples, will be determined +automatically, based on the sampling rate of the individual input audio file. + +@item g +Set the Gaussian filter window size. In range from 3 to 301, must be odd +number. Default is 31. +Probably the most important parameter of the Dynamic Audio Normalizer is the +@code{window size} of the Gaussian smoothing filter. The filter's window size +is specified in frames, centered around the current frame. For the sake of +simplicity, this must be an odd number. Consequently, the default value of 31 +takes into account the current frame, as well as the 15 preceding frames and +the 15 subsequent frames. Using a larger window results in a stronger +smoothing effect and thus in less gain variation, i.e. slower gain +adaptation. Conversely, using a smaller window results in a weaker smoothing +effect and thus in more gain variation, i.e. faster gain adaptation. +In other words, the more you increase this value, the more the Dynamic Audio +Normalizer will behave like a "traditional" normalization filter. On the +contrary, the more you decrease this value, the more the Dynamic Audio +Normalizer will behave like a dynamic range compressor. + +@item p +Set the target peak value. This specifies the highest permissible magnitude +level for the normalized audio input. This filter will try to approach the +target peak magnitude as closely as possible, but at the same time it also +makes sure that the normalized signal will never exceed the peak magnitude. +A frame's maximum local gain factor is imposed directly by the target peak +magnitude. The default value is 0.95 and thus leaves a headroom of 5%*. +It is not recommended to go above this value. + +@item m +Set the maximum gain factor. In range from 1.0 to 100.0. Default is 10.0. +The Dynamic Audio Normalizer determines the maximum possible (local) gain +factor for each input frame, i.e. the maximum gain factor that does not +result in clipping or distortion. The maximum gain factor is determined by +the frame's highest magnitude sample. However, the Dynamic Audio Normalizer +additionally bounds the frame's maximum gain factor by a predetermined +(global) maximum gain factor. This is done in order to avoid excessive gain +factors in "silent" or almost silent frames. By default, the maximum gain +factor is 10.0, For most inputs the default value should be sufficient and +it usually is not recommended to increase this value. Though, for input +with an extremely low overall volume level, it may be necessary to allow even +higher gain factors. Note, however, that the Dynamic Audio Normalizer does +not simply apply a "hard" threshold (i.e. cut off values above the threshold). +Instead, a "sigmoid" threshold function will be applied. This way, the +gain factors will smoothly approach the threshold value, but never exceed that +value. + +@item r +Set the target RMS. In range from 0.0 to 1.0. Default is 0.0 - disabled. +By default, the Dynamic Audio Normalizer performs "peak" normalization. +This means that the maximum local gain factor for each frame is defined +(only) by the frame's highest magnitude sample. This way, the samples can +be amplified as much as possible without exceeding the maximum signal +level, i.e. without clipping. Optionally, however, the Dynamic Audio +Normalizer can also take into account the frame's root mean square, +abbreviated RMS. In electrical engineering, the RMS is commonly used to +determine the power of a time-varying signal. It is therefore considered +that the RMS is a better approximation of the "perceived loudness" than +just looking at the signal's peak magnitude. Consequently, by adjusting all +frames to a constant RMS value, a uniform "perceived loudness" can be +established. If a target RMS value has been specified, a frame's local gain +factor is defined as the factor that would result in exactly that RMS value. +Note, however, that the maximum local gain factor is still restricted by the +frame's highest magnitude sample, in order to prevent clipping. + +@item n +Enable channels coupling. By default is enabled. +By default, the Dynamic Audio Normalizer will amplify all channels by the same +amount. This means the same gain factor will be applied to all channels, i.e. +the maximum possible gain factor is determined by the "loudest" channel. +However, in some recordings, it may happen that the volume of the different +channels is uneven, e.g. one channel may be "quieter" than the other one(s). +In this case, this option can be used to disable the channel coupling. This way, +the gain factor will be determined independently for each channel, depending +only on the individual channel's highest magnitude sample. This allows for +harmonizing the volume of the different channels. + +@item c +Enable DC bias correction. By default is disabled. +An audio signal (in the time domain) is a sequence of sample values. +In the Dynamic Audio Normalizer these sample values are represented in the +-1.0 to 1.0 range, regardless of the original input format. Normally, the +audio signal, or "waveform", should be centered around the zero point. +That means if we calculate the mean value of all samples in a file, or in a +single frame, then the result should be 0.0 or at least very close to that +value. If, however, there is a significant deviation of the mean value from +0.0, in either positive or negative direction, this is referred to as a +DC bias or DC offset. Since a DC bias is clearly undesirable, the Dynamic +Audio Normalizer provides optional DC bias correction. +With DC bias correction enabled, the Dynamic Audio Normalizer will determine +the mean value, or "DC correction" offset, of each input frame and subtract +that value from all of the frame's sample values which ensures those samples +are centered around 0.0 again. Also, in order to avoid "gaps" at the frame +boundaries, the DC correction offset values will be interpolated smoothly +between neighbouring frames. + +@item b +Enable alternative boundary mode. By default is disabled. +The Dynamic Audio Normalizer takes into account a certain neighbourhood +around each frame. This includes the preceding frames as well as the +subsequent frames. However, for the "boundary" frames, located at the very +beginning and at the very end of the audio file, not all neighbouring +frames are available. In particular, for the first few frames in the audio +file, the preceding frames are not known. And, similarly, for the last few +frames in the audio file, the subsequent frames are not known. Thus, the +question arises which gain factors should be assumed for the missing frames +in the "boundary" region. The Dynamic Audio Normalizer implements two modes +to deal with this situation. The default boundary mode assumes a gain factor +of exactly 1.0 for the missing frames, resulting in a smooth "fade in" and +"fade out" at the beginning and at the end of the input, respectively. + +@item s +Set the compress factor. In range from 0.0 to 30.0. Default is 0.0. +By default, the Dynamic Audio Normalizer does not apply "traditional" +compression. This means that signal peaks will not be pruned and thus the +full dynamic range will be retained within each local neighbourhood. However, +in some cases it may be desirable to combine the Dynamic Audio Normalizer's +normalization algorithm with a more "traditional" compression. +For this purpose, the Dynamic Audio Normalizer provides an optional compression +(thresholding) function. If (and only if) the compression feature is enabled, +all input frames will be processed by a soft knee thresholding function prior +to the actual normalization process. Put simply, the thresholding function is +going to prune all samples whose magnitude exceeds a certain threshold value. +However, the Dynamic Audio Normalizer does not simply apply a fixed threshold +value. Instead, the threshold value will be adjusted for each individual +frame. +In general, smaller parameters result in stronger compression, and vice versa. +Values below 3.0 are not recommended, because audible distortion may appear. +@end table + +@section earwax + +Make audio easier to listen to on headphones. + +This filter adds `cues' to 44.1kHz stereo (i.e. audio CD format) audio +so that when listened to on headphones the stereo image is moved from +inside your head (standard for headphones) to outside and in front of +the listener (standard for speakers). + +Ported from SoX. + +@section equalizer + +Apply a two-pole peaking equalisation (EQ) filter. With this +filter, the signal-level at and around a selected frequency can +be increased or decreased, whilst (unlike bandpass and bandreject +filters) that at all other frequencies is unchanged. + +In order to produce complex equalisation curves, this filter can +be given several times, each with a different central frequency. + +The filter accepts the following options: + +@table @option +@item frequency, f +Set the filter's central frequency in Hz. + +@item width_type +Set method to specify band-width of filter. +@table @option +@item h +Hz +@item q +Q-Factor +@item o +octave +@item s +slope +@end table + +@item width, w +Specify the band-width of a filter in width_type units. + +@item gain, g +Set the required gain or attenuation in dB. +Beware of clipping when using a positive gain. +@end table + +@subsection Examples +@itemize +@item +Attenuate 10 dB at 1000 Hz, with a bandwidth of 200 Hz: +@example +equalizer=f=1000:width_type=h:width=200:g=-10 +@end example + +@item +Apply 2 dB gain at 1000 Hz with Q 1 and attenuate 5 dB at 100 Hz with Q 2: +@example +equalizer=f=1000:width_type=q:width=1:g=2,equalizer=f=100:width_type=q:width=2:g=-5 +@end example +@end itemize + +@section extrastereo + +Linearly increases the difference between left and right channels which +adds some sort of "live" effect to playback. + +The filter accepts the following option: + +@table @option +@item m +Sets the difference coefficient (default: 2.5). 0.0 means mono sound +(average of both channels), with 1.0 sound will be unchanged, with +-1.0 left and right channels will be swapped. + +@item c +Enable clipping. By default is enabled. +@end table + +@section flanger +Apply a flanging effect to the audio. + +The filter accepts the following options: + +@table @option +@item delay +Set base delay in milliseconds. Range from 0 to 30. Default value is 0. + +@item depth +Set added swep delay in milliseconds. Range from 0 to 10. Default value is 2. + +@item regen +Set percentage regeneration (delayed signal feedback). Range from -95 to 95. +Default value is 0. + +@item width +Set percentage of delayed signal mixed with original. Range from 0 to 100. +Default value is 71. + +@item speed +Set sweeps per second (Hz). Range from 0.1 to 10. Default value is 0.5. + +@item shape +Set swept wave shape, can be @var{triangular} or @var{sinusoidal}. +Default value is @var{sinusoidal}. + +@item phase +Set swept wave percentage-shift for multi channel. Range from 0 to 100. +Default value is 25. + +@item interp +Set delay-line interpolation, @var{linear} or @var{quadratic}. +Default is @var{linear}. +@end table + +@section highpass + +Apply a high-pass filter with 3dB point frequency. +The filter can be either single-pole, or double-pole (the default). +The filter roll off at 6dB per pole per octave (20dB per pole per decade). + +The filter accepts the following options: + +@table @option +@item frequency, f +Set frequency in Hz. Default is 3000. + +@item poles, p +Set number of poles. Default is 2. + +@item width_type +Set method to specify band-width of filter. +@table @option +@item h +Hz +@item q +Q-Factor +@item o +octave +@item s +slope +@end table + +@item width, w +Specify the band-width of a filter in width_type units. +Applies only to double-pole filter. +The default is 0.707q and gives a Butterworth response. +@end table + @section join + Join multiple input streams into one multi-channel stream. It accepts the following parameters: @@ -637,21 +2464,849 @@ and if that fails it picks the first unused input channel. Join 3 inputs (with properly set channel layouts): @example -avconv -i INPUT1 -i INPUT2 -i INPUT3 -filter_complex join=inputs=3 OUTPUT +ffmpeg -i INPUT1 -i INPUT2 -i INPUT3 -filter_complex join=inputs=3 OUTPUT @end example Build a 5.1 output from 6 single-channel streams: @example -avconv -i fl -i fr -i fc -i sl -i sr -i lfe -filter_complex +ffmpeg -i fl -i fr -i fc -i sl -i sr -i lfe -filter_complex 'join=inputs=6:channel_layout=5.1:map=0.0-FL|1.0-FR|2.0-FC|3.0-SL|4.0-SR|5.0-LFE' out @end example +@section ladspa + +Load a LADSPA (Linux Audio Developer's Simple Plugin API) plugin. + +To enable compilation of this filter you need to configure FFmpeg with +@code{--enable-ladspa}. + +@table @option +@item file, f +Specifies the name of LADSPA plugin library to load. If the environment +variable @env{LADSPA_PATH} is defined, the LADSPA plugin is searched in +each one of the directories specified by the colon separated list in +@env{LADSPA_PATH}, otherwise in the standard LADSPA paths, which are in +this order: @file{HOME/.ladspa/lib/}, @file{/usr/local/lib/ladspa/}, +@file{/usr/lib/ladspa/}. + +@item plugin, p +Specifies the plugin within the library. Some libraries contain only +one plugin, but others contain many of them. If this is not set filter +will list all available plugins within the specified library. + +@item controls, c +Set the '|' separated list of controls which are zero or more floating point +values that determine the behavior of the loaded plugin (for example delay, +threshold or gain). +Controls need to be defined using the following syntax: +c0=@var{value0}|c1=@var{value1}|c2=@var{value2}|..., where +@var{valuei} is the value set on the @var{i}-th control. +Alternatively they can be also defined using the following syntax: +@var{value0}|@var{value1}|@var{value2}|..., where +@var{valuei} is the value set on the @var{i}-th control. +If @option{controls} is set to @code{help}, all available controls and +their valid ranges are printed. + +@item sample_rate, s +Specify the sample rate, default to 44100. Only used if plugin have +zero inputs. + +@item nb_samples, n +Set the number of samples per channel per each output frame, default +is 1024. Only used if plugin have zero inputs. + +@item duration, d +Set the minimum duration of the sourced audio. See +@ref{time duration syntax,,the Time duration section in the ffmpeg-utils(1) manual,ffmpeg-utils} +for the accepted syntax. +Note that the resulting duration may be greater than the specified duration, +as the generated audio is always cut at the end of a complete frame. +If not specified, or the expressed duration is negative, the audio is +supposed to be generated forever. +Only used if plugin have zero inputs. + +@end table + +@subsection Examples + +@itemize +@item +List all available plugins within amp (LADSPA example plugin) library: +@example +ladspa=file=amp +@end example + +@item +List all available controls and their valid ranges for @code{vcf_notch} +plugin from @code{VCF} library: +@example +ladspa=f=vcf:p=vcf_notch:c=help +@end example + +@item +Simulate low quality audio equipment using @code{Computer Music Toolkit} (CMT) +plugin library: +@example +ladspa=file=cmt:plugin=lofi:controls=c0=22|c1=12|c2=12 +@end example + +@item +Add reverberation to the audio using TAP-plugins +(Tom's Audio Processing plugins): +@example +ladspa=file=tap_reverb:tap_reverb +@end example + +@item +Generate white noise, with 0.2 amplitude: +@example +ladspa=file=cmt:noise_source_white:c=c0=.2 +@end example + +@item +Generate 20 bpm clicks using plugin @code{C* Click - Metronome} from the +@code{C* Audio Plugin Suite} (CAPS) library: +@example +ladspa=file=caps:Click:c=c1=20' +@end example + +@item +Apply @code{C* Eq10X2 - Stereo 10-band equaliser} effect: +@example +ladspa=caps:Eq10X2:c=c0=-48|c9=-24|c3=12|c4=2 +@end example + +@item +Increase volume by 20dB using fast lookahead limiter from Steve Harris +@code{SWH Plugins} collection: +@example +ladspa=fast_lookahead_limiter_1913:fastLookaheadLimiter:20|0|2 +@end example + +@item +Attenuate low frequencies using Multiband EQ from Steve Harris +@code{SWH Plugins} collection: +@example +ladspa=mbeq_1197:mbeq:-24|-24|-24|0|0|0|0|0|0|0|0|0|0|0|0 +@end example +@end itemize + +@subsection Commands + +This filter supports the following commands: +@table @option +@item cN +Modify the @var{N}-th control value. + +If the specified value is not valid, it is ignored and prior one is kept. +@end table + +@section lowpass + +Apply a low-pass filter with 3dB point frequency. +The filter can be either single-pole or double-pole (the default). +The filter roll off at 6dB per pole per octave (20dB per pole per decade). + +The filter accepts the following options: + +@table @option +@item frequency, f +Set frequency in Hz. Default is 500. + +@item poles, p +Set number of poles. Default is 2. + +@item width_type +Set method to specify band-width of filter. +@table @option +@item h +Hz +@item q +Q-Factor +@item o +octave +@item s +slope +@end table + +@item width, w +Specify the band-width of a filter in width_type units. +Applies only to double-pole filter. +The default is 0.707q and gives a Butterworth response. +@end table + +@anchor{pan} +@section pan + +Mix channels with specific gain levels. The filter accepts the output +channel layout followed by a set of channels definitions. + +This filter is also designed to efficiently remap the channels of an audio +stream. + +The filter accepts parameters of the form: +"@var{l}|@var{outdef}|@var{outdef}|..." + +@table @option +@item l +output channel layout or number of channels + +@item outdef +output channel specification, of the form: +"@var{out_name}=[@var{gain}*]@var{in_name}[+[@var{gain}*]@var{in_name}...]" + +@item out_name +output channel to define, either a channel name (FL, FR, etc.) or a channel +number (c0, c1, etc.) + +@item gain +multiplicative coefficient for the channel, 1 leaving the volume unchanged + +@item in_name +input channel to use, see out_name for details; it is not possible to mix +named and numbered input channels +@end table + +If the `=' in a channel specification is replaced by `<', then the gains for +that specification will be renormalized so that the total is 1, thus +avoiding clipping noise. + +@subsection Mixing examples + +For example, if you want to down-mix from stereo to mono, but with a bigger +factor for the left channel: +@example +pan=1c|c0=0.9*c0+0.1*c1 +@end example + +A customized down-mix to stereo that works automatically for 3-, 4-, 5- and +7-channels surround: +@example +pan=stereo| FL < FL + 0.5*FC + 0.6*BL + 0.6*SL | FR < FR + 0.5*FC + 0.6*BR + 0.6*SR +@end example + +Note that @command{ffmpeg} integrates a default down-mix (and up-mix) system +that should be preferred (see "-ac" option) unless you have very specific +needs. + +@subsection Remapping examples + +The channel remapping will be effective if, and only if: + +@itemize +@item gain coefficients are zeroes or ones, +@item only one input per channel output, +@end itemize + +If all these conditions are satisfied, the filter will notify the user ("Pure +channel mapping detected"), and use an optimized and lossless method to do the +remapping. + +For example, if you have a 5.1 source and want a stereo audio stream by +dropping the extra channels: +@example +pan="stereo| c0=FL | c1=FR" +@end example + +Given the same source, you can also switch front left and front right channels +and keep the input channel layout: +@example +pan="5.1| c0=c1 | c1=c0 | c2=c2 | c3=c3 | c4=c4 | c5=c5" +@end example + +If the input is a stereo audio stream, you can mute the front left channel (and +still keep the stereo channel layout) with: +@example +pan="stereo|c1=c1" +@end example + +Still with a stereo audio stream input, you can copy the right channel in both +front left and right: +@example +pan="stereo| c0=FR | c1=FR" +@end example + +@section replaygain + +ReplayGain scanner filter. This filter takes an audio stream as an input and +outputs it unchanged. +At end of filtering it displays @code{track_gain} and @code{track_peak}. + @section resample + Convert the audio sample format, sample rate and channel layout. It is -not meant to be used directly; it is inserted automatically by libavfilter -whenever conversion is needed. Use the @var{aformat} filter to force a specific -conversion. +not meant to be used directly. + +@section rubberband +Apply time-stretching and pitch-shifting with librubberband. + +The filter accepts the following options: + +@table @option +@item tempo +Set tempo scale factor. + +@item pitch +Set pitch scale factor. + +@item transients +Set transients detector. +Possible values are: +@table @var +@item crisp +@item mixed +@item smooth +@end table + +@item detector +Set detector. +Possible values are: +@table @var +@item compound +@item percussive +@item soft +@end table + +@item phase +Set phase. +Possible values are: +@table @var +@item laminar +@item independent +@end table + +@item window +Set processing window size. +Possible values are: +@table @var +@item standard +@item short +@item long +@end table + +@item smoothing +Set smoothing. +Possible values are: +@table @var +@item off +@item on +@end table + +@item formant +Enable formant preservation when shift pitching. +Possible values are: +@table @var +@item shifted +@item preserved +@end table + +@item pitchq +Set pitch quality. +Possible values are: +@table @var +@item quality +@item speed +@item consistency +@end table + +@item channels +Set channels. +Possible values are: +@table @var +@item apart +@item together +@end table +@end table + +@section sidechaincompress + +This filter acts like normal compressor but has the ability to compress +detected signal using second input signal. +It needs two input streams and returns one output stream. +First input stream will be processed depending on second stream signal. +The filtered signal then can be filtered with other filters in later stages of +processing. See @ref{pan} and @ref{amerge} filter. + +The filter accepts the following options: + +@table @option +@item level_in +Set input gain. Default is 1. Range is between 0.015625 and 64. + +@item threshold +If a signal of second stream raises above this level it will affect the gain +reduction of first stream. +By default is 0.125. Range is between 0.00097563 and 1. + +@item ratio +Set a ratio about which the signal is reduced. 1:2 means that if the level +raised 4dB above the threshold, it will be only 2dB above after the reduction. +Default is 2. Range is between 1 and 20. + +@item attack +Amount of milliseconds the signal has to rise above the threshold before gain +reduction starts. Default is 20. Range is between 0.01 and 2000. + +@item release +Amount of milliseconds the signal has to fall below the threshold before +reduction is decreased again. Default is 250. Range is between 0.01 and 9000. + +@item makeup +Set the amount by how much signal will be amplified after processing. +Default is 2. Range is from 1 and 64. + +@item knee +Curve the sharp knee around the threshold to enter gain reduction more softly. +Default is 2.82843. Range is between 1 and 8. + +@item link +Choose if the @code{average} level between all channels of side-chain stream +or the louder(@code{maximum}) channel of side-chain stream affects the +reduction. Default is @code{average}. + +@item detection +Should the exact signal be taken in case of @code{peak} or an RMS one in case +of @code{rms}. Default is @code{rms} which is mainly smoother. + +@item level_sc +Set sidechain gain. Default is 1. Range is between 0.015625 and 64. + +@item mix +How much to use compressed signal in output. Default is 1. +Range is between 0 and 1. +@end table + +@subsection Examples + +@itemize +@item +Full ffmpeg example taking 2 audio inputs, 1st input to be compressed +depending on the signal of 2nd input and later compressed signal to be +merged with 2nd input: +@example +ffmpeg -i main.flac -i sidechain.flac -filter_complex "[1:a]asplit=2[sc][mix];[0:a][sc]sidechaincompress[compr];[compr][mix]amerge" +@end example +@end itemize + +@section sidechaingate + +A sidechain gate acts like a normal (wideband) gate but has the ability to +filter the detected signal before sending it to the gain reduction stage. +Normally a gate uses the full range signal to detect a level above the +threshold. +For example: If you cut all lower frequencies from your sidechain signal +the gate will decrease the volume of your track only if not enough highs +appear. With this technique you are able to reduce the resonation of a +natural drum or remove "rumbling" of muted strokes from a heavily distorted +guitar. +It needs two input streams and returns one output stream. +First input stream will be processed depending on second stream signal. + +The filter accepts the following options: + +@table @option +@item level_in +Set input level before filtering. +Default is 1. Allowed range is from 0.015625 to 64. + +@item range +Set the level of gain reduction when the signal is below the threshold. +Default is 0.06125. Allowed range is from 0 to 1. + +@item threshold +If a signal rises above this level the gain reduction is released. +Default is 0.125. Allowed range is from 0 to 1. + +@item ratio +Set a ratio about which the signal is reduced. +Default is 2. Allowed range is from 1 to 9000. + +@item attack +Amount of milliseconds the signal has to rise above the threshold before gain +reduction stops. +Default is 20 milliseconds. Allowed range is from 0.01 to 9000. + +@item release +Amount of milliseconds the signal has to fall below the threshold before the +reduction is increased again. Default is 250 milliseconds. +Allowed range is from 0.01 to 9000. + +@item makeup +Set amount of amplification of signal after processing. +Default is 1. Allowed range is from 1 to 64. + +@item knee +Curve the sharp knee around the threshold to enter gain reduction more softly. +Default is 2.828427125. Allowed range is from 1 to 8. + +@item detection +Choose if exact signal should be taken for detection or an RMS like one. +Default is rms. Can be peak or rms. + +@item link +Choose if the average level between all channels or the louder channel affects +the reduction. +Default is average. Can be average or maximum. + +@item level_sc +Set sidechain gain. Default is 1. Range is from 0.015625 to 64. +@end table + +@section silencedetect + +Detect silence in an audio stream. + +This filter logs a message when it detects that the input audio volume is less +or equal to a noise tolerance value for a duration greater or equal to the +minimum detected noise duration. + +The printed times and duration are expressed in seconds. + +The filter accepts the following options: + +@table @option +@item duration, d +Set silence duration until notification (default is 2 seconds). + +@item noise, n +Set noise tolerance. Can be specified in dB (in case "dB" is appended to the +specified value) or amplitude ratio. Default is -60dB, or 0.001. +@end table + +@subsection Examples + +@itemize +@item +Detect 5 seconds of silence with -50dB noise tolerance: +@example +silencedetect=n=-50dB:d=5 +@end example + +@item +Complete example with @command{ffmpeg} to detect silence with 0.0001 noise +tolerance in @file{silence.mp3}: +@example +ffmpeg -i silence.mp3 -af silencedetect=noise=0.0001 -f null - +@end example +@end itemize + +@section silenceremove + +Remove silence from the beginning, middle or end of the audio. + +The filter accepts the following options: + +@table @option +@item start_periods +This value is used to indicate if audio should be trimmed at beginning of +the audio. A value of zero indicates no silence should be trimmed from the +beginning. When specifying a non-zero value, it trims audio up until it +finds non-silence. Normally, when trimming silence from beginning of audio +the @var{start_periods} will be @code{1} but it can be increased to higher +values to trim all audio up to specific count of non-silence periods. +Default value is @code{0}. + +@item start_duration +Specify the amount of time that non-silence must be detected before it stops +trimming audio. By increasing the duration, bursts of noises can be treated +as silence and trimmed off. Default value is @code{0}. + +@item start_threshold +This indicates what sample value should be treated as silence. For digital +audio, a value of @code{0} may be fine but for audio recorded from analog, +you may wish to increase the value to account for background noise. +Can be specified in dB (in case "dB" is appended to the specified value) +or amplitude ratio. Default value is @code{0}. + +@item stop_periods +Set the count for trimming silence from the end of audio. +To remove silence from the middle of a file, specify a @var{stop_periods} +that is negative. This value is then treated as a positive value and is +used to indicate the effect should restart processing as specified by +@var{start_periods}, making it suitable for removing periods of silence +in the middle of the audio. +Default value is @code{0}. + +@item stop_duration +Specify a duration of silence that must exist before audio is not copied any +more. By specifying a higher duration, silence that is wanted can be left in +the audio. +Default value is @code{0}. + +@item stop_threshold +This is the same as @option{start_threshold} but for trimming silence from +the end of audio. +Can be specified in dB (in case "dB" is appended to the specified value) +or amplitude ratio. Default value is @code{0}. + +@item leave_silence +This indicate that @var{stop_duration} length of audio should be left intact +at the beginning of each period of silence. +For example, if you want to remove long pauses between words but do not want +to remove the pauses completely. Default value is @code{0}. + +@item detection +Set how is silence detected. Can be @code{rms} or @code{peak}. Second is faster +and works better with digital silence which is exactly 0. +Default value is @code{rms}. + +@item window +Set ratio used to calculate size of window for detecting silence. +Default value is @code{0.02}. Allowed range is from @code{0} to @code{10}. +@end table + +@subsection Examples + +@itemize +@item +The following example shows how this filter can be used to start a recording +that does not contain the delay at the start which usually occurs between +pressing the record button and the start of the performance: +@example +silenceremove=1:5:0.02 +@end example + +@item +Trim all silence encountered from begining to end where there is more than 1 +second of silence in audio: +@example +silenceremove=0:0:0:-1:1:-90dB +@end example +@end itemize + +@section sofalizer + +SOFAlizer uses head-related transfer functions (HRTFs) to create virtual +loudspeakers around the user for binaural listening via headphones (audio +formats up to 9 channels supported). +The HRTFs are stored in SOFA files (see @url{http://www.sofacoustics.org/} for a database). +SOFAlizer is developed at the Acoustics Research Institute (ARI) of the +Austrian Academy of Sciences. + +To enable compilation of this filter you need to configure FFmpeg with +@code{--enable-netcdf}. + +The filter accepts the following options: + +@table @option +@item sofa +Set the SOFA file used for rendering. + +@item gain +Set gain applied to audio. Value is in dB. Default is 0. + +@item rotation +Set rotation of virtual loudspeakers in deg. Default is 0. + +@item elevation +Set elevation of virtual speakers in deg. Default is 0. + +@item radius +Set distance in meters between loudspeakers and the listener with near-field +HRTFs. Default is 1. + +@item type +Set processing type. Can be @var{time} or @var{freq}. @var{time} is +processing audio in time domain which is slow but gives high quality output. +@var{freq} is processing audio in frequency domain which is fast but gives +mediocre output. Default is @var{freq}. +@end table + +@section stereotools + +This filter has some handy utilities to manage stereo signals, for converting +M/S stereo recordings to L/R signal while having control over the parameters +or spreading the stereo image of master track. + +The filter accepts the following options: + +@table @option +@item level_in +Set input level before filtering for both channels. Defaults is 1. +Allowed range is from 0.015625 to 64. + +@item level_out +Set output level after filtering for both channels. Defaults is 1. +Allowed range is from 0.015625 to 64. + +@item balance_in +Set input balance between both channels. Default is 0. +Allowed range is from -1 to 1. + +@item balance_out +Set output balance between both channels. Default is 0. +Allowed range is from -1 to 1. + +@item softclip +Enable softclipping. Results in analog distortion instead of harsh digital 0dB +clipping. Disabled by default. + +@item mutel +Mute the left channel. Disabled by default. + +@item muter +Mute the right channel. Disabled by default. + +@item phasel +Change the phase of the left channel. Disabled by default. + +@item phaser +Change the phase of the right channel. Disabled by default. + +@item mode +Set stereo mode. Available values are: + +@table @samp +@item lr>lr +Left/Right to Left/Right, this is default. + +@item lr>ms +Left/Right to Mid/Side. + +@item ms>lr +Mid/Side to Left/Right. + +@item lr>ll +Left/Right to Left/Left. + +@item lr>rr +Left/Right to Right/Right. + +@item lr>l+r +Left/Right to Left + Right. + +@item lr>rl +Left/Right to Right/Left. +@end table + +@item slev +Set level of side signal. Default is 1. +Allowed range is from 0.015625 to 64. + +@item sbal +Set balance of side signal. Default is 0. +Allowed range is from -1 to 1. + +@item mlev +Set level of the middle signal. Default is 1. +Allowed range is from 0.015625 to 64. + +@item mpan +Set middle signal pan. Default is 0. Allowed range is from -1 to 1. + +@item base +Set stereo base between mono and inversed channels. Default is 0. +Allowed range is from -1 to 1. + +@item delay +Set delay in milliseconds how much to delay left from right channel and +vice versa. Default is 0. Allowed range is from -20 to 20. + +@item sclevel +Set S/C level. Default is 1. Allowed range is from 1 to 100. + +@item phase +Set the stereo phase in degrees. Default is 0. Allowed range is from 0 to 360. +@end table + +@section stereowiden + +This filter enhance the stereo effect by suppressing signal common to both +channels and by delaying the signal of left into right and vice versa, +thereby widening the stereo effect. + +The filter accepts the following options: + +@table @option +@item delay +Time in milliseconds of the delay of left signal into right and vice versa. +Default is 20 milliseconds. + +@item feedback +Amount of gain in delayed signal into right and vice versa. Gives a delay +effect of left signal in right output and vice versa which gives widening +effect. Default is 0.3. + +@item crossfeed +Cross feed of left into right with inverted phase. This helps in suppressing +the mono. If the value is 1 it will cancel all the signal common to both +channels. Default is 0.3. + +@item drymix +Set level of input signal of original channel. Default is 0.8. +@end table + +@section treble + +Boost or cut treble (upper) frequencies of the audio using a two-pole +shelving filter with a response similar to that of a standard +hi-fi's tone-controls. This is also known as shelving equalisation (EQ). + +The filter accepts the following options: + +@table @option +@item gain, g +Give the gain at whichever is the lower of ~22 kHz and the +Nyquist frequency. Its useful range is about -20 (for a large cut) +to +20 (for a large boost). Beware of clipping when using a positive gain. + +@item frequency, f +Set the filter's central frequency and so can be used +to extend or reduce the frequency range to be boosted or cut. +The default value is @code{3000} Hz. + +@item width_type +Set method to specify band-width of filter. +@table @option +@item h +Hz +@item q +Q-Factor +@item o +octave +@item s +slope +@end table + +@item width, w +Determine how steep is the filter's shelf transition. +@end table + +@section tremolo + +Sinusoidal amplitude modulation. + +The filter accepts the following options: + +@table @option +@item f +Modulation frequency in Hertz. Modulation frequencies in the subharmonic range +(20 Hz or lower) will result in a tremolo effect. +This filter may also be used as a ring modulator by specifying +a modulation frequency higher than 20 Hz. +Range is 0.1 - 20000.0. Default value is 5.0 Hz. + +@item d +Depth of modulation as a percentage. Range is 0.0 - 1.0. +Default value is 0.5. +@end table + +@section vibrato + +Sinusoidal phase modulation. + +The filter accepts the following options: + +@table @option +@item f +Modulation frequency in Hertz. +Range is 0.1 - 20000.0. Default value is 5.0 Hz. + +@item d +Depth of modulation as a percentage. Range is 0.0 - 1.0. +Default value is 0.5. +@end table @section volume @@ -661,7 +3316,7 @@ It accepts the following parameters: @table @option @item volume -This expresses how the audio volume will be increased or decreased. +Set audio volume expression. Output values are clipped to the maximum value. @@ -670,7 +3325,7 @@ The output audio volume is given by the relation: @var{output_volume} = @var{volume} * @var{input_volume} @end example -The default value for @var{volume} is 1.0. +The default value for @var{volume} is "1.0". @item precision This parameter represents the mathematical precision. @@ -709,6 +3364,65 @@ Pre-amplification gain in dB to apply to the selected replaygain gain. Default value for @var{replaygain_preamp} is 0.0. +@item eval +Set when the volume expression is evaluated. + +It accepts the following values: +@table @samp +@item once +only evaluate expression once during the filter initialization, or +when the @samp{volume} command is sent + +@item frame +evaluate expression for each incoming frame +@end table + +Default value is @samp{once}. +@end table + +The volume expression can contain the following parameters. + +@table @option +@item n +frame number (starting at zero) +@item nb_channels +number of channels +@item nb_consumed_samples +number of samples consumed by the filter +@item nb_samples +number of samples in the current frame +@item pos +original frame position in the file +@item pts +frame PTS +@item sample_rate +sample rate +@item startpts +PTS at start of stream +@item startt +time at start of stream +@item t +frame time +@item tb +timestamp timebase +@item volume +last set volume value +@end table + +Note that when @option{eval} is set to @samp{once} only the +@var{sample_rate} and @var{tb} variables are available, all other +variables will evaluate to NAN. + +@subsection Commands + +This filter supports the following commands: +@table @option +@item volume +Modify the volume expression. +The command accepts the same syntax of the corresponding option. + +If the specified expression is not valid, it is kept at its current +value. @item replaygain_noclip Prevent clipping by limiting the gain applied. @@ -727,53 +3441,80 @@ volume=volume=1/2 volume=volume=-6.0206dB @end example +In all the above example the named key for @option{volume} can be +omitted, for example like in: +@example +volume=0.5 +@end example + @item Increase input audio power by 6 decibels using fixed-point precision: @example volume=volume=6dB:precision=fixed @end example + +@item +Fade volume after time 10 with an annihilation period of 5 seconds: +@example +volume='if(lt(t,10),1,max(1-(t-10)/5,0))':eval=frame +@end example @end itemize -@c man end AUDIO FILTERS +@section volumedetect -@chapter Audio Sources -@c man begin AUDIO SOURCES +Detect the volume of the input video. -Below is a description of the currently available audio sources. +The filter has no parameters. The input is not modified. Statistics about +the volume will be printed in the log when the input stream end is reached. -@section anullsrc +In particular it will show the mean volume (root mean square), maximum +volume (on a per-sample basis), and the beginning of a histogram of the +registered volume values (from the maximum value to a cumulated 1/1000 of +the samples). -The null audio source; it never returns audio frames. It is mainly useful as a -template and for use in analysis / debugging tools. +All volumes are in decibels relative to the maximum PCM value. -It accepts, as an optional parameter, a string of the form -@var{sample_rate}:@var{channel_layout}. +@subsection Examples -@var{sample_rate} specifies the sample rate, and defaults to 44100. +Here is an excerpt of the output: +@example +[Parsed_volumedetect_0 @ 0xa23120] mean_volume: -27 dB +[Parsed_volumedetect_0 @ 0xa23120] max_volume: -4 dB +[Parsed_volumedetect_0 @ 0xa23120] histogram_4db: 6 +[Parsed_volumedetect_0 @ 0xa23120] histogram_5db: 62 +[Parsed_volumedetect_0 @ 0xa23120] histogram_6db: 286 +[Parsed_volumedetect_0 @ 0xa23120] histogram_7db: 1042 +[Parsed_volumedetect_0 @ 0xa23120] histogram_8db: 2551 +[Parsed_volumedetect_0 @ 0xa23120] histogram_9db: 4609 +[Parsed_volumedetect_0 @ 0xa23120] histogram_10db: 8409 +@end example -@var{channel_layout} specifies the channel layout, and can be either an -integer or a string representing a channel layout. The default value -of @var{channel_layout} is 3, which corresponds to CH_LAYOUT_STEREO. +It means that: +@itemize +@item +The mean square energy is approximately -27 dB, or 10^-2.7. +@item +The largest sample is at -4 dB, or more precisely between -4 dB and -5 dB. +@item +There are 6 samples at -4 dB, 62 at -5 dB, 286 at -6 dB, etc. +@end itemize -Check the channel_layout_map definition in -@file{libavutil/channel_layout.c} for the mapping between strings and -channel layout values. +In other words, raising the volume by +4 dB does not cause any clipping, +raising it by +5 dB causes clipping for 6 samples, etc. -Some examples: -@example -# Set the sample rate to 48000 Hz and the channel layout to CH_LAYOUT_MONO -anullsrc=48000:4 +@c man end AUDIO FILTERS -# The same as above -anullsrc=48000:mono -@end example +@chapter Audio Sources +@c man begin AUDIO SOURCES + +Below is a description of the currently available audio sources. @section abuffer + Buffer audio frames, and make them available to the filter chain. -This source is not intended to be part of user-supplied graph descriptions; it -is for insertion by calling programs, through the interface defined in -@file{libavfilter/buffersrc.h}. +This source is mainly intended for a programmatic use, in particular +through the interface defined in @file{libavfilter/asrc_abuffer.h}. It accepts the following parameters: @table @option @@ -783,18 +3524,360 @@ The timebase which will be used for timestamps of submitted frames. It must be either a floating-point number or in @var{numerator}/@var{denominator} form. @item sample_rate -The audio sample rate. +The sample rate of the incoming audio buffers. @item sample_fmt -The name of the sample format, as returned by @code{av_get_sample_fmt_name()}. +The sample format of the incoming audio buffers. +Either a sample format name or its corresponding integer representation from +the enum AVSampleFormat in @file{libavutil/samplefmt.h} @item channel_layout -The channel layout of the audio data, in the form that can be accepted by -@code{av_get_channel_layout()}. +The channel layout of the incoming audio buffers. +Either a channel layout name from channel_layout_map in +@file{libavutil/channel_layout.c} or its corresponding integer representation +from the AV_CH_LAYOUT_* macros in @file{libavutil/channel_layout.h} + +@item channels +The number of channels of the incoming audio buffers. +If both @var{channels} and @var{channel_layout} are specified, then they +must be consistent. + +@end table + +@subsection Examples + +@example +abuffer=sample_rate=44100:sample_fmt=s16p:channel_layout=stereo +@end example + +will instruct the source to accept planar 16bit signed stereo at 44100Hz. +Since the sample format with name "s16p" corresponds to the number +6 and the "stereo" channel layout corresponds to the value 0x3, this is +equivalent to: +@example +abuffer=sample_rate=44100:sample_fmt=6:channel_layout=0x3 +@end example + +@section aevalsrc + +Generate an audio signal specified by an expression. + +This source accepts in input one or more expressions (one for each +channel), which are evaluated and used to generate a corresponding +audio signal. + +This source accepts the following options: + +@table @option +@item exprs +Set the '|'-separated expressions list for each separate channel. In case the +@option{channel_layout} option is not specified, the selected channel layout +depends on the number of provided expressions. Otherwise the last +specified expression is applied to the remaining output channels. + +@item channel_layout, c +Set the channel layout. The number of channels in the specified layout +must be equal to the number of specified expressions. + +@item duration, d +Set the minimum duration of the sourced audio. See +@ref{time duration syntax,,the Time duration section in the ffmpeg-utils(1) manual,ffmpeg-utils} +for the accepted syntax. +Note that the resulting duration may be greater than the specified +duration, as the generated audio is always cut at the end of a +complete frame. + +If not specified, or the expressed duration is negative, the audio is +supposed to be generated forever. + +@item nb_samples, n +Set the number of samples per channel per each output frame, +default to 1024. + +@item sample_rate, s +Specify the sample rate, default to 44100. @end table +Each expression in @var{exprs} can contain the following constants: + +@table @option +@item n +number of the evaluated sample, starting from 0 + +@item t +time of the evaluated sample expressed in seconds, starting from 0 + +@item s +sample rate + +@end table + +@subsection Examples + +@itemize +@item +Generate silence: +@example +aevalsrc=0 +@end example + +@item +Generate a sin signal with frequency of 440 Hz, set sample rate to +8000 Hz: +@example +aevalsrc="sin(440*2*PI*t):s=8000" +@end example + +@item +Generate a two channels signal, specify the channel layout (Front +Center + Back Center) explicitly: +@example +aevalsrc="sin(420*2*PI*t)|cos(430*2*PI*t):c=FC|BC" +@end example + +@item +Generate white noise: +@example +aevalsrc="-2+random(0)" +@end example + +@item +Generate an amplitude modulated signal: +@example +aevalsrc="sin(10*2*PI*t)*sin(880*2*PI*t)" +@end example + +@item +Generate 2.5 Hz binaural beats on a 360 Hz carrier: +@example +aevalsrc="0.1*sin(2*PI*(360-2.5/2)*t) | 0.1*sin(2*PI*(360+2.5/2)*t)" +@end example + +@end itemize + +@section anullsrc + +The null audio source, return unprocessed audio frames. It is mainly useful +as a template and to be employed in analysis / debugging tools, or as +the source for filters which ignore the input data (for example the sox +synth filter). + +This source accepts the following options: + +@table @option + +@item channel_layout, cl + +Specifies the channel layout, and can be either an integer or a string +representing a channel layout. The default value of @var{channel_layout} +is "stereo". + +Check the channel_layout_map definition in +@file{libavutil/channel_layout.c} for the mapping between strings and +channel layout values. + +@item sample_rate, r +Specifies the sample rate, and defaults to 44100. + +@item nb_samples, n +Set the number of samples per requested frames. + +@end table + +@subsection Examples + +@itemize +@item +Set the sample rate to 48000 Hz and the channel layout to AV_CH_LAYOUT_MONO. +@example +anullsrc=r=48000:cl=4 +@end example + +@item +Do the same operation with a more obvious syntax: +@example +anullsrc=r=48000:cl=mono +@end example +@end itemize + All the parameters need to be explicitly defined. +@section flite + +Synthesize a voice utterance using the libflite library. + +To enable compilation of this filter you need to configure FFmpeg with +@code{--enable-libflite}. + +Note that the flite library is not thread-safe. + +The filter accepts the following options: + +@table @option + +@item list_voices +If set to 1, list the names of the available voices and exit +immediately. Default value is 0. + +@item nb_samples, n +Set the maximum number of samples per frame. Default value is 512. + +@item textfile +Set the filename containing the text to speak. + +@item text +Set the text to speak. + +@item voice, v +Set the voice to use for the speech synthesis. Default value is +@code{kal}. See also the @var{list_voices} option. +@end table + +@subsection Examples + +@itemize +@item +Read from file @file{speech.txt}, and synthesize the text using the +standard flite voice: +@example +flite=textfile=speech.txt +@end example + +@item +Read the specified text selecting the @code{slt} voice: +@example +flite=text='So fare thee well, poor devil of a Sub-Sub, whose commentator I am':voice=slt +@end example + +@item +Input text to ffmpeg: +@example +ffmpeg -f lavfi -i flite=text='So fare thee well, poor devil of a Sub-Sub, whose commentator I am':voice=slt +@end example + +@item +Make @file{ffplay} speak the specified text, using @code{flite} and +the @code{lavfi} device: +@example +ffplay -f lavfi flite=text='No more be grieved for which that thou hast done.' +@end example +@end itemize + +For more information about libflite, check: +@url{http://www.speech.cs.cmu.edu/flite/} + +@section anoisesrc + +Generate a noise audio signal. + +The filter accepts the following options: + +@table @option +@item sample_rate, r +Specify the sample rate. Default value is 48000 Hz. + +@item amplitude, a +Specify the amplitude (0.0 - 1.0) of the generated audio stream. Default value +is 1.0. + +@item duration, d +Specify the duration of the generated audio stream. Not specifying this option +results in noise with an infinite length. + +@item color, colour, c +Specify the color of noise. Available noise colors are white, pink, and brown. +Default color is white. + +@item seed, s +Specify a value used to seed the PRNG. + +@item nb_samples, n +Set the number of samples per each output frame, default is 1024. +@end table + +@subsection Examples + +@itemize + +@item +Generate 60 seconds of pink noise, with a 44.1 kHz sampling rate and an amplitude of 0.5: +@example +anoisesrc=d=60:c=pink:r=44100:a=0.5 +@end example +@end itemize + +@section sine + +Generate an audio signal made of a sine wave with amplitude 1/8. + +The audio signal is bit-exact. + +The filter accepts the following options: + +@table @option + +@item frequency, f +Set the carrier frequency. Default is 440 Hz. + +@item beep_factor, b +Enable a periodic beep every second with frequency @var{beep_factor} times +the carrier frequency. Default is 0, meaning the beep is disabled. + +@item sample_rate, r +Specify the sample rate, default is 44100. + +@item duration, d +Specify the duration of the generated audio stream. + +@item samples_per_frame +Set the number of samples per output frame. + +The expression can contain the following constants: + +@table @option +@item n +The (sequential) number of the output audio frame, starting from 0. + +@item pts +The PTS (Presentation TimeStamp) of the output audio frame, +expressed in @var{TB} units. + +@item t +The PTS of the output audio frame, expressed in seconds. + +@item TB +The timebase of the output audio frames. +@end table + +Default is @code{1024}. +@end table + +@subsection Examples + +@itemize + +@item +Generate a simple 440 Hz sine wave: +@example +sine +@end example + +@item +Generate a 220 Hz sine wave with a 880 Hz beep each second, for 5 seconds: +@example +sine=220:4:d=5 +sine=f=220:b=4:d=5 +sine=frequency=220:beep_factor=4:duration=5 +@end example + +@item +Generate a 1 kHz sine wave following @code{1602,1601,1602,1601,1602} NTSC +pattern: +@example +sine=1000:samples_per_frame='st(0,mod(n,5)); 1602-not(not(eq(ld(0),1)+eq(ld(0),3)))' +@end example +@end itemize + @c man end AUDIO SOURCES @chapter Audio Sinks @@ -802,31 +3885,193 @@ All the parameters need to be explicitly defined. Below is a description of the currently available audio sinks. +@section abuffersink + +Buffer audio frames, and make them available to the end of filter chain. + +This sink is mainly intended for programmatic use, in particular +through the interface defined in @file{libavfilter/buffersink.h} +or the options system. + +It accepts a pointer to an AVABufferSinkContext structure, which +defines the incoming buffers' formats, to be passed as the opaque +parameter to @code{avfilter_init_filter} for initialization. @section anullsink Null audio sink; do absolutely nothing with the input audio. It is mainly useful as a template and for use in analysis / debugging tools. -@section abuffersink -This sink is intended for programmatic use. Frames that arrive on this sink can -be retrieved by the calling program, using the interface defined in -@file{libavfilter/buffersink.h}. - -It does not accept any parameters. - @c man end AUDIO SINKS @chapter Video Filters @c man begin VIDEO FILTERS -When you configure your Libav build, you can disable any of the -existing filters using --disable-filters. +When you configure your FFmpeg build, you can disable any of the +existing filters using @code{--disable-filters}. The configure output will show the video filters included in your build. Below is a description of the currently available video filters. +@section alphaextract + +Extract the alpha component from the input as a grayscale video. This +is especially useful with the @var{alphamerge} filter. + +@section alphamerge + +Add or replace the alpha component of the primary input with the +grayscale value of a second input. This is intended for use with +@var{alphaextract} to allow the transmission or storage of frame +sequences that have alpha in a format that doesn't support an alpha +channel. + +For example, to reconstruct full frames from a normal YUV-encoded video +and a separate video created with @var{alphaextract}, you might use: +@example +movie=in_alpha.mkv [alpha]; [in][alpha] alphamerge [out] +@end example + +Since this filter is designed for reconstruction, it operates on frame +sequences without considering timestamps, and terminates when either +input reaches end of stream. This will cause problems if your encoding +pipeline drops frames. If you're trying to apply an image as an +overlay to a video stream, consider the @var{overlay} filter instead. + +@section ass + +Same as the @ref{subtitles} filter, except that it doesn't require libavcodec +and libavformat to work. On the other hand, it is limited to ASS (Advanced +Substation Alpha) subtitles files. + +This filter accepts the following option in addition to the common options from +the @ref{subtitles} filter: + +@table @option +@item shaping +Set the shaping engine + +Available values are: +@table @samp +@item auto +The default libass shaping engine, which is the best available. +@item simple +Fast, font-agnostic shaper that can do only substitutions +@item complex +Slower shaper using OpenType for substitutions and positioning +@end table + +The default is @code{auto}. +@end table + +@section atadenoise +Apply an Adaptive Temporal Averaging Denoiser to the video input. + +The filter accepts the following options: + +@table @option +@item 0a +Set threshold A for 1st plane. Default is 0.02. +Valid range is 0 to 0.3. + +@item 0b +Set threshold B for 1st plane. Default is 0.04. +Valid range is 0 to 5. + +@item 1a +Set threshold A for 2nd plane. Default is 0.02. +Valid range is 0 to 0.3. + +@item 1b +Set threshold B for 2nd plane. Default is 0.04. +Valid range is 0 to 5. + +@item 2a +Set threshold A for 3rd plane. Default is 0.02. +Valid range is 0 to 0.3. + +@item 2b +Set threshold B for 3rd plane. Default is 0.04. +Valid range is 0 to 5. + +Threshold A is designed to react on abrupt changes in the input signal and +threshold B is designed to react on continuous changes in the input signal. + +@item s +Set number of frames filter will use for averaging. Default is 33. Must be odd +number in range [5, 129]. +@end table + +@section bbox + +Compute the bounding box for the non-black pixels in the input frame +luminance plane. + +This filter computes the bounding box containing all the pixels with a +luminance value greater than the minimum allowed value. +The parameters describing the bounding box are printed on the filter +log. + +The filter accepts the following option: + +@table @option +@item min_val +Set the minimal luminance value. Default is @code{16}. +@end table + +@section blackdetect + +Detect video intervals that are (almost) completely black. Can be +useful to detect chapter transitions, commercials, or invalid +recordings. Output lines contains the time for the start, end and +duration of the detected black interval expressed in seconds. + +In order to display the output lines, you need to set the loglevel at +least to the AV_LOG_INFO value. + +The filter accepts the following options: + +@table @option +@item black_min_duration, d +Set the minimum detected black duration expressed in seconds. It must +be a non-negative floating point number. + +Default value is 2.0. + +@item picture_black_ratio_th, pic_th +Set the threshold for considering a picture "black". +Express the minimum value for the ratio: +@example +@var{nb_black_pixels} / @var{nb_pixels} +@end example + +for which a picture is considered black. +Default value is 0.98. + +@item pixel_black_th, pix_th +Set the threshold for considering a pixel "black". + +The threshold expresses the maximum pixel luminance value for which a +pixel is considered "black". The provided value is scaled according to +the following equation: +@example +@var{absolute_threshold} = @var{luminance_minimum_value} + @var{pixel_black_th} * @var{luminance_range_size} +@end example + +@var{luminance_range_size} and @var{luminance_minimum_value} depend on +the input video format, the range is [0-255] for YUV full-range +formats and [16-235] for YUV non full-range formats. + +Default value is 0.10. +@end table + +The following example sets the maximum pixel threshold to the minimum +value, and detects only black intervals of 2 or more seconds: +@example +blackdetect=d=2:pix_th=0.00 +@end example + @section blackframe Detect frames that are (almost) completely black. Can be useful to @@ -843,13 +4088,168 @@ It accepts the following parameters: @item amount The percentage of the pixels that have to be below the threshold; it defaults to -98. +@code{98}. -@item threshold -The threshold below which a pixel value is considered black; it defaults to 32. +@item threshold, thresh +The threshold below which a pixel value is considered black; it defaults to +@code{32}. + +@end table + +@section blend, tblend + +Blend two video frames into each other. + +The @code{blend} filter takes two input streams and outputs one +stream, the first input is the "top" layer and second input is +"bottom" layer. Output terminates when shortest input terminates. + +The @code{tblend} (time blend) filter takes two consecutive frames +from one single stream, and outputs the result obtained by blending +the new frame on top of the old frame. + +A description of the accepted options follows. + +@table @option +@item c0_mode +@item c1_mode +@item c2_mode +@item c3_mode +@item all_mode +Set blend mode for specific pixel component or all pixel components in case +of @var{all_mode}. Default value is @code{normal}. + +Available values for component modes are: +@table @samp +@item addition +@item addition128 +@item and +@item average +@item burn +@item darken +@item difference +@item difference128 +@item divide +@item dodge +@item exclusion +@item glow +@item hardlight +@item hardmix +@item lighten +@item linearlight +@item multiply +@item multiply128 +@item negation +@item normal +@item or +@item overlay +@item phoenix +@item pinlight +@item reflect +@item screen +@item softlight +@item subtract +@item vividlight +@item xor +@end table + +@item c0_opacity +@item c1_opacity +@item c2_opacity +@item c3_opacity +@item all_opacity +Set blend opacity for specific pixel component or all pixel components in case +of @var{all_opacity}. Only used in combination with pixel component blend modes. +@item c0_expr +@item c1_expr +@item c2_expr +@item c3_expr +@item all_expr +Set blend expression for specific pixel component or all pixel components in case +of @var{all_expr}. Note that related mode options will be ignored if those are set. + +The expressions can use the following variables: + +@table @option +@item N +The sequential number of the filtered frame, starting from @code{0}. + +@item X +@item Y +the coordinates of the current sample + +@item W +@item H +the width and height of currently filtered plane + +@item SW +@item SH +Width and height scale depending on the currently filtered plane. It is the +ratio between the corresponding luma plane number of pixels and the current +plane ones. E.g. for YUV4:2:0 the values are @code{1,1} for the luma plane, and +@code{0.5,0.5} for chroma planes. + +@item T +Time of the current frame, expressed in seconds. + +@item TOP, A +Value of pixel component at current location for first video frame (top layer). + +@item BOTTOM, B +Value of pixel component at current location for second video frame (bottom layer). +@end table + +@item shortest +Force termination when the shortest input terminates. Default is +@code{0}. This option is only defined for the @code{blend} filter. + +@item repeatlast +Continue applying the last bottom frame after the end of the stream. A value of +@code{0} disable the filter after the last frame of the bottom layer is reached. +Default is @code{1}. This option is only defined for the @code{blend} filter. @end table +@subsection Examples + +@itemize +@item +Apply transition from bottom layer to top layer in first 10 seconds: +@example +blend=all_expr='A*(if(gte(T,10),1,T/10))+B*(1-(if(gte(T,10),1,T/10)))' +@end example + +@item +Apply 1x1 checkerboard effect: +@example +blend=all_expr='if(eq(mod(X,2),mod(Y,2)),A,B)' +@end example + +@item +Apply uncover left effect: +@example +blend=all_expr='if(gte(N*SW+X,W),A,B)' +@end example + +@item +Apply uncover down effect: +@example +blend=all_expr='if(gte(Y-N*SH,0),A,B)' +@end example + +@item +Apply uncover up-left effect: +@example +blend=all_expr='if(gte(T*SH*40+Y,H)*gte((T*40*SW+X)*W/H,W),A,B)' +@end example + +@item +Display differences between the current and the previous frame: +@example +tblend=all_mode=difference128 +@end example +@end itemize + @section boxblur Apply a boxblur algorithm to the input video. @@ -858,58 +4258,77 @@ It accepts the following parameters: @table @option -@item luma_radius -@item luma_power -@item chroma_radius -@item chroma_power -@item alpha_radius -@item alpha_power +@item luma_radius, lr +@item luma_power, lp +@item chroma_radius, cr +@item chroma_power, cp +@item alpha_radius, ar +@item alpha_power, ap @end table -The chroma and alpha parameters are optional. If not specified, they default -to the corresponding values set for @var{luma_radius} and -@var{luma_power}. +A description of the accepted options follows. -@var{luma_radius}, @var{chroma_radius}, and @var{alpha_radius} represent -the radius in pixels of the box used for blurring the corresponding -input plane. They are expressions, and can contain the following -constants: @table @option -@item w, h +@item luma_radius, lr +@item chroma_radius, cr +@item alpha_radius, ar +Set an expression for the box radius in pixels used for blurring the +corresponding input plane. + +The radius value must be a non-negative number, and must not be +greater than the value of the expression @code{min(w,h)/2} for the +luma and alpha planes, and of @code{min(cw,ch)/2} for the chroma +planes. + +Default value for @option{luma_radius} is "2". If not specified, +@option{chroma_radius} and @option{alpha_radius} default to the +corresponding value set for @option{luma_radius}. + +The expressions can contain the following constants: +@table @option +@item w +@item h The input width and height in pixels. -@item cw, ch +@item cw +@item ch The input chroma image width and height in pixels. -@item hsub, vsub +@item hsub +@item vsub The horizontal and vertical chroma subsample values. For example, for the pixel format "yuv422p", @var{hsub} is 2 and @var{vsub} is 1. @end table -The radius must be a non-negative number, and must not be greater than -the value of the expression @code{min(w,h)/2} for the luma and alpha planes, -and of @code{min(cw,ch)/2} for the chroma planes. +@item luma_power, lp +@item chroma_power, cp +@item alpha_power, ap +Specify how many times the boxblur filter is applied to the +corresponding plane. -@var{luma_power}, @var{chroma_power}, and @var{alpha_power} represent -how many times the boxblur filter is applied to the corresponding -plane. +Default value for @option{luma_power} is 2. If not specified, +@option{chroma_power} and @option{alpha_power} default to the +corresponding value set for @option{luma_power}. -Some examples: +A value of 0 will disable the effect. +@end table -@itemize +@subsection Examples +@itemize @item Apply a boxblur filter with the luma, chroma, and alpha radii set to 2: @example boxblur=luma_radius=2:luma_power=1 +boxblur=2:1 @end example @item Set the luma radius to 2, and alpha and chroma radius to 0: @example -boxblur=2:1:0:0:0:0 +boxblur=2:1:cr=0:ar=0 @end example @item @@ -917,7 +4336,393 @@ Set the luma and chroma radii to a fraction of the video dimension: @example boxblur=luma_radius=min(h\,w)/10:luma_power=1:chroma_radius=min(cw\,ch)/10:chroma_power=1 @end example +@end itemize + +@section chromakey +YUV colorspace color/chroma keying. +The filter accepts the following options: + +@table @option +@item color +The color which will be replaced with transparency. + +@item similarity +Similarity percentage with the key color. + +0.01 matches only the exact key color, while 1.0 matches everything. + +@item blend +Blend percentage. + +0.0 makes pixels either fully transparent, or not transparent at all. + +Higher values result in semi-transparent pixels, with a higher transparency +the more similar the pixels color is to the key color. + +@item yuv +Signals that the color passed is already in YUV instead of RGB. + +Litteral colors like "green" or "red" don't make sense with this enabled anymore. +This can be used to pass exact YUV values as hexadecimal numbers. +@end table + +@subsection Examples + +@itemize +@item +Make every green pixel in the input image transparent: +@example +ffmpeg -i input.png -vf chromakey=green out.png +@end example + +@item +Overlay a greenscreen-video on top of a static black background. +@example +ffmpeg -f lavfi -i color=c=black:s=1280x720 -i video.mp4 -shortest -filter_complex "[1:v]chromakey=0x70de77:0.1:0.2[ckout];[0:v][ckout]overlay[out]" -map "[out]" output.mkv +@end example +@end itemize + +@section codecview + +Visualize information exported by some codecs. + +Some codecs can export information through frames using side-data or other +means. For example, some MPEG based codecs export motion vectors through the +@var{export_mvs} flag in the codec @option{flags2} option. + +The filter accepts the following option: + +@table @option +@item mv +Set motion vectors to visualize. + +Available flags for @var{mv} are: + +@table @samp +@item pf +forward predicted MVs of P-frames +@item bf +forward predicted MVs of B-frames +@item bb +backward predicted MVs of B-frames +@end table + +@item qp +Display quantization parameters using the chroma planes +@end table + +@subsection Examples + +@itemize +@item +Visualizes multi-directionals MVs from P and B-Frames using @command{ffplay}: +@example +ffplay -flags2 +export_mvs input.mpg -vf codecview=mv=pf+bf+bb +@end example +@end itemize + +@section colorbalance +Modify intensity of primary colors (red, green and blue) of input frames. + +The filter allows an input frame to be adjusted in the shadows, midtones or highlights +regions for the red-cyan, green-magenta or blue-yellow balance. + +A positive adjustment value shifts the balance towards the primary color, a negative +value towards the complementary color. + +The filter accepts the following options: + +@table @option +@item rs +@item gs +@item bs +Adjust red, green and blue shadows (darkest pixels). + +@item rm +@item gm +@item bm +Adjust red, green and blue midtones (medium pixels). + +@item rh +@item gh +@item bh +Adjust red, green and blue highlights (brightest pixels). + +Allowed ranges for options are @code{[-1.0, 1.0]}. Defaults are @code{0}. +@end table + +@subsection Examples + +@itemize +@item +Add red color cast to shadows: +@example +colorbalance=rs=.3 +@end example +@end itemize + +@section colorkey +RGB colorspace color keying. + +The filter accepts the following options: + +@table @option +@item color +The color which will be replaced with transparency. + +@item similarity +Similarity percentage with the key color. + +0.01 matches only the exact key color, while 1.0 matches everything. + +@item blend +Blend percentage. + +0.0 makes pixels either fully transparent, or not transparent at all. + +Higher values result in semi-transparent pixels, with a higher transparency +the more similar the pixels color is to the key color. +@end table + +@subsection Examples + +@itemize +@item +Make every green pixel in the input image transparent: +@example +ffmpeg -i input.png -vf colorkey=green out.png +@end example + +@item +Overlay a greenscreen-video on top of a static background image. +@example +ffmpeg -i background.png -i video.mp4 -filter_complex "[1:v]colorkey=0x3BBD1E:0.3:0.2[ckout];[0:v][ckout]overlay[out]" -map "[out]" output.flv +@end example +@end itemize + +@section colorlevels + +Adjust video input frames using levels. + +The filter accepts the following options: + +@table @option +@item rimin +@item gimin +@item bimin +@item aimin +Adjust red, green, blue and alpha input black point. +Allowed ranges for options are @code{[-1.0, 1.0]}. Defaults are @code{0}. + +@item rimax +@item gimax +@item bimax +@item aimax +Adjust red, green, blue and alpha input white point. +Allowed ranges for options are @code{[-1.0, 1.0]}. Defaults are @code{1}. + +Input levels are used to lighten highlights (bright tones), darken shadows +(dark tones), change the balance of bright and dark tones. + +@item romin +@item gomin +@item bomin +@item aomin +Adjust red, green, blue and alpha output black point. +Allowed ranges for options are @code{[0, 1.0]}. Defaults are @code{0}. + +@item romax +@item gomax +@item bomax +@item aomax +Adjust red, green, blue and alpha output white point. +Allowed ranges for options are @code{[0, 1.0]}. Defaults are @code{1}. + +Output levels allows manual selection of a constrained output level range. +@end table + +@subsection Examples + +@itemize +@item +Make video output darker: +@example +colorlevels=rimin=0.058:gimin=0.058:bimin=0.058 +@end example + +@item +Increase contrast: +@example +colorlevels=rimin=0.039:gimin=0.039:bimin=0.039:rimax=0.96:gimax=0.96:bimax=0.96 +@end example + +@item +Make video output lighter: +@example +colorlevels=rimax=0.902:gimax=0.902:bimax=0.902 +@end example + +@item +Increase brightness: +@example +colorlevels=romin=0.5:gomin=0.5:bomin=0.5 +@end example +@end itemize + +@section colorchannelmixer + +Adjust video input frames by re-mixing color channels. + +This filter modifies a color channel by adding the values associated to +the other channels of the same pixels. For example if the value to +modify is red, the output value will be: +@example +@var{red}=@var{red}*@var{rr} + @var{blue}*@var{rb} + @var{green}*@var{rg} + @var{alpha}*@var{ra} +@end example + +The filter accepts the following options: + +@table @option +@item rr +@item rg +@item rb +@item ra +Adjust contribution of input red, green, blue and alpha channels for output red channel. +Default is @code{1} for @var{rr}, and @code{0} for @var{rg}, @var{rb} and @var{ra}. + +@item gr +@item gg +@item gb +@item ga +Adjust contribution of input red, green, blue and alpha channels for output green channel. +Default is @code{1} for @var{gg}, and @code{0} for @var{gr}, @var{gb} and @var{ga}. + +@item br +@item bg +@item bb +@item ba +Adjust contribution of input red, green, blue and alpha channels for output blue channel. +Default is @code{1} for @var{bb}, and @code{0} for @var{br}, @var{bg} and @var{ba}. + +@item ar +@item ag +@item ab +@item aa +Adjust contribution of input red, green, blue and alpha channels for output alpha channel. +Default is @code{1} for @var{aa}, and @code{0} for @var{ar}, @var{ag} and @var{ab}. + +Allowed ranges for options are @code{[-2.0, 2.0]}. +@end table + +@subsection Examples + +@itemize +@item +Convert source to grayscale: +@example +colorchannelmixer=.3:.4:.3:0:.3:.4:.3:0:.3:.4:.3 +@end example +@item +Simulate sepia tones: +@example +colorchannelmixer=.393:.769:.189:0:.349:.686:.168:0:.272:.534:.131 +@end example +@end itemize + +@section colormatrix + +Convert color matrix. + +The filter accepts the following options: + +@table @option +@item src +@item dst +Specify the source and destination color matrix. Both values must be +specified. + +The accepted values are: +@table @samp +@item bt709 +BT.709 + +@item bt601 +BT.601 + +@item smpte240m +SMPTE-240M + +@item fcc +FCC +@end table +@end table + +For example to convert from BT.601 to SMPTE-240M, use the command: +@example +colormatrix=bt601:smpte240m +@end example + +@section convolution + +Apply convolution 3x3 or 5x5 filter. + +The filter accepts the following options: + +@table @option +@item 0m +@item 1m +@item 2m +@item 3m +Set matrix for each plane. +Matrix is sequence of 9 or 25 signed integers. + +@item 0rdiv +@item 1rdiv +@item 2rdiv +@item 3rdiv +Set multiplier for calculated value for each plane. + +@item 0bias +@item 1bias +@item 2bias +@item 3bias +Set bias for each plane. This value is added to the result of the multiplication. +Useful for making the overall image brighter or darker. Default is 0.0. +@end table + +@subsection Examples + +@itemize +@item +Apply sharpen: +@example +convolution="0 -1 0 -1 5 -1 0 -1 0:0 -1 0 -1 5 -1 0 -1 0:0 -1 0 -1 5 -1 0 -1 0:0 -1 0 -1 5 -1 0 -1 0" +@end example + +@item +Apply blur: +@example +convolution="1 1 1 1 1 1 1 1 1:1 1 1 1 1 1 1 1 1:1 1 1 1 1 1 1 1 1:1 1 1 1 1 1 1 1 1:1/9:1/9:1/9:1/9" +@end example + +@item +Apply edge enhance: +@example +convolution="0 0 0 -1 1 0 0 0 0:0 0 0 -1 1 0 0 0 0:0 0 0 -1 1 0 0 0 0:0 0 0 -1 1 0 0 0 0:5:1:1:1:0:128:128:128" +@end example + +@item +Apply edge detect: +@example +convolution="0 1 0 1 -4 1 0 1 0:0 1 0 1 -4 1 0 1 0:0 1 0 1 -4 1 0 1 0:0 1 0 1 -4 1 0 1 0:5:5:5:1:0:128:128:128" +@end example + +@item +Apply emboss: +@example +convolution="-2 -1 0 -1 1 1 0 1 2:-2 -1 0 -1 1 1 0 1 2:-2 -1 0 -1 1 1 0 1 2:-2 -1 0 -1 1 1 0 1 2" +@end example @end itemize @section copy @@ -932,60 +4737,82 @@ Crop the input video to given dimensions. It accepts the following parameters: @table @option +@item w, out_w +The width of the output video. It defaults to @code{iw}. +This expression is evaluated only once during the filter +configuration, or when the @samp{w} or @samp{out_w} command is sent. -@item out_w -The width of the output video. - -@item out_h -The height of the output video. +@item h, out_h +The height of the output video. It defaults to @code{ih}. +This expression is evaluated only once during the filter +configuration, or when the @samp{h} or @samp{out_h} command is sent. @item x The horizontal position, in the input video, of the left edge of the output -video. +video. It defaults to @code{(in_w-out_w)/2}. +This expression is evaluated per-frame. @item y The vertical position, in the input video, of the top edge of the output video. +It defaults to @code{(in_h-out_h)/2}. +This expression is evaluated per-frame. +@item keep_aspect +If set to 1 will force the output display aspect ratio +to be the same of the input, by changing the output sample aspect +ratio. It defaults to 0. @end table -The parameters are expressions containing the following constants: +The @var{out_w}, @var{out_h}, @var{x}, @var{y} parameters are +expressions containing the following constants: @table @option -@item E, PI, PHI -These are approximated values for the mathematical constants e -(Euler's number), pi (Greek pi), and phi (the golden ratio). - -@item x, y +@item x +@item y The computed values for @var{x} and @var{y}. They are evaluated for each new frame. -@item in_w, in_h +@item in_w +@item in_h The input width and height. -@item iw, ih +@item iw +@item ih These are the same as @var{in_w} and @var{in_h}. -@item out_w, out_h +@item out_w +@item out_h The output (cropped) width and height. -@item ow, oh +@item ow +@item oh These are the same as @var{out_w} and @var{out_h}. +@item a +same as @var{iw} / @var{ih} + +@item sar +input sample aspect ratio + +@item dar +input display aspect ratio, it is the same as (@var{iw} / @var{ih}) * @var{sar} + +@item hsub +@item vsub +horizontal and vertical chroma subsample values. For example for the +pixel format "yuv422p" @var{hsub} is 2 and @var{vsub} is 1. + @item n The number of the input frame, starting from 0. +@item pos +the position in the file of the input frame, NAN if unknown + @item t The timestamp expressed in seconds. It's NAN if the input timestamp is unknown. @end table -The @var{out_w} and @var{out_h} parameters specify the expressions for -the width and height of the output (cropped) video. They are only -evaluated during the configuration of the filter. - -The default value of @var{out_w} is "in_w", and the default value of -@var{out_h} is "in_h". - The expression for @var{out_w} may depend on the value of @var{out_h}, and the expression for @var{out_h} may depend on @var{out_w}, but they cannot depend on @var{x} and @var{y}, as @var{x} and @var{y} are @@ -996,48 +4823,103 @@ position of the top-left corner of the output (non-cropped) area. They are evaluated for each frame. If the evaluated value is not valid, it is approximated to the nearest valid value. -The default value of @var{x} is "(in_w-out_w)/2", and the default -value for @var{y} is "(in_h-out_h)/2", which set the cropped area at -the center of the input image. - The expression for @var{x} may depend on @var{y}, and the expression for @var{y} may depend on @var{x}. -Some examples: +@subsection Examples + +@itemize +@item +Crop area with size 100x100 at position (12,34). +@example +crop=100:100:12:34 +@end example + +Using named options, the example above becomes: @example -# Crop the central input area with size 100x100 -crop=out_w=100:out_h=100 +crop=w=100:h=100:x=12:y=34 +@end example + +@item +Crop the central input area with size 100x100: +@example +crop=100:100 +@end example -# Crop the central input area with size 2/3 of the input video -"crop=out_w=2/3*in_w:out_h=2/3*in_h" +@item +Crop the central input area with size 2/3 of the input video: +@example +crop=2/3*in_w:2/3*in_h +@end example -# Crop the input video central square +@item +Crop the input video central square: +@example crop=out_w=in_h +crop=in_h +@end example -# Delimit the rectangle with the top-left corner placed at position -# 100:100 and the right-bottom corner corresponding to the right-bottom -# corner of the input image -crop=out_w=in_w-100:out_h=in_h-100:x=100:y=100 +@item +Delimit the rectangle with the top-left corner placed at position +100:100 and the right-bottom corner corresponding to the right-bottom +corner of the input image. +@example +crop=in_w-100:in_h-100:100:100 +@end example -# Crop 10 pixels from the left and right borders, and 20 pixels from -# the top and bottom borders -"crop=out_w=in_w-2*10:out_h=in_h-2*20" +@item +Crop 10 pixels from the left and right borders, and 20 pixels from +the top and bottom borders +@example +crop=in_w-2*10:in_h-2*20 +@end example -# Keep only the bottom right quarter of the input image -"crop=out_w=in_w/2:out_h=in_h/2:x=in_w/2:y=in_h/2" +@item +Keep only the bottom right quarter of the input image: +@example +crop=in_w/2:in_h/2:in_w/2:in_h/2 +@end example -# Crop height for getting Greek harmony -"crop=out_w=in_w:out_h=1/PHI*in_w" +@item +Crop height for getting Greek harmony: +@example +crop=in_w:1/PHI*in_w +@end example -# Trembling effect -"crop=in_w/2:in_h/2:(in_w-out_w)/2+((in_w-out_w)/2)*sin(n/10):(in_h-out_h)/2 +((in_h-out_h)/2)*sin(n/7)" +@item +Apply trembling effect: +@example +crop=in_w/2:in_h/2:(in_w-out_w)/2+((in_w-out_w)/2)*sin(n/10):(in_h-out_h)/2 +((in_h-out_h)/2)*sin(n/7) +@end example -# Erratic camera effect depending on timestamp -"crop=out_w=in_w/2:out_h=in_h/2:x=(in_w-out_w)/2+((in_w-out_w)/2)*sin(t*10):y=(in_h-out_h)/2 +((in_h-out_h)/2)*sin(t*13)" +@item +Apply erratic camera effect depending on timestamp: +@example +crop=in_w/2:in_h/2:(in_w-out_w)/2+((in_w-out_w)/2)*sin(t*10):(in_h-out_h)/2 +((in_h-out_h)/2)*sin(t*13)" +@end example -# Set x depending on the value of y -"crop=in_w/2:in_h/2:y:10+10*sin(n/10)" +@item +Set x depending on the value of y: +@example +crop=in_w/2:in_h/2:y:10+10*sin(n/10) @end example +@end itemize + +@subsection Commands + +This filter supports the following commands: +@table @option +@item w, out_w +@item h, out_h +@item x +@item y +Set width/height of the output video and the horizontal/vertical position +in the input video. +The command accepts the same syntax of the corresponding option. + +If the specified expression is not valid, it is kept at its current +value. +@end table @section cropdetect @@ -1052,8 +4934,11 @@ It accepts the following parameters: @table @option @item limit -The threshold, an optional parameter between nothing (0) and -everything (255). It defaults to 24. +Set higher black value threshold, which can be optionally specified +from nothing (0) to everything (255 for 8bit based formats). An intensity +value greater to the set value is considered non-black. It defaults to 24. +You can also specify a value between 0.0 and 1.0 which will be scaled depending +on the bitdepth of the pixel format. @item round The value which the width/height should be divisible by. It defaults to @@ -1061,16 +4946,324 @@ The value which the width/height should be divisible by. It defaults to get only even dimensions (needed for 4:2:2 video). 16 is best when encoding to most video codecs. -@item reset -A counter that determines how many frames cropdetect will reset -the previously detected largest video area after. It will then start over -and detect the current optimal crop area. It defaults to 0. +@item reset_count, reset +Set the counter that determines after how many frames cropdetect will +reset the previously detected largest video area and start over to +detect the current optimal crop area. Default value is 0. This can be useful when channel logos distort the video area. 0 indicates 'never reset', and returns the largest area encountered during playback. @end table +@anchor{curves} +@section curves + +Apply color adjustments using curves. + +This filter is similar to the Adobe Photoshop and GIMP curves tools. Each +component (red, green and blue) has its values defined by @var{N} key points +tied from each other using a smooth curve. The x-axis represents the pixel +values from the input frame, and the y-axis the new pixel values to be set for +the output frame. + +By default, a component curve is defined by the two points @var{(0;0)} and +@var{(1;1)}. This creates a straight line where each original pixel value is +"adjusted" to its own value, which means no change to the image. + +The filter allows you to redefine these two points and add some more. A new +curve (using a natural cubic spline interpolation) will be define to pass +smoothly through all these new coordinates. The new defined points needs to be +strictly increasing over the x-axis, and their @var{x} and @var{y} values must +be in the @var{[0;1]} interval. If the computed curves happened to go outside +the vector spaces, the values will be clipped accordingly. + +If there is no key point defined in @code{x=0}, the filter will automatically +insert a @var{(0;0)} point. In the same way, if there is no key point defined +in @code{x=1}, the filter will automatically insert a @var{(1;1)} point. + +The filter accepts the following options: + +@table @option +@item preset +Select one of the available color presets. This option can be used in addition +to the @option{r}, @option{g}, @option{b} parameters; in this case, the later +options takes priority on the preset values. +Available presets are: +@table @samp +@item none +@item color_negative +@item cross_process +@item darker +@item increase_contrast +@item lighter +@item linear_contrast +@item medium_contrast +@item negative +@item strong_contrast +@item vintage +@end table +Default is @code{none}. +@item master, m +Set the master key points. These points will define a second pass mapping. It +is sometimes called a "luminance" or "value" mapping. It can be used with +@option{r}, @option{g}, @option{b} or @option{all} since it acts like a +post-processing LUT. +@item red, r +Set the key points for the red component. +@item green, g +Set the key points for the green component. +@item blue, b +Set the key points for the blue component. +@item all +Set the key points for all components (not including master). +Can be used in addition to the other key points component +options. In this case, the unset component(s) will fallback on this +@option{all} setting. +@item psfile +Specify a Photoshop curves file (@code{.acv}) to import the settings from. +@end table + +To avoid some filtergraph syntax conflicts, each key points list need to be +defined using the following syntax: @code{x0/y0 x1/y1 x2/y2 ...}. + +@subsection Examples + +@itemize +@item +Increase slightly the middle level of blue: +@example +curves=blue='0.5/0.58' +@end example + +@item +Vintage effect: +@example +curves=r='0/0.11 .42/.51 1/0.95':g='0.50/0.48':b='0/0.22 .49/.44 1/0.8' +@end example +Here we obtain the following coordinates for each components: +@table @var +@item red +@code{(0;0.11) (0.42;0.51) (1;0.95)} +@item green +@code{(0;0) (0.50;0.48) (1;1)} +@item blue +@code{(0;0.22) (0.49;0.44) (1;0.80)} +@end table + +@item +The previous example can also be achieved with the associated built-in preset: +@example +curves=preset=vintage +@end example + +@item +Or simply: +@example +curves=vintage +@end example + +@item +Use a Photoshop preset and redefine the points of the green component: +@example +curves=psfile='MyCurvesPresets/purple.acv':green='0.45/0.53' +@end example +@end itemize + +@section dctdnoiz + +Denoise frames using 2D DCT (frequency domain filtering). + +This filter is not designed for real time. + +The filter accepts the following options: + +@table @option +@item sigma, s +Set the noise sigma constant. + +This @var{sigma} defines a hard threshold of @code{3 * sigma}; every DCT +coefficient (absolute value) below this threshold with be dropped. + +If you need a more advanced filtering, see @option{expr}. + +Default is @code{0}. + +@item overlap +Set number overlapping pixels for each block. Since the filter can be slow, you +may want to reduce this value, at the cost of a less effective filter and the +risk of various artefacts. + +If the overlapping value doesn't permit processing the whole input width or +height, a warning will be displayed and according borders won't be denoised. + +Default value is @var{blocksize}-1, which is the best possible setting. + +@item expr, e +Set the coefficient factor expression. + +For each coefficient of a DCT block, this expression will be evaluated as a +multiplier value for the coefficient. + +If this is option is set, the @option{sigma} option will be ignored. + +The absolute value of the coefficient can be accessed through the @var{c} +variable. + +@item n +Set the @var{blocksize} using the number of bits. @code{1<<@var{n}} defines the +@var{blocksize}, which is the width and height of the processed blocks. + +The default value is @var{3} (8x8) and can be raised to @var{4} for a +@var{blocksize} of 16x16. Note that changing this setting has huge consequences +on the speed processing. Also, a larger block size does not necessarily means a +better de-noising. +@end table + +@subsection Examples + +Apply a denoise with a @option{sigma} of @code{4.5}: +@example +dctdnoiz=4.5 +@end example + +The same operation can be achieved using the expression system: +@example +dctdnoiz=e='gte(c, 4.5*3)' +@end example + +Violent denoise using a block size of @code{16x16}: +@example +dctdnoiz=15:n=4 +@end example + +@section deband + +Remove banding artifacts from input video. +It works by replacing banded pixels with average value of referenced pixels. + +The filter accepts the following options: + +@table @option +@item 1thr +@item 2thr +@item 3thr +@item 4thr +Set banding detection threshold for each plane. Default is 0.02. +Valid range is 0.00003 to 0.5. +If difference between current pixel and reference pixel is less than threshold, +it will be considered as banded. + +@item range, r +Banding detection range in pixels. Default is 16. If positive, random number +in range 0 to set value will be used. If negative, exact absolute value +will be used. +The range defines square of four pixels around current pixel. + +@item direction, d +Set direction in radians from which four pixel will be compared. If positive, +random direction from 0 to set direction will be picked. If negative, exact of +absolute value will be picked. For example direction 0, -PI or -2*PI radians +will pick only pixels on same row and -PI/2 will pick only pixels on same +column. + +@item blur +If enabled, current pixel is compared with average value of all four +surrounding pixels. The default is enabled. If disabled current pixel is +compared with all four surrounding pixels. The pixel is considered banded +if only all four differences with surrounding pixels are less than threshold. +@end table + +@anchor{decimate} +@section decimate + +Drop duplicated frames at regular intervals. + +The filter accepts the following options: + +@table @option +@item cycle +Set the number of frames from which one will be dropped. Setting this to +@var{N} means one frame in every batch of @var{N} frames will be dropped. +Default is @code{5}. + +@item dupthresh +Set the threshold for duplicate detection. If the difference metric for a frame +is less than or equal to this value, then it is declared as duplicate. Default +is @code{1.1} + +@item scthresh +Set scene change threshold. Default is @code{15}. + +@item blockx +@item blocky +Set the size of the x and y-axis blocks used during metric calculations. +Larger blocks give better noise suppression, but also give worse detection of +small movements. Must be a power of two. Default is @code{32}. + +@item ppsrc +Mark main input as a pre-processed input and activate clean source input +stream. This allows the input to be pre-processed with various filters to help +the metrics calculation while keeping the frame selection lossless. When set to +@code{1}, the first stream is for the pre-processed input, and the second +stream is the clean source from where the kept frames are chosen. Default is +@code{0}. + +@item chroma +Set whether or not chroma is considered in the metric calculations. Default is +@code{1}. +@end table + +@section deflate + +Apply deflate effect to the video. + +This filter replaces the pixel by the local(3x3) average by taking into account +only values lower than the pixel. + +It accepts the following options: + +@table @option +@item threshold0 +@item threshold1 +@item threshold2 +@item threshold3 +Limit the maximum change for each plane, default is 65535. +If 0, plane will remain unchanged. +@end table + +@section dejudder + +Remove judder produced by partially interlaced telecined content. + +Judder can be introduced, for instance, by @ref{pullup} filter. If the original +source was partially telecined content then the output of @code{pullup,dejudder} +will have a variable frame rate. May change the recorded frame rate of the +container. Aside from that change, this filter will not affect constant frame +rate video. + +The option available in this filter is: +@table @option + +@item cycle +Specify the length of the window over which the judder repeats. + +Accepts any integer greater than 1. Useful values are: +@table @samp + +@item 4 +If the original was telecined from 24 to 30 fps (Film to NTSC). + +@item 5 +If the original was telecined from 25 to 30 fps (PAL to NTSC). + +@item 20 +If a mixture of the two. +@end table + +The default is @samp{4}. +@end table + @section delogo Suppress a TV station logo by a simple interpolation of the surrounding @@ -1080,29 +5273,37 @@ pixels. Just set a rectangle covering the logo and watch it disappear It accepts the following parameters: @table @option -@item x, y +@item x +@item y Specify the top left corner coordinates of the logo. They must be specified. -@item w, h +@item w +@item h Specify the width and height of the logo to clear. They must be specified. @item band, t Specify the thickness of the fuzzy edge of the rectangle (added to -@var{w} and @var{h}). The default value is 4. +@var{w} and @var{h}). The default value is 1. This option is +deprecated, setting higher values should no longer be necessary and +is not recommended. @item show When set to 1, a green rectangle is drawn on the screen to simplify -finding the right @var{x}, @var{y}, @var{w}, @var{h} parameters, and -@var{band} is set to 4. The default value is 0. +finding the right @var{x}, @var{y}, @var{w}, and @var{h} parameters. +The default value is 0. + +The rectangle is drawn on the outermost pixels which will be (partly) +replaced with interpolated values. The values of the next pixels +immediately outside this rectangle in each direction will be used to +compute the interpolated pixel values inside the rectangle. @end table -An example: +@subsection Examples @itemize - @item Set a rectangle covering the area with top left corner coordinates 0,0 and size 100x77, and a band of size 10: @@ -1112,6 +5313,192 @@ delogo=x=0:y=0:w=100:h=77:band=10 @end itemize +@section deshake + +Attempt to fix small changes in horizontal and/or vertical shift. This +filter helps remove camera shake from hand-holding a camera, bumping a +tripod, moving on a vehicle, etc. + +The filter accepts the following options: + +@table @option + +@item x +@item y +@item w +@item h +Specify a rectangular area where to limit the search for motion +vectors. +If desired the search for motion vectors can be limited to a +rectangular area of the frame defined by its top left corner, width +and height. These parameters have the same meaning as the drawbox +filter which can be used to visualise the position of the bounding +box. + +This is useful when simultaneous movement of subjects within the frame +might be confused for camera motion by the motion vector search. + +If any or all of @var{x}, @var{y}, @var{w} and @var{h} are set to -1 +then the full frame is used. This allows later options to be set +without specifying the bounding box for the motion vector search. + +Default - search the whole frame. + +@item rx +@item ry +Specify the maximum extent of movement in x and y directions in the +range 0-64 pixels. Default 16. + +@item edge +Specify how to generate pixels to fill blanks at the edge of the +frame. Available values are: +@table @samp +@item blank, 0 +Fill zeroes at blank locations +@item original, 1 +Original image at blank locations +@item clamp, 2 +Extruded edge value at blank locations +@item mirror, 3 +Mirrored edge at blank locations +@end table +Default value is @samp{mirror}. + +@item blocksize +Specify the blocksize to use for motion search. Range 4-128 pixels, +default 8. + +@item contrast +Specify the contrast threshold for blocks. Only blocks with more than +the specified contrast (difference between darkest and lightest +pixels) will be considered. Range 1-255, default 125. + +@item search +Specify the search strategy. Available values are: +@table @samp +@item exhaustive, 0 +Set exhaustive search +@item less, 1 +Set less exhaustive search. +@end table +Default value is @samp{exhaustive}. + +@item filename +If set then a detailed log of the motion search is written to the +specified file. + +@item opencl +If set to 1, specify using OpenCL capabilities, only available if +FFmpeg was configured with @code{--enable-opencl}. Default value is 0. + +@end table + +@section detelecine + +Apply an exact inverse of the telecine operation. It requires a predefined +pattern specified using the pattern option which must be the same as that passed +to the telecine filter. + +This filter accepts the following options: + +@table @option +@item first_field +@table @samp +@item top, t +top field first +@item bottom, b +bottom field first +The default value is @code{top}. +@end table + +@item pattern +A string of numbers representing the pulldown pattern you wish to apply. +The default value is @code{23}. + +@item start_frame +A number representing position of the first frame with respect to the telecine +pattern. This is to be used if the stream is cut. The default value is @code{0}. +@end table + +@section dilation + +Apply dilation effect to the video. + +This filter replaces the pixel by the local(3x3) maximum. + +It accepts the following options: + +@table @option +@item threshold0 +@item threshold1 +@item threshold2 +@item threshold3 +Limit the maximum change for each plane, default is 65535. +If 0, plane will remain unchanged. + +@item coordinates +Flag which specifies the pixel to refer to. Default is 255 i.e. all eight +pixels are used. + +Flags to local 3x3 coordinates maps like this: + + 1 2 3 + 4 5 + 6 7 8 +@end table + +@section displace + +Displace pixels as indicated by second and third input stream. + +It takes three input streams and outputs one stream, the first input is the +source, and second and third input are displacement maps. + +The second input specifies how much to displace pixels along the +x-axis, while the third input specifies how much to displace pixels +along the y-axis. +If one of displacement map streams terminates, last frame from that +displacement map will be used. + +Note that once generated, displacements maps can be reused over and over again. + +A description of the accepted options follows. + +@table @option +@item edge +Set displace behavior for pixels that are out of range. + +Available values are: +@table @samp +@item blank +Missing pixels are replaced by black pixels. + +@item smear +Adjacent pixels will spread out to replace missing pixels. + +@item wrap +Out of range pixels are wrapped so they point to pixels of other side. +@end table +Default is @samp{smear}. + +@end table + +@subsection Examples + +@itemize +@item +Add ripple effect to rgb input of video size hd720: +@example +ffmpeg -i INPUT -f lavfi -i nullsrc=s=hd720,lutrgb=128:128:128 -f lavfi -i nullsrc=s=hd720,geq='r=128+30*sin(2*PI*X/400+T):g=128+30*sin(2*PI*X/400+T):b=128+30*sin(2*PI*X/400+T)' -lavfi '[0][1][2]displace' OUTPUT +@end example + +@item +Add wave effect to rgb input of video size hd720: +@example +ffmpeg -i INPUT -f lavfi -i nullsrc=hd720,geq='r=128+80*(sin(sqrt((X-W/2)*(X-W/2)+(Y-H/2)*(Y-H/2))/220*2*PI+T)):g=128+80*(sin(sqrt((X-W/2)*(X-W/2)+(Y-H/2)*(Y-H/2))/220*2*PI+T)):b=128+80*(sin(sqrt((X-W/2)*(X-W/2)+(Y-H/2)*(Y-H/2))/220*2*PI+T))' -lavfi '[1]split[x][y],[0][x][y]displace' OUTPUT +@end example +@end itemize + @section drawbox Draw a colored box on the input image. @@ -1119,141 +5506,367 @@ Draw a colored box on the input image. It accepts the following parameters: @table @option +@item x +@item y +The expressions which specify the top left corner coordinates of the box. It defaults to 0. -@item x, y -Specify the top left corner coordinates of the box. It defaults to 0. - -@item width, height -Specify the width and height of the box; if 0 they are interpreted as +@item width, w +@item height, h +The expressions which specify the width and height of the box; if 0 they are interpreted as the input width and height. It defaults to 0. -@item color -Specify the color of the box to write. It can be the name of a color -(case insensitive match) or a 0xRRGGBB[AA] sequence. +@item color, c +Specify the color of the box to write. For the general syntax of this option, +check the "Color" section in the ffmpeg-utils manual. If the special +value @code{invert} is used, the box edge color is the same as the +video with inverted luma. + +@item thickness, t +The expression which sets the thickness of the box edge. Default value is @code{3}. + +See below for the list of accepted constants. @end table -Some examples: +The parameters for @var{x}, @var{y}, @var{w} and @var{h} and @var{t} are expressions containing the +following constants: + +@table @option +@item dar +The input display aspect ratio, it is the same as (@var{w} / @var{h}) * @var{sar}. + +@item hsub +@item vsub +horizontal and vertical chroma subsample values. For example for the +pixel format "yuv422p" @var{hsub} is 2 and @var{vsub} is 1. + +@item in_h, ih +@item in_w, iw +The input width and height. + +@item sar +The input sample aspect ratio. + +@item x +@item y +The x and y offset coordinates where the box is drawn. + +@item w +@item h +The width and height of the drawn box. + +@item t +The thickness of the drawn box. + +These constants allow the @var{x}, @var{y}, @var{w}, @var{h} and @var{t} expressions to refer to +each other, so you may for example specify @code{y=x/dar} or @code{h=w/dar}. + +@end table + +@subsection Examples + +@itemize +@item +Draw a black box around the edge of the input image: @example -# Draw a black box around the edge of the input image drawbox +@end example -# Draw a box with color red and an opacity of 50% -drawbox=x=10:y=20:width=200:height=60:color=red@@0.5" +@item +Draw a box with color red and an opacity of 50%: +@example +drawbox=10:20:200:60:red@@0.5 @end example -@section drawtext +The previous example can be specified as: +@example +drawbox=x=10:y=20:w=200:h=60:color=red@@0.5 +@end example -Draw a text string or text from a specified file on top of a video, using the -libfreetype library. +@item +Fill the box with pink color: +@example +drawbox=x=10:y=10:w=100:h=100:color=pink@@0.5:t=max +@end example -To enable compilation of this filter, you need to configure Libav with -@code{--enable-libfreetype}. -To enable default font fallback and the @var{font} option you need to -configure Libav with @code{--enable-libfontconfig}. +@item +Draw a 2-pixel red 2.40:1 mask: +@example +drawbox=x=-t:y=0.5*(ih-iw/2.4)-t:w=iw+t*2:h=iw/2.4+t*2:t=2:c=red +@end example +@end itemize -The filter also recognizes strftime() sequences in the provided text -and expands them accordingly. Check the documentation of strftime(). +@section drawgraph, adrawgraph + +Draw a graph using input video or audio metadata. It accepts the following parameters: @table @option +@item m1 +Set 1st frame metadata key from which metadata values will be used to draw a graph. -@item font -The font family to be used for drawing text. By default Sans. +@item fg1 +Set 1st foreground color expression. -@item fontfile -The font file to be used for drawing text. The path must be included. -This parameter is mandatory if the fontconfig support is disabled. +@item m2 +Set 2nd frame metadata key from which metadata values will be used to draw a graph. -@item text -The text string to be drawn. The text must be a sequence of UTF-8 -encoded characters. -This parameter is mandatory if no file is specified with the parameter -@var{textfile}. +@item fg2 +Set 2nd foreground color expression. -@item textfile -A text file containing text to be drawn. The text must be a sequence -of UTF-8 encoded characters. +@item m3 +Set 3rd frame metadata key from which metadata values will be used to draw a graph. -This parameter is mandatory if no text string is specified with the -parameter @var{text}. +@item fg3 +Set 3rd foreground color expression. + +@item m4 +Set 4th frame metadata key from which metadata values will be used to draw a graph. + +@item fg4 +Set 4th foreground color expression. + +@item min +Set minimal value of metadata value. + +@item max +Set maximal value of metadata value. + +@item bg +Set graph background color. Default is white. + +@item mode +Set graph mode. + +Available values for mode is: +@table @samp +@item bar +@item dot +@item line +@end table + +Default is @code{line}. -If both text and textfile are specified, an error is thrown. +@item slide +Set slide mode. -@item x, y -The offsets where text will be drawn within the video frame. -It is relative to the top/left border of the output image. -They accept expressions similar to the @ref{overlay} filter: +Available values for slide is: +@table @samp +@item frame +Draw new frame when right border is reached. + +@item replace +Replace old columns with new ones. + +@item scroll +Scroll from right to left. + +@item rscroll +Scroll from left to right. +@end table + +Default is @code{frame}. + +@item size +Set size of graph video. For the syntax of this option, check the +@ref{video size syntax,,"Video size" section in the ffmpeg-utils manual,ffmpeg-utils}. +The default value is @code{900x256}. + +The foreground color expressions can use the following variables: @table @option +@item MIN +Minimal value of metadata value. -@item x, y -The computed values for @var{x} and @var{y}. They are evaluated for -each new frame. +@item MAX +Maximal value of metadata value. -@item main_w, main_h -The main input width and height. +@item VAL +Current metadata key value. +@end table -@item W, H -These are the same as @var{main_w} and @var{main_h}. +The color is defined as 0xAABBGGRR. +@end table -@item text_w, text_h -The rendered text's width and height. +Example using metadata from @ref{signalstats} filter: +@example +signalstats,drawgraph=lavfi.signalstats.YAVG:min=0:max=255 +@end example -@item w, h -These are the same as @var{text_w} and @var{text_h}. +Example using metadata from @ref{ebur128} filter: +@example +ebur128=metadata=1,adrawgraph=lavfi.r128.M:min=-120:max=5 +@end example -@item n -The number of frames processed, starting from 0. +@section drawgrid + +Draw a grid on the input image. + +It accepts the following parameters: + +@table @option +@item x +@item y +The expressions which specify the coordinates of some point of grid intersection (meant to configure offset). Both default to 0. + +@item width, w +@item height, h +The expressions which specify the width and height of the grid cell, if 0 they are interpreted as the +input width and height, respectively, minus @code{thickness}, so image gets +framed. Default to 0. + +@item color, c +Specify the color of the grid. For the general syntax of this option, +check the "Color" section in the ffmpeg-utils manual. If the special +value @code{invert} is used, the grid color is the same as the +video with inverted luma. + +@item thickness, t +The expression which sets the thickness of the grid line. Default value is @code{1}. + +See below for the list of accepted constants. +@end table + +The parameters for @var{x}, @var{y}, @var{w} and @var{h} and @var{t} are expressions containing the +following constants: + +@table @option +@item dar +The input display aspect ratio, it is the same as (@var{w} / @var{h}) * @var{sar}. + +@item hsub +@item vsub +horizontal and vertical chroma subsample values. For example for the +pixel format "yuv422p" @var{hsub} is 2 and @var{vsub} is 1. + +@item in_h, ih +@item in_w, iw +The input grid cell width and height. + +@item sar +The input sample aspect ratio. + +@item x +@item y +The x and y coordinates of some point of grid intersection (meant to configure offset). + +@item w +@item h +The width and height of the drawn cell. @item t -The timestamp, expressed in seconds. It's NAN if the input timestamp is unknown. +The thickness of the drawn cell. + +These constants allow the @var{x}, @var{y}, @var{w}, @var{h} and @var{t} expressions to refer to +each other, so you may for example specify @code{y=x/dar} or @code{h=w/dar}. @end table -The default value of @var{x} and @var{y} is 0. +@subsection Examples + +@itemize +@item +Draw a grid with cell 100x100 pixels, thickness 2 pixels, with color red and an opacity of 50%: +@example +drawgrid=width=100:height=100:thickness=2:color=red@@0.5 +@end example + +@item +Draw a white 3x3 grid with an opacity of 50%: +@example +drawgrid=w=iw/3:h=ih/3:t=2:c=white@@0.5 +@end example +@end itemize + +@anchor{drawtext} +@section drawtext + +Draw a text string or text from a specified file on top of a video, using the +libfreetype library. + +To enable compilation of this filter, you need to configure FFmpeg with +@code{--enable-libfreetype}. +To enable default font fallback and the @var{font} option you need to +configure FFmpeg with @code{--enable-libfontconfig}. +To enable the @var{text_shaping} option, you need to configure FFmpeg with +@code{--enable-libfribidi}. + +@subsection Syntax + +It accepts the following parameters: + +@table @option + +@item box +Used to draw a box around text using the background color. +The value must be either 1 (enable) or 0 (disable). +The default value of @var{box} is 0. + +@item boxborderw +Set the width of the border to be drawn around the box using @var{boxcolor}. +The default value of @var{boxborderw} is 0. + +@item boxcolor +The color to be used for drawing box around text. For the syntax of this +option, check the "Color" section in the ffmpeg-utils manual. + +The default value of @var{boxcolor} is "white". + +@item borderw +Set the width of the border to be drawn around the text using @var{bordercolor}. +The default value of @var{borderw} is 0. + +@item bordercolor +Set the color to be used for drawing border around text. For the syntax of this +option, check the "Color" section in the ffmpeg-utils manual. + +The default value of @var{bordercolor} is "black". + +@item expansion +Select how the @var{text} is expanded. Can be either @code{none}, +@code{strftime} (deprecated) or +@code{normal} (default). See the @ref{drawtext_expansion, Text expansion} section +below for details. + +@item fix_bounds +If true, check and fix text coords to avoid clipping. + +@item fontcolor +The color to be used for drawing fonts. For the syntax of this option, check +the "Color" section in the ffmpeg-utils manual. + +The default value of @var{fontcolor} is "black". + +@item fontcolor_expr +String which is expanded the same way as @var{text} to obtain dynamic +@var{fontcolor} value. By default this option has empty value and is not +processed. When this option is set, it overrides @var{fontcolor} option. + +@item font +The font family to be used for drawing text. By default Sans. + +@item fontfile +The font file to be used for drawing text. The path must be included. +This parameter is mandatory if the fontconfig support is disabled. @item draw -Draw the text only if the expression evaluates as non-zero. -The expression accepts the same variables @var{x, y} do. -The default value is 1. +This option does not exist, please see the timeline system @item alpha Draw the text applying alpha blending. The value can be either a number between 0.0 and 1.0 The expression accepts the same variables @var{x, y} do. The default value is 1. +Please see fontcolor_expr @item fontsize The font size to be used for drawing text. The default value of @var{fontsize} is 16. -@item fontcolor -The color to be used for drawing fonts. -It is either a string (e.g. "red"), or in 0xRRGGBB[AA] format -(e.g. "0xff000033"), possibly followed by an alpha specifier. -The default value of @var{fontcolor} is "black". - -@item boxcolor -The color to be used for drawing box around text. -It is either a string (e.g. "yellow") or in 0xRRGGBB[AA] format -(e.g. "0xff00ff"), possibly followed by an alpha specifier. -The default value of @var{boxcolor} is "white". - -@item box -Used to draw a box around text using the background color. -The value must be either 1 (enable) or 0 (disable). -The default value of @var{box} is 0. - -@item shadowx, shadowy -The x and y offsets for the text shadow position with respect to the -position of the text. They can be either positive or negative -values. The default value for both is "0". - -@item shadowcolor -The color to be used for drawing a shadow behind the drawn text. It -can be a color name (e.g. "yellow") or a string in the 0xRRGGBB[AA] -form (e.g. "0xff00ff"), possibly followed by an alpha specifier. -The default value of @var{shadowcolor} is "black". +@item text_shaping +If set to 1, attempt to shape the text (for example, reverse the order of +right-to-left text and join Arabic characters) before drawing it. +Otherwise, just draw the text exactly as given. +By default 1 (if supported). @item ft_load_flags The flags to be used for loading the fonts. @@ -1276,47 +5889,575 @@ a combination of the following values: @item monochrome @item linear_design @item no_autohint -@item end table @end table -Default value is "render". +Default value is "default". For more information consult the documentation for the FT_LOAD_* libfreetype flags. +@item shadowcolor +The color to be used for drawing a shadow behind the drawn text. For the +syntax of this option, check the "Color" section in the ffmpeg-utils manual. + +The default value of @var{shadowcolor} is "black". + +@item shadowx +@item shadowy +The x and y offsets for the text shadow position with respect to the +position of the text. They can be either positive or negative +values. The default value for both is "0". + +@item start_number +The starting frame number for the n/frame_num variable. The default value +is "0". + @item tabsize The size in number of spaces to use for rendering the tab. Default value is 4. -@item fix_bounds -If true, check and fix text coords to avoid clipping. +@item timecode +Set the initial timecode representation in "hh:mm:ss[:;.]ff" +format. It can be used with or without text parameter. @var{timecode_rate} +option must be specified. + +@item timecode_rate, rate, r +Set the timecode frame rate (timecode only). + +@item text +The text string to be drawn. The text must be a sequence of UTF-8 +encoded characters. +This parameter is mandatory if no file is specified with the parameter +@var{textfile}. + +@item textfile +A text file containing text to be drawn. The text must be a sequence +of UTF-8 encoded characters. + +This parameter is mandatory if no text string is specified with the +parameter @var{text}. + +If both @var{text} and @var{textfile} are specified, an error is thrown. + +@item reload +If set to 1, the @var{textfile} will be reloaded before each frame. +Be sure to update it atomically, or it may be read partially, or even fail. + +@item x +@item y +The expressions which specify the offsets where text will be drawn +within the video frame. They are relative to the top/left border of the +output image. + +The default value of @var{x} and @var{y} is "0". + +See below for the list of accepted constants and functions. +@end table + +The parameters for @var{x} and @var{y} are expressions containing the +following constants and functions: + +@table @option +@item dar +input display aspect ratio, it is the same as (@var{w} / @var{h}) * @var{sar} + +@item hsub +@item vsub +horizontal and vertical chroma subsample values. For example for the +pixel format "yuv422p" @var{hsub} is 2 and @var{vsub} is 1. + +@item line_h, lh +the height of each text line + +@item main_h, h, H +the input height + +@item main_w, w, W +the input width + +@item max_glyph_a, ascent +the maximum distance from the baseline to the highest/upper grid +coordinate used to place a glyph outline point, for all the rendered +glyphs. +It is a positive value, due to the grid's orientation with the Y axis +upwards. + +@item max_glyph_d, descent +the maximum distance from the baseline to the lowest grid coordinate +used to place a glyph outline point, for all the rendered glyphs. +This is a negative value, due to the grid's orientation, with the Y axis +upwards. + +@item max_glyph_h +maximum glyph height, that is the maximum height for all the glyphs +contained in the rendered text, it is equivalent to @var{ascent} - +@var{descent}. + +@item max_glyph_w +maximum glyph width, that is the maximum width for all the glyphs +contained in the rendered text + +@item n +the number of input frame, starting from 0 + +@item rand(min, max) +return a random number included between @var{min} and @var{max} + +@item sar +The input sample aspect ratio. + +@item t +timestamp expressed in seconds, NAN if the input timestamp is unknown + +@item text_h, th +the height of the rendered text + +@item text_w, tw +the width of the rendered text + +@item x +@item y +the x and y offset coordinates where the text is drawn. + +These parameters allow the @var{x} and @var{y} expressions to refer +each other, so you can for example specify @code{y=x/dar}. @end table -For example the command: +@anchor{drawtext_expansion} +@subsection Text expansion + +If @option{expansion} is set to @code{strftime}, +the filter recognizes strftime() sequences in the provided text and +expands them accordingly. Check the documentation of strftime(). This +feature is deprecated. + +If @option{expansion} is set to @code{none}, the text is printed verbatim. + +If @option{expansion} is set to @code{normal} (which is the default), +the following expansion mechanism is used. + +The backslash character @samp{\}, followed by any character, always expands to +the second character. + +Sequence of the form @code{%@{...@}} are expanded. The text between the +braces is a function name, possibly followed by arguments separated by ':'. +If the arguments contain special characters or delimiters (':' or '@}'), +they should be escaped. + +Note that they probably must also be escaped as the value for the +@option{text} option in the filter argument string and as the filter +argument in the filtergraph description, and possibly also for the shell, +that makes up to four levels of escaping; using a text file avoids these +problems. + +The following functions are available: + +@table @command + +@item expr, e +The expression evaluation result. + +It must take one argument specifying the expression to be evaluated, +which accepts the same constants and functions as the @var{x} and +@var{y} values. Note that not all constants should be used, for +example the text size is not known when evaluating the expression, so +the constants @var{text_w} and @var{text_h} will have an undefined +value. + +@item expr_int_format, eif +Evaluate the expression's value and output as formatted integer. + +The first argument is the expression to be evaluated, just as for the @var{expr} function. +The second argument specifies the output format. Allowed values are @samp{x}, +@samp{X}, @samp{d} and @samp{u}. They are treated exactly as in the +@code{printf} function. +The third parameter is optional and sets the number of positions taken by the output. +It can be used to add padding with zeros from the left. + +@item gmtime +The time at which the filter is running, expressed in UTC. +It can accept an argument: a strftime() format string. + +@item localtime +The time at which the filter is running, expressed in the local time zone. +It can accept an argument: a strftime() format string. + +@item metadata +Frame metadata. It must take one argument specifying metadata key. + +@item n, frame_num +The frame number, starting from 0. + +@item pict_type +A 1 character description of the current picture type. + +@item pts +The timestamp of the current frame. +It can take up to three arguments. + +The first argument is the format of the timestamp; it defaults to @code{flt} +for seconds as a decimal number with microsecond accuracy; @code{hms} stands +for a formatted @var{[-]HH:MM:SS.mmm} timestamp with millisecond accuracy. +@code{gmtime} stands for the timestamp of the frame formatted as UTC time; +@code{localtime} stands for the timestamp of the frame formatted as +local time zone time. + +The second argument is an offset added to the timestamp. + +If the format is set to @code{localtime} or @code{gmtime}, +a third argument may be supplied: a strftime() format string. +By default, @var{YYYY-MM-DD HH:MM:SS} format will be used. +@end table + +@subsection Examples + +@itemize +@item +Draw "Test Text" with font FreeSerif, using the default values for the +optional parameters. + @example drawtext="fontfile=/usr/share/fonts/truetype/freefont/FreeSerif.ttf: text='Test Text'" @end example -will draw "Test Text" with font FreeSerif, using the default values -for the optional parameters. +@item +Draw 'Test Text' with font FreeSerif of size 24 at position x=100 +and y=50 (counting from the top-left corner of the screen), text is +yellow with a red box around it. Both the text and the box have an +opacity of 20%. -The command: @example drawtext="fontfile=/usr/share/fonts/truetype/freefont/FreeSerif.ttf: text='Test Text':\ x=100: y=50: fontsize=24: fontcolor=yellow@@0.2: box=1: boxcolor=red@@0.2" @end example -will draw 'Test Text' with font FreeSerif of size 24 at position x=100 -and y=50 (counting from the top-left corner of the screen), text is -yellow with a red box around it. Both the text and the box have an -opacity of 20%. - Note that the double quotes are not necessary if spaces are not used within the parameter list. +@item +Show the text at the center of the video frame: +@example +drawtext="fontsize=30:fontfile=FreeSerif.ttf:text='hello world':x=(w-text_w)/2:y=(h-text_h)/2" +@end example + +@item +Show a text line sliding from right to left in the last row of the video +frame. The file @file{LONG_LINE} is assumed to contain a single line +with no newlines. +@example +drawtext="fontsize=15:fontfile=FreeSerif.ttf:text=LONG_LINE:y=h-line_h:x=-50*t" +@end example + +@item +Show the content of file @file{CREDITS} off the bottom of the frame and scroll up. +@example +drawtext="fontsize=20:fontfile=FreeSerif.ttf:textfile=CREDITS:y=h-20*t" +@end example + +@item +Draw a single green letter "g", at the center of the input video. +The glyph baseline is placed at half screen height. +@example +drawtext="fontsize=60:fontfile=FreeSerif.ttf:fontcolor=green:text=g:x=(w-max_glyph_w)/2:y=h/2-ascent" +@end example + +@item +Show text for 1 second every 3 seconds: +@example +drawtext="fontfile=FreeSerif.ttf:fontcolor=white:x=100:y=x/dar:enable=lt(mod(t\,3)\,1):text='blink'" +@end example + +@item +Use fontconfig to set the font. Note that the colons need to be escaped. +@example +drawtext='fontfile=Linux Libertine O-40\:style=Semibold:text=FFmpeg' +@end example + +@item +Print the date of a real-time encoding (see strftime(3)): +@example +drawtext='fontfile=FreeSans.ttf:text=%@{localtime\:%a %b %d %Y@}' +@end example + +@item +Show text fading in and out (appearing/disappearing): +@example +#!/bin/sh +DS=1.0 # display start +DE=10.0 # display end +FID=1.5 # fade in duration +FOD=5 # fade out duration +ffplay -f lavfi "color,drawtext=text=TEST:fontsize=50:fontfile=FreeSerif.ttf:fontcolor_expr=ff0000%@{eif\\\\: clip(255*(1*between(t\\, $DS + $FID\\, $DE - $FOD) + ((t - $DS)/$FID)*between(t\\, $DS\\, $DS + $FID) + (-(t - $DE)/$FOD)*between(t\\, $DE - $FOD\\, $DE) )\\, 0\\, 255) \\\\: x\\\\: 2 @}" +@end example + +@end itemize + For more information about libfreetype, check: @url{http://www.freetype.org/}. +For more information about fontconfig, check: +@url{http://freedesktop.org/software/fontconfig/fontconfig-user.html}. + +For more information about libfribidi, check: +@url{http://fribidi.org/}. + +@section edgedetect + +Detect and draw edges. The filter uses the Canny Edge Detection algorithm. + +The filter accepts the following options: + +@table @option +@item low +@item high +Set low and high threshold values used by the Canny thresholding +algorithm. + +The high threshold selects the "strong" edge pixels, which are then +connected through 8-connectivity with the "weak" edge pixels selected +by the low threshold. + +@var{low} and @var{high} threshold values must be chosen in the range +[0,1], and @var{low} should be lesser or equal to @var{high}. + +Default value for @var{low} is @code{20/255}, and default value for @var{high} +is @code{50/255}. + +@item mode +Define the drawing mode. + +@table @samp +@item wires +Draw white/gray wires on black background. + +@item colormix +Mix the colors to create a paint/cartoon effect. +@end table + +Default value is @var{wires}. +@end table + +@subsection Examples + +@itemize +@item +Standard edge detection with custom values for the hysteresis thresholding: +@example +edgedetect=low=0.1:high=0.4 +@end example + +@item +Painting effect without thresholding: +@example +edgedetect=mode=colormix:high=0 +@end example +@end itemize + +@section eq +Set brightness, contrast, saturation and approximate gamma adjustment. + +The filter accepts the following options: + +@table @option +@item contrast +Set the contrast expression. The value must be a float value in range +@code{-2.0} to @code{2.0}. The default value is "1". + +@item brightness +Set the brightness expression. The value must be a float value in +range @code{-1.0} to @code{1.0}. The default value is "0". + +@item saturation +Set the saturation expression. The value must be a float in +range @code{0.0} to @code{3.0}. The default value is "1". + +@item gamma +Set the gamma expression. The value must be a float in range +@code{0.1} to @code{10.0}. The default value is "1". + +@item gamma_r +Set the gamma expression for red. The value must be a float in +range @code{0.1} to @code{10.0}. The default value is "1". + +@item gamma_g +Set the gamma expression for green. The value must be a float in range +@code{0.1} to @code{10.0}. The default value is "1". + +@item gamma_b +Set the gamma expression for blue. The value must be a float in range +@code{0.1} to @code{10.0}. The default value is "1". + +@item gamma_weight +Set the gamma weight expression. It can be used to reduce the effect +of a high gamma value on bright image areas, e.g. keep them from +getting overamplified and just plain white. The value must be a float +in range @code{0.0} to @code{1.0}. A value of @code{0.0} turns the +gamma correction all the way down while @code{1.0} leaves it at its +full strength. Default is "1". + +@item eval +Set when the expressions for brightness, contrast, saturation and +gamma expressions are evaluated. + +It accepts the following values: +@table @samp +@item init +only evaluate expressions once during the filter initialization or +when a command is processed + +@item frame +evaluate expressions for each incoming frame +@end table + +Default value is @samp{init}. +@end table + +The expressions accept the following parameters: +@table @option +@item n +frame count of the input frame starting from 0 + +@item pos +byte position of the corresponding packet in the input file, NAN if +unspecified + +@item r +frame rate of the input video, NAN if the input frame rate is unknown + +@item t +timestamp expressed in seconds, NAN if the input timestamp is unknown +@end table + +@subsection Commands +The filter supports the following commands: + +@table @option +@item contrast +Set the contrast expression. + +@item brightness +Set the brightness expression. + +@item saturation +Set the saturation expression. + +@item gamma +Set the gamma expression. + +@item gamma_r +Set the gamma_r expression. + +@item gamma_g +Set gamma_g expression. + +@item gamma_b +Set gamma_b expression. + +@item gamma_weight +Set gamma_weight expression. + +The command accepts the same syntax of the corresponding option. + +If the specified expression is not valid, it is kept at its current +value. + +@end table + +@section erosion + +Apply erosion effect to the video. + +This filter replaces the pixel by the local(3x3) minimum. + +It accepts the following options: + +@table @option +@item threshold0 +@item threshold1 +@item threshold2 +@item threshold3 +Limit the maximum change for each plane, default is 65535. +If 0, plane will remain unchanged. + +@item coordinates +Flag which specifies the pixel to refer to. Default is 255 i.e. all eight +pixels are used. + +Flags to local 3x3 coordinates maps like this: + + 1 2 3 + 4 5 + 6 7 8 +@end table + +@section extractplanes + +Extract color channel components from input video stream into +separate grayscale video streams. + +The filter accepts the following option: + +@table @option +@item planes +Set plane(s) to extract. + +Available values for planes are: +@table @samp +@item y +@item u +@item v +@item a +@item r +@item g +@item b +@end table + +Choosing planes not available in the input will result in an error. +That means you cannot select @code{r}, @code{g}, @code{b} planes +with @code{y}, @code{u}, @code{v} planes at same time. +@end table + +@subsection Examples + +@itemize +@item +Extract luma, u and v color channel component from input video frame +into 3 grayscale outputs: +@example +ffmpeg -i video.avi -filter_complex 'extractplanes=y+u+v[y][u][v]' -map '[y]' y.avi -map '[u]' u.avi -map '[v]' v.avi +@end example +@end itemize + +@section elbg + +Apply a posterize effect using the ELBG (Enhanced LBG) algorithm. + +For each input image, the filter will compute the optimal mapping from +the input to the output given the codebook length, that is the number +of distinct output colors. + +This filter accepts the following options. + +@table @option +@item codebook_length, l +Set codebook length. The value must be a positive integer, and +represents the number of distinct output colors. Default value is 256. + +@item nb_steps, n +Set the maximum number of iterations to apply for computing the optimal +mapping. The higher the value the better the result and the higher the +computation time. Default value is 1. + +@item seed, s +Set a random seed, must be an integer included between 0 and +UINT32_MAX. If not specified, or if explicitly set to -1, the filter +will try to use a good random seed on a best effort basis. + +@item pal8 +Set pal8 output pixel format. This option does not work with codebook +length greater than 256. +@end table + @section fade Apply a fade-in/out effect to the input video. @@ -1324,34 +6465,497 @@ Apply a fade-in/out effect to the input video. It accepts the following parameters: @table @option - -@item type +@item type, t The effect type can be either "in" for a fade-in, or "out" for a fade-out effect. +Default is @code{in}. -@item start_frame -The number of the frame to start applying the fade effect at. +@item start_frame, s +Specify the number of the frame to start applying the fade +effect at. Default is 0. -@item nb_frames +@item nb_frames, n The number of frames that the fade effect lasts. At the end of the fade-in effect, the output video will have the same intensity as the input video. -At the end of the fade-out transition, the output video will be completely black. +At the end of the fade-out transition, the output video will be filled with the +selected @option{color}. +Default is 25. +@item alpha +If set to 1, fade only alpha channel, if one exists on the input. +Default value is 0. + +@item start_time, st +Specify the timestamp (in seconds) of the frame to start to apply the fade +effect. If both start_frame and start_time are specified, the fade will start at +whichever comes last. Default is 0. + +@item duration, d +The number of seconds for which the fade effect has to last. At the end of the +fade-in effect the output video will have the same intensity as the input video, +at the end of the fade-out transition the output video will be filled with the +selected @option{color}. +If both duration and nb_frames are specified, duration is used. Default is 0 +(nb_frames is used by default). + +@item color, c +Specify the color of the fade. Default is "black". @end table -Some examples: +@subsection Examples + +@itemize +@item +Fade in the first 30 frames of video: +@example +fade=in:0:30 +@end example + +The command above is equivalent to: @example -# Fade in the first 30 frames of video -fade=type=in:nb_frames=30 +fade=t=in:s=0:n=30 +@end example -# Fade out the last 45 frames of a 200-frame video +@item +Fade out the last 45 frames of a 200-frame video: +@example +fade=out:155:45 fade=type=out:start_frame=155:nb_frames=45 +@end example + +@item +Fade in the first 25 frames and fade out the last 25 frames of a 1000-frame video: +@example +fade=in:0:25, fade=out:975:25 +@end example + +@item +Make the first 5 frames yellow, then fade in from frame 5-24: +@example +fade=in:5:20:color=yellow +@end example + +@item +Fade in alpha over first 25 frames of video: +@example +fade=in:0:25:alpha=1 +@end example + +@item +Make the first 5.5 seconds black, then fade in for 0.5 seconds: +@example +fade=t=in:st=5.5:d=0.5 +@end example + +@end itemize + +@section fftfilt +Apply arbitrary expressions to samples in frequency domain + +@table @option +@item dc_Y +Adjust the dc value (gain) of the luma plane of the image. The filter +accepts an integer value in range @code{0} to @code{1000}. The default +value is set to @code{0}. + +@item dc_U +Adjust the dc value (gain) of the 1st chroma plane of the image. The +filter accepts an integer value in range @code{0} to @code{1000}. The +default value is set to @code{0}. + +@item dc_V +Adjust the dc value (gain) of the 2nd chroma plane of the image. The +filter accepts an integer value in range @code{0} to @code{1000}. The +default value is set to @code{0}. + +@item weight_Y +Set the frequency domain weight expression for the luma plane. + +@item weight_U +Set the frequency domain weight expression for the 1st chroma plane. + +@item weight_V +Set the frequency domain weight expression for the 2nd chroma plane. + +The filter accepts the following variables: +@item X +@item Y +The coordinates of the current sample. + +@item W +@item H +The width and height of the image. +@end table + +@subsection Examples + +@itemize +@item +High-pass: +@example +fftfilt=dc_Y=128:weight_Y='squish(1-(Y+X)/100)' +@end example + +@item +Low-pass: +@example +fftfilt=dc_Y=0:weight_Y='squish((Y+X)/100-1)' +@end example + +@item +Sharpen: +@example +fftfilt=dc_Y=0:weight_Y='1+squish(1-(Y+X)/100)' +@end example + +@item +Blur: +@example +fftfilt=dc_Y=0:weight_Y='exp(-4 * ((Y+X)/(W+H)))' +@end example + +@end itemize + +@section field + +Extract a single field from an interlaced image using stride +arithmetic to avoid wasting CPU time. The output frames are marked as +non-interlaced. + +The filter accepts the following options: + +@table @option +@item type +Specify whether to extract the top (if the value is @code{0} or +@code{top}) or the bottom field (if the value is @code{1} or +@code{bottom}). +@end table + +@section fieldmatch + +Field matching filter for inverse telecine. It is meant to reconstruct the +progressive frames from a telecined stream. The filter does not drop duplicated +frames, so to achieve a complete inverse telecine @code{fieldmatch} needs to be +followed by a decimation filter such as @ref{decimate} in the filtergraph. + +The separation of the field matching and the decimation is notably motivated by +the possibility of inserting a de-interlacing filter fallback between the two. +If the source has mixed telecined and real interlaced content, +@code{fieldmatch} will not be able to match fields for the interlaced parts. +But these remaining combed frames will be marked as interlaced, and thus can be +de-interlaced by a later filter such as @ref{yadif} before decimation. + +In addition to the various configuration options, @code{fieldmatch} can take an +optional second stream, activated through the @option{ppsrc} option. If +enabled, the frames reconstruction will be based on the fields and frames from +this second stream. This allows the first input to be pre-processed in order to +help the various algorithms of the filter, while keeping the output lossless +(assuming the fields are matched properly). Typically, a field-aware denoiser, +or brightness/contrast adjustments can help. + +Note that this filter uses the same algorithms as TIVTC/TFM (AviSynth project) +and VIVTC/VFM (VapourSynth project). The later is a light clone of TFM from +which @code{fieldmatch} is based on. While the semantic and usage are very +close, some behaviour and options names can differ. + +The @ref{decimate} filter currently only works for constant frame rate input. +If your input has mixed telecined (30fps) and progressive content with a lower +framerate like 24fps use the following filterchain to produce the necessary cfr +stream: @code{dejudder,fps=30000/1001,fieldmatch,decimate}. + +The filter accepts the following options: + +@table @option +@item order +Specify the assumed field order of the input stream. Available values are: + +@table @samp +@item auto +Auto detect parity (use FFmpeg's internal parity value). +@item bff +Assume bottom field first. +@item tff +Assume top field first. +@end table + +Note that it is sometimes recommended not to trust the parity announced by the +stream. + +Default value is @var{auto}. + +@item mode +Set the matching mode or strategy to use. @option{pc} mode is the safest in the +sense that it won't risk creating jerkiness due to duplicate frames when +possible, but if there are bad edits or blended fields it will end up +outputting combed frames when a good match might actually exist. On the other +hand, @option{pcn_ub} mode is the most risky in terms of creating jerkiness, +but will almost always find a good frame if there is one. The other values are +all somewhere in between @option{pc} and @option{pcn_ub} in terms of risking +jerkiness and creating duplicate frames versus finding good matches in sections +with bad edits, orphaned fields, blended fields, etc. + +More details about p/c/n/u/b are available in @ref{p/c/n/u/b meaning} section. + +Available values are: + +@table @samp +@item pc +2-way matching (p/c) +@item pc_n +2-way matching, and trying 3rd match if still combed (p/c + n) +@item pc_u +2-way matching, and trying 3rd match (same order) if still combed (p/c + u) +@item pc_n_ub +2-way matching, trying 3rd match if still combed, and trying 4th/5th matches if +still combed (p/c + n + u/b) +@item pcn +3-way matching (p/c/n) +@item pcn_ub +3-way matching, and trying 4th/5th matches if all 3 of the original matches are +detected as combed (p/c/n + u/b) +@end table + +The parenthesis at the end indicate the matches that would be used for that +mode assuming @option{order}=@var{tff} (and @option{field} on @var{auto} or +@var{top}). + +In terms of speed @option{pc} mode is by far the fastest and @option{pcn_ub} is +the slowest. + +Default value is @var{pc_n}. + +@item ppsrc +Mark the main input stream as a pre-processed input, and enable the secondary +input stream as the clean source to pick the fields from. See the filter +introduction for more details. It is similar to the @option{clip2} feature from +VFM/TFM. + +Default value is @code{0} (disabled). -# Fade in the first 25 frames and fade out the last 25 frames of a 1000-frame video -fade=type=in:start_frame=0:nb_frames=25, fade=type=out:start_frame=975:nb_frames=25 +@item field +Set the field to match from. It is recommended to set this to the same value as +@option{order} unless you experience matching failures with that setting. In +certain circumstances changing the field that is used to match from can have a +large impact on matching performance. Available values are: -# Make the first 5 frames black, then fade in from frame 5-24 -fade=type=in:start_frame=5:nb_frames=20 +@table @samp +@item auto +Automatic (same value as @option{order}). +@item bottom +Match from the bottom field. +@item top +Match from the top field. +@end table + +Default value is @var{auto}. + +@item mchroma +Set whether or not chroma is included during the match comparisons. In most +cases it is recommended to leave this enabled. You should set this to @code{0} +only if your clip has bad chroma problems such as heavy rainbowing or other +artifacts. Setting this to @code{0} could also be used to speed things up at +the cost of some accuracy. + +Default value is @code{1}. + +@item y0 +@item y1 +These define an exclusion band which excludes the lines between @option{y0} and +@option{y1} from being included in the field matching decision. An exclusion +band can be used to ignore subtitles, a logo, or other things that may +interfere with the matching. @option{y0} sets the starting scan line and +@option{y1} sets the ending line; all lines in between @option{y0} and +@option{y1} (including @option{y0} and @option{y1}) will be ignored. Setting +@option{y0} and @option{y1} to the same value will disable the feature. +@option{y0} and @option{y1} defaults to @code{0}. + +@item scthresh +Set the scene change detection threshold as a percentage of maximum change on +the luma plane. Good values are in the @code{[8.0, 14.0]} range. Scene change +detection is only relevant in case @option{combmatch}=@var{sc}. The range for +@option{scthresh} is @code{[0.0, 100.0]}. + +Default value is @code{12.0}. + +@item combmatch +When @option{combatch} is not @var{none}, @code{fieldmatch} will take into +account the combed scores of matches when deciding what match to use as the +final match. Available values are: + +@table @samp +@item none +No final matching based on combed scores. +@item sc +Combed scores are only used when a scene change is detected. +@item full +Use combed scores all the time. +@end table + +Default is @var{sc}. + +@item combdbg +Force @code{fieldmatch} to calculate the combed metrics for certain matches and +print them. This setting is known as @option{micout} in TFM/VFM vocabulary. +Available values are: + +@table @samp +@item none +No forced calculation. +@item pcn +Force p/c/n calculations. +@item pcnub +Force p/c/n/u/b calculations. +@end table + +Default value is @var{none}. + +@item cthresh +This is the area combing threshold used for combed frame detection. This +essentially controls how "strong" or "visible" combing must be to be detected. +Larger values mean combing must be more visible and smaller values mean combing +can be less visible or strong and still be detected. Valid settings are from +@code{-1} (every pixel will be detected as combed) to @code{255} (no pixel will +be detected as combed). This is basically a pixel difference value. A good +range is @code{[8, 12]}. + +Default value is @code{9}. + +@item chroma +Sets whether or not chroma is considered in the combed frame decision. Only +disable this if your source has chroma problems (rainbowing, etc.) that are +causing problems for the combed frame detection with chroma enabled. Actually, +using @option{chroma}=@var{0} is usually more reliable, except for the case +where there is chroma only combing in the source. + +Default value is @code{0}. + +@item blockx +@item blocky +Respectively set the x-axis and y-axis size of the window used during combed +frame detection. This has to do with the size of the area in which +@option{combpel} pixels are required to be detected as combed for a frame to be +declared combed. See the @option{combpel} parameter description for more info. +Possible values are any number that is a power of 2 starting at 4 and going up +to 512. + +Default value is @code{16}. + +@item combpel +The number of combed pixels inside any of the @option{blocky} by +@option{blockx} size blocks on the frame for the frame to be detected as +combed. While @option{cthresh} controls how "visible" the combing must be, this +setting controls "how much" combing there must be in any localized area (a +window defined by the @option{blockx} and @option{blocky} settings) on the +frame. Minimum value is @code{0} and maximum is @code{blocky x blockx} (at +which point no frames will ever be detected as combed). This setting is known +as @option{MI} in TFM/VFM vocabulary. + +Default value is @code{80}. +@end table + +@anchor{p/c/n/u/b meaning} +@subsection p/c/n/u/b meaning + +@subsubsection p/c/n + +We assume the following telecined stream: + +@example +Top fields: 1 2 2 3 4 +Bottom fields: 1 2 3 4 4 +@end example + +The numbers correspond to the progressive frame the fields relate to. Here, the +first two frames are progressive, the 3rd and 4th are combed, and so on. + +When @code{fieldmatch} is configured to run a matching from bottom +(@option{field}=@var{bottom}) this is how this input stream get transformed: + +@example +Input stream: + T 1 2 2 3 4 + B 1 2 3 4 4 <-- matching reference + +Matches: c c n n c + +Output stream: + T 1 2 3 4 4 + B 1 2 3 4 4 +@end example + +As a result of the field matching, we can see that some frames get duplicated. +To perform a complete inverse telecine, you need to rely on a decimation filter +after this operation. See for instance the @ref{decimate} filter. + +The same operation now matching from top fields (@option{field}=@var{top}) +looks like this: + +@example +Input stream: + T 1 2 2 3 4 <-- matching reference + B 1 2 3 4 4 + +Matches: c c p p c + +Output stream: + T 1 2 2 3 4 + B 1 2 2 3 4 +@end example + +In these examples, we can see what @var{p}, @var{c} and @var{n} mean; +basically, they refer to the frame and field of the opposite parity: + +@itemize +@item @var{p} matches the field of the opposite parity in the previous frame +@item @var{c} matches the field of the opposite parity in the current frame +@item @var{n} matches the field of the opposite parity in the next frame +@end itemize + +@subsubsection u/b + +The @var{u} and @var{b} matching are a bit special in the sense that they match +from the opposite parity flag. In the following examples, we assume that we are +currently matching the 2nd frame (Top:2, bottom:2). According to the match, a +'x' is placed above and below each matched fields. + +With bottom matching (@option{field}=@var{bottom}): +@example +Match: c p n b u + + x x x x x + Top 1 2 2 1 2 2 1 2 2 1 2 2 1 2 2 + Bottom 1 2 3 1 2 3 1 2 3 1 2 3 1 2 3 + x x x x x + +Output frames: + 2 1 2 2 2 + 2 2 2 1 3 +@end example + +With top matching (@option{field}=@var{top}): +@example +Match: c p n b u + + x x x x x + Top 1 2 2 1 2 2 1 2 2 1 2 2 1 2 2 + Bottom 1 2 3 1 2 3 1 2 3 1 2 3 1 2 3 + x x x x x + +Output frames: + 2 2 2 1 2 + 2 1 3 2 2 +@end example + +@subsection Examples + +Simple IVTC of a top field first telecined stream: +@example +fieldmatch=order=tff:combmatch=none, decimate +@end example + +Advanced IVTC, with fallback on @ref{yadif} for still combed frames: +@example +fieldmatch=order=tff:combmatch=full, yadif=deint=interlaced, decimate @end example @section fieldorder @@ -1367,7 +6971,7 @@ The output field order. Valid values are @var{tff} for top field first or @var{b for bottom field first. @end table -The default value is "tff". +The default value is @samp{tff}. The transformation is done by shifting the picture content up or down by one line, and filling the remaining line with appropriate picture content. @@ -1382,10 +6986,10 @@ which is bottom field first. For example: @example -./avconv -i in.vob -vf "fieldorder=order=bff" out.dv +ffmpeg -i in.vob -vf "fieldorder=bff" out.dv @end example -@section fifo +@section fifo, afifo Buffer input images and send them when they are requested. @@ -1394,6 +6998,71 @@ framework. It does not take parameters. +@section find_rect + +Find a rectangular object + +It accepts the following options: + +@table @option +@item object +Filepath of the object image, needs to be in gray8. + +@item threshold +Detection threshold, default is 0.5. + +@item mipmaps +Number of mipmaps, default is 3. + +@item xmin, ymin, xmax, ymax +Specifies the rectangle in which to search. +@end table + +@subsection Examples + +@itemize +@item +Generate a representative palette of a given video using @command{ffmpeg}: +@example +ffmpeg -i file.ts -vf find_rect=newref.pgm,cover_rect=cover.jpg:mode=cover new.mkv +@end example +@end itemize + +@section cover_rect + +Cover a rectangular object + +It accepts the following options: + +@table @option +@item cover +Filepath of the optional cover image, needs to be in yuv420. + +@item mode +Set covering mode. + +It accepts the following values: +@table @samp +@item cover +cover it by the supplied image +@item blur +cover it by interpolating the surrounding pixels +@end table + +Default value is @var{blur}. +@end table + +@subsection Examples + +@itemize +@item +Generate a representative palette of a given video using @command{ffmpeg}: +@example +ffmpeg -i file.ts -vf find_rect=newref.pgm,cover_rect=cover.jpg:mode=cover new.mkv +@end example +@end itemize + +@anchor{format} @section format Convert the input video to one of the specified pixel formats. @@ -1409,26 +7078,50 @@ A '|'-separated list of pixel format names, such as @end table -Some examples: +@subsection Examples + +@itemize +@item +Convert the input video to the @var{yuv420p} format @example -# Convert the input video to the "yuv420p" format format=pix_fmts=yuv420p +@end example -# Convert the input video to any of the formats in the list +Convert the input video to any of the formats in the list +@example format=pix_fmts=yuv420p|yuv444p|yuv410p @end example +@end itemize @anchor{fps} @section fps -Convert the video to specified constant framerate by duplicating or dropping +Convert the video to specified constant frame rate by duplicating or dropping frames as necessary. It accepts the following parameters: @table @option @item fps -The desired output framerate. +The desired output frame rate. The default is @code{25}. + +@item round +Rounding method. + +Possible values are: +@table @option +@item zero +zero round towards 0 +@item inf +round away from 0 +@item down +round towards -infinity +@item up +round towards +infinity +@item near +round to nearest +@end table +The default is @code{near}. @item start_time Assume the first PTS should be the given value, in seconds. This allows for @@ -1440,6 +7133,27 @@ frames with a negative PTS. @end table +Alternatively, the options can be specified as a flat string: +@var{fps}[:@var{round}]. + +See also the @ref{setpts} filter. + +@subsection Examples + +@itemize +@item +A typical usage in order to set the fps to 25: +@example +fps=fps=25 +@end example + +@item +Sets the fps to 24, using abbreviation and rounding method to round to nearest: +@example +fps=fps=film:round=near +@end example +@end itemize + @section framepack Pack two different video streams into a stereoscopic video, setting proper @@ -1479,19 +7193,75 @@ Some examples: @example # Convert left and right views into a frame-sequential video -avconv -i LEFT -i RIGHT -filter_complex framepack=frameseq OUTPUT +ffmpeg -i LEFT -i RIGHT -filter_complex framepack=frameseq OUTPUT # Convert views into a side-by-side video with the same output resolution as the input -avconv -i LEFT -i RIGHT -filter_complex [0:v]scale=w=iw/2[left],[1:v]scale=w=iw/2[right],[left][right]framepack=sbs OUTPUT +ffmpeg -i LEFT -i RIGHT -filter_complex [0:v]scale=w=iw/2[left],[1:v]scale=w=iw/2[right],[left][right]framepack=sbs OUTPUT @end example +@section framerate + +Change the frame rate by interpolating new video output frames from the source +frames. + +This filter is not designed to function correctly with interlaced media. If +you wish to change the frame rate of interlaced media then you are required +to deinterlace before this filter and re-interlace after this filter. + +A description of the accepted options follows. + +@table @option +@item fps +Specify the output frames per second. This option can also be specified +as a value alone. The default is @code{50}. + +@item interp_start +Specify the start of a range where the output frame will be created as a +linear interpolation of two frames. The range is [@code{0}-@code{255}], +the default is @code{15}. + +@item interp_end +Specify the end of a range where the output frame will be created as a +linear interpolation of two frames. The range is [@code{0}-@code{255}], +the default is @code{240}. + +@item scene +Specify the level at which a scene change is detected as a value between +0 and 100 to indicate a new scene; a low value reflects a low +probability for the current frame to introduce a new scene, while a higher +value means the current frame is more likely to be one. +The default is @code{7}. + +@item flags +Specify flags influencing the filter process. + +Available value for @var{flags} is: + +@table @option +@item scene_change_detect, scd +Enable scene change detection using the value of the option @var{scene}. +This flag is enabled by default. +@end table +@end table + +@section framestep + +Select one frame every N-th frame. + +This filter accepts the following option: +@table @option +@item step +Select frame after every @code{step} frames. +Allowed values are positive integers higher than 0. Default value is @code{1}. +@end table + @anchor{frei0r} @section frei0r Apply a frei0r effect to the input video. To enable the compilation of this filter, you need to install the frei0r -header and configure Libav with --enable-frei0r. +header and configure FFmpeg with @code{--enable-frei0r}. It accepts the following parameters: @@ -1513,35 +7283,214 @@ A '|'-separated list of parameters to pass to the frei0r effect. A frei0r effect parameter can be a boolean (its value is either "y" or "n"), a double, a color (specified as @var{R}/@var{G}/@var{B}, where @var{R}, @var{G}, and @var{B} are floating point -numbers between 0.0 and 1.0, inclusive) or by an @code{av_parse_color()} color -description), a position (specified as @var{X}/@var{Y}, where +numbers between 0.0 and 1.0, inclusive) or by a color description specified in the "Color" +section in the ffmpeg-utils manual), a position (specified as @var{X}/@var{Y}, where @var{X} and @var{Y} are floating point numbers) and/or a string. The number and types of parameters depend on the loaded effect. If an effect parameter is not specified, the default value is set. -Some examples: +@subsection Examples + +@itemize +@item +Apply the distort0r effect, setting the first two double parameters: @example -# Apply the distort0r effect, setting the first two double parameters frei0r=filter_name=distort0r:filter_params=0.5|0.01 +@end example -# Apply the colordistance effect, taking a color as the first parameter +@item +Apply the colordistance effect, taking a color as the first parameter: +@example frei0r=colordistance:0.2/0.3/0.4 frei0r=colordistance:violet frei0r=colordistance:0x112233 +@end example -# Apply the perspective effect, specifying the top left and top right -# image positions +@item +Apply the perspective effect, specifying the top left and top right image +positions: +@example frei0r=perspective:0.2/0.2|0.8/0.2 @end example +@end itemize For more information, see -@url{http://piksel.org/frei0r} +@url{http://frei0r.dyne.org} + +@section fspp + +Apply fast and simple postprocessing. It is a faster version of @ref{spp}. + +It splits (I)DCT into horizontal/vertical passes. Unlike the simple post- +processing filter, one of them is performed once per block, not per pixel. +This allows for much higher speed. + +The filter accepts the following options: + +@table @option +@item quality +Set quality. This option defines the number of levels for averaging. It accepts +an integer in the range 4-5. Default value is @code{4}. + +@item qp +Force a constant quantization parameter. It accepts an integer in range 0-63. +If not set, the filter will use the QP from the video stream (if available). + +@item strength +Set filter strength. It accepts an integer in range -15 to 32. Lower values mean +more details but also more artifacts, while higher values make the image smoother +but also blurrier. Default value is @code{0} − PSNR optimal. + +@item use_bframe_qp +Enable the use of the QP from the B-Frames if set to @code{1}. Using this +option may cause flicker since the B-Frames have often larger QP. Default is +@code{0} (not enabled). + +@end table + +@section geq + +The filter accepts the following options: + +@table @option +@item lum_expr, lum +Set the luminance expression. +@item cb_expr, cb +Set the chrominance blue expression. +@item cr_expr, cr +Set the chrominance red expression. +@item alpha_expr, a +Set the alpha expression. +@item red_expr, r +Set the red expression. +@item green_expr, g +Set the green expression. +@item blue_expr, b +Set the blue expression. +@end table + +The colorspace is selected according to the specified options. If one +of the @option{lum_expr}, @option{cb_expr}, or @option{cr_expr} +options is specified, the filter will automatically select a YCbCr +colorspace. If one of the @option{red_expr}, @option{green_expr}, or +@option{blue_expr} options is specified, it will select an RGB +colorspace. + +If one of the chrominance expression is not defined, it falls back on the other +one. If no alpha expression is specified it will evaluate to opaque value. +If none of chrominance expressions are specified, they will evaluate +to the luminance expression. + +The expressions can use the following variables and functions: + +@table @option +@item N +The sequential number of the filtered frame, starting from @code{0}. + +@item X +@item Y +The coordinates of the current sample. + +@item W +@item H +The width and height of the image. + +@item SW +@item SH +Width and height scale depending on the currently filtered plane. It is the +ratio between the corresponding luma plane number of pixels and the current +plane ones. E.g. for YUV4:2:0 the values are @code{1,1} for the luma plane, and +@code{0.5,0.5} for chroma planes. + +@item T +Time of the current frame, expressed in seconds. + +@item p(x, y) +Return the value of the pixel at location (@var{x},@var{y}) of the current +plane. + +@item lum(x, y) +Return the value of the pixel at location (@var{x},@var{y}) of the luminance +plane. + +@item cb(x, y) +Return the value of the pixel at location (@var{x},@var{y}) of the +blue-difference chroma plane. Return 0 if there is no such plane. + +@item cr(x, y) +Return the value of the pixel at location (@var{x},@var{y}) of the +red-difference chroma plane. Return 0 if there is no such plane. + +@item r(x, y) +@item g(x, y) +@item b(x, y) +Return the value of the pixel at location (@var{x},@var{y}) of the +red/green/blue component. Return 0 if there is no such component. + +@item alpha(x, y) +Return the value of the pixel at location (@var{x},@var{y}) of the alpha +plane. Return 0 if there is no such plane. +@end table + +For functions, if @var{x} and @var{y} are outside the area, the value will be +automatically clipped to the closer edge. + +@subsection Examples + +@itemize +@item +Flip the image horizontally: +@example +geq=p(W-X\,Y) +@end example + +@item +Generate a bidimensional sine wave, with angle @code{PI/3} and a +wavelength of 100 pixels: +@example +geq=128 + 100*sin(2*(PI/100)*(cos(PI/3)*(X-50*T) + sin(PI/3)*Y)):128:128 +@end example + +@item +Generate a fancy enigmatic moving light: +@example +nullsrc=s=256x256,geq=random(1)/hypot(X-cos(N*0.07)*W/2-W/2\,Y-sin(N*0.09)*H/2-H/2)^2*1000000*sin(N*0.02):128:128 +@end example + +@item +Generate a quick emboss effect: +@example +format=gray,geq=lum_expr='(p(X,Y)+(256-p(X-4,Y-4)))/2' +@end example + +@item +Modify RGB components depending on pixel position: +@example +geq=r='X/W*r(X,Y)':g='(1-X/W)*g(X,Y)':b='(H-Y)/H*b(X,Y)' +@end example + +@item +Create a radial gradient that is the same size as the input (also see +the @ref{vignette} filter): +@example +geq=lum=255*gauss((X/W-0.5)*3)*gauss((Y/H-0.5)*3)/gauss(0)/gauss(0),format=gray +@end example + +@item +Create a linear gradient to use as a mask for another filter, then +compose with @ref{overlay}. In this example the video will gradually +become more blurry from the top to the bottom of the y-axis as defined +by the linear gradient: +@example +ffmpeg -i input.mp4 -filter_complex "geq=lum=255*(Y/H),format=gray[grad];[0:v]boxblur=4[blur];[blur][grad]alphamerge[alpha];[0:v][alpha]overlay" output.mp4 +@end example +@end itemize @section gradfun Fix the banding artifacts that are sometimes introduced into nearly flat -regions by truncation to 8bit colordepth. +regions by truncation to 8bit color depth. Interpolate the gradients that should go where the bands are, and dither them. @@ -1567,23 +7516,201 @@ values will be clipped to the valid range. @end table +Alternatively, the options can be specified as a flat string: +@var{strength}[:@var{radius}] + +@subsection Examples + +@itemize +@item +Apply the filter with a @code{3.5} strength and radius of @code{8}: @example -# Default parameters -gradfun=strength=1.2:radius=16 +gradfun=3.5:8 +@end example -# Omitting the radius -gradfun=1.2 +@item +Specify radius, omitting the strength (which will fall-back to the default +value): +@example +gradfun=radius=8 +@end example + +@end itemize + +@anchor{haldclut} +@section haldclut + +Apply a Hald CLUT to a video stream. + +First input is the video stream to process, and second one is the Hald CLUT. +The Hald CLUT input can be a simple picture or a complete video stream. + +The filter accepts the following options: + +@table @option +@item shortest +Force termination when the shortest input terminates. Default is @code{0}. +@item repeatlast +Continue applying the last CLUT after the end of the stream. A value of +@code{0} disable the filter after the last frame of the CLUT is reached. +Default is @code{1}. +@end table + +@code{haldclut} also has the same interpolation options as @ref{lut3d} (both +filters share the same internals). + +More information about the Hald CLUT can be found on Eskil Steenberg's website +(Hald CLUT author) at @url{http://www.quelsolaar.com/technology/clut.html}. + +@subsection Workflow examples + +@subsubsection Hald CLUT video stream + +Generate an identity Hald CLUT stream altered with various effects: +@example +ffmpeg -f lavfi -i @ref{haldclutsrc}=8 -vf "hue=H=2*PI*t:s=sin(2*PI*t)+1, curves=cross_process" -t 10 -c:v ffv1 clut.nut +@end example + +Note: make sure you use a lossless codec. + +Then use it with @code{haldclut} to apply it on some random stream: +@example +ffmpeg -f lavfi -i mandelbrot -i clut.nut -filter_complex '[0][1] haldclut' -t 20 mandelclut.mkv +@end example + +The Hald CLUT will be applied to the 10 first seconds (duration of +@file{clut.nut}), then the latest picture of that CLUT stream will be applied +to the remaining frames of the @code{mandelbrot} stream. + +@subsubsection Hald CLUT with preview + +A Hald CLUT is supposed to be a squared image of @code{Level*Level*Level} by +@code{Level*Level*Level} pixels. For a given Hald CLUT, FFmpeg will select the +biggest possible square starting at the top left of the picture. The remaining +padding pixels (bottom or right) will be ignored. This area can be used to add +a preview of the Hald CLUT. + +Typically, the following generated Hald CLUT will be supported by the +@code{haldclut} filter: + +@example +ffmpeg -f lavfi -i @ref{haldclutsrc}=8 -vf " + pad=iw+320 [padded_clut]; + smptebars=s=320x256, split [a][b]; + [padded_clut][a] overlay=W-320:h, curves=color_negative [main]; + [main][b] overlay=W-320" -frames:v 1 clut.png +@end example + +It contains the original and a preview of the effect of the CLUT: SMPTE color +bars are displayed on the right-top, and below the same color bars processed by +the color changes. + +Then, the effect of this Hald CLUT can be visualized with: +@example +ffplay input.mkv -vf "movie=clut.png, [in] haldclut" @end example @section hflip Flip the input video horizontally. -For example, to horizontally flip the input video with @command{avconv}: +For example, to horizontally flip the input video with @command{ffmpeg}: @example -avconv -i in.avi -vf "hflip" out.avi +ffmpeg -i in.avi -vf "hflip" out.avi @end example +@section histeq +This filter applies a global color histogram equalization on a +per-frame basis. + +It can be used to correct video that has a compressed range of pixel +intensities. The filter redistributes the pixel intensities to +equalize their distribution across the intensity range. It may be +viewed as an "automatically adjusting contrast filter". This filter is +useful only for correcting degraded or poorly captured source +video. + +The filter accepts the following options: + +@table @option +@item strength +Determine the amount of equalization to be applied. As the strength +is reduced, the distribution of pixel intensities more-and-more +approaches that of the input frame. The value must be a float number +in the range [0,1] and defaults to 0.200. + +@item intensity +Set the maximum intensity that can generated and scale the output +values appropriately. The strength should be set as desired and then +the intensity can be limited if needed to avoid washing-out. The value +must be a float number in the range [0,1] and defaults to 0.210. + +@item antibanding +Set the antibanding level. If enabled the filter will randomly vary +the luminance of output pixels by a small amount to avoid banding of +the histogram. Possible values are @code{none}, @code{weak} or +@code{strong}. It defaults to @code{none}. +@end table + +@section histogram + +Compute and draw a color distribution histogram for the input video. + +The computed histogram is a representation of the color component +distribution in an image. + +Standard histogram displays the color components distribution in an image. +Displays color graph for each color component. Shows distribution of +the Y, U, V, A or R, G, B components, depending on input format, in the +current frame. Below each graph a color component scale meter is shown. + +The filter accepts the following options: + +@table @option +@item level_height +Set height of level. Default value is @code{200}. +Allowed range is [50, 2048]. + +@item scale_height +Set height of color scale. Default value is @code{12}. +Allowed range is [0, 40]. + +@item display_mode +Set display mode. +It accepts the following values: +@table @samp +@item parade +Per color component graphs are placed below each other. + +@item overlay +Presents information identical to that in the @code{parade}, except +that the graphs representing color components are superimposed directly +over one another. +@end table +Default is @code{parade}. + +@item levels_mode +Set mode. Can be either @code{linear}, or @code{logarithmic}. +Default is @code{linear}. + +@item components +Set what color components to display. +Default is @code{7}. +@end table + +@subsection Examples + +@itemize + +@item +Calculate and draw histogram: +@example +ffplay -i input -vf histogram +@end example + +@end itemize + +@anchor{hqdn3d} @section hqdn3d This is a high precision/quality 3d denoise filter. It aims to reduce @@ -1610,6 +7737,286 @@ A floating point number which specifies chroma temporal strength. It defaults to @var{luma_tmp}*@var{chroma_spatial}/@var{luma_spatial}. @end table +@section hqx + +Apply a high-quality magnification filter designed for pixel art. This filter +was originally created by Maxim Stepin. + +It accepts the following option: + +@table @option +@item n +Set the scaling dimension: @code{2} for @code{hq2x}, @code{3} for +@code{hq3x} and @code{4} for @code{hq4x}. +Default is @code{3}. +@end table + +@section hstack +Stack input videos horizontally. + +All streams must be of same pixel format and of same height. + +Note that this filter is faster than using @ref{overlay} and @ref{pad} filter +to create same output. + +The filter accept the following option: + +@table @option +@item inputs +Set number of input streams. Default is 2. + +@item shortest +If set to 1, force the output to terminate when the shortest input +terminates. Default value is 0. +@end table + +@section hue + +Modify the hue and/or the saturation of the input. + +It accepts the following parameters: + +@table @option +@item h +Specify the hue angle as a number of degrees. It accepts an expression, +and defaults to "0". + +@item s +Specify the saturation in the [-10,10] range. It accepts an expression and +defaults to "1". + +@item H +Specify the hue angle as a number of radians. It accepts an +expression, and defaults to "0". + +@item b +Specify the brightness in the [-10,10] range. It accepts an expression and +defaults to "0". +@end table + +@option{h} and @option{H} are mutually exclusive, and can't be +specified at the same time. + +The @option{b}, @option{h}, @option{H} and @option{s} option values are +expressions containing the following constants: + +@table @option +@item n +frame count of the input frame starting from 0 + +@item pts +presentation timestamp of the input frame expressed in time base units + +@item r +frame rate of the input video, NAN if the input frame rate is unknown + +@item t +timestamp expressed in seconds, NAN if the input timestamp is unknown + +@item tb +time base of the input video +@end table + +@subsection Examples + +@itemize +@item +Set the hue to 90 degrees and the saturation to 1.0: +@example +hue=h=90:s=1 +@end example + +@item +Same command but expressing the hue in radians: +@example +hue=H=PI/2:s=1 +@end example + +@item +Rotate hue and make the saturation swing between 0 +and 2 over a period of 1 second: +@example +hue="H=2*PI*t: s=sin(2*PI*t)+1" +@end example + +@item +Apply a 3 seconds saturation fade-in effect starting at 0: +@example +hue="s=min(t/3\,1)" +@end example + +The general fade-in expression can be written as: +@example +hue="s=min(0\, max((t-START)/DURATION\, 1))" +@end example + +@item +Apply a 3 seconds saturation fade-out effect starting at 5 seconds: +@example +hue="s=max(0\, min(1\, (8-t)/3))" +@end example + +The general fade-out expression can be written as: +@example +hue="s=max(0\, min(1\, (START+DURATION-t)/DURATION))" +@end example + +@end itemize + +@subsection Commands + +This filter supports the following commands: +@table @option +@item b +@item s +@item h +@item H +Modify the hue and/or the saturation and/or brightness of the input video. +The command accepts the same syntax of the corresponding option. + +If the specified expression is not valid, it is kept at its current +value. +@end table + +@section idet + +Detect video interlacing type. + +This filter tries to detect if the input frames as interlaced, progressive, +top or bottom field first. It will also try and detect fields that are +repeated between adjacent frames (a sign of telecine). + +Single frame detection considers only immediately adjacent frames when classifying each frame. +Multiple frame detection incorporates the classification history of previous frames. + +The filter will log these metadata values: + +@table @option +@item single.current_frame +Detected type of current frame using single-frame detection. One of: +``tff'' (top field first), ``bff'' (bottom field first), +``progressive'', or ``undetermined'' + +@item single.tff +Cumulative number of frames detected as top field first using single-frame detection. + +@item multiple.tff +Cumulative number of frames detected as top field first using multiple-frame detection. + +@item single.bff +Cumulative number of frames detected as bottom field first using single-frame detection. + +@item multiple.current_frame +Detected type of current frame using multiple-frame detection. One of: +``tff'' (top field first), ``bff'' (bottom field first), +``progressive'', or ``undetermined'' + +@item multiple.bff +Cumulative number of frames detected as bottom field first using multiple-frame detection. + +@item single.progressive +Cumulative number of frames detected as progressive using single-frame detection. + +@item multiple.progressive +Cumulative number of frames detected as progressive using multiple-frame detection. + +@item single.undetermined +Cumulative number of frames that could not be classified using single-frame detection. + +@item multiple.undetermined +Cumulative number of frames that could not be classified using multiple-frame detection. + +@item repeated.current_frame +Which field in the current frame is repeated from the last. One of ``neither'', ``top'', or ``bottom''. + +@item repeated.neither +Cumulative number of frames with no repeated field. + +@item repeated.top +Cumulative number of frames with the top field repeated from the previous frame's top field. + +@item repeated.bottom +Cumulative number of frames with the bottom field repeated from the previous frame's bottom field. +@end table + +The filter accepts the following options: + +@table @option +@item intl_thres +Set interlacing threshold. +@item prog_thres +Set progressive threshold. +@item repeat_thres +Threshold for repeated field detection. +@item half_life +Number of frames after which a given frame's contribution to the +statistics is halved (i.e., it contributes only 0.5 to it's +classification). The default of 0 means that all frames seen are given +full weight of 1.0 forever. +@item analyze_interlaced_flag +When this is not 0 then idet will use the specified number of frames to determine +if the interlaced flag is accurate, it will not count undetermined frames. +If the flag is found to be accurate it will be used without any further +computations, if it is found to be inaccurate it will be cleared without any +further computations. This allows inserting the idet filter as a low computational +method to clean up the interlaced flag +@end table + +@section il + +Deinterleave or interleave fields. + +This filter allows one to process interlaced images fields without +deinterlacing them. Deinterleaving splits the input frame into 2 +fields (so called half pictures). Odd lines are moved to the top +half of the output image, even lines to the bottom half. +You can process (filter) them independently and then re-interleave them. + +The filter accepts the following options: + +@table @option +@item luma_mode, l +@item chroma_mode, c +@item alpha_mode, a +Available values for @var{luma_mode}, @var{chroma_mode} and +@var{alpha_mode} are: + +@table @samp +@item none +Do nothing. + +@item deinterleave, d +Deinterleave fields, placing one above the other. + +@item interleave, i +Interleave fields. Reverse the effect of deinterleaving. +@end table +Default value is @code{none}. + +@item luma_swap, ls +@item chroma_swap, cs +@item alpha_swap, as +Swap luma/chroma/alpha fields. Exchange even & odd lines. Default value is @code{0}. +@end table + +@section inflate + +Apply inflate effect to the video. + +This filter replaces the pixel by the local(3x3) average by taking into account +only values higher than the pixel. + +It accepts the following options: + +@table @option +@item threshold0 +@item threshold1 +@item threshold2 +@item threshold3 +Limit the maximum change for each plane, default is 65535. +If 0, plane will remain unchanged. +@end table + @section interlace Simple interlacing filter from progressive contents. This interleaves upper (or @@ -1640,6 +8047,140 @@ Enable (default) or disable the vertical lowpass filter to avoid twitter interlacing and reduce moire patterns. @end table +@section kerndeint + +Deinterlace input video by applying Donald Graft's adaptive kernel +deinterling. Work on interlaced parts of a video to produce +progressive frames. + +The description of the accepted parameters follows. + +@table @option +@item thresh +Set the threshold which affects the filter's tolerance when +determining if a pixel line must be processed. It must be an integer +in the range [0,255] and defaults to 10. A value of 0 will result in +applying the process on every pixels. + +@item map +Paint pixels exceeding the threshold value to white if set to 1. +Default is 0. + +@item order +Set the fields order. Swap fields if set to 1, leave fields alone if +0. Default is 0. + +@item sharp +Enable additional sharpening if set to 1. Default is 0. + +@item twoway +Enable twoway sharpening if set to 1. Default is 0. +@end table + +@subsection Examples + +@itemize +@item +Apply default values: +@example +kerndeint=thresh=10:map=0:order=0:sharp=0:twoway=0 +@end example + +@item +Enable additional sharpening: +@example +kerndeint=sharp=1 +@end example + +@item +Paint processed pixels in white: +@example +kerndeint=map=1 +@end example +@end itemize + +@section lenscorrection + +Correct radial lens distortion + +This filter can be used to correct for radial distortion as can result from the use +of wide angle lenses, and thereby re-rectify the image. To find the right parameters +one can use tools available for example as part of opencv or simply trial-and-error. +To use opencv use the calibration sample (under samples/cpp) from the opencv sources +and extract the k1 and k2 coefficients from the resulting matrix. + +Note that effectively the same filter is available in the open-source tools Krita and +Digikam from the KDE project. + +In contrast to the @ref{vignette} filter, which can also be used to compensate lens errors, +this filter corrects the distortion of the image, whereas @ref{vignette} corrects the +brightness distribution, so you may want to use both filters together in certain +cases, though you will have to take care of ordering, i.e. whether vignetting should +be applied before or after lens correction. + +@subsection Options + +The filter accepts the following options: + +@table @option +@item cx +Relative x-coordinate of the focal point of the image, and thereby the center of the +distortion. This value has a range [0,1] and is expressed as fractions of the image +width. +@item cy +Relative y-coordinate of the focal point of the image, and thereby the center of the +distortion. This value has a range [0,1] and is expressed as fractions of the image +height. +@item k1 +Coefficient of the quadratic correction term. 0.5 means no correction. +@item k2 +Coefficient of the double quadratic correction term. 0.5 means no correction. +@end table + +The formula that generates the correction is: + +@var{r_src} = @var{r_tgt} * (1 + @var{k1} * (@var{r_tgt} / @var{r_0})^2 + @var{k2} * (@var{r_tgt} / @var{r_0})^4) + +where @var{r_0} is halve of the image diagonal and @var{r_src} and @var{r_tgt} are the +distances from the focal point in the source and target images, respectively. + +@anchor{lut3d} +@section lut3d + +Apply a 3D LUT to an input video. + +The filter accepts the following options: + +@table @option +@item file +Set the 3D LUT file name. + +Currently supported formats: +@table @samp +@item 3dl +AfterEffects +@item cube +Iridas +@item dat +DaVinci +@item m3d +Pandora +@end table +@item interp +Select interpolation mode. + +Available values are: + +@table @samp +@item nearest +Use values from the nearest defined point. +@item trilinear +Interpolate values using the 8 points defining a cube. +@item tetrahedral +Interpolate values using a tetrahedron. +@end table +@end table + @section lut, lutrgb, lutyuv Compute a look-up table for binding each pixel component input value @@ -1650,19 +8191,30 @@ to an RGB input video. These filters accept the following parameters: @table @option -@item @var{c0} (first pixel component) -@item @var{c1} (second pixel component) -@item @var{c2} (third pixel component) -@item @var{c3} (fourth pixel component, corresponds to the alpha component) - -@item @var{r} (red component) -@item @var{g} (green component) -@item @var{b} (blue component) -@item @var{a} (alpha component) +@item c0 +set first pixel component expression +@item c1 +set second pixel component expression +@item c2 +set third pixel component expression +@item c3 +set fourth pixel component expression, corresponds to the alpha component + +@item r +set red component expression +@item g +set green component expression +@item b +set blue component expression +@item a +alpha component expression -@item @var{y} (Y/luminance component) -@item @var{u} (U/Cb component) -@item @var{v} (V/Cr component) +@item y +set Y/luminance component expression +@item u +set U/Cb component expression +@item v +set V/Cr component expression @end table Each of them specifies the expression to use for computing the lookup table for @@ -1677,11 +8229,8 @@ The @var{lut} filter requires either YUV or RGB pixel formats in input, The expressions can contain the following constants and functions: @table @option -@item E, PI, PHI -These are approximated values for the mathematical constants e -(Euler's number), pi (Greek pi), and phi (the golden ratio). - -@item w, h +@item w +@item h The input width and height. @item val @@ -1715,35 +8264,320 @@ expression All expressions default to "val". -Some examples: +@subsection Examples + +@itemize +@item +Negate input video: @example -# Negate input video lutrgb="r=maxval+minval-val:g=maxval+minval-val:b=maxval+minval-val" lutyuv="y=maxval+minval-val:u=maxval+minval-val:v=maxval+minval-val" +@end example -# The above is the same as +The above is the same as: +@example lutrgb="r=negval:g=negval:b=negval" lutyuv="y=negval:u=negval:v=negval" +@end example -# Negate luminance -lutyuv=negval +@item +Negate luminance: +@example +lutyuv=y=negval +@end example -# Remove chroma components, turning the video into a graytone image +@item +Remove chroma components, turning the video into a graytone image: +@example lutyuv="u=128:v=128" +@end example -# Apply a luma burning effect +@item +Apply a luma burning effect: +@example lutyuv="y=2*val" +@end example -# Remove green and blue components +@item +Remove green and blue components: +@example lutrgb="g=0:b=0" +@end example -# Set a constant alpha channel value on input +@item +Set a constant alpha channel value on input: +@example format=rgba,lutrgb=a="maxval-minval/2" +@end example -# Correct luminance gamma by a factor of 0.5 +@item +Correct luminance gamma by a factor of 0.5: +@example lutyuv=y=gammaval(0.5) @end example +@item +Discard least significant bits of luma: +@example +lutyuv=y='bitand(val, 128+64+32)' +@end example +@end itemize + +@section maskedmerge + +Merge the first input stream with the second input stream using per pixel +weights in the third input stream. + +A value of 0 in the third stream pixel component means that pixel component +from first stream is returned unchanged, while maximum value (eg. 255 for +8-bit videos) means that pixel component from second stream is returned +unchanged. Intermediate values define the amount of merging between both +input stream's pixel components. + +This filter accepts the following options: +@table @option +@item planes +Set which planes will be processed as bitmap, unprocessed planes will be +copied from first stream. +By default value 0xf, all planes will be processed. +@end table + +@section mcdeint + +Apply motion-compensation deinterlacing. + +It needs one field per frame as input and must thus be used together +with yadif=1/3 or equivalent. + +This filter accepts the following options: +@table @option +@item mode +Set the deinterlacing mode. + +It accepts one of the following values: +@table @samp +@item fast +@item medium +@item slow +use iterative motion estimation +@item extra_slow +like @samp{slow}, but use multiple reference frames. +@end table +Default value is @samp{fast}. + +@item parity +Set the picture field parity assumed for the input video. It must be +one of the following values: + +@table @samp +@item 0, tff +assume top field first +@item 1, bff +assume bottom field first +@end table + +Default value is @samp{bff}. + +@item qp +Set per-block quantization parameter (QP) used by the internal +encoder. + +Higher values should result in a smoother motion vector field but less +optimal individual vectors. Default value is 1. +@end table + +@section mergeplanes + +Merge color channel components from several video streams. + +The filter accepts up to 4 input streams, and merge selected input +planes to the output video. + +This filter accepts the following options: +@table @option +@item mapping +Set input to output plane mapping. Default is @code{0}. + +The mappings is specified as a bitmap. It should be specified as a +hexadecimal number in the form 0xAa[Bb[Cc[Dd]]]. 'Aa' describes the +mapping for the first plane of the output stream. 'A' sets the number of +the input stream to use (from 0 to 3), and 'a' the plane number of the +corresponding input to use (from 0 to 3). The rest of the mappings is +similar, 'Bb' describes the mapping for the output stream second +plane, 'Cc' describes the mapping for the output stream third plane and +'Dd' describes the mapping for the output stream fourth plane. + +@item format +Set output pixel format. Default is @code{yuva444p}. +@end table + +@subsection Examples + +@itemize +@item +Merge three gray video streams of same width and height into single video stream: +@example +[a0][a1][a2]mergeplanes=0x001020:yuv444p +@end example + +@item +Merge 1st yuv444p stream and 2nd gray video stream into yuva444p video stream: +@example +[a0][a1]mergeplanes=0x00010210:yuva444p +@end example + +@item +Swap Y and A plane in yuva444p stream: +@example +format=yuva444p,mergeplanes=0x03010200:yuva444p +@end example + +@item +Swap U and V plane in yuv420p stream: +@example +format=yuv420p,mergeplanes=0x000201:yuv420p +@end example + +@item +Cast a rgb24 clip to yuv444p: +@example +format=rgb24,mergeplanes=0x000102:yuv444p +@end example +@end itemize + +@section metadata, ametadata + +Manipulate frame metadata. + +This filter accepts the following options: + +@table @option +@item mode +Set mode of operation of the filter. + +Can be one of the following: + +@table @samp +@item select +If both @code{value} and @code{key} is set, select frames +which have such metadata. If only @code{key} is set, select +every frame that has such key in metadata. + +@item add +Add new metadata @code{key} and @code{value}. If key is already available +do nothing. + +@item modify +Modify value of already present key. + +@item delete +If @code{value} is set, delete only keys that have such value. +Otherwise, delete key. + +@item print +Print key and its value if metadata was found. If @code{key} is not set print all +metadata values available in frame. +@end table + +@item key +Set key used with all modes. Must be set for all modes except @code{print}. + +@item value +Set metadata value which will be used. This option is mandatory for +@code{modify} and @code{add} mode. + +@item length +Set length of how many characters of two metadata values need to match to be +considered same. Default is all available characters. + +@item function +Which function to use when comparing metadata value and @code{value}. + +Can be one of following: + +@table @samp +@item string +Values are interpreted as strings, returns true if @code{value} is same as metadata value up +to N chars as set in @code{length} option. + +@item less +Values are interpreted as floats, returns true if metadata value is less than @code{value}. + +@item equal +Values are interpreted as floats, returns true if @code{value} is equal with metadata value. + +@item greater +Values are interpreted as floats, returns true if metadata value is greater than @code{value}. + +@item expr +Values are interpreted as floats, returns true if expression from option @code{expr} +evaluates to true. +@end table + +@item expr +Set expression which is used when @code{function} is set to @code{expr}. +The expression is evaluated through the eval API and can contain the following +constants: + +@table @option +@item VALUE1 +Float representation of @code{value} from metadata key. + +@item VALUE2 +Float representation of @code{value} as supplied by user in @code{value} option. +@end table +@end table + +@subsection Examples + +@itemize +@item +Print all metadata values for frames with key @code{lavfi.singnalstats.YDIF} with values +between 0 and 1. +@example +@end example +signalstats,metadata=print:key=lavfi.signalstats.YDIF:function=expr:expr='between(VALUE1,0,1)' +@end itemize + +@section mpdecimate + +Drop frames that do not differ greatly from the previous frame in +order to reduce frame rate. + +The main use of this filter is for very-low-bitrate encoding +(e.g. streaming over dialup modem), but it could in theory be used for +fixing movies that were inverse-telecined incorrectly. + +A description of the accepted options follows. + +@table @option +@item max +Set the maximum number of consecutive frames which can be dropped (if +positive), or the minimum interval between dropped frames (if +negative). If the value is 0, the frame is dropped unregarding the +number of previous sequentially dropped frames. + +Default value is 0. + +@item hi +@item lo +@item frac +Set the dropping threshold values. + +Values for @option{hi} and @option{lo} are for 8x8 pixel blocks and +represent actual pixel value differences, so a threshold of 64 +corresponds to 1 unit of difference for each pixel, or the same spread +out differently over the block. + +A frame is a candidate for dropping if no 8x8 blocks differ by more +than a threshold of @option{hi}, and if no more than @option{frac} blocks (1 +meaning the whole image) differ by more than a threshold of @option{lo}. + +Default value for @option{hi} is 64*12, default value for @option{lo} is +64*5, and default value for @option{frac} is 0.33. +@end table + + @section negate Negate input video. @@ -1751,6 +8585,115 @@ Negate input video. It accepts an integer in input; if non-zero it negates the alpha component (if available). The default value in input is 0. +@section nnedi + +Deinterlace video using neural network edge directed interpolation. + +This filter accepts the following options: + +@table @option +@item weights +Mandatory option, without binary file filter can not work. +Currently file can be found here: +https://github.com/dubhater/vapoursynth-nnedi3/blob/master/src/nnedi3_weights.bin + +@item deint +Set which frames to deinterlace, by default it is @code{all}. +Can be @code{all} or @code{interlaced}. + +@item field +Set mode of operation. + +Can be one of the following: + +@table @samp +@item af +Use frame flags, both fields. +@item a +Use frame flags, single field. +@item t +Use top field only. +@item b +Use bottom field only. +@item ft +Use both fields, top first. +@item fb +Use both fields, bottom first. +@end table + +@item planes +Set which planes to process, by default filter process all frames. + +@item nsize +Set size of local neighborhood around each pixel, used by the predictor neural +network. + +Can be one of the following: + +@table @samp +@item s8x6 +@item s16x6 +@item s32x6 +@item s48x6 +@item s8x4 +@item s16x4 +@item s32x4 +@end table + +@item nns +Set the number of neurons in predicctor neural network. +Can be one of the following: + +@table @samp +@item n16 +@item n32 +@item n64 +@item n128 +@item n256 +@end table + +@item qual +Controls the number of different neural network predictions that are blended +together to compute the final output value. Can be @code{fast}, default or +@code{slow}. + +@item etype +Set which set of weights to use in the predictor. +Can be one of the following: + +@table @samp +@item a +weights trained to minimize absolute error +@item s +weights trained to minimize squared error +@end table + +@item pscrn +Controls whether or not the prescreener neural network is used to decide +which pixels should be processed by the predictor neural network and which +can be handled by simple cubic interpolation. +The prescreener is trained to know whether cubic interpolation will be +sufficient for a pixel or whether it should be predicted by the predictor nn. +The computational complexity of the prescreener nn is much less than that of +the predictor nn. Since most pixels can be handled by cubic interpolation, +using the prescreener generally results in much faster processing. +The prescreener is pretty accurate, so the difference between using it and not +using it is almost always unnoticeable. + +Can be one of the following: + +@table @samp +@item none +@item original +@item new +@end table + +Default is @code{new}. + +@item fapprox +Set various debugging flags. +@end table + @section noformat Force libavfilter not to use any of the specified pixel formats for the @@ -1765,26 +8708,106 @@ apix_fmts=yuv420p|monow|rgb24". @end table -Some examples: +@subsection Examples + +@itemize +@item +Force libavfilter to use a format different from @var{yuv420p} for the +input to the vflip filter: @example -# Force libavfilter to use a format different from "yuv420p" for the -# input to the vflip filter noformat=pix_fmts=yuv420p,vflip +@end example -# Convert the input video to any of the formats not contained in the list +@item +Convert the input video to any of the formats not contained in the list: +@example noformat=yuv420p|yuv444p|yuv410p @end example +@end itemize + +@section noise + +Add noise on video input frame. + +The filter accepts the following options: + +@table @option +@item all_seed +@item c0_seed +@item c1_seed +@item c2_seed +@item c3_seed +Set noise seed for specific pixel component or all pixel components in case +of @var{all_seed}. Default value is @code{123457}. + +@item all_strength, alls +@item c0_strength, c0s +@item c1_strength, c1s +@item c2_strength, c2s +@item c3_strength, c3s +Set noise strength for specific pixel component or all pixel components in case +@var{all_strength}. Default value is @code{0}. Allowed range is [0, 100]. + +@item all_flags, allf +@item c0_flags, c0f +@item c1_flags, c1f +@item c2_flags, c2f +@item c3_flags, c3f +Set pixel component flags or set flags for all components if @var{all_flags}. +Available values for component flags are: +@table @samp +@item a +averaged temporal noise (smoother) +@item p +mix random noise with a (semi)regular pattern +@item t +temporal noise (noise pattern changes between frames) +@item u +uniform noise (gaussian otherwise) +@end table +@end table + +@subsection Examples + +Add temporal and uniform noise to input video: +@example +noise=alls=20:allf=t+u +@end example @section null Pass the video source unchanged to the output. +@section ocr +Optical Character Recognition + +This filter uses Tesseract for optical character recognition. + +It accepts the following options: + +@table @option +@item datapath +Set datapath to tesseract data. Default is to use whatever was +set at installation. + +@item language +Set language, default is "eng". + +@item whitelist +Set character whitelist. + +@item blacklist +Set character blacklist. +@end table + +The filter exports recognized text as the frame metadata @code{lavfi.ocr.text}. + @section ocv Apply a video transform using libopencv. To enable this filter, install the libopencv library and headers and -configure Libav with --enable-libopencv. +configure FFmpeg with @code{--enable-libopencv}. It accepts the following parameters: @@ -1801,7 +8824,7 @@ values are assumed. Refer to the official libopencv documentation for more precise information: -@url{http://opencv.willowgarage.com/documentation/c/image_filtering.html} +@url{http://docs.opencv.org/master/modules/imgproc/doc/filtering.html} Several libopencv filters are supported; see the following subsections. @@ -1889,34 +8912,19 @@ libopencv function @code{cvSmooth}. Overlay one video on top of another. It takes two inputs and has one output. The first input is the "main" -video on which the second input is overlayed. +video on which the second input is overlaid. It accepts the following parameters: -@table @option +A description of the accepted options follows. +@table @option @item x -The horizontal position of the left edge of the overlaid video on the main video. - @item y -The vertical position of the top edge of the overlaid video on the main video. - -@end table - -The parameters are expressions containing the following parameters: - -@table @option -@item main_w, main_h -The main input width and height. - -@item W, H -These are the same as @var{main_w} and @var{main_h}. - -@item overlay_w, overlay_h -The overlay input width and height. - -@item w, h -These are the same as @var{overlay_w} and @var{overlay_h}. +Set the expression for the x and y coordinates of the overlaid video +on the main video. Default value is "0" for both expressions. In case +the expression is invalid, it is set to a huge value (meaning that the +overlay will not be displayed within the output visible area). @item eof_action The action to take when EOF is encountered on the secondary input; it accepts @@ -1931,41 +8939,231 @@ End both streams. Pass the main input through. @end table +@item eval +Set when the expressions for @option{x}, and @option{y} are evaluated. + +It accepts the following values: +@table @samp +@item init +only evaluate expressions once during the filter initialization or +when a command is processed + +@item frame +evaluate expressions for each incoming frame +@end table + +Default value is @samp{frame}. + +@item shortest +If set to 1, force the output to terminate when the shortest input +terminates. Default value is 0. + +@item format +Set the format for the output video. + +It accepts the following values: +@table @samp +@item yuv420 +force YUV420 output + +@item yuv422 +force YUV422 output + +@item yuv444 +force YUV444 output + +@item rgb +force RGB output @end table +Default value is @samp{yuv420}. + +@item rgb @emph{(deprecated)} +If set to 1, force the filter to accept inputs in the RGB +color space. Default value is 0. This option is deprecated, use +@option{format} instead. + +@item repeatlast +If set to 1, force the filter to draw the last overlay frame over the +main input until the end of the stream. A value of 0 disables this +behavior. Default value is 1. +@end table + +The @option{x}, and @option{y} expressions can contain the following +parameters. + +@table @option +@item main_w, W +@item main_h, H +The main input width and height. + +@item overlay_w, w +@item overlay_h, h +The overlay input width and height. + +@item x +@item y +The computed values for @var{x} and @var{y}. They are evaluated for +each new frame. + +@item hsub +@item vsub +horizontal and vertical chroma subsample values of the output +format. For example for the pixel format "yuv422p" @var{hsub} is 2 and +@var{vsub} is 1. + +@item n +the number of input frame, starting from 0 + +@item pos +the position in the file of the input frame, NAN if unknown + +@item t +The timestamp, expressed in seconds. It's NAN if the input timestamp is unknown. + +@end table + +Note that the @var{n}, @var{pos}, @var{t} variables are available only +when evaluation is done @emph{per frame}, and will evaluate to NAN +when @option{eval} is set to @samp{init}. + Be aware that frames are taken from each input video in timestamp -order, hence, if their initial timestamps differ, it is a a good idea +order, hence, if their initial timestamps differ, it is a good idea to pass the two inputs through a @var{setpts=PTS-STARTPTS} filter to have them begin in the same zero timestamp, as the example for the @var{movie} filter does. -Some examples: +You can chain together more overlays but you should test the +efficiency of such approach. + +@subsection Commands + +This filter supports the following commands: +@table @option +@item x +@item y +Modify the x and y of the overlay input. +The command accepts the same syntax of the corresponding option. + +If the specified expression is not valid, it is kept at its current +value. +@end table + +@subsection Examples + +@itemize +@item +Draw the overlay at 10 pixels from the bottom right corner of the main +video: +@example +overlay=main_w-overlay_w-10:main_h-overlay_h-10 +@end example + +Using named options the example above becomes: @example -# Draw the overlay at 10 pixels from the bottom right -# corner of the main video overlay=x=main_w-overlay_w-10:y=main_h-overlay_h-10 +@end example -# Insert a transparent PNG logo in the bottom left corner of the input -avconv -i input -i logo -filter_complex 'overlay=x=10:y=main_h-overlay_h-10' output +@item +Insert a transparent PNG logo in the bottom left corner of the input, +using the @command{ffmpeg} tool with the @code{-filter_complex} option: +@example +ffmpeg -i input -i logo -filter_complex 'overlay=10:main_h-overlay_h-10' output +@end example -# Insert 2 different transparent PNG logos (second logo on bottom -# right corner) -avconv -i input -i logo1 -i logo2 -filter_complex -'overlay=x=10:y=H-h-10,overlay=x=W-w-10:y=H-h-10' output +@item +Insert 2 different transparent PNG logos (second logo on bottom +right corner) using the @command{ffmpeg} tool: +@example +ffmpeg -i input -i logo1 -i logo2 -filter_complex 'overlay=x=10:y=H-h-10,overlay=x=W-w-10:y=H-h-10' output +@end example -# Add a transparent color layer on top of the main video; -# WxH specifies the size of the main input to the overlay filter -color=red@.3:WxH [over]; [in][over] overlay [out] +@item +Add a transparent color layer on top of the main video; @code{WxH} +must specify the size of the main input to the overlay filter: +@example +color=color=red@@.3:size=WxH [over]; [in][over] overlay [out] +@end example -# Mask 10-20 seconds of a video by applying the delogo filter to a section -avconv -i test.avi -codec:v:0 wmv2 -ar 11025 -b:v 9000k +@item +Play an original video and a filtered version (here with the deshake +filter) side by side using the @command{ffplay} tool: +@example +ffplay input.avi -vf 'split[a][b]; [a]pad=iw*2:ih[src]; [b]deshake[filt]; [src][filt]overlay=w' +@end example + +The above command is the same as: +@example +ffplay input.avi -vf 'split[b], pad=iw*2[src], [b]deshake, [src]overlay=w' +@end example + +@item +Make a sliding overlay appearing from the left to the right top part of the +screen starting since time 2: +@example +overlay=x='if(gte(t,2), -w+(t-2)*20, NAN)':y=0 +@end example + +@item +Compose output by putting two input videos side to side: +@example +ffmpeg -i left.avi -i right.avi -filter_complex " +nullsrc=size=200x100 [background]; +[0:v] setpts=PTS-STARTPTS, scale=100x100 [left]; +[1:v] setpts=PTS-STARTPTS, scale=100x100 [right]; +[background][left] overlay=shortest=1 [background+left]; +[background+left][right] overlay=shortest=1:x=100 [left+right] +" +@end example + +@item +Mask 10-20 seconds of a video by applying the delogo filter to a section +@example +ffmpeg -i test.avi -codec:v:0 wmv2 -ar 11025 -b:v 9000k -vf '[in]split[split_main][split_delogo];[split_delogo]trim=start=360:end=371,delogo=0:0:640:480[delogoed];[split_main][delogoed]overlay=eof_action=pass[out]' masked.avi @end example -You can chain together more overlays but the efficiency of such -approach is yet to be tested. +@item +Chain several overlays in cascade: +@example +nullsrc=s=200x200 [bg]; +testsrc=s=100x100, split=4 [in0][in1][in2][in3]; +[in0] lutrgb=r=0, [bg] overlay=0:0 [mid0]; +[in1] lutrgb=g=0, [mid0] overlay=100:0 [mid1]; +[in2] lutrgb=b=0, [mid1] overlay=0:100 [mid2]; +[in3] null, [mid2] overlay=100:100 [out0] +@end example + +@end itemize + +@section owdenoise + +Apply Overcomplete Wavelet denoiser. + +The filter accepts the following options: + +@table @option +@item depth +Set depth. +Larger depth values will denoise lower frequency components more, but +slow down filtering. + +Must be an int in the range 8-16, default is @code{8}. + +@item luma_strength, ls +Set luma strength. + +Must be a double value in the range 0-1000, default is @code{1.0}. + +@item chroma_strength, cs +Set chroma strength. + +Must be a double value in the range 0-1000, default is @code{1.0}. +@end table + +@anchor{pad} @section pad Add paddings to the input image, and place the original input at the @@ -1974,19 +9172,19 @@ provided @var{x}, @var{y} coordinates. It accepts the following parameters: @table @option -@item width, height - -Specify the size of the output image with the paddings added. If the -value for @var{width} or @var{height} is 0, the corresponding input size -is used for the output. +@item width, w +@item height, h +Specify an expression for the size of the output image with the +paddings added. If the value for @var{width} or @var{height} is 0, the +corresponding input size is used for the output. The @var{width} expression can reference the value set by the @var{height} expression, and vice versa. The default value of @var{width} and @var{height} is 0. -@item x, y - +@item x +@item y Specify the offsets to place the input image at within the padded area, with respect to the top/left border of the output image. @@ -1996,71 +9194,346 @@ expression, and vice versa. The default value of @var{x} and @var{y} is 0. @item color - -Specify the color of the padded area. It can be the name of a color -(case insensitive match) or an 0xRRGGBB[AA] sequence. +Specify the color of the padded area. For the syntax of this option, +check the "Color" section in the ffmpeg-utils manual. The default value of @var{color} is "black". - @end table -The parameters @var{width}, @var{height}, @var{x}, and @var{y} are -expressions containing the following constants: +The value for the @var{width}, @var{height}, @var{x}, and @var{y} +options are expressions containing the following constants: @table @option -@item E, PI, PHI -These are approximated values for the mathematical constants e -(Euler's number), pi (Greek pi), and phi (the golden ratio). - -@item in_w, in_h +@item in_w +@item in_h The input video width and height. -@item iw, ih +@item iw +@item ih These are the same as @var{in_w} and @var{in_h}. -@item out_w, out_h +@item out_w +@item out_h The output width and height (the size of the padded area), as specified by the @var{width} and @var{height} expressions. -@item ow, oh +@item ow +@item oh These are the same as @var{out_w} and @var{out_h}. -@item x, y +@item x +@item y The x and y offsets as specified by the @var{x} and @var{y} expressions, or NAN if not yet specified. @item a -The input display aspect ratio, same as @var{iw} / @var{ih}. +same as @var{iw} / @var{ih} -@item hsub, vsub +@item sar +input sample aspect ratio + +@item dar +input display aspect ratio, it is the same as (@var{iw} / @var{ih}) * @var{sar} + +@item hsub +@item vsub The horizontal and vertical chroma subsample values. For example for the pixel format "yuv422p" @var{hsub} is 2 and @var{vsub} is 1. @end table -Some examples: +@subsection Examples + +@itemize +@item +Add paddings with the color "violet" to the input video. The output video +size is 640x480, and the top-left corner of the input video is placed at +column 0, row 40 +@example +pad=640:480:0:40:violet +@end example +The example above is equivalent to the following command: @example -# Add paddings with the color "violet" to the input video. The output video -# size is 640x480, and the top-left corner of the input video is placed at -# column 0, row 40 pad=width=640:height=480:x=0:y=40:color=violet +@end example -# Pad the input to get an output with dimensions increased by 3/2, -# and put the input video at the center of the padded area +@item +Pad the input to get an output with dimensions increased by 3/2, +and put the input video at the center of the padded area: +@example pad="3/2*iw:3/2*ih:(ow-iw)/2:(oh-ih)/2" +@end example -# Pad the input to get a squared output with size equal to the maximum -# value between the input width and height, and put the input video at -# the center of the padded area +@item +Pad the input to get a squared output with size equal to the maximum +value between the input width and height, and put the input video at +the center of the padded area: +@example pad="max(iw\,ih):ow:(ow-iw)/2:(oh-ih)/2" +@end example -# Pad the input to get a final w/h ratio of 16:9 +@item +Pad the input to get a final w/h ratio of 16:9: +@example pad="ih*16/9:ih:(ow-iw)/2:(oh-ih)/2" +@end example -# Double the output size and put the input video in the bottom-right -# corner of the output padded area +@item +In case of anamorphic video, in order to set the output display aspect +correctly, it is necessary to use @var{sar} in the expression, +according to the relation: +@example +(ih * X / ih) * sar = output_dar +X = output_dar / sar +@end example + +Thus the previous example needs to be modified to: +@example +pad="ih*16/9/sar:ih:(ow-iw)/2:(oh-ih)/2" +@end example + +@item +Double the output size and put the input video in the bottom-right +corner of the output padded area: +@example pad="2*iw:2*ih:ow-iw:oh-ih" @end example +@end itemize + +@anchor{palettegen} +@section palettegen + +Generate one palette for a whole video stream. + +It accepts the following options: + +@table @option +@item max_colors +Set the maximum number of colors to quantize in the palette. +Note: the palette will still contain 256 colors; the unused palette entries +will be black. + +@item reserve_transparent +Create a palette of 255 colors maximum and reserve the last one for +transparency. Reserving the transparency color is useful for GIF optimization. +If not set, the maximum of colors in the palette will be 256. You probably want +to disable this option for a standalone image. +Set by default. + +@item stats_mode +Set statistics mode. + +It accepts the following values: +@table @samp +@item full +Compute full frame histograms. +@item diff +Compute histograms only for the part that differs from previous frame. This +might be relevant to give more importance to the moving part of your input if +the background is static. +@end table + +Default value is @var{full}. +@end table + +The filter also exports the frame metadata @code{lavfi.color_quant_ratio} +(@code{nb_color_in / nb_color_out}) which you can use to evaluate the degree of +color quantization of the palette. This information is also visible at +@var{info} logging level. + +@subsection Examples + +@itemize +@item +Generate a representative palette of a given video using @command{ffmpeg}: +@example +ffmpeg -i input.mkv -vf palettegen palette.png +@end example +@end itemize + +@section paletteuse + +Use a palette to downsample an input video stream. + +The filter takes two inputs: one video stream and a palette. The palette must +be a 256 pixels image. + +It accepts the following options: + +@table @option +@item dither +Select dithering mode. Available algorithms are: +@table @samp +@item bayer +Ordered 8x8 bayer dithering (deterministic) +@item heckbert +Dithering as defined by Paul Heckbert in 1982 (simple error diffusion). +Note: this dithering is sometimes considered "wrong" and is included as a +reference. +@item floyd_steinberg +Floyd and Steingberg dithering (error diffusion) +@item sierra2 +Frankie Sierra dithering v2 (error diffusion) +@item sierra2_4a +Frankie Sierra dithering v2 "Lite" (error diffusion) +@end table + +Default is @var{sierra2_4a}. + +@item bayer_scale +When @var{bayer} dithering is selected, this option defines the scale of the +pattern (how much the crosshatch pattern is visible). A low value means more +visible pattern for less banding, and higher value means less visible pattern +at the cost of more banding. + +The option must be an integer value in the range [0,5]. Default is @var{2}. + +@item diff_mode +If set, define the zone to process + +@table @samp +@item rectangle +Only the changing rectangle will be reprocessed. This is similar to GIF +cropping/offsetting compression mechanism. This option can be useful for speed +if only a part of the image is changing, and has use cases such as limiting the +scope of the error diffusal @option{dither} to the rectangle that bounds the +moving scene (it leads to more deterministic output if the scene doesn't change +much, and as a result less moving noise and better GIF compression). +@end table + +Default is @var{none}. +@end table + +@subsection Examples + +@itemize +@item +Use a palette (generated for example with @ref{palettegen}) to encode a GIF +using @command{ffmpeg}: +@example +ffmpeg -i input.mkv -i palette.png -lavfi paletteuse output.gif +@end example +@end itemize + +@section perspective + +Correct perspective of video not recorded perpendicular to the screen. + +A description of the accepted parameters follows. + +@table @option +@item x0 +@item y0 +@item x1 +@item y1 +@item x2 +@item y2 +@item x3 +@item y3 +Set coordinates expression for top left, top right, bottom left and bottom right corners. +Default values are @code{0:0:W:0:0:H:W:H} with which perspective will remain unchanged. +If the @code{sense} option is set to @code{source}, then the specified points will be sent +to the corners of the destination. If the @code{sense} option is set to @code{destination}, +then the corners of the source will be sent to the specified coordinates. + +The expressions can use the following variables: + +@table @option +@item W +@item H +the width and height of video frame. +@end table + +@item interpolation +Set interpolation for perspective correction. + +It accepts the following values: +@table @samp +@item linear +@item cubic +@end table + +Default value is @samp{linear}. + +@item sense +Set interpretation of coordinate options. + +It accepts the following values: +@table @samp +@item 0, source + +Send point in the source specified by the given coordinates to +the corners of the destination. + +@item 1, destination + +Send the corners of the source to the point in the destination specified +by the given coordinates. + +Default value is @samp{source}. +@end table +@end table + +@section phase + +Delay interlaced video by one field time so that the field order changes. + +The intended use is to fix PAL movies that have been captured with the +opposite field order to the film-to-video transfer. + +A description of the accepted parameters follows. + +@table @option +@item mode +Set phase mode. + +It accepts the following values: +@table @samp +@item t +Capture field order top-first, transfer bottom-first. +Filter will delay the bottom field. + +@item b +Capture field order bottom-first, transfer top-first. +Filter will delay the top field. + +@item p +Capture and transfer with the same field order. This mode only exists +for the documentation of the other options to refer to, but if you +actually select it, the filter will faithfully do nothing. + +@item a +Capture field order determined automatically by field flags, transfer +opposite. +Filter selects among @samp{t} and @samp{b} modes on a frame by frame +basis using field flags. If no field information is available, +then this works just like @samp{u}. + +@item u +Capture unknown or varying, transfer opposite. +Filter selects among @samp{t} and @samp{b} on a frame by frame basis by +analyzing the images and selecting the alternative that produces best +match between the fields. + +@item T +Capture top-first, transfer unknown or varying. +Filter selects among @samp{t} and @samp{p} using image analysis. + +@item B +Capture bottom-first, transfer unknown or varying. +Filter selects among @samp{b} and @samp{p} using image analysis. + +@item A +Capture determined by field flags, transfer unknown or varying. +Filter selects among @samp{t}, @samp{b} and @samp{p} using field flags and +image analysis. If no field information is available, then this works just +like @samp{U}. This is the default mode. + +@item U +Both capture and transfer unknown or varying. +Filter selects among @samp{t}, @samp{b} and @samp{p} using image analysis only. +@end table +@end table @section pixdesctest @@ -2074,367 +9547,1177 @@ format=monow, pixdesctest can be used to test the monowhite pixel format descriptor definition. -@anchor{scale} -@section scale +@section pp -Scale the input video and/or convert the image format. +Enable the specified chain of postprocessing subfilters using libpostproc. This +library should be automatically selected with a GPL build (@code{--enable-gpl}). +Subfilters must be separated by '/' and can be disabled by prepending a '-'. +Each subfilter and some options have a short and a long name that can be used +interchangeably, i.e. dr/dering are the same. -It accepts the following parameters: +The filters accept the following options: @table @option +@item subfilters +Set postprocessing subfilters string. +@end table -@item w -The output video width. +All subfilters share common options to determine their scope: -@item h -The output video height. +@table @option +@item a/autoq +Honor the quality commands for this subfilter. + +@item c/chrom +Do chrominance filtering, too (default). +@item y/nochrom +Do luminance filtering only (no chrominance). + +@item n/noluma +Do chrominance filtering only (no luminance). @end table -The parameters @var{w} and @var{h} are expressions containing -the following constants: +These options can be appended after the subfilter name, separated by a '|'. + +Available subfilters are: @table @option -@item E, PI, PHI -These are approximated values for the mathematical constants e -(Euler's number), pi (Greek pi), and phi (the golden ratio). +@item hb/hdeblock[|difference[|flatness]] +Horizontal deblocking filter +@table @option +@item difference +Difference factor where higher values mean more deblocking (default: @code{32}). +@item flatness +Flatness threshold where lower values mean more deblocking (default: @code{39}). +@end table -@item in_w, in_h -The input width and height. +@item vb/vdeblock[|difference[|flatness]] +Vertical deblocking filter +@table @option +@item difference +Difference factor where higher values mean more deblocking (default: @code{32}). +@item flatness +Flatness threshold where lower values mean more deblocking (default: @code{39}). +@end table -@item iw, ih -These are the same as @var{in_w} and @var{in_h}. +@item ha/hadeblock[|difference[|flatness]] +Accurate horizontal deblocking filter +@table @option +@item difference +Difference factor where higher values mean more deblocking (default: @code{32}). +@item flatness +Flatness threshold where lower values mean more deblocking (default: @code{39}). +@end table -@item out_w, out_h -The output (cropped) width and height. +@item va/vadeblock[|difference[|flatness]] +Accurate vertical deblocking filter +@table @option +@item difference +Difference factor where higher values mean more deblocking (default: @code{32}). +@item flatness +Flatness threshold where lower values mean more deblocking (default: @code{39}). +@end table +@end table -@item ow, oh -These are the same as @var{out_w} and @var{out_h}. +The horizontal and vertical deblocking filters share the difference and +flatness values so you cannot set different horizontal and vertical +thresholds. -@item a -This is the same as @var{iw} / @var{ih}. +@table @option +@item h1/x1hdeblock +Experimental horizontal deblocking filter -@item sar -input sample aspect ratio +@item v1/x1vdeblock +Experimental vertical deblocking filter -@item dar -The input display aspect ratio; it is the same as -(@var{iw} / @var{ih}) * @var{sar}. +@item dr/dering +Deringing filter -@item hsub, vsub -The horizontal and vertical chroma subsample values. For example, for the -pixel format "yuv422p" @var{hsub} is 2 and @var{vsub} is 1. +@item tn/tmpnoise[|threshold1[|threshold2[|threshold3]]], temporal noise reducer +@table @option +@item threshold1 +larger -> stronger filtering +@item threshold2 +larger -> stronger filtering +@item threshold3 +larger -> stronger filtering @end table -If the input image format is different from the format requested by -the next filter, the scale filter will convert the input to the -requested format. +@item al/autolevels[:f/fullyrange], automatic brightness / contrast correction +@table @option +@item f/fullyrange +Stretch luminance to @code{0-255}. +@end table -If the value for @var{w} or @var{h} is 0, the respective input -size is used for the output. +@item lb/linblenddeint +Linear blend deinterlacing filter that deinterlaces the given block by +filtering all lines with a @code{(1 2 1)} filter. -If the value for @var{w} or @var{h} is -1, the scale filter will use, for the -respective output size, a value that maintains the aspect ratio of the input -image. +@item li/linipoldeint +Linear interpolating deinterlacing filter that deinterlaces the given block by +linearly interpolating every second line. -The default value of @var{w} and @var{h} is 0. +@item ci/cubicipoldeint +Cubic interpolating deinterlacing filter deinterlaces the given block by +cubically interpolating every second line. -Some examples: -@example -# Scale the input video to a size of 200x100 -scale=w=200:h=100 +@item md/mediandeint +Median deinterlacing filter that deinterlaces the given block by applying a +median filter to every second line. -# Scale the input to 2x -scale=w=2*iw:h=2*ih -# The above is the same as -scale=2*in_w:2*in_h +@item fd/ffmpegdeint +FFmpeg deinterlacing filter that deinterlaces the given block by filtering every +second line with a @code{(-1 4 2 4 -1)} filter. -# Scale the input to half the original size -scale=w=iw/2:h=ih/2 +@item l5/lowpass5 +Vertically applied FIR lowpass deinterlacing filter that deinterlaces the given +block by filtering all lines with a @code{(-1 2 6 2 -1)} filter. -# Increase the width, and set the height to the same size -scale=3/2*iw:ow +@item fq/forceQuant[|quantizer] +Overrides the quantizer table from the input with the constant quantizer you +specify. +@table @option +@item quantizer +Quantizer to use +@end table -# Seek Greek harmony -scale=iw:1/PHI*iw -scale=ih*PHI:ih +@item de/default +Default pp filter combination (@code{hb|a,vb|a,dr|a}) -# Increase the height, and set the width to 3/2 of the height -scale=w=3/2*oh:h=3/5*ih +@item fa/fast +Fast pp filter combination (@code{h1|a,v1|a,dr|a}) -# Increase the size, making the size a multiple of the chroma -scale="trunc(3/2*iw/hsub)*hsub:trunc(3/2*ih/vsub)*vsub" +@item ac +High quality pp filter combination (@code{ha|a|128|7,va|a,dr|a}) +@end table -# Increase the width to a maximum of 500 pixels, -# keeping the same aspect ratio as the input -scale=w='min(500\, iw*3/2):h=-1' +@subsection Examples + +@itemize +@item +Apply horizontal and vertical deblocking, deringing and automatic +brightness/contrast: +@example +pp=hb/vb/dr/al @end example -@section select -Select frames to pass in output. +@item +Apply default filters without brightness/contrast correction: +@example +pp=de/-al +@end example -It accepts the following parameters: +@item +Apply default filters and temporal denoiser: +@example +pp=default/tmpnoise|1|2|3 +@end example + +@item +Apply deblocking on luminance only, and switch vertical deblocking on or off +automatically depending on available CPU time: +@example +pp=hb|y/vb|a +@end example +@end itemize + +@section pp7 +Apply Postprocessing filter 7. It is variant of the @ref{spp} filter, +similar to spp = 6 with 7 point DCT, where only the center sample is +used after IDCT. + +The filter accepts the following options: @table @option +@item qp +Force a constant quantization parameter. It accepts an integer in range +0 to 63. If not set, the filter will use the QP from the video stream +(if available). -@item expr -An expression, which is evaluated for each input frame. If the expression is -evaluated to a non-zero value, the frame is selected and passed to the output, -otherwise it is discarded. +@item mode +Set thresholding mode. Available modes are: +@table @samp +@item hard +Set hard thresholding. +@item soft +Set soft thresholding (better de-ringing effect, but likely blurrier). +@item medium +Set medium thresholding (good results, default). +@end table @end table -The expression can contain the following constants: +@section psnr + +Obtain the average, maximum and minimum PSNR (Peak Signal to Noise +Ratio) between two input videos. + +This filter takes in input two input videos, the first input is +considered the "main" source and is passed unchanged to the +output. The second input is used as a "reference" video for computing +the PSNR. + +Both video inputs must have the same resolution and pixel format for +this filter to work correctly. Also it assumes that both inputs +have the same number of frames, which are compared one by one. + +The obtained average PSNR is printed through the logging system. + +The filter stores the accumulated MSE (mean squared error) of each +frame, and at the end of the processing it is averaged across all frames +equally, and the following formula is applied to obtain the PSNR: + +@example +PSNR = 10*log10(MAX^2/MSE) +@end example + +Where MAX is the average of the maximum values of each component of the +image. + +The description of the accepted parameters follows. @table @option -@item E, PI, PHI -These are approximated values for the mathematical constants e -(Euler's number), pi (Greek pi), and phi (the golden ratio). +@item stats_file, f +If specified the filter will use the named file to save the PSNR of +each individual frame. When filename equals "-" the data is sent to +standard output. +@end table +The file printed if @var{stats_file} is selected, contains a sequence of +key/value pairs of the form @var{key}:@var{value} for each compared +couple of frames. + +A description of each shown parameter follows: + +@table @option @item n -The (sequential) number of the filtered frame, starting from 0. +sequential number of the input frame, starting from 1 -@item selected_n -The (sequential) number of the selected frame, starting from 0. +@item mse_avg +Mean Square Error pixel-by-pixel average difference of the compared +frames, averaged over all the image components. -@item prev_selected_n -The sequential number of the last selected frame. It's NAN if undefined. +@item mse_y, mse_u, mse_v, mse_r, mse_g, mse_g, mse_a +Mean Square Error pixel-by-pixel average difference of the compared +frames for the component specified by the suffix. -@item TB -The timebase of the input timestamps. +@item psnr_y, psnr_u, psnr_v, psnr_r, psnr_g, psnr_b, psnr_a +Peak Signal to Noise ratio of the compared frames for the component +specified by the suffix. +@end table -@item pts -The PTS (Presentation TimeStamp) of the filtered video frame, -expressed in @var{TB} units. It's NAN if undefined. +For example: +@example +movie=ref_movie.mpg, setpts=PTS-STARTPTS [main]; +[main][ref] psnr="stats_file=stats.log" [out] +@end example -@item t -The PTS of the filtered video frame, -expressed in seconds. It's NAN if undefined. +On this example the input file being processed is compared with the +reference file @file{ref_movie.mpg}. The PSNR of each individual frame +is stored in @file{stats.log}. -@item prev_pts -The PTS of the previously filtered video frame. It's NAN if undefined. +@anchor{pullup} +@section pullup -@item prev_selected_pts -The PTS of the last previously filtered video frame. It's NAN if undefined. +Pulldown reversal (inverse telecine) filter, capable of handling mixed +hard-telecine, 24000/1001 fps progressive, and 30000/1001 fps progressive +content. -@item prev_selected_t -The PTS of the last previously selected video frame. It's NAN if undefined. +The pullup filter is designed to take advantage of future context in making +its decisions. This filter is stateless in the sense that it does not lock +onto a pattern to follow, but it instead looks forward to the following +fields in order to identify matches and rebuild progressive frames. -@item start_pts -The PTS of the first video frame in the video. It's NAN if undefined. +To produce content with an even framerate, insert the fps filter after +pullup, use @code{fps=24000/1001} if the input frame rate is 29.97fps, +@code{fps=24} for 30fps and the (rare) telecined 25fps input. -@item start_t -The time of the first video frame in the video. It's NAN if undefined. +The filter accepts the following options: -@item pict_type -The type of the filtered frame. It can assume one of the following -values: @table @option -@item I -@item P -@item B -@item S -@item SI -@item SP -@item BI +@item jl +@item jr +@item jt +@item jb +These options set the amount of "junk" to ignore at the left, right, top, and +bottom of the image, respectively. Left and right are in units of 8 pixels, +while top and bottom are in units of 2 lines. +The default is 8 pixels on each side. + +@item sb +Set the strict breaks. Setting this option to 1 will reduce the chances of +filter generating an occasional mismatched frame, but it may also cause an +excessive number of frames to be dropped during high motion sequences. +Conversely, setting it to -1 will make filter match fields more easily. +This may help processing of video where there is slight blurring between +the fields, but may also cause there to be interlaced frames in the output. +Default value is @code{0}. + +@item mp +Set the metric plane to use. It accepts the following values: +@table @samp +@item l +Use luma plane. + +@item u +Use chroma blue plane. + +@item v +Use chroma red plane. @end table -@item interlace_type -The frame interlace type. It can assume one of the following values: +This option may be set to use chroma plane instead of the default luma plane +for doing filter's computations. This may improve accuracy on very clean +source material, but more likely will decrease accuracy, especially if there +is chroma noise (rainbow effect) or any grayscale video. +The main purpose of setting @option{mp} to a chroma plane is to reduce CPU +load and make pullup usable in realtime on slow machines. +@end table + +For best results (without duplicated frames in the output file) it is +necessary to change the output frame rate. For example, to inverse +telecine NTSC input: +@example +ffmpeg -i input -vf pullup -r 24000/1001 ... +@end example + +@section qp + +Change video quantization parameters (QP). + +The filter accepts the following option: + @table @option -@item PROGRESSIVE -The frame is progressive (not interlaced). -@item TOPFIRST -The frame is top-field-first. -@item BOTTOMFIRST -The frame is bottom-field-first. +@item qp +Set expression for quantization parameter. @end table -@item key -This is 1 if the filtered frame is a key-frame, 0 otherwise. +The expression is evaluated through the eval API and can contain, among others, +the following constants: +@table @var +@item known +1 if index is not 129, 0 otherwise. + +@item qp +Sequentional index starting from -129 to 128. @end table -The default value of the select expression is "1". +@subsection Examples -Some examples: +@itemize +@item +Some equation like: +@example +qp=2+2*sin(PI*qp) +@end example +@end itemize + +@section random + +Flush video frames from internal cache of frames into a random order. +No frame is discarded. +Inspired by @ref{frei0r} nervous filter. + +@table @option +@item frames +Set size in number of frames of internal cache, in range from @code{2} to +@code{512}. Default is @code{30}. + +@item seed +Set seed for random number generator, must be an integer included between +@code{0} and @code{UINT32_MAX}. If not specified, or if explicitly set to +less than @code{0}, the filter will try to use a good random seed on a +best effort basis. +@end table + +@section removegrain + +The removegrain filter is a spatial denoiser for progressive video. + +@table @option +@item m0 +Set mode for the first plane. + +@item m1 +Set mode for the second plane. + +@item m2 +Set mode for the third plane. + +@item m3 +Set mode for the fourth plane. +@end table + +Range of mode is from 0 to 24. Description of each mode follows: + +@table @var +@item 0 +Leave input plane unchanged. Default. + +@item 1 +Clips the pixel with the minimum and maximum of the 8 neighbour pixels. + +@item 2 +Clips the pixel with the second minimum and maximum of the 8 neighbour pixels. + +@item 3 +Clips the pixel with the third minimum and maximum of the 8 neighbour pixels. + +@item 4 +Clips the pixel with the fourth minimum and maximum of the 8 neighbour pixels. +This is equivalent to a median filter. + +@item 5 +Line-sensitive clipping giving the minimal change. + +@item 6 +Line-sensitive clipping, intermediate. + +@item 7 +Line-sensitive clipping, intermediate. + +@item 8 +Line-sensitive clipping, intermediate. + +@item 9 +Line-sensitive clipping on a line where the neighbours pixels are the closest. + +@item 10 +Replaces the target pixel with the closest neighbour. + +@item 11 +[1 2 1] horizontal and vertical kernel blur. + +@item 12 +Same as mode 11. + +@item 13 +Bob mode, interpolates top field from the line where the neighbours +pixels are the closest. + +@item 14 +Bob mode, interpolates bottom field from the line where the neighbours +pixels are the closest. + +@item 15 +Bob mode, interpolates top field. Same as 13 but with a more complicated +interpolation formula. + +@item 16 +Bob mode, interpolates bottom field. Same as 14 but with a more complicated +interpolation formula. + +@item 17 +Clips the pixel with the minimum and maximum of respectively the maximum and +minimum of each pair of opposite neighbour pixels. + +@item 18 +Line-sensitive clipping using opposite neighbours whose greatest distance from +the current pixel is minimal. + +@item 19 +Replaces the pixel with the average of its 8 neighbours. + +@item 20 +Averages the 9 pixels ([1 1 1] horizontal and vertical blur). + +@item 21 +Clips pixels using the averages of opposite neighbour. + +@item 22 +Same as mode 21 but simpler and faster. + +@item 23 +Small edge and halo removal, but reputed useless. + +@item 24 +Similar as 23. +@end table + +@section removelogo + +Suppress a TV station logo, using an image file to determine which +pixels comprise the logo. It works by filling in the pixels that +comprise the logo with neighboring pixels. + +The filter accepts the following options: + +@table @option +@item filename, f +Set the filter bitmap file, which can be any image format supported by +libavformat. The width and height of the image file must match those of the +video stream being processed. +@end table +Pixels in the provided bitmap image with a value of zero are not +considered part of the logo, non-zero pixels are considered part of +the logo. If you use white (255) for the logo and black (0) for the +rest, you will be safe. For making the filter bitmap, it is +recommended to take a screen capture of a black frame with the logo +visible, and then using a threshold filter followed by the erode +filter once or twice. + +If needed, little splotches can be fixed manually. Remember that if +logo pixels are not covered, the filter quality will be much +reduced. Marking too many pixels as part of the logo does not hurt as +much, but it will increase the amount of blurring needed to cover over +the image and will destroy more information than necessary, and extra +pixels will slow things down on a large logo. + +@section repeatfields + +This filter uses the repeat_field flag from the Video ES headers and hard repeats +fields based on its value. + +@section reverse, areverse + +Reverse a clip. + +Warning: This filter requires memory to buffer the entire clip, so trimming +is suggested. + +@subsection Examples + +@itemize +@item +Take the first 5 seconds of a clip, and reverse it. @example -# Select all the frames in input -select +trim=end=5,reverse +@end example +@end itemize -# The above is the same as -select=expr=1 +@section rotate -# Skip all frames -select=expr=0 +Rotate video by an arbitrary angle expressed in radians. -# Select only I-frames -select='expr=eq(pict_type\,I)' +The filter accepts the following options: -# Select one frame per 100 -select='not(mod(n\,100))' +A description of the optional parameters follows. +@table @option +@item angle, a +Set an expression for the angle by which to rotate the input video +clockwise, expressed as a number of radians. A negative value will +result in a counter-clockwise rotation. By default it is set to "0". -# Select only frames contained in the 10-20 time interval -select='gte(t\,10)*lte(t\,20)' +This expression is evaluated for each frame. -# Select only I frames contained in the 10-20 time interval -select='gte(t\,10)*lte(t\,20)*eq(pict_type\,I)' +@item out_w, ow +Set the output width expression, default value is "iw". +This expression is evaluated just once during configuration. -# Select frames with a minimum distance of 10 seconds -select='isnan(prev_selected_t)+gte(t-prev_selected_t\,10)' +@item out_h, oh +Set the output height expression, default value is "ih". +This expression is evaluated just once during configuration. + +@item bilinear +Enable bilinear interpolation if set to 1, a value of 0 disables +it. Default value is 1. + +@item fillcolor, c +Set the color used to fill the output area not covered by the rotated +image. For the general syntax of this option, check the "Color" section in the +ffmpeg-utils manual. If the special value "none" is selected then no +background is printed (useful for example if the background is never shown). + +Default value is "black". +@end table + +The expressions for the angle and the output size can contain the +following constants and functions: + +@table @option +@item n +sequential number of the input frame, starting from 0. It is always NAN +before the first frame is filtered. + +@item t +time in seconds of the input frame, it is set to 0 when the filter is +configured. It is always NAN before the first frame is filtered. + +@item hsub +@item vsub +horizontal and vertical chroma subsample values. For example for the +pixel format "yuv422p" @var{hsub} is 2 and @var{vsub} is 1. + +@item in_w, iw +@item in_h, ih +the input video width and height + +@item out_w, ow +@item out_h, oh +the output width and height, that is the size of the padded area as +specified by the @var{width} and @var{height} expressions + +@item rotw(a) +@item roth(a) +the minimal width/height required for completely containing the input +video rotated by @var{a} radians. + +These are only available when computing the @option{out_w} and +@option{out_h} expressions. +@end table + +@subsection Examples + +@itemize +@item +Rotate the input by PI/6 radians clockwise: +@example +rotate=PI/6 @end example -@anchor{setdar} -@section setdar +@item +Rotate the input by PI/6 radians counter-clockwise: +@example +rotate=-PI/6 +@end example -Set the Display Aspect Ratio for the filter output video. +@item +Rotate the input by 45 degrees clockwise: +@example +rotate=45*PI/180 +@end example -This is done by changing the specified Sample (aka Pixel) Aspect -Ratio, according to the following equation: -@math{DAR = HORIZONTAL_RESOLUTION / VERTICAL_RESOLUTION * SAR} +@item +Apply a constant rotation with period T, starting from an angle of PI/3: +@example +rotate=PI/3+2*PI*t/T +@end example -Keep in mind that this filter does not modify the pixel dimensions of -the video frame. Also, the display aspect ratio set by this filter may -be changed by later filters in the filterchain, e.g. in case of -scaling or if another "setdar" or a "setsar" filter is applied. +@item +Make the input video rotation oscillating with a period of T +seconds and an amplitude of A radians: +@example +rotate=A*sin(2*PI/T*t) +@end example -It accepts the following parameters: +@item +Rotate the video, output size is chosen so that the whole rotating +input video is always completely contained in the output: +@example +rotate='2*PI*t:ow=hypot(iw,ih):oh=ow' +@end example + +@item +Rotate the video, reduce the output size so that no background is ever +shown: +@example +rotate=2*PI*t:ow='min(iw,ih)/sqrt(2)':oh=ow:c=none +@end example +@end itemize + +@subsection Commands + +The filter supports the following commands: @table @option +@item a, angle +Set the angle expression. +The command accepts the same syntax of the corresponding option. -@item dar -The output display aspect ratio. +If the specified expression is not valid, it is kept at its current +value. +@end table +@section sab + +Apply Shape Adaptive Blur. + +The filter accepts the following options: + +@table @option +@item luma_radius, lr +Set luma blur filter strength, must be a value in range 0.1-4.0, default +value is 1.0. A greater value will result in a more blurred image, and +in slower processing. + +@item luma_pre_filter_radius, lpfr +Set luma pre-filter radius, must be a value in the 0.1-2.0 range, default +value is 1.0. + +@item luma_strength, ls +Set luma maximum difference between pixels to still be considered, must +be a value in the 0.1-100.0 range, default value is 1.0. + +@item chroma_radius, cr +Set chroma blur filter strength, must be a value in range 0.1-4.0. A +greater value will result in a more blurred image, and in slower +processing. + +@item chroma_pre_filter_radius, cpfr +Set chroma pre-filter radius, must be a value in the 0.1-2.0 range. + +@item chroma_strength, cs +Set chroma maximum difference between pixels to still be considered, +must be a value in the 0.1-100.0 range. @end table -The parameter @var{dar} is an expression containing -the following constants: +Each chroma option value, if not explicitly specified, is set to the +corresponding luma option value. + +@anchor{scale} +@section scale + +Scale (resize) the input video, using the libswscale library. + +The scale filter forces the output display aspect ratio to be the same +of the input, by changing the output sample aspect ratio. + +If the input image format is different from the format requested by +the next filter, the scale filter will convert the input to the +requested format. + +@subsection Options +The filter accepts the following options, or any of the options +supported by the libswscale scaler. + +See @ref{scaler_options,,the ffmpeg-scaler manual,ffmpeg-scaler} for +the complete list of scaler options. @table @option -@item E, PI, PHI -These are approximated values for the mathematical constants e -(Euler's number), pi (Greek pi), and phi (the golden ratio). +@item width, w +@item height, h +Set the output video dimension expression. Default value is the input +dimension. -@item w, h -The input width and height. +If the value is 0, the input width is used for the output. + +If one of the values is -1, the scale filter will use a value that +maintains the aspect ratio of the input image, calculated from the +other specified dimension. If both of them are -1, the input size is +used + +If one of the values is -n with n > 1, the scale filter will also use a value +that maintains the aspect ratio of the input image, calculated from the other +specified dimension. After that it will, however, make sure that the calculated +dimension is divisible by n and adjust the value if necessary. + +See below for the list of accepted constants for use in the dimension +expression. + +@item eval +Specify when to evaluate @var{width} and @var{height} expression. It accepts the following values: + +@table @samp +@item init +Only evaluate expressions once during the filter initialization or when a command is processed. + +@item frame +Evaluate expressions for each incoming frame. + +@end table + +Default value is @samp{init}. + + +@item interl +Set the interlacing mode. It accepts the following values: + +@table @samp +@item 1 +Force interlaced aware scaling. + +@item 0 +Do not apply interlaced scaling. + +@item -1 +Select interlaced aware scaling depending on whether the source frames +are flagged as interlaced or not. +@end table + +Default value is @samp{0}. + +@item flags +Set libswscale scaling flags. See +@ref{sws_flags,,the ffmpeg-scaler manual,ffmpeg-scaler} for the +complete list of values. If not explicitly specified the filter applies +the default flags. + + +@item param0, param1 +Set libswscale input parameters for scaling algorithms that need them. See +@ref{sws_params,,the ffmpeg-scaler manual,ffmpeg-scaler} for the +complete documentation. If not explicitly specified the filter applies +empty parameters. + + + +@item size, s +Set the video size. For the syntax of this option, check the +@ref{video size syntax,,"Video size" section in the ffmpeg-utils manual,ffmpeg-utils}. + +@item in_color_matrix +@item out_color_matrix +Set in/output YCbCr color space type. + +This allows the autodetected value to be overridden as well as allows forcing +a specific value used for the output and encoder. + +If not specified, the color space type depends on the pixel format. + +Possible values: + +@table @samp +@item auto +Choose automatically. + +@item bt709 +Format conforming to International Telecommunication Union (ITU) +Recommendation BT.709. + +@item fcc +Set color space conforming to the United States Federal Communications +Commission (FCC) Code of Federal Regulations (CFR) Title 47 (2003) 73.682 (a). + +@item bt601 +Set color space conforming to: + +@itemize +@item +ITU Radiocommunication Sector (ITU-R) Recommendation BT.601 + +@item +ITU-R Rec. BT.470-6 (1998) Systems B, B1, and G + +@item +Society of Motion Picture and Television Engineers (SMPTE) ST 170:2004 + +@end itemize + +@item smpte240m +Set color space conforming to SMPTE ST 240:1999. +@end table + +@item in_range +@item out_range +Set in/output YCbCr sample range. + +This allows the autodetected value to be overridden as well as allows forcing +a specific value used for the output and encoder. If not specified, the +range depends on the pixel format. Possible values: + +@table @samp +@item auto +Choose automatically. + +@item jpeg/full/pc +Set full range (0-255 in case of 8-bit luma). + +@item mpeg/tv +Set "MPEG" range (16-235 in case of 8-bit luma). +@end table + +@item force_original_aspect_ratio +Enable decreasing or increasing output video width or height if necessary to +keep the original aspect ratio. Possible values: + +@table @samp +@item disable +Scale the video as specified and disable this feature. + +@item decrease +The output video dimensions will automatically be decreased if needed. + +@item increase +The output video dimensions will automatically be increased if needed. + +@end table + +One useful instance of this option is that when you know a specific device's +maximum allowed resolution, you can use this to limit the output video to +that, while retaining the aspect ratio. For example, device A allows +1280x720 playback, and your video is 1920x800. Using this option (set it to +decrease) and specifying 1280x720 to the command line makes the output +1280x533. + +Please note that this is a different thing than specifying -1 for @option{w} +or @option{h}, you still need to specify the output resolution for this option +to work. + +@end table + +The values of the @option{w} and @option{h} options are expressions +containing the following constants: + +@table @var +@item in_w +@item in_h +The input width and height + +@item iw +@item ih +These are the same as @var{in_w} and @var{in_h}. + +@item out_w +@item out_h +The output (scaled) width and height + +@item ow +@item oh +These are the same as @var{out_w} and @var{out_h} @item a -This is the same as @var{w} / @var{h}. +The same as @var{iw} / @var{ih} @item sar -The input sample aspect ratio. +input sample aspect ratio @item dar -The input display aspect ratio. It is the same as -(@var{w} / @var{h}) * @var{sar}. +The input display aspect ratio. Calculated from @code{(iw / ih) * sar}. -@item hsub, vsub -The horizontal and vertical chroma subsample values. For example, for the +@item hsub +@item vsub +horizontal and vertical input chroma subsample values. For example for the +pixel format "yuv422p" @var{hsub} is 2 and @var{vsub} is 1. + +@item ohsub +@item ovsub +horizontal and vertical output chroma subsample values. For example for the pixel format "yuv422p" @var{hsub} is 2 and @var{vsub} is 1. @end table -To change the display aspect ratio to 16:9, specify: +@subsection Examples + +@itemize +@item +Scale the input video to a size of 200x100 @example -setdar=dar=16/9 -# The above is equivalent to -setdar=dar=1.77777 +scale=w=200:h=100 @end example -Also see the the @ref{setsar} filter documentation. +This is equivalent to: +@example +scale=200:100 +@end example -@section setpts +or: +@example +scale=200x100 +@end example -Change the PTS (presentation timestamp) of the input video frames. +@item +Specify a size abbreviation for the output size: +@example +scale=qcif +@end example -It accepts the following parameters: +which can also be written as: +@example +scale=size=qcif +@end example -@table @option +@item +Scale the input to 2x: +@example +scale=w=2*iw:h=2*ih +@end example -@item expr -The expression which is evaluated for each frame to construct its timestamp. +@item +The above is the same as: +@example +scale=2*in_w:2*in_h +@end example -@end table +@item +Scale the input to 2x with forced interlaced scaling: +@example +scale=2*iw:2*ih:interl=1 +@end example -The expression is evaluated through the eval API and can contain the following -constants: +@item +Scale the input to half size: +@example +scale=w=iw/2:h=ih/2 +@end example + +@item +Increase the width, and set the height to the same size: +@example +scale=3/2*iw:ow +@end example +@item +Seek Greek harmony: +@example +scale=iw:1/PHI*iw +scale=ih*PHI:ih +@end example + +@item +Increase the height, and set the width to 3/2 of the height: +@example +scale=w=3/2*oh:h=3/5*ih +@end example + +@item +Increase the size, making the size a multiple of the chroma +subsample values: +@example +scale="trunc(3/2*iw/hsub)*hsub:trunc(3/2*ih/vsub)*vsub" +@end example + +@item +Increase the width to a maximum of 500 pixels, +keeping the same aspect ratio as the input: +@example +scale=w='min(500\, iw*3/2):h=-1' +@end example +@end itemize + +@subsection Commands + +This filter supports the following commands: @table @option -@item PTS -The presentation timestamp in input. +@item width, w +@item height, h +Set the output video dimension expression. +The command accepts the same syntax of the corresponding option. -@item E, PI, PHI -These are approximated values for the mathematical constants e -(Euler's number), pi (Greek pi), and phi (the golden ratio). +If the specified expression is not valid, it is kept at its current +value. +@end table -@item N -The count of the input frame, starting from 0. +@section scale2ref -@item STARTPTS -The PTS of the first video frame. +Scale (resize) the input video, based on a reference video. -@item INTERLACED -State whether the current frame is interlaced. +See the scale filter for available options, scale2ref supports the same but +uses the reference video instead of the main input as basis. -@item PREV_INPTS -The previous input PTS. +@subsection Examples -@item PREV_OUTPTS -The previous output PTS. +@itemize +@item +Scale a subtitle stream to match the main video in size before overlaying +@example +'scale2ref[b][a];[a][b]overlay' +@end example +@end itemize -@item RTCTIME -The wallclock (RTC) time in microseconds. +@section selectivecolor -@item RTCSTART -The wallclock (RTC) time at the start of the movie in microseconds. +Adjust cyan, magenta, yellow and black (CMYK) to certain ranges of colors (such +as "reds", "yellows", "greens", "cyans", ...). The adjustment range is defined +by the "purity" of the color (that is, how saturated it already is). -@item TB -The timebase of the input timestamps. +This filter is similar to the Adobe Photoshop Selective Color tool. +The filter accepts the following options: + +@table @option +@item correction_method +Select color correction method. + +Available values are: +@table @samp +@item absolute +Specified adjustments are applied "as-is" (added/subtracted to original pixel +component value). +@item relative +Specified adjustments are relative to the original component value. +@end table +Default is @code{absolute}. +@item reds +Adjustments for red pixels (pixels where the red component is the maximum) +@item yellows +Adjustments for yellow pixels (pixels where the blue component is the minimum) +@item greens +Adjustments for green pixels (pixels where the green component is the maximum) +@item cyans +Adjustments for cyan pixels (pixels where the red component is the minimum) +@item blues +Adjustments for blue pixels (pixels where the blue component is the maximum) +@item magentas +Adjustments for magenta pixels (pixels where the green component is the minimum) +@item whites +Adjustments for white pixels (pixels where all components are greater than 128) +@item neutrals +Adjustments for all pixels except pure black and pure white +@item blacks +Adjustments for black pixels (pixels where all components are lesser than 128) +@item psfile +Specify a Photoshop selective color file (@code{.asv}) to import the settings from. @end table -Some examples: +All the adjustment settings (@option{reds}, @option{yellows}, ...) accept up to +4 space separated floating point adjustment values in the [-1,1] range, +respectively to adjust the amount of cyan, magenta, yellow and black for the +pixels of its range. + +@subsection Examples +@itemize +@item +Increase cyan by 50% and reduce yellow by 33% in every green areas, and +increase magenta by 27% in blue areas: @example -# Start counting the PTS from zero -setpts=expr=PTS-STARTPTS +selectivecolor=greens=.5 0 -.33 0:blues=0 .27 +@end example -# Fast motion -setpts=expr=0.5*PTS +@item +Use a Photoshop selective color preset: +@example +selectivecolor=psfile=MySelectiveColorPresets/Misty.asv +@end example +@end itemize -# Slow motion -setpts=2.0*PTS +@section separatefields -# Fixed rate 25 fps -setpts=N/(25*TB) +The @code{separatefields} takes a frame-based video input and splits +each frame into its components fields, producing a new half height clip +with twice the frame rate and twice the frame count. -# Fixed rate 25 fps with some jitter -setpts='1/(25*TB) * (N + 0.05 * sin(N*2*PI/25))' +This filter use field-dominance information in frame to decide which +of each pair of fields to place first in the output. +If it gets it wrong use @ref{setfield} filter before @code{separatefields} filter. + +@section setdar, setsar + +The @code{setdar} filter sets the Display Aspect Ratio for the filter +output video. -# Generate timestamps from a "live source" and rebase onto the current timebase -setpts='(RTCTIME - RTCSTART) / (TB * 1000000)" +This is done by changing the specified Sample (aka Pixel) Aspect +Ratio, according to the following equation: +@example +@var{DAR} = @var{HORIZONTAL_RESOLUTION} / @var{VERTICAL_RESOLUTION} * @var{SAR} @end example -@anchor{setsar} -@section setsar +Keep in mind that the @code{setdar} filter does not modify the pixel +dimensions of the video frame. Also, the display aspect ratio set by +this filter may be changed by later filters in the filterchain, +e.g. in case of scaling or if another "setdar" or a "setsar" filter is +applied. -Set the Sample (aka Pixel) Aspect Ratio for the filter output video. +The @code{setsar} filter sets the Sample (aka Pixel) Aspect Ratio for +the filter output video. Note that as a consequence of the application of this filter, the -output display aspect ratio will change according to the following -equation: -@math{DAR = HORIZONTAL_RESOLUTION / VERTICAL_RESOLUTION * SAR} +output display aspect ratio will change according to the equation +above. -Keep in mind that the sample aspect ratio set by this filter may be -changed by later filters in the filterchain, e.g. if another "setsar" -or a "setdar" filter is applied. +Keep in mind that the sample aspect ratio set by the @code{setsar} +filter may be changed by later filters in the filterchain, e.g. if +another "setsar" or a "setdar" filter is applied. It accepts the following parameters: @table @option +@item r, ratio, dar (@code{setdar} only), sar (@code{setsar} only) +Set the aspect ratio used by the filter. -@item sar -The output sample aspect ratio. +The parameter can be a floating point number string, an expression, or +a string of the form @var{num}:@var{den}, where @var{num} and +@var{den} are the numerator and denominator of the aspect ratio. If +the parameter is not specified, it is assumed the value "0". +In case the form "@var{num}:@var{den}" is used, the @code{:} character +should be escaped. + +@item max +Set the maximum integer value to use for expressing numerator and +denominator when reducing the expressed aspect ratio to a rational. +Default value is @code{100}. @end table @@ -2464,48 +10747,64 @@ Horizontal and vertical chroma subsample values. For example, for the pixel format "yuv422p" @var{hsub} is 2 and @var{vsub} is 1. @end table +@subsection Examples + +@itemize + +@item +To change the display aspect ratio to 16:9, specify one of the following: +@example +setdar=dar=1.77777 +setdar=dar=16/9 +setdar=dar=1.77777 +@end example + +@item To change the sample aspect ratio to 10:11, specify: @example setsar=sar=10/11 @end example -@section settb - -Set the timebase to use for the output frames timestamps. -It is mainly useful for testing timebase configuration. - -It accepts the following parameters: +@item +To set a display aspect ratio of 16:9, and specify a maximum integer value of +1000 in the aspect ratio reduction, use the command: +@example +setdar=ratio=16/9:max=1000 +@end example -@table @option +@end itemize -@item expr -The expression which is evaluated into the output timebase. +@anchor{setfield} +@section setfield -@end table +Force field for the output video frame. -The expression can contain the constants "PI", "E", "PHI", "AVTB" (the -default timebase), and "intb" (the input timebase). +The @code{setfield} filter marks the interlace type field for the +output frames. It does not change the input frame, but only sets the +corresponding property, which affects how the frame is treated by +following filters (e.g. @code{fieldorder} or @code{yadif}). -The default value for the input is "intb". +The filter accepts the following options: -Some examples: +@table @option -@example -# Set the timebase to 1/25 -settb=expr=1/25 +@item mode +Available values are: -# Set the timebase to 1/10 -settb=expr=0.1 +@table @samp +@item auto +Keep the same field property. -# Set the timebase to 1001/1000 -settb=1+0.001 +@item bff +Mark the frame as bottom-field-first. -#Set the timebase to 2*intb -settb=2*intb +@item tff +Mark the frame as top-field-first. -#Set the default timebase value -settb=AVTB -@end example +@item prog +Mark the frame as progressive. +@end table +@end table @section showinfo @@ -2515,7 +10814,7 @@ The input video is not modified. The shown line contains a sequence of key/value pairs of the form @var{key}:@var{value}. -It accepts the following parameters: +The following values are shown in the output: @table @option @item n @@ -2541,8 +10840,8 @@ The sample aspect ratio of the input frame, expressed in the form @var{num}/@var{den}. @item s -The size of the input frame, expressed in the form -@var{width}x@var{height}. +The size of the input frame. For the syntax of this option, check the +@ref{video size syntax,,"Video size" section in the ffmpeg-utils manual,ffmpeg-utils}. @item i The type of interlaced mode ("P" for "progressive", "T" for top field first, "B" @@ -2559,13 +10858,46 @@ the @code{av_get_picture_type_char} function defined in @file{libavutil/avutil.h}. @item checksum -The Adler-32 checksum of all the planes of the input frame. +The Adler-32 checksum (printed in hexadecimal) of all the planes of the input frame. @item plane_checksum -The Adler-32 checksum of each plane of the input frame, expressed in the form -"[@var{c0} @var{c1} @var{c2} @var{c3}]". +The Adler-32 checksum (printed in hexadecimal) of each plane of the input frame, +expressed in the form "[@var{c0} @var{c1} @var{c2} @var{c3}]". @end table +@section showpalette + +Displays the 256 colors palette of each frame. This filter is only relevant for +@var{pal8} pixel format frames. + +It accepts the following option: + +@table @option +@item s +Set the size of the box used to represent one palette color entry. Default is +@code{30} (for a @code{30x30} pixel box). +@end table + +@section shuffleframes + +Reorder and/or duplicate video frames. + +It accepts the following parameters: + +@table @option +@item mapping +Set the destination indexes of input frames. +This is space or '|' separated list of indexes that maps input frames to output +frames. Number of indexes also sets maximal value that each index may have. +@end table + +The first frame has the index 0. The default is to keep the input unchanged. + +Swap second and third frame of every three frames of the input: +@example +ffmpeg -i INPUT -vf "shuffleframes=0 2 1" OUTPUT +@end example + @section shuffleplanes Reorder and/or duplicate video planes. @@ -2592,21 +10924,1058 @@ The first plane has the index 0. The default is to keep the input unchanged. Swap the second and third planes of the input: @example -avconv -i INPUT -vf shuffleplanes=0:2:1:3 OUTPUT +ffmpeg -i INPUT -vf shuffleplanes=0:2:1:3 OUTPUT @end example -@section split +@anchor{signalstats} +@section signalstats +Evaluate various visual metrics that assist in determining issues associated +with the digitization of analog video media. -Split input video into several identical outputs. +By default the filter will log these metadata values: -It accepts a single parameter, which specifies the number of outputs. If -unspecified, it defaults to 2. +@table @option +@item YMIN +Display the minimal Y value contained within the input frame. Expressed in +range of [0-255]. + +@item YLOW +Display the Y value at the 10% percentile within the input frame. Expressed in +range of [0-255]. + +@item YAVG +Display the average Y value within the input frame. Expressed in range of +[0-255]. + +@item YHIGH +Display the Y value at the 90% percentile within the input frame. Expressed in +range of [0-255]. + +@item YMAX +Display the maximum Y value contained within the input frame. Expressed in +range of [0-255]. + +@item UMIN +Display the minimal U value contained within the input frame. Expressed in +range of [0-255]. + +@item ULOW +Display the U value at the 10% percentile within the input frame. Expressed in +range of [0-255]. + +@item UAVG +Display the average U value within the input frame. Expressed in range of +[0-255]. + +@item UHIGH +Display the U value at the 90% percentile within the input frame. Expressed in +range of [0-255]. + +@item UMAX +Display the maximum U value contained within the input frame. Expressed in +range of [0-255]. + +@item VMIN +Display the minimal V value contained within the input frame. Expressed in +range of [0-255]. + +@item VLOW +Display the V value at the 10% percentile within the input frame. Expressed in +range of [0-255]. + +@item VAVG +Display the average V value within the input frame. Expressed in range of +[0-255]. + +@item VHIGH +Display the V value at the 90% percentile within the input frame. Expressed in +range of [0-255]. + +@item VMAX +Display the maximum V value contained within the input frame. Expressed in +range of [0-255]. + +@item SATMIN +Display the minimal saturation value contained within the input frame. +Expressed in range of [0-~181.02]. + +@item SATLOW +Display the saturation value at the 10% percentile within the input frame. +Expressed in range of [0-~181.02]. + +@item SATAVG +Display the average saturation value within the input frame. Expressed in range +of [0-~181.02]. + +@item SATHIGH +Display the saturation value at the 90% percentile within the input frame. +Expressed in range of [0-~181.02]. + +@item SATMAX +Display the maximum saturation value contained within the input frame. +Expressed in range of [0-~181.02]. + +@item HUEMED +Display the median value for hue within the input frame. Expressed in range of +[0-360]. + +@item HUEAVG +Display the average value for hue within the input frame. Expressed in range of +[0-360]. + +@item YDIF +Display the average of sample value difference between all values of the Y +plane in the current frame and corresponding values of the previous input frame. +Expressed in range of [0-255]. + +@item UDIF +Display the average of sample value difference between all values of the U +plane in the current frame and corresponding values of the previous input frame. +Expressed in range of [0-255]. + +@item VDIF +Display the average of sample value difference between all values of the V +plane in the current frame and corresponding values of the previous input frame. +Expressed in range of [0-255]. +@end table + +The filter accepts the following options: + +@table @option +@item stat +@item out + +@option{stat} specify an additional form of image analysis. +@option{out} output video with the specified type of pixel highlighted. -Create 5 copies of the input video: +Both options accept the following values: + +@table @samp +@item tout +Identify @var{temporal outliers} pixels. A @var{temporal outlier} is a pixel +unlike the neighboring pixels of the same field. Examples of temporal outliers +include the results of video dropouts, head clogs, or tape tracking issues. + +@item vrep +Identify @var{vertical line repetition}. Vertical line repetition includes +similar rows of pixels within a frame. In born-digital video vertical line +repetition is common, but this pattern is uncommon in video digitized from an +analog source. When it occurs in video that results from the digitization of an +analog source it can indicate concealment from a dropout compensator. + +@item brng +Identify pixels that fall outside of legal broadcast range. +@end table + +@item color, c +Set the highlight color for the @option{out} option. The default color is +yellow. +@end table + +@subsection Examples + +@itemize +@item +Output data of various video metrics: +@example +ffprobe -f lavfi movie=example.mov,signalstats="stat=tout+vrep+brng" -show_frames +@end example + +@item +Output specific data about the minimum and maximum values of the Y plane per frame: @example -avconv -i INPUT -filter_complex split=5 OUTPUT +ffprobe -f lavfi movie=example.mov,signalstats -show_entries frame_tags=lavfi.signalstats.YMAX,lavfi.signalstats.YMIN @end example +@item +Playback video while highlighting pixels that are outside of broadcast range in red. +@example +ffplay example.mov -vf signalstats="out=brng:color=red" +@end example + +@item +Playback video with signalstats metadata drawn over the frame. +@example +ffplay example.mov -vf signalstats=stat=brng+vrep+tout,drawtext=fontfile=FreeSerif.ttf:textfile=signalstat_drawtext.txt +@end example + +The contents of signalstat_drawtext.txt used in the command are: +@example +time %@{pts:hms@} +Y (%@{metadata:lavfi.signalstats.YMIN@}-%@{metadata:lavfi.signalstats.YMAX@}) +U (%@{metadata:lavfi.signalstats.UMIN@}-%@{metadata:lavfi.signalstats.UMAX@}) +V (%@{metadata:lavfi.signalstats.VMIN@}-%@{metadata:lavfi.signalstats.VMAX@}) +saturation maximum: %@{metadata:lavfi.signalstats.SATMAX@} + +@end example +@end itemize + +@anchor{smartblur} +@section smartblur + +Blur the input video without impacting the outlines. + +It accepts the following options: + +@table @option +@item luma_radius, lr +Set the luma radius. The option value must be a float number in +the range [0.1,5.0] that specifies the variance of the gaussian filter +used to blur the image (slower if larger). Default value is 1.0. + +@item luma_strength, ls +Set the luma strength. The option value must be a float number +in the range [-1.0,1.0] that configures the blurring. A value included +in [0.0,1.0] will blur the image whereas a value included in +[-1.0,0.0] will sharpen the image. Default value is 1.0. + +@item luma_threshold, lt +Set the luma threshold used as a coefficient to determine +whether a pixel should be blurred or not. The option value must be an +integer in the range [-30,30]. A value of 0 will filter all the image, +a value included in [0,30] will filter flat areas and a value included +in [-30,0] will filter edges. Default value is 0. + +@item chroma_radius, cr +Set the chroma radius. The option value must be a float number in +the range [0.1,5.0] that specifies the variance of the gaussian filter +used to blur the image (slower if larger). Default value is 1.0. + +@item chroma_strength, cs +Set the chroma strength. The option value must be a float number +in the range [-1.0,1.0] that configures the blurring. A value included +in [0.0,1.0] will blur the image whereas a value included in +[-1.0,0.0] will sharpen the image. Default value is 1.0. + +@item chroma_threshold, ct +Set the chroma threshold used as a coefficient to determine +whether a pixel should be blurred or not. The option value must be an +integer in the range [-30,30]. A value of 0 will filter all the image, +a value included in [0,30] will filter flat areas and a value included +in [-30,0] will filter edges. Default value is 0. +@end table + +If a chroma option is not explicitly set, the corresponding luma value +is set. + +@section ssim + +Obtain the SSIM (Structural SImilarity Metric) between two input videos. + +This filter takes in input two input videos, the first input is +considered the "main" source and is passed unchanged to the +output. The second input is used as a "reference" video for computing +the SSIM. + +Both video inputs must have the same resolution and pixel format for +this filter to work correctly. Also it assumes that both inputs +have the same number of frames, which are compared one by one. + +The filter stores the calculated SSIM of each frame. + +The description of the accepted parameters follows. + +@table @option +@item stats_file, f +If specified the filter will use the named file to save the SSIM of +each individual frame. When filename equals "-" the data is sent to +standard output. +@end table + +The file printed if @var{stats_file} is selected, contains a sequence of +key/value pairs of the form @var{key}:@var{value} for each compared +couple of frames. + +A description of each shown parameter follows: + +@table @option +@item n +sequential number of the input frame, starting from 1 + +@item Y, U, V, R, G, B +SSIM of the compared frames for the component specified by the suffix. + +@item All +SSIM of the compared frames for the whole frame. + +@item dB +Same as above but in dB representation. +@end table + +For example: +@example +movie=ref_movie.mpg, setpts=PTS-STARTPTS [main]; +[main][ref] ssim="stats_file=stats.log" [out] +@end example + +On this example the input file being processed is compared with the +reference file @file{ref_movie.mpg}. The SSIM of each individual frame +is stored in @file{stats.log}. + +Another example with both psnr and ssim at same time: +@example +ffmpeg -i main.mpg -i ref.mpg -lavfi "ssim;[0:v][1:v]psnr" -f null - +@end example + +@section stereo3d + +Convert between different stereoscopic image formats. + +The filters accept the following options: + +@table @option +@item in +Set stereoscopic image format of input. + +Available values for input image formats are: +@table @samp +@item sbsl +side by side parallel (left eye left, right eye right) + +@item sbsr +side by side crosseye (right eye left, left eye right) + +@item sbs2l +side by side parallel with half width resolution +(left eye left, right eye right) + +@item sbs2r +side by side crosseye with half width resolution +(right eye left, left eye right) + +@item abl +above-below (left eye above, right eye below) + +@item abr +above-below (right eye above, left eye below) + +@item ab2l +above-below with half height resolution +(left eye above, right eye below) + +@item ab2r +above-below with half height resolution +(right eye above, left eye below) + +@item al +alternating frames (left eye first, right eye second) + +@item ar +alternating frames (right eye first, left eye second) + +@item irl +interleaved rows (left eye has top row, right eye starts on next row) + +@item irr +interleaved rows (right eye has top row, left eye starts on next row) + +@item icl +interleaved columns, left eye first + +@item icr +interleaved columns, right eye first + +Default value is @samp{sbsl}. +@end table + +@item out +Set stereoscopic image format of output. + +@table @samp +@item sbsl +side by side parallel (left eye left, right eye right) + +@item sbsr +side by side crosseye (right eye left, left eye right) + +@item sbs2l +side by side parallel with half width resolution +(left eye left, right eye right) + +@item sbs2r +side by side crosseye with half width resolution +(right eye left, left eye right) + +@item abl +above-below (left eye above, right eye below) + +@item abr +above-below (right eye above, left eye below) + +@item ab2l +above-below with half height resolution +(left eye above, right eye below) + +@item ab2r +above-below with half height resolution +(right eye above, left eye below) + +@item al +alternating frames (left eye first, right eye second) + +@item ar +alternating frames (right eye first, left eye second) + +@item irl +interleaved rows (left eye has top row, right eye starts on next row) + +@item irr +interleaved rows (right eye has top row, left eye starts on next row) + +@item arbg +anaglyph red/blue gray +(red filter on left eye, blue filter on right eye) + +@item argg +anaglyph red/green gray +(red filter on left eye, green filter on right eye) + +@item arcg +anaglyph red/cyan gray +(red filter on left eye, cyan filter on right eye) + +@item arch +anaglyph red/cyan half colored +(red filter on left eye, cyan filter on right eye) + +@item arcc +anaglyph red/cyan color +(red filter on left eye, cyan filter on right eye) + +@item arcd +anaglyph red/cyan color optimized with the least squares projection of dubois +(red filter on left eye, cyan filter on right eye) + +@item agmg +anaglyph green/magenta gray +(green filter on left eye, magenta filter on right eye) + +@item agmh +anaglyph green/magenta half colored +(green filter on left eye, magenta filter on right eye) + +@item agmc +anaglyph green/magenta colored +(green filter on left eye, magenta filter on right eye) + +@item agmd +anaglyph green/magenta color optimized with the least squares projection of dubois +(green filter on left eye, magenta filter on right eye) + +@item aybg +anaglyph yellow/blue gray +(yellow filter on left eye, blue filter on right eye) + +@item aybh +anaglyph yellow/blue half colored +(yellow filter on left eye, blue filter on right eye) + +@item aybc +anaglyph yellow/blue colored +(yellow filter on left eye, blue filter on right eye) + +@item aybd +anaglyph yellow/blue color optimized with the least squares projection of dubois +(yellow filter on left eye, blue filter on right eye) + +@item ml +mono output (left eye only) + +@item mr +mono output (right eye only) + +@item chl +checkerboard, left eye first + +@item chr +checkerboard, right eye first + +@item icl +interleaved columns, left eye first + +@item icr +interleaved columns, right eye first +@end table + +Default value is @samp{arcd}. +@end table + +@subsection Examples + +@itemize +@item +Convert input video from side by side parallel to anaglyph yellow/blue dubois: +@example +stereo3d=sbsl:aybd +@end example + +@item +Convert input video from above below (left eye above, right eye below) to side by side crosseye. +@example +stereo3d=abl:sbsr +@end example +@end itemize + +@section streamselect, astreamselect +Select video or audio streams. + +The filter accepts the following options: + +@table @option +@item inputs +Set number of inputs. Default is 2. + +@item map +Set input indexes to remap to outputs. +@end table + +@subsection Commands + +The @code{streamselect} and @code{astreamselect} filter supports the following +commands: + +@table @option +@item map +Set input indexes to remap to outputs. +@end table + +@subsection Examples + +@itemize +@item +Select first 5 seconds 1st stream and rest of time 2nd stream: +@example +sendcmd='5.0 streamselect map 1',streamselect=inputs=2:map=0 +@end example + +@item +Same as above, but for audio: +@example +asendcmd='5.0 astreamselect map 1',astreamselect=inputs=2:map=0 +@end example +@end itemize + +@anchor{spp} +@section spp + +Apply a simple postprocessing filter that compresses and decompresses the image +at several (or - in the case of @option{quality} level @code{6} - all) shifts +and average the results. + +The filter accepts the following options: + +@table @option +@item quality +Set quality. This option defines the number of levels for averaging. It accepts +an integer in the range 0-6. If set to @code{0}, the filter will have no +effect. A value of @code{6} means the higher quality. For each increment of +that value the speed drops by a factor of approximately 2. Default value is +@code{3}. + +@item qp +Force a constant quantization parameter. If not set, the filter will use the QP +from the video stream (if available). + +@item mode +Set thresholding mode. Available modes are: + +@table @samp +@item hard +Set hard thresholding (default). +@item soft +Set soft thresholding (better de-ringing effect, but likely blurrier). +@end table + +@item use_bframe_qp +Enable the use of the QP from the B-Frames if set to @code{1}. Using this +option may cause flicker since the B-Frames have often larger QP. Default is +@code{0} (not enabled). +@end table + +@anchor{subtitles} +@section subtitles + +Draw subtitles on top of input video using the libass library. + +To enable compilation of this filter you need to configure FFmpeg with +@code{--enable-libass}. This filter also requires a build with libavcodec and +libavformat to convert the passed subtitles file to ASS (Advanced Substation +Alpha) subtitles format. + +The filter accepts the following options: + +@table @option +@item filename, f +Set the filename of the subtitle file to read. It must be specified. + +@item original_size +Specify the size of the original video, the video for which the ASS file +was composed. For the syntax of this option, check the +@ref{video size syntax,,"Video size" section in the ffmpeg-utils manual,ffmpeg-utils}. +Due to a misdesign in ASS aspect ratio arithmetic, this is necessary to +correctly scale the fonts if the aspect ratio has been changed. + +@item fontsdir +Set a directory path containing fonts that can be used by the filter. +These fonts will be used in addition to whatever the font provider uses. + +@item charenc +Set subtitles input character encoding. @code{subtitles} filter only. Only +useful if not UTF-8. + +@item stream_index, si +Set subtitles stream index. @code{subtitles} filter only. + +@item force_style +Override default style or script info parameters of the subtitles. It accepts a +string containing ASS style format @code{KEY=VALUE} couples separated by ",". +@end table + +If the first key is not specified, it is assumed that the first value +specifies the @option{filename}. + +For example, to render the file @file{sub.srt} on top of the input +video, use the command: +@example +subtitles=sub.srt +@end example + +which is equivalent to: +@example +subtitles=filename=sub.srt +@end example + +To render the default subtitles stream from file @file{video.mkv}, use: +@example +subtitles=video.mkv +@end example + +To render the second subtitles stream from that file, use: +@example +subtitles=video.mkv:si=1 +@end example + +To make the subtitles stream from @file{sub.srt} appear in transparent green +@code{DejaVu Serif}, use: +@example +subtitles=sub.srt:force_style='FontName=DejaVu Serif,PrimaryColour=&HAA00FF00' +@end example + +@section super2xsai + +Scale the input by 2x and smooth using the Super2xSaI (Scale and +Interpolate) pixel art scaling algorithm. + +Useful for enlarging pixel art images without reducing sharpness. + +@section swaprect + +Swap two rectangular objects in video. + +This filter accepts the following options: + +@table @option +@item w +Set object width. + +@item h +Set object height. + +@item x1 +Set 1st rect x coordinate. + +@item y1 +Set 1st rect y coordinate. + +@item x2 +Set 2nd rect x coordinate. + +@item y2 +Set 2nd rect y coordinate. + +All expressions are evaluated once for each frame. +@end table + +The all options are expressions containing the following constants: + +@table @option +@item w +@item h +The input width and height. + +@item a +same as @var{w} / @var{h} + +@item sar +input sample aspect ratio + +@item dar +input display aspect ratio, it is the same as (@var{w} / @var{h}) * @var{sar} + +@item n +The number of the input frame, starting from 0. + +@item t +The timestamp expressed in seconds. It's NAN if the input timestamp is unknown. + +@item pos +the position in the file of the input frame, NAN if unknown +@end table + +@section swapuv +Swap U & V plane. + +@section telecine + +Apply telecine process to the video. + +This filter accepts the following options: + +@table @option +@item first_field +@table @samp +@item top, t +top field first +@item bottom, b +bottom field first +The default value is @code{top}. +@end table + +@item pattern +A string of numbers representing the pulldown pattern you wish to apply. +The default value is @code{23}. +@end table + +@example +Some typical patterns: + +NTSC output (30i): +27.5p: 32222 +24p: 23 (classic) +24p: 2332 (preferred) +20p: 33 +18p: 334 +16p: 3444 + +PAL output (25i): +27.5p: 12222 +24p: 222222222223 ("Euro pulldown") +16.67p: 33 +16p: 33333334 +@end example + +@section thumbnail +Select the most representative frame in a given sequence of consecutive frames. + +The filter accepts the following options: + +@table @option +@item n +Set the frames batch size to analyze; in a set of @var{n} frames, the filter +will pick one of them, and then handle the next batch of @var{n} frames until +the end. Default is @code{100}. +@end table + +Since the filter keeps track of the whole frames sequence, a bigger @var{n} +value will result in a higher memory usage, so a high value is not recommended. + +@subsection Examples + +@itemize +@item +Extract one picture each 50 frames: +@example +thumbnail=50 +@end example + +@item +Complete example of a thumbnail creation with @command{ffmpeg}: +@example +ffmpeg -i in.avi -vf thumbnail,scale=300:200 -frames:v 1 out.png +@end example +@end itemize + +@section tile + +Tile several successive frames together. + +The filter accepts the following options: + +@table @option + +@item layout +Set the grid size (i.e. the number of lines and columns). For the syntax of +this option, check the +@ref{video size syntax,,"Video size" section in the ffmpeg-utils manual,ffmpeg-utils}. + +@item nb_frames +Set the maximum number of frames to render in the given area. It must be less +than or equal to @var{w}x@var{h}. The default value is @code{0}, meaning all +the area will be used. + +@item margin +Set the outer border margin in pixels. + +@item padding +Set the inner border thickness (i.e. the number of pixels between frames). For +more advanced padding options (such as having different values for the edges), +refer to the pad video filter. + +@item color +Specify the color of the unused area. For the syntax of this option, check the +"Color" section in the ffmpeg-utils manual. The default value of @var{color} +is "black". +@end table + +@subsection Examples + +@itemize +@item +Produce 8x8 PNG tiles of all keyframes (@option{-skip_frame nokey}) in a movie: +@example +ffmpeg -skip_frame nokey -i file.avi -vf 'scale=128:72,tile=8x8' -an -vsync 0 keyframes%03d.png +@end example +The @option{-vsync 0} is necessary to prevent @command{ffmpeg} from +duplicating each output frame to accommodate the originally detected frame +rate. + +@item +Display @code{5} pictures in an area of @code{3x2} frames, +with @code{7} pixels between them, and @code{2} pixels of initial margin, using +mixed flat and named options: +@example +tile=3x2:nb_frames=5:padding=7:margin=2 +@end example +@end itemize + +@section tinterlace + +Perform various types of temporal field interlacing. + +Frames are counted starting from 1, so the first input frame is +considered odd. + +The filter accepts the following options: + +@table @option + +@item mode +Specify the mode of the interlacing. This option can also be specified +as a value alone. See below for a list of values for this option. + +Available values are: + +@table @samp +@item merge, 0 +Move odd frames into the upper field, even into the lower field, +generating a double height frame at half frame rate. +@example + ------> time +Input: +Frame 1 Frame 2 Frame 3 Frame 4 + +11111 22222 33333 44444 +11111 22222 33333 44444 +11111 22222 33333 44444 +11111 22222 33333 44444 + +Output: +11111 33333 +22222 44444 +11111 33333 +22222 44444 +11111 33333 +22222 44444 +11111 33333 +22222 44444 +@end example + +@item drop_odd, 1 +Only output even frames, odd frames are dropped, generating a frame with +unchanged height at half frame rate. + +@example + ------> time +Input: +Frame 1 Frame 2 Frame 3 Frame 4 + +11111 22222 33333 44444 +11111 22222 33333 44444 +11111 22222 33333 44444 +11111 22222 33333 44444 + +Output: + 22222 44444 + 22222 44444 + 22222 44444 + 22222 44444 +@end example + +@item drop_even, 2 +Only output odd frames, even frames are dropped, generating a frame with +unchanged height at half frame rate. + +@example + ------> time +Input: +Frame 1 Frame 2 Frame 3 Frame 4 + +11111 22222 33333 44444 +11111 22222 33333 44444 +11111 22222 33333 44444 +11111 22222 33333 44444 + +Output: +11111 33333 +11111 33333 +11111 33333 +11111 33333 +@end example + +@item pad, 3 +Expand each frame to full height, but pad alternate lines with black, +generating a frame with double height at the same input frame rate. + +@example + ------> time +Input: +Frame 1 Frame 2 Frame 3 Frame 4 + +11111 22222 33333 44444 +11111 22222 33333 44444 +11111 22222 33333 44444 +11111 22222 33333 44444 + +Output: +11111 ..... 33333 ..... +..... 22222 ..... 44444 +11111 ..... 33333 ..... +..... 22222 ..... 44444 +11111 ..... 33333 ..... +..... 22222 ..... 44444 +11111 ..... 33333 ..... +..... 22222 ..... 44444 +@end example + + +@item interleave_top, 4 +Interleave the upper field from odd frames with the lower field from +even frames, generating a frame with unchanged height at half frame rate. + +@example + ------> time +Input: +Frame 1 Frame 2 Frame 3 Frame 4 + +11111<- 22222 33333<- 44444 +11111 22222<- 33333 44444<- +11111<- 22222 33333<- 44444 +11111 22222<- 33333 44444<- + +Output: +11111 33333 +22222 44444 +11111 33333 +22222 44444 +@end example + + +@item interleave_bottom, 5 +Interleave the lower field from odd frames with the upper field from +even frames, generating a frame with unchanged height at half frame rate. + +@example + ------> time +Input: +Frame 1 Frame 2 Frame 3 Frame 4 + +11111 22222<- 33333 44444<- +11111<- 22222 33333<- 44444 +11111 22222<- 33333 44444<- +11111<- 22222 33333<- 44444 + +Output: +22222 44444 +11111 33333 +22222 44444 +11111 33333 +@end example + + +@item interlacex2, 6 +Double frame rate with unchanged height. Frames are inserted each +containing the second temporal field from the previous input frame and +the first temporal field from the next input frame. This mode relies on +the top_field_first flag. Useful for interlaced video displays with no +field synchronisation. + +@example + ------> time +Input: +Frame 1 Frame 2 Frame 3 Frame 4 + +11111 22222 33333 44444 + 11111 22222 33333 44444 +11111 22222 33333 44444 + 11111 22222 33333 44444 + +Output: +11111 22222 22222 33333 33333 44444 44444 + 11111 11111 22222 22222 33333 33333 44444 +11111 22222 22222 33333 33333 44444 44444 + 11111 11111 22222 22222 33333 33333 44444 +@end example + +@item mergex2, 7 +Move odd frames into the upper field, even into the lower field, +generating a double height frame at same frame rate. +@example + ------> time +Input: +Frame 1 Frame 2 Frame 3 Frame 4 + +11111 22222 33333 44444 +11111 22222 33333 44444 +11111 22222 33333 44444 +11111 22222 33333 44444 + +Output: +11111 33333 33333 55555 +22222 22222 44444 44444 +11111 33333 33333 55555 +22222 22222 44444 44444 +11111 33333 33333 55555 +22222 22222 44444 44444 +11111 33333 33333 55555 +22222 22222 44444 44444 +@end example + +@end table + +Numeric values are deprecated but are accepted for backward +compatibility reasons. + +Default mode is @code{merge}. + +@item flags +Specify flags influencing the filter process. + +Available value for @var{flags} is: + +@table @option +@item low_pass_filter, vlfp +Enable vertical low-pass filtering in the filter. +Vertical low-pass filtering is required when creating an interlaced +destination from a progressive source which contains high-frequency +vertical detail. Filtering will reduce interlace 'twitter' and Moire +patterning. + +Vertical low-pass filtering can only be enabled for @option{mode} +@var{interleave_top} and @var{interleave_bottom}. + +@end table +@end table + @section transpose Transpose rows with columns in the input video and optionally flip it. @@ -2616,14 +11985,11 @@ It accepts the following parameters: @table @option @item dir -The direction of the transpose. - -@end table - -The direction can assume the following values: +Specify the transposition direction. +Can assume the following values: @table @samp -@item cclock_flip +@item 0, 4, cclock_flip Rotate by 90 degrees counterclockwise and vertically flip (default), that is: @example L.R L.l @@ -2631,7 +11997,7 @@ L.R L.l l.r R.r @end example -@item clock +@item 1, 5, clock Rotate by 90 degrees clockwise, that is: @example L.R l.L @@ -2639,7 +12005,7 @@ L.R l.L l.r r.R @end example -@item cclock +@item 2, 6, cclock Rotate by 90 degrees counterclockwise, that is: @example L.R R.r @@ -2647,7 +12013,7 @@ L.R R.r l.r L.l @end example -@item clock_flip +@item 3, 7, clock_flip Rotate by 90 degrees clockwise and vertically flip, that is: @example L.R r.R @@ -2656,17 +12022,50 @@ l.r l.L @end example @end table +For values between 4-7, the transposition is only done if the input +video geometry is portrait and not landscape. These values are +deprecated, the @code{passthrough} option should be used instead. + +Numerical values are deprecated, and should be dropped in favor of +symbolic constants. + +@item passthrough +Do not apply the transposition if the input geometry matches the one +specified by the specified value. It accepts the following values: +@table @samp +@item none +Always apply transposition. +@item portrait +Preserve portrait geometry (when @var{height} >= @var{width}). +@item landscape +Preserve landscape geometry (when @var{width} >= @var{height}). +@end table + +Default value is @code{none}. +@end table + +For example to rotate by 90 degrees clockwise and preserve portrait +layout: +@example +transpose=dir=1:passthrough=portrait +@end example + +The command above can also be specified as: +@example +transpose=1:portrait +@end example + @section trim Trim the input so that the output contains one continuous subpart of the input. It accepts the following parameters: @table @option @item start -The timestamp (in seconds) of the start of the kept section. The frame with the +Specify the time of the start of the kept section, i.e. the frame with the timestamp @var{start} will be the first frame in the output. @item end -The timestamp (in seconds) of the first frame that will be dropped. The frame +Specify the time of the first frame that will be dropped, i.e. the frame immediately preceding the one with the timestamp @var{end} will be the last frame in the output. @@ -2688,6 +12087,11 @@ The number of the first frame that should be passed to the output. The number of the first frame that should be dropped. @end table +@option{start}, @option{end}, and @option{duration} are expressed as time +duration specifications; see +@ref{time duration syntax,,the Time duration section in the ffmpeg-utils(1) manual,ffmpeg-utils} +for the accepted syntax. + Note that the first two sets of the start/end options and the @option{duration} option look at the frame timestamp, while the _frame variants simply count the frames that pass through the filter. Also note that this filter does not modify @@ -2707,16 +12111,19 @@ Examples: @item Drop everything except the second minute of input: @example -avconv -i INPUT -vf trim=60:120 +ffmpeg -i INPUT -vf trim=60:120 @end example @item Keep only the first second: @example -avconv -i INPUT -vf trim=duration=1 +ffmpeg -i INPUT -vf trim=duration=1 @end example @end itemize + + +@anchor{unsharp} @section unsharp Sharpen or blur the input video. @@ -2724,56 +12131,676 @@ Sharpen or blur the input video. It accepts the following parameters: @table @option +@item luma_msize_x, lx +Set the luma matrix horizontal size. It must be an odd integer between +3 and 63. The default value is 5. + +@item luma_msize_y, ly +Set the luma matrix vertical size. It must be an odd integer between 3 +and 63. The default value is 5. + +@item luma_amount, la +Set the luma effect strength. It must be a floating point number, reasonable +values lay between -1.5 and 1.5. -@item luma_msize_x -Set the luma matrix horizontal size. It must be an integer between 3 -and 13. The default value is 5. +Negative values will blur the input video, while positive values will +sharpen it, a value of zero will disable the effect. -@item luma_msize_y -Set the luma matrix vertical size. It must be an integer between 3 -and 13. The default value is 5. +Default value is 1.0. -@item luma_amount -Set the luma effect strength. It must be a floating point number between -2.0 -and 5.0. The default value is 1.0. +@item chroma_msize_x, cx +Set the chroma matrix horizontal size. It must be an odd integer +between 3 and 63. The default value is 5. -@item chroma_msize_x -Set the chroma matrix horizontal size. It must be an integer between 3 -and 13. The default value is 5. +@item chroma_msize_y, cy +Set the chroma matrix vertical size. It must be an odd integer +between 3 and 63. The default value is 5. -@item chroma_msize_y -Set the chroma matrix vertical size. It must be an integer between 3 -and 13. The default value is 5. +@item chroma_amount, ca +Set the chroma effect strength. It must be a floating point number, reasonable +values lay between -1.5 and 1.5. -@item chroma_amount -Set the chroma effect strength. It must be a floating point number between -2.0 -and 5.0. The default value is 0.0. +Negative values will blur the input video, while positive values will +sharpen it, a value of zero will disable the effect. + +Default value is 0.0. + +@item opencl +If set to 1, specify using OpenCL capabilities, only available if +FFmpeg was configured with @code{--enable-opencl}. Default value is 0. @end table -Negative values for the amount will blur the input video, while positive -values will sharpen. All parameters are optional and default to the -equivalent of the string '5:5:1.0:5:5:0.0'. +All parameters are optional and default to the equivalent of the +string '5:5:1.0:5:5:0.0'. +@subsection Examples + +@itemize +@item +Apply strong luma sharpen effect: @example -# Strong luma sharpen effect parameters unsharp=luma_msize_x=7:luma_msize_y=7:luma_amount=2.5 +@end example -# A strong blur of both luma and chroma parameters +@item +Apply a strong blur of both luma and chroma parameters: +@example unsharp=7:7:-2:7:7:-2 +@end example +@end itemize + +@section uspp + +Apply ultra slow/simple postprocessing filter that compresses and decompresses +the image at several (or - in the case of @option{quality} level @code{8} - all) +shifts and average the results. + +The way this differs from the behavior of spp is that uspp actually encodes & +decodes each case with libavcodec Snow, whereas spp uses a simplified intra only 8x8 +DCT similar to MJPEG. + +The filter accepts the following options: + +@table @option +@item quality +Set quality. This option defines the number of levels for averaging. It accepts +an integer in the range 0-8. If set to @code{0}, the filter will have no +effect. A value of @code{8} means the higher quality. For each increment of +that value the speed drops by a factor of approximately 2. Default value is +@code{3}. + +@item qp +Force a constant quantization parameter. If not set, the filter will use the QP +from the video stream (if available). +@end table + +@section vectorscope + +Display 2 color component values in the two dimensional graph (which is called +a vectorscope). + +This filter accepts the following options: + +@table @option +@item mode, m +Set vectorscope mode. + +It accepts the following values: +@table @samp +@item gray +Gray values are displayed on graph, higher brightness means more pixels have +same component color value on location in graph. This is the default mode. + +@item color +Gray values are displayed on graph. Surrounding pixels values which are not +present in video frame are drawn in gradient of 2 color components which are +set by option @code{x} and @code{y}. + +@item color2 +Actual color components values present in video frame are displayed on graph. + +@item color3 +Similar as color2 but higher frequency of same values @code{x} and @code{y} +on graph increases value of another color component, which is luminance by +default values of @code{x} and @code{y}. + +@item color4 +Actual colors present in video frame are displayed on graph. If two different +colors map to same position on graph then color with higher value of component +not present in graph is picked. +@end table + +@item x +Set which color component will be represented on X-axis. Default is @code{1}. + +@item y +Set which color component will be represented on Y-axis. Default is @code{2}. + +@item intensity, i +Set intensity, used by modes: gray, color and color3 for increasing brightness +of color component which represents frequency of (X, Y) location in graph. + +@item envelope, e +@table @samp +@item none +No envelope, this is default. + +@item instant +Instant envelope, even darkest single pixel will be clearly highlighted. + +@item peak +Hold maximum and minimum values presented in graph over time. This way you +can still spot out of range values without constantly looking at vectorscope. + +@item peak+instant +Peak and instant envelope combined together. +@end table +@end table + +@anchor{vidstabdetect} +@section vidstabdetect + +Analyze video stabilization/deshaking. Perform pass 1 of 2, see +@ref{vidstabtransform} for pass 2. + +This filter generates a file with relative translation and rotation +transform information about subsequent frames, which is then used by +the @ref{vidstabtransform} filter. + +To enable compilation of this filter you need to configure FFmpeg with +@code{--enable-libvidstab}. + +This filter accepts the following options: + +@table @option +@item result +Set the path to the file used to write the transforms information. +Default value is @file{transforms.trf}. + +@item shakiness +Set how shaky the video is and how quick the camera is. It accepts an +integer in the range 1-10, a value of 1 means little shakiness, a +value of 10 means strong shakiness. Default value is 5. + +@item accuracy +Set the accuracy of the detection process. It must be a value in the +range 1-15. A value of 1 means low accuracy, a value of 15 means high +accuracy. Default value is 15. + +@item stepsize +Set stepsize of the search process. The region around minimum is +scanned with 1 pixel resolution. Default value is 6. + +@item mincontrast +Set minimum contrast. Below this value a local measurement field is +discarded. Must be a floating point value in the range 0-1. Default +value is 0.3. + +@item tripod +Set reference frame number for tripod mode. + +If enabled, the motion of the frames is compared to a reference frame +in the filtered stream, identified by the specified number. The idea +is to compensate all movements in a more-or-less static scene and keep +the camera view absolutely still. + +If set to 0, it is disabled. The frames are counted starting from 1. + +@item show +Show fields and transforms in the resulting frames. It accepts an +integer in the range 0-2. Default value is 0, which disables any +visualization. +@end table + +@subsection Examples + +@itemize +@item +Use default values: +@example +vidstabdetect +@end example + +@item +Analyze strongly shaky movie and put the results in file +@file{mytransforms.trf}: +@example +vidstabdetect=shakiness=10:accuracy=15:result="mytransforms.trf" +@end example + +@item +Visualize the result of internal transformations in the resulting +video: +@example +vidstabdetect=show=1 +@end example + +@item +Analyze a video with medium shakiness using @command{ffmpeg}: +@example +ffmpeg -i input -vf vidstabdetect=shakiness=5:show=1 dummy.avi +@end example +@end itemize + +@anchor{vidstabtransform} +@section vidstabtransform + +Video stabilization/deshaking: pass 2 of 2, +see @ref{vidstabdetect} for pass 1. + +Read a file with transform information for each frame and +apply/compensate them. Together with the @ref{vidstabdetect} +filter this can be used to deshake videos. See also +@url{http://public.hronopik.de/vid.stab}. It is important to also use +the @ref{unsharp} filter, see below. -# Use the default values with @command{avconv} -./avconv -i in.avi -vf "unsharp" out.mp4 +To enable compilation of this filter you need to configure FFmpeg with +@code{--enable-libvidstab}. + +@subsection Options + +@table @option +@item input +Set path to the file used to read the transforms. Default value is +@file{transforms.trf}. + +@item smoothing +Set the number of frames (value*2 + 1) used for lowpass filtering the +camera movements. Default value is 10. + +For example a number of 10 means that 21 frames are used (10 in the +past and 10 in the future) to smoothen the motion in the video. A +larger value leads to a smoother video, but limits the acceleration of +the camera (pan/tilt movements). 0 is a special case where a static +camera is simulated. + +@item optalgo +Set the camera path optimization algorithm. + +Accepted values are: +@table @samp +@item gauss +gaussian kernel low-pass filter on camera motion (default) +@item avg +averaging on transformations +@end table + +@item maxshift +Set maximal number of pixels to translate frames. Default value is -1, +meaning no limit. + +@item maxangle +Set maximal angle in radians (degree*PI/180) to rotate frames. Default +value is -1, meaning no limit. + +@item crop +Specify how to deal with borders that may be visible due to movement +compensation. + +Available values are: +@table @samp +@item keep +keep image information from previous frame (default) +@item black +fill the border black +@end table + +@item invert +Invert transforms if set to 1. Default value is 0. + +@item relative +Consider transforms as relative to previous frame if set to 1, +absolute if set to 0. Default value is 0. + +@item zoom +Set percentage to zoom. A positive value will result in a zoom-in +effect, a negative value in a zoom-out effect. Default value is 0 (no +zoom). + +@item optzoom +Set optimal zooming to avoid borders. + +Accepted values are: +@table @samp +@item 0 +disabled +@item 1 +optimal static zoom value is determined (only very strong movements +will lead to visible borders) (default) +@item 2 +optimal adaptive zoom value is determined (no borders will be +visible), see @option{zoomspeed} +@end table + +Note that the value given at zoom is added to the one calculated here. + +@item zoomspeed +Set percent to zoom maximally each frame (enabled when +@option{optzoom} is set to 2). Range is from 0 to 5, default value is +0.25. + +@item interpol +Specify type of interpolation. + +Available values are: +@table @samp +@item no +no interpolation +@item linear +linear only horizontal +@item bilinear +linear in both directions (default) +@item bicubic +cubic in both directions (slow) +@end table + +@item tripod +Enable virtual tripod mode if set to 1, which is equivalent to +@code{relative=0:smoothing=0}. Default value is 0. + +Use also @code{tripod} option of @ref{vidstabdetect}. + +@item debug +Increase log verbosity if set to 1. Also the detected global motions +are written to the temporary file @file{global_motions.trf}. Default +value is 0. +@end table + +@subsection Examples + +@itemize +@item +Use @command{ffmpeg} for a typical stabilization with default values: +@example +ffmpeg -i inp.mpeg -vf vidstabtransform,unsharp=5:5:0.8:3:3:0.4 inp_stabilized.mpeg +@end example + +Note the use of the @ref{unsharp} filter which is always recommended. + +@item +Zoom in a bit more and load transform data from a given file: +@example +vidstabtransform=zoom=5:input="mytransforms.trf" +@end example + +@item +Smoothen the video even more: +@example +vidstabtransform=smoothing=30 @end example +@end itemize @section vflip Flip the input video vertically. +For example, to vertically flip a video with @command{ffmpeg}: @example -./avconv -i in.avi -vf "vflip" out.avi +ffmpeg -i in.avi -vf "vflip" out.avi @end example +@anchor{vignette} +@section vignette + +Make or reverse a natural vignetting effect. + +The filter accepts the following options: + +@table @option +@item angle, a +Set lens angle expression as a number of radians. + +The value is clipped in the @code{[0,PI/2]} range. + +Default value: @code{"PI/5"} + +@item x0 +@item y0 +Set center coordinates expressions. Respectively @code{"w/2"} and @code{"h/2"} +by default. + +@item mode +Set forward/backward mode. + +Available modes are: +@table @samp +@item forward +The larger the distance from the central point, the darker the image becomes. + +@item backward +The larger the distance from the central point, the brighter the image becomes. +This can be used to reverse a vignette effect, though there is no automatic +detection to extract the lens @option{angle} and other settings (yet). It can +also be used to create a burning effect. +@end table + +Default value is @samp{forward}. + +@item eval +Set evaluation mode for the expressions (@option{angle}, @option{x0}, @option{y0}). + +It accepts the following values: +@table @samp +@item init +Evaluate expressions only once during the filter initialization. + +@item frame +Evaluate expressions for each incoming frame. This is way slower than the +@samp{init} mode since it requires all the scalers to be re-computed, but it +allows advanced dynamic expressions. +@end table + +Default value is @samp{init}. + +@item dither +Set dithering to reduce the circular banding effects. Default is @code{1} +(enabled). + +@item aspect +Set vignette aspect. This setting allows one to adjust the shape of the vignette. +Setting this value to the SAR of the input will make a rectangular vignetting +following the dimensions of the video. + +Default is @code{1/1}. +@end table + +@subsection Expressions + +The @option{alpha}, @option{x0} and @option{y0} expressions can contain the +following parameters. + +@table @option +@item w +@item h +input width and height + +@item n +the number of input frame, starting from 0 + +@item pts +the PTS (Presentation TimeStamp) time of the filtered video frame, expressed in +@var{TB} units, NAN if undefined + +@item r +frame rate of the input video, NAN if the input frame rate is unknown + +@item t +the PTS (Presentation TimeStamp) of the filtered video frame, +expressed in seconds, NAN if undefined + +@item tb +time base of the input video +@end table + + +@subsection Examples + +@itemize +@item +Apply simple strong vignetting effect: +@example +vignette=PI/4 +@end example + +@item +Make a flickering vignetting: +@example +vignette='PI/4+random(1)*PI/50':eval=frame +@end example + +@end itemize + +@section vstack +Stack input videos vertically. + +All streams must be of same pixel format and of same width. + +Note that this filter is faster than using @ref{overlay} and @ref{pad} filter +to create same output. + +The filter accept the following option: + +@table @option +@item inputs +Set number of input streams. Default is 2. + +@item shortest +If set to 1, force the output to terminate when the shortest input +terminates. Default value is 0. +@end table + +@section w3fdif + +Deinterlace the input video ("w3fdif" stands for "Weston 3 Field +Deinterlacing Filter"). + +Based on the process described by Martin Weston for BBC R&D, and +implemented based on the de-interlace algorithm written by Jim +Easterbrook for BBC R&D, the Weston 3 field deinterlacing filter +uses filter coefficients calculated by BBC R&D. + +There are two sets of filter coefficients, so called "simple": +and "complex". Which set of filter coefficients is used can +be set by passing an optional parameter: + +@table @option +@item filter +Set the interlacing filter coefficients. Accepts one of the following values: + +@table @samp +@item simple +Simple filter coefficient set. +@item complex +More-complex filter coefficient set. +@end table +Default value is @samp{complex}. + +@item deint +Specify which frames to deinterlace. Accept one of the following values: + +@table @samp +@item all +Deinterlace all frames, +@item interlaced +Only deinterlace frames marked as interlaced. +@end table + +Default value is @samp{all}. +@end table + +@section waveform +Video waveform monitor. + +The waveform monitor plots color component intensity. By default luminance +only. Each column of the waveform corresponds to a column of pixels in the +source video. + +It accepts the following options: + +@table @option +@item mode, m +Can be either @code{row}, or @code{column}. Default is @code{column}. +In row mode, the graph on the left side represents color component value 0 and +the right side represents value = 255. In column mode, the top side represents +color component value = 0 and bottom side represents value = 255. + +@item intensity, i +Set intensity. Smaller values are useful to find out how many values of the same +luminance are distributed across input rows/columns. +Default value is @code{0.04}. Allowed range is [0, 1]. + +@item mirror, r +Set mirroring mode. @code{0} means unmirrored, @code{1} means mirrored. +In mirrored mode, higher values will be represented on the left +side for @code{row} mode and at the top for @code{column} mode. Default is +@code{1} (mirrored). + +@item display, d +Set display mode. +It accepts the following values: +@table @samp +@item overlay +Presents information identical to that in the @code{parade}, except +that the graphs representing color components are superimposed directly +over one another. + +This display mode makes it easier to spot relative differences or similarities +in overlapping areas of the color components that are supposed to be identical, +such as neutral whites, grays, or blacks. + +@item parade +Display separate graph for the color components side by side in +@code{row} mode or one below the other in @code{column} mode. + +Using this display mode makes it easy to spot color casts in the highlights +and shadows of an image, by comparing the contours of the top and the bottom +graphs of each waveform. Since whites, grays, and blacks are characterized +by exactly equal amounts of red, green, and blue, neutral areas of the picture +should display three waveforms of roughly equal width/height. If not, the +correction is easy to perform by making level adjustments the three waveforms. +@end table +Default is @code{parade}. + +@item components, c +Set which color components to display. Default is 1, which means only luminance +or red color component if input is in RGB colorspace. If is set for example to +7 it will display all 3 (if) available color components. + +@item envelope, e +@table @samp +@item none +No envelope, this is default. + +@item instant +Instant envelope, minimum and maximum values presented in graph will be easily +visible even with small @code{step} value. + +@item peak +Hold minimum and maximum values presented in graph across time. This way you +can still spot out of range values without constantly looking at waveforms. + +@item peak+instant +Peak and instant envelope combined together. +@end table + +@item filter, f +@table @samp +@item lowpass +No filtering, this is default. + +@item flat +Luma and chroma combined together. + +@item aflat +Similar as above, but shows difference between blue and red chroma. + +@item chroma +Displays only chroma. + +@item achroma +Similar as above, but shows difference between blue and red chroma. + +@item color +Displays actual color value on waveform. +@end table +@end table + +@section xbr +Apply the xBR high-quality magnification filter which is designed for pixel +art. It follows a set of edge-detection rules, see +@url{http://www.libretro.com/forums/viewtopic.php?f=6&t=134}. + +It accepts the following option: + +@table @option +@item n +Set the scaling dimension: @code{2} for @code{2xBR}, @code{3} for +@code{3xBR} and @code{4} for @code{4xBR}. +Default is @code{3}. +@end table + +@anchor{yadif} @section yadif Deinterlace the input video ("yadif" means "yet another deinterlacing @@ -2781,54 +12808,379 @@ filter"). It accepts the following parameters: + @table @option @item mode The interlacing mode to adopt. It accepts one of the following values: @table @option -@item 0 +@item 0, send_frame Output one frame for each frame. -@item 1 +@item 1, send_field Output one frame for each field. -@item 2 -Like 0, but it skips the spatial interlacing check. -@item 3 -Like 1, but it skips the spatial interlacing check. +@item 2, send_frame_nospatial +Like @code{send_frame}, but it skips the spatial interlacing check. +@item 3, send_field_nospatial +Like @code{send_field}, but it skips the spatial interlacing check. @end table -The default value is 0. +The default value is @code{send_frame}. @item parity The picture field parity assumed for the input interlaced video. It accepts one of the following values: @table @option -@item 0 +@item 0, tff Assume the top field is first. -@item 1 +@item 1, bff Assume the bottom field is first. -@item -1 +@item -1, auto Enable automatic detection of field parity. @end table -The default value is -1. +The default value is @code{auto}. If the interlacing is unknown or the decoder does not export this information, top field first will be assumed. -@item auto -Whether the deinterlacer should trust the interlaced flag and only deinterlace -frames marked as interlaced. +@item deint +Specify which frames to deinterlace. Accept one of the following +values: @table @option -@item 0 +@item 0, all Deinterlace all frames. -@item 1 +@item 1, interlaced Only deinterlace frames marked as interlaced. @end table -The default value is 0. +The default value is @code{all}. +@end table + +@section zoompan + +Apply Zoom & Pan effect. + +This filter accepts the following options: +@table @option +@item zoom, z +Set the zoom expression. Default is 1. + +@item x +@item y +Set the x and y expression. Default is 0. + +@item d +Set the duration expression in number of frames. +This sets for how many number of frames effect will last for +single input image. + +@item s +Set the output image size, default is 'hd720'. + +@item fps +Set the output frame rate, default is '25'. +@end table + +Each expression can contain the following constants: + +@table @option +@item in_w, iw +Input width. + +@item in_h, ih +Input height. + +@item out_w, ow +Output width. + +@item out_h, oh +Output height. + +@item in +Input frame count. + +@item on +Output frame count. + +@item x +@item y +Last calculated 'x' and 'y' position from 'x' and 'y' expression +for current input frame. + +@item px +@item py +'x' and 'y' of last output frame of previous input frame or 0 when there was +not yet such frame (first input frame). + +@item zoom +Last calculated zoom from 'z' expression for current input frame. + +@item pzoom +Last calculated zoom of last output frame of previous input frame. + +@item duration +Number of output frames for current input frame. Calculated from 'd' expression +for each input frame. + +@item pduration +number of output frames created for previous input frame + +@item a +Rational number: input width / input height + +@item sar +sample aspect ratio + +@item dar +display aspect ratio + +@end table + +@subsection Examples + +@itemize +@item +Zoom-in up to 1.5 and pan at same time to some spot near center of picture: +@example +zoompan=z='min(zoom+0.0015,1.5)':d=700:x='if(gte(zoom,1.5),x,x+1/a)':y='if(gte(zoom,1.5),y,y+1)':s=640x360 +@end example + +@item +Zoom-in up to 1.5 and pan always at center of picture: +@example +zoompan=z='min(zoom+0.0015,1.5)':d=700:x='iw/2-(iw/zoom/2)':y='ih/2-(ih/zoom/2)' +@end example +@end itemize + +@section zscale +Scale (resize) the input video, using the z.lib library: +https://github.com/sekrit-twc/zimg. + +The zscale filter forces the output display aspect ratio to be the same +as the input, by changing the output sample aspect ratio. + +If the input image format is different from the format requested by +the next filter, the zscale filter will convert the input to the +requested format. + +@subsection Options +The filter accepts the following options. + +@table @option +@item width, w +@item height, h +Set the output video dimension expression. Default value is the input +dimension. + +If the @var{width} or @var{w} is 0, the input width is used for the output. +If the @var{height} or @var{h} is 0, the input height is used for the output. + +If one of the values is -1, the zscale filter will use a value that +maintains the aspect ratio of the input image, calculated from the +other specified dimension. If both of them are -1, the input size is +used + +If one of the values is -n with n > 1, the zscale filter will also use a value +that maintains the aspect ratio of the input image, calculated from the other +specified dimension. After that it will, however, make sure that the calculated +dimension is divisible by n and adjust the value if necessary. + +See below for the list of accepted constants for use in the dimension +expression. + +@item size, s +Set the video size. For the syntax of this option, check the +@ref{video size syntax,,"Video size" section in the ffmpeg-utils manual,ffmpeg-utils}. + +@item dither, d +Set the dither type. + +Possible values are: +@table @var +@item none +@item ordered +@item random +@item error_diffusion +@end table + +Default is none. + +@item filter, f +Set the resize filter type. + +Possible values are: +@table @var +@item point +@item bilinear +@item bicubic +@item spline16 +@item spline36 +@item lanczos +@end table + +Default is bilinear. + +@item range, r +Set the color range. + +Possible values are: +@table @var +@item input +@item limited +@item full +@end table + +Default is same as input. + +@item primaries, p +Set the color primaries. + +Possible values are: +@table @var +@item input +@item 709 +@item unspecified +@item 170m +@item 240m +@item 2020 +@end table + +Default is same as input. + +@item transfer, t +Set the transfer characteristics. + +Possible values are: +@table @var +@item input +@item 709 +@item unspecified +@item 601 +@item linear +@item 2020_10 +@item 2020_12 +@end table + +Default is same as input. + +@item matrix, m +Set the colorspace matrix. + +Possible value are: +@table @var +@item input +@item 709 +@item unspecified +@item 470bg +@item 170m +@item 2020_ncl +@item 2020_cl +@end table + +Default is same as input. + +@item rangein, rin +Set the input color range. + +Possible values are: +@table @var +@item input +@item limited +@item full +@end table + +Default is same as input. + +@item primariesin, pin +Set the input color primaries. + +Possible values are: +@table @var +@item input +@item 709 +@item unspecified +@item 170m +@item 240m +@item 2020 +@end table + +Default is same as input. + +@item transferin, tin +Set the input transfer characteristics. + +Possible values are: +@table @var +@item input +@item 709 +@item unspecified +@item 601 +@item linear +@item 2020_10 +@item 2020_12 +@end table + +Default is same as input. + +@item matrixin, min +Set the input colorspace matrix. + +Possible value are: +@table @var +@item input +@item 709 +@item unspecified +@item 470bg +@item 170m +@item 2020_ncl +@item 2020_cl +@end table +@end table + +The values of the @option{w} and @option{h} options are expressions +containing the following constants: + +@table @var +@item in_w +@item in_h +The input width and height + +@item iw +@item ih +These are the same as @var{in_w} and @var{in_h}. + +@item out_w +@item out_h +The output (scaled) width and height + +@item ow +@item oh +These are the same as @var{out_w} and @var{out_h} + +@item a +The same as @var{iw} / @var{ih} + +@item sar +input sample aspect ratio + +@item dar +The input display aspect ratio. Calculated from @code{(iw / ih) * sar}. + +@item hsub +@item vsub +horizontal and vertical input chroma subsample values. For example for the +pixel format "yuv422p" @var{hsub} is 2 and @var{vsub} is 1. + +@item ohsub +@item ovsub +horizontal and vertical output chroma subsample values. For example for the +pixel format "yuv422p" @var{hsub} is 2 and @var{vsub} is 1. +@end table + +@table @option @end table @c man end VIDEO FILTERS @@ -2849,6 +13201,11 @@ It accepts the following parameters: @table @option +@item video_size +Specify the size (width and height) of the buffered video frames. For the +syntax of this option, check the +@ref{video size syntax,,"Video size" section in the ffmpeg-utils manual,ffmpeg-utils}. + @item width The input video width. @@ -2856,14 +13213,23 @@ The input video width. The input video height. @item pix_fmt -The name of the input video pixel format. +A string representing the pixel format of the buffered video frames. +It may be a number corresponding to a pixel format, or a pixel format +name. @item time_base -The time base used for input timestamps. +Specify the timebase assumed by the timestamps of the buffered frames. -@item sar +@item frame_rate +Specify the frame rate expected for the video stream. + +@item pixel_aspect, sar The sample (pixel) aspect ratio of the input video. +@item sws_param +Specify the optional parameters to be used for the scale filter which +is automatically inserted when an input change is detected in the +input size or format. @end table For example: @@ -2874,131 +13240,272 @@ buffer=width=320:height=240:pix_fmt=yuv410p:time_base=1/24:sar=1 will instruct the source to accept video frames with size 320x240 and with format "yuv410p", assuming 1/24 as the timestamps timebase and square pixels (1:1 sample aspect ratio). +Since the pixel format with name "yuv410p" corresponds to the number 6 +(check the enum AVPixelFormat definition in @file{libavutil/pixfmt.h}), +this example corresponds to: +@example +buffer=size=320x240:pixfmt=6:time_base=1/24:pixel_aspect=1/1 +@end example -@section color +Alternatively, the options can be specified as a flat string, but this +syntax is deprecated: -Provide an uniformly colored input. +@var{width}:@var{height}:@var{pix_fmt}:@var{time_base.num}:@var{time_base.den}:@var{pixel_aspect.num}:@var{pixel_aspect.den}[:@var{sws_param}] -It accepts the following parameters: +@section cellauto + +Create a pattern generated by an elementary cellular automaton. + +The initial state of the cellular automaton can be defined through the +@option{filename}, and @option{pattern} options. If such options are +not specified an initial state is created randomly. + +At each new frame a new row in the video is filled with the result of +the cellular automaton next generation. The behavior when the whole +frame is filled is defined by the @option{scroll} option. + +This source accepts the following options: @table @option +@item filename, f +Read the initial cellular automaton state, i.e. the starting row, from +the specified file. +In the file, each non-whitespace character is considered an alive +cell, a newline will terminate the row, and further characters in the +file will be ignored. -@item color -Specify the color of the source. It can be the name of a color (case -insensitive match) or a 0xRRGGBB[AA] sequence, possibly followed by an -alpha specifier. The default value is "black". +@item pattern, p +Read the initial cellular automaton state, i.e. the starting row, from +the specified string. -@item size -Specify the size of the sourced video, it may be a string of the form -@var{width}x@var{height}, or the name of a size abbreviation. The -default value is "320x240". +Each non-whitespace character in the string is considered an alive +cell, a newline will terminate the row, and further characters in the +string will be ignored. -@item framerate -Specify the frame rate of the sourced video, as the number of frames -generated per second. It has to be a string in the format -@var{frame_rate_num}/@var{frame_rate_den}, an integer number, a floating point -number or a valid video frame rate abbreviation. The default value is -"25". +@item rate, r +Set the video rate, that is the number of frames generated per second. +Default is 25. + +@item random_fill_ratio, ratio +Set the random fill ratio for the initial cellular automaton row. It +is a floating point number value ranging from 0 to 1, defaults to +1/PHI. + +This option is ignored when a file or a pattern is specified. + +@item random_seed, seed +Set the seed for filling randomly the initial row, must be an integer +included between 0 and UINT32_MAX. If not specified, or if explicitly +set to -1, the filter will try to use a good random seed on a best +effort basis. + +@item rule +Set the cellular automaton rule, it is a number ranging from 0 to 255. +Default value is 110. +@item size, s +Set the size of the output video. For the syntax of this option, check the +@ref{video size syntax,,"Video size" section in the ffmpeg-utils manual,ffmpeg-utils}. + +If @option{filename} or @option{pattern} is specified, the size is set +by default to the width of the specified initial state row, and the +height is set to @var{width} * PHI. + +If @option{size} is set, it must contain the width of the specified +pattern string, and the specified pattern will be centered in the +larger row. + +If a filename or a pattern string is not specified, the size value +defaults to "320x518" (used for a randomly generated initial state). + +@item scroll +If set to 1, scroll the output upward when all the rows in the output +have been already filled. If set to 0, the new generated row will be +written over the top row just after the bottom row is filled. +Defaults to 1. + +@item start_full, full +If set to 1, completely fill the output with generated rows before +outputting the first frame. +This is the default behavior, for disabling set the value to 0. + +@item stitch +If set to 1, stitch the left and right row edges together. +This is the default behavior, for disabling set the value to 0. @end table -The following graph description will generate a red source -with an opacity of 0.2, with size "qcif" and a frame rate of 10 -frames per second, which will be overlayed over the source connected -to the pad with identifier "in": +@subsection Examples +@itemize +@item +Read the initial state from @file{pattern}, and specify an output of +size 200x400. @example -"color=red@@0.2:qcif:10 [color]; [in][color] overlay [out]" +cellauto=f=pattern:s=200x400 @end example -@section movie +@item +Generate a random initial row with a width of 200 cells, with a fill +ratio of 2/3: +@example +cellauto=ratio=2/3:s=200x200 +@end example + +@item +Create a pattern generated by rule 18 starting by a single alive cell +centered on an initial row with width 100: +@example +cellauto=p=@@:s=100x400:full=0:rule=18 +@end example -Read a video stream from a movie container. +@item +Specify a more elaborated initial pattern: +@example +cellauto=p='@@@@ @@ @@@@':s=100x400:full=0:rule=18 +@end example -Note that this source is a hack that bypasses the standard input path. It can be -useful in applications that do not support arbitrary filter graphs, but its use -is discouraged in those that do. It should never be used with -@command{avconv}; the @option{-filter_complex} option fully replaces it. +@end itemize -It accepts the following parameters: +@section mandelbrot + +Generate a Mandelbrot set fractal, and progressively zoom towards the +point specified with @var{start_x} and @var{start_y}. + +This source accepts the following options: @table @option -@item filename -The name of the resource to read (not necessarily a file; it can also be a -device or a stream accessed through some protocol). +@item end_pts +Set the terminal pts value. Default value is 400. -@item format_name, f -Specifies the format assumed for the movie to read, and can be either -the name of a container or an input device. If not specified, the -format is guessed from @var{movie_name} or by probing. +@item end_scale +Set the terminal scale value. +Must be a floating point value. Default value is 0.3. -@item seek_point, sp -Specifies the seek point in seconds. The frames will be output -starting from this seek point. The parameter is evaluated with -@code{av_strtod}, so the numerical value may be suffixed by an IS -postfix. The default value is "0". +@item inner +Set the inner coloring mode, that is the algorithm used to draw the +Mandelbrot fractal internal region. -@item stream_index, si -Specifies the index of the video stream to read. If the value is -1, -the most suitable video stream will be automatically selected. The default -value is "-1". +It shall assume one of the following values: +@table @option +@item black +Set black mode. +@item convergence +Show time until convergence. +@item mincol +Set color based on point closest to the origin of the iterations. +@item period +Set period mode. +@end table + +Default value is @var{mincol}. +@item bailout +Set the bailout value. Default value is 10.0. + +@item maxiter +Set the maximum of iterations performed by the rendering +algorithm. Default value is 7189. + +@item outer +Set outer coloring mode. +It shall assume one of following values: +@table @option +@item iteration_count +Set iteration cound mode. +@item normalized_iteration_count +set normalized iteration count mode. @end table +Default value is @var{normalized_iteration_count}. -It allows overlaying a second video on top of the main input of -a filtergraph, as shown in this graph: -@example -input -----------> deltapts0 --> overlay --> output - ^ - | -movie --> scale--> deltapts1 -------+ -@end example +@item rate, r +Set frame rate, expressed as number of frames per second. Default +value is "25". -Some examples: -@example -# Skip 3.2 seconds from the start of the AVI file in.avi, and overlay it -# on top of the input labelled "in" -movie=in.avi:seek_point=3.2, scale=180:-1, setpts=PTS-STARTPTS [movie]; -[in] setpts=PTS-STARTPTS, [movie] overlay=16:16 [out] +@item size, s +Set frame size. For the syntax of this option, check the "Video +size" section in the ffmpeg-utils manual. Default value is "640x480". -# Read from a video4linux2 device, and overlay it on top of the input -# labelled "in" -movie=/dev/video0:f=video4linux2, scale=180:-1, setpts=PTS-STARTPTS [movie]; -[in] setpts=PTS-STARTPTS, [movie] overlay=16:16 [out] +@item start_scale +Set the initial scale value. Default value is 3.0. -@end example +@item start_x +Set the initial x position. Must be a floating point value between +-100 and 100. Default value is -0.743643887037158704752191506114774. + +@item start_y +Set the initial y position. Must be a floating point value between +-100 and 100. Default value is -0.131825904205311970493132056385139. +@end table + +@section mptestsrc + +Generate various test patterns, as generated by the MPlayer test filter. + +The size of the generated video is fixed, and is 256x256. +This source is useful in particular for testing encoding features. + +This source accepts the following options: + +@table @option + +@item rate, r +Specify the frame rate of the sourced video, as the number of frames +generated per second. It has to be a string in the format +@var{frame_rate_num}/@var{frame_rate_den}, an integer number, a floating point +number or a valid video frame rate abbreviation. The default value is +"25". + +@item duration, d +Set the duration of the sourced video. See +@ref{time duration syntax,,the Time duration section in the ffmpeg-utils(1) manual,ffmpeg-utils} +for the accepted syntax. -@section nullsrc +If not specified, or the expressed duration is negative, the video is +supposed to be generated forever. -Null video source: never return images. It is mainly useful as a -template and to be employed in analysis / debugging tools. +@item test, t -It accepts a string of the form -@var{width}:@var{height}:@var{timebase} as an optional parameter. +Set the number or the name of the test to perform. Supported tests are: +@table @option +@item dc_luma +@item dc_chroma +@item freq_luma +@item freq_chroma +@item amp_luma +@item amp_chroma +@item cbp +@item mv +@item ring1 +@item ring2 +@item all -@var{width} and @var{height} specify the size of the configured -source. The default values of @var{width} and @var{height} are -respectively 352 and 288 (corresponding to the CIF size format). +@end table -@var{timebase} specifies an arithmetic expression representing a -timebase. The expression can contain the constants "PI", "E", "PHI", and -"AVTB" (the default timebase), and defaults to the value "AVTB". +Default value is "all", which will cycle through the list of all tests. +@end table + +Some examples: +@example +mptestsrc=t=dc_luma +@end example + +will generate a "dc_luma" test pattern. @section frei0r_src Provide a frei0r source. To enable compilation of this filter you need to install the frei0r -header and configure Libav with --enable-frei0r. +header and configure FFmpeg with @code{--enable-frei0r}. This source accepts the following parameters: @table @option @item size -The size of the video to generate. It may be a string of the form -@var{width}x@var{height} or a frame size abbreviation. +The size of the video to generate. For the syntax of this option, check the +@ref{video size syntax,,"Video size" section in the ffmpeg-utils manual,ffmpeg-utils}. @item framerate The framerate of the generated video. It may be a string of the form @@ -3014,19 +13521,175 @@ A '|'-separated list of parameters to pass to the frei0r source. @end table -An example: +For example, to generate a frei0r partik0l source with size 200x200 +and frame rate 10 which is overlaid on the overlay filter main input: @example -# Generate a frei0r partik0l source with size 200x200 and framerate 10 -# which is overlayed on the overlay filter main input frei0r_src=size=200x200:framerate=10:filter_name=partik0l:filter_params=1234 [overlay]; [in][overlay] overlay @end example -@section rgbtestsrc, testsrc +@section life + +Generate a life pattern. + +This source is based on a generalization of John Conway's life game. + +The sourced input represents a life grid, each pixel represents a cell +which can be in one of two possible states, alive or dead. Every cell +interacts with its eight neighbours, which are the cells that are +horizontally, vertically, or diagonally adjacent. + +At each interaction the grid evolves according to the adopted rule, +which specifies the number of neighbor alive cells which will make a +cell stay alive or born. The @option{rule} option allows one to specify +the rule to adopt. + +This source accepts the following options: + +@table @option +@item filename, f +Set the file from which to read the initial grid state. In the file, +each non-whitespace character is considered an alive cell, and newline +is used to delimit the end of each row. + +If this option is not specified, the initial grid is generated +randomly. + +@item rate, r +Set the video rate, that is the number of frames generated per second. +Default is 25. + +@item random_fill_ratio, ratio +Set the random fill ratio for the initial random grid. It is a +floating point number value ranging from 0 to 1, defaults to 1/PHI. +It is ignored when a file is specified. + +@item random_seed, seed +Set the seed for filling the initial random grid, must be an integer +included between 0 and UINT32_MAX. If not specified, or if explicitly +set to -1, the filter will try to use a good random seed on a best +effort basis. + +@item rule +Set the life rule. + +A rule can be specified with a code of the kind "S@var{NS}/B@var{NB}", +where @var{NS} and @var{NB} are sequences of numbers in the range 0-8, +@var{NS} specifies the number of alive neighbor cells which make a +live cell stay alive, and @var{NB} the number of alive neighbor cells +which make a dead cell to become alive (i.e. to "born"). +"s" and "b" can be used in place of "S" and "B", respectively. + +Alternatively a rule can be specified by an 18-bits integer. The 9 +high order bits are used to encode the next cell state if it is alive +for each number of neighbor alive cells, the low order bits specify +the rule for "borning" new cells. Higher order bits encode for an +higher number of neighbor cells. +For example the number 6153 = @code{(12<<9)+9} specifies a stay alive +rule of 12 and a born rule of 9, which corresponds to "S23/B03". + +Default value is "S23/B3", which is the original Conway's game of life +rule, and will keep a cell alive if it has 2 or 3 neighbor alive +cells, and will born a new cell if there are three alive cells around +a dead cell. + +@item size, s +Set the size of the output video. For the syntax of this option, check the +@ref{video size syntax,,"Video size" section in the ffmpeg-utils manual,ffmpeg-utils}. + +If @option{filename} is specified, the size is set by default to the +same size of the input file. If @option{size} is set, it must contain +the size specified in the input file, and the initial grid defined in +that file is centered in the larger resulting area. + +If a filename is not specified, the size value defaults to "320x240" +(used for a randomly generated initial grid). + +@item stitch +If set to 1, stitch the left and right grid edges together, and the +top and bottom edges also. Defaults to 1. + +@item mold +Set cell mold speed. If set, a dead cell will go from @option{death_color} to +@option{mold_color} with a step of @option{mold}. @option{mold} can have a +value from 0 to 255. + +@item life_color +Set the color of living (or new born) cells. + +@item death_color +Set the color of dead cells. If @option{mold} is set, this is the first color +used to represent a dead cell. + +@item mold_color +Set mold color, for definitely dead and moldy cells. + +For the syntax of these 3 color options, check the "Color" section in the +ffmpeg-utils manual. +@end table + +@subsection Examples + +@itemize +@item +Read a grid from @file{pattern}, and center it on a grid of size +300x300 pixels: +@example +life=f=pattern:s=300x300 +@end example + +@item +Generate a random grid of size 200x200, with a fill ratio of 2/3: +@example +life=ratio=2/3:s=200x200 +@end example + +@item +Specify a custom rule for evolving a randomly generated grid: +@example +life=rule=S14/B34 +@end example + +@item +Full example with slow death effect (mold) using @command{ffplay}: +@example +ffplay -f lavfi life=s=300x200:mold=10:r=60:ratio=0.1:death_color=#C83232:life_color=#00ff00,scale=1200:800:flags=16 +@end example +@end itemize + +@anchor{allrgb} +@anchor{allyuv} +@anchor{color} +@anchor{haldclutsrc} +@anchor{nullsrc} +@anchor{rgbtestsrc} +@anchor{smptebars} +@anchor{smptehdbars} +@anchor{testsrc} +@section allrgb, allyuv, color, haldclutsrc, nullsrc, rgbtestsrc, smptebars, smptehdbars, testsrc + +The @code{allrgb} source returns frames of size 4096x4096 of all rgb colors. + +The @code{allyuv} source returns frames of size 4096x4096 of all yuv colors. + +The @code{color} source provides an uniformly colored input. + +The @code{haldclutsrc} source provides an identity Hald CLUT. See also +@ref{haldclut} filter. + +The @code{nullsrc} source returns unprocessed video frames. It is +mainly useful to be employed in analysis / debugging tools, or as the +source for filters which ignore the input data. The @code{rgbtestsrc} source generates an RGB test pattern useful for detecting RGB vs BGR issues. You should see a red, green and blue stripe from top to bottom. +The @code{smptebars} source generates a color bars pattern, based on +the SMPTE Engineering Guideline EG 1-1990. + +The @code{smptehdbars} source generates a color bars pattern, based on +the SMPTE RP 219-2002. + The @code{testsrc} source generates a test video pattern, showing a color pattern, a scrolling gradient and a timestamp. This is mainly intended for testing purposes. @@ -3035,10 +13698,23 @@ The sources accept the following parameters: @table @option +@item color, c +Specify the color of the source, only available in the @code{color} +source. For the syntax of this option, check the "Color" section in the +ffmpeg-utils manual. + +@item level +Specify the level of the Hald CLUT, only available in the @code{haldclutsrc} +source. A level of @code{N} generates a picture of @code{N*N*N} by @code{N*N*N} +pixels to be used as identity matrix for 3D lookup tables. Each component is +coded on a @code{1/(N*N)} scale. + @item size, s -Specify the size of the sourced video, it may be a string of the form -@var{width}x@var{height}, or the name of a size abbreviation. The -default value is "320x240". +Specify the size of the sourced video. For the syntax of this option, check the +@ref{video size syntax,,"Video size" section in the ffmpeg-utils manual,ffmpeg-utils}. +The default value is @code{320x240}. + +This option is not available with the @code{haldclutsrc} filter. @item rate, r Specify the frame rate of the sourced video, as the number of frames @@ -3050,16 +13726,21 @@ number or a valid video frame rate abbreviation. The default value is @item sar Set the sample aspect ratio of the sourced video. -@item duration -Set the video duration of the sourced video. The accepted syntax is: -@example -[-]HH[:MM[:SS[.m...]]] -[-]S+[.m...] -@end example -Also see the the @code{av_parse_time()} function. +@item duration, d +Set the duration of the sourced video. See +@ref{time duration syntax,,the Time duration section in the ffmpeg-utils(1) manual,ffmpeg-utils} +for the accepted syntax. If not specified, or the expressed duration is negative, the video is supposed to be generated forever. + +@item decimals, n +Set the number of decimals to show in the timestamp, only available in the +@code{testsrc} source. + +The displayed timestamp value will correspond to the original +timestamp value multiplied by the power of 10 of the specified +value. Default value is 0. @end table For example the following: @@ -3068,7 +13749,31 @@ testsrc=duration=5.3:size=qcif:rate=10 @end example will generate a video with a duration of 5.3 seconds, with size -176x144 and a framerate of 10 frames per second. +176x144 and a frame rate of 10 frames per second. + +The following graph description will generate a red source +with an opacity of 0.2, with size "qcif" and a frame rate of 10 +frames per second. +@example +color=c=red@@0.2:s=qcif:r=10 +@end example + +If the input content is to be ignored, @code{nullsrc} can be used. The +following command generates noise in the luminance plane by employing +the @code{geq} filter: +@example +nullsrc=s=256x256, geq=random(1)*255:128:128 +@end example + +@subsection Commands + +The @code{color} source supports the following commands: + +@table @option +@item c, color +Set the color of the created image. Accepts the same syntax of the +corresponding @option{color} option. +@end table @c man end VIDEO SOURCES @@ -3082,8 +13787,13 @@ Below is a description of the currently available video sinks. Buffer video frames, and make them available to the end of the filter graph. -This sink is intended for programmatic use through the interface defined in -@file{libavfilter/buffersink.h}. +This sink is mainly intended for programmatic use, in particular +through the interface defined in @file{libavfilter/buffersink.h} +or the options system. + +It accepts a pointer to an AVBufferSinkContext structure, which +defines the incoming buffers' formats, to be passed as the opaque +parameter to @code{avfilter_init_filter} for initialization. @section nullsink @@ -3092,3 +13802,2091 @@ mainly useful as a template and for use in analysis / debugging tools. @c man end VIDEO SINKS + +@chapter Multimedia Filters +@c man begin MULTIMEDIA FILTERS + +Below is a description of the currently available multimedia filters. + +@section ahistogram + +Convert input audio to a video output, displaying the volume histogram. + +The filter accepts the following options: + +@table @option +@item dmode +Specify how histogram is calculated. + +It accepts the following values: +@table @samp +@item single +Use single histogram for all channels. +@item separate +Use separate histogram for each channel. +@end table +Default is @code{single}. + +@item rate, r +Set frame rate, expressed as number of frames per second. Default +value is "25". + +@item size, s +Specify the video size for the output. For the syntax of this option, check the +@ref{video size syntax,,"Video size" section in the ffmpeg-utils manual,ffmpeg-utils}. +Default value is @code{hd720}. + +@item scale +Set display scale. + +It accepts the following values: +@table @samp +@item log +logarithmic +@item sqrt +square root +@item cbrt +cubic root +@item lin +linear +@item rlog +reverse logarithmic +@end table +Default is @code{log}. + +@item ascale +Set amplitude scale. + +It accepts the following values: +@table @samp +@item log +logarithmic +@item lin +linear +@end table +Default is @code{log}. + +@item acount +Set how much frames to accumulate in histogram. +Defauls is 1. Setting this to -1 accumulates all frames. + +@item rheight +Set histogram ratio of window height. + +@item slide +Set sonogram sliding. + +It accepts the following values: +@table @samp +@item replace +replace old rows with new ones. +@item scroll +scroll from top to bottom. +@end table +Default is @code{replace}. +@end table + +@section aphasemeter + +Convert input audio to a video output, displaying the audio phase. + +The filter accepts the following options: + +@table @option +@item rate, r +Set the output frame rate. Default value is @code{25}. + +@item size, s +Set the video size for the output. For the syntax of this option, check the +@ref{video size syntax,,"Video size" section in the ffmpeg-utils manual,ffmpeg-utils}. +Default value is @code{800x400}. + +@item rc +@item gc +@item bc +Specify the red, green, blue contrast. Default values are @code{2}, +@code{7} and @code{1}. +Allowed range is @code{[0, 255]}. + +@item mpc +Set color which will be used for drawing median phase. If color is +@code{none} which is default, no median phase value will be drawn. +@end table + +The filter also exports the frame metadata @code{lavfi.aphasemeter.phase} which +represents mean phase of current audio frame. Value is in range @code{[-1, 1]}. +The @code{-1} means left and right channels are completely out of phase and +@code{1} means channels are in phase. + +@section avectorscope + +Convert input audio to a video output, representing the audio vector +scope. + +The filter is used to measure the difference between channels of stereo +audio stream. A monoaural signal, consisting of identical left and right +signal, results in straight vertical line. Any stereo separation is visible +as a deviation from this line, creating a Lissajous figure. +If the straight (or deviation from it) but horizontal line appears this +indicates that the left and right channels are out of phase. + +The filter accepts the following options: + +@table @option +@item mode, m +Set the vectorscope mode. + +Available values are: +@table @samp +@item lissajous +Lissajous rotated by 45 degrees. + +@item lissajous_xy +Same as above but not rotated. + +@item polar +Shape resembling half of circle. +@end table + +Default value is @samp{lissajous}. + +@item size, s +Set the video size for the output. For the syntax of this option, check the +@ref{video size syntax,,"Video size" section in the ffmpeg-utils manual,ffmpeg-utils}. +Default value is @code{400x400}. + +@item rate, r +Set the output frame rate. Default value is @code{25}. + +@item rc +@item gc +@item bc +@item ac +Specify the red, green, blue and alpha contrast. Default values are @code{40}, +@code{160}, @code{80} and @code{255}. +Allowed range is @code{[0, 255]}. + +@item rf +@item gf +@item bf +@item af +Specify the red, green, blue and alpha fade. Default values are @code{15}, +@code{10}, @code{5} and @code{5}. +Allowed range is @code{[0, 255]}. + +@item zoom +Set the zoom factor. Default value is @code{1}. Allowed range is @code{[1, 10]}. + +@item draw +Set the vectorscope drawing mode. + +Available values are: +@table @samp +@item dot +Draw dot for each sample. + +@item line +Draw line between previous and current sample. +@end table + +Default value is @samp{dot}. +@end table + +@subsection Examples + +@itemize +@item +Complete example using @command{ffplay}: +@example +ffplay -f lavfi 'amovie=input.mp3, asplit [a][out1]; + [a] avectorscope=zoom=1.3:rc=2:gc=200:bc=10:rf=1:gf=8:bf=7 [out0]' +@end example +@end itemize + +@section concat + +Concatenate audio and video streams, joining them together one after the +other. + +The filter works on segments of synchronized video and audio streams. All +segments must have the same number of streams of each type, and that will +also be the number of streams at output. + +The filter accepts the following options: + +@table @option + +@item n +Set the number of segments. Default is 2. + +@item v +Set the number of output video streams, that is also the number of video +streams in each segment. Default is 1. + +@item a +Set the number of output audio streams, that is also the number of audio +streams in each segment. Default is 0. + +@item unsafe +Activate unsafe mode: do not fail if segments have a different format. + +@end table + +The filter has @var{v}+@var{a} outputs: first @var{v} video outputs, then +@var{a} audio outputs. + +There are @var{n}x(@var{v}+@var{a}) inputs: first the inputs for the first +segment, in the same order as the outputs, then the inputs for the second +segment, etc. + +Related streams do not always have exactly the same duration, for various +reasons including codec frame size or sloppy authoring. For that reason, +related synchronized streams (e.g. a video and its audio track) should be +concatenated at once. The concat filter will use the duration of the longest +stream in each segment (except the last one), and if necessary pad shorter +audio streams with silence. + +For this filter to work correctly, all segments must start at timestamp 0. + +All corresponding streams must have the same parameters in all segments; the +filtering system will automatically select a common pixel format for video +streams, and a common sample format, sample rate and channel layout for +audio streams, but other settings, such as resolution, must be converted +explicitly by the user. + +Different frame rates are acceptable but will result in variable frame rate +at output; be sure to configure the output file to handle it. + +@subsection Examples + +@itemize +@item +Concatenate an opening, an episode and an ending, all in bilingual version +(video in stream 0, audio in streams 1 and 2): +@example +ffmpeg -i opening.mkv -i episode.mkv -i ending.mkv -filter_complex \ + '[0:0] [0:1] [0:2] [1:0] [1:1] [1:2] [2:0] [2:1] [2:2] + concat=n=3:v=1:a=2 [v] [a1] [a2]' \ + -map '[v]' -map '[a1]' -map '[a2]' output.mkv +@end example + +@item +Concatenate two parts, handling audio and video separately, using the +(a)movie sources, and adjusting the resolution: +@example +movie=part1.mp4, scale=512:288 [v1] ; amovie=part1.mp4 [a1] ; +movie=part2.mp4, scale=512:288 [v2] ; amovie=part2.mp4 [a2] ; +[v1] [v2] concat [outv] ; [a1] [a2] concat=v=0:a=1 [outa] +@end example +Note that a desync will happen at the stitch if the audio and video streams +do not have exactly the same duration in the first file. + +@end itemize + +@anchor{ebur128} +@section ebur128 + +EBU R128 scanner filter. This filter takes an audio stream as input and outputs +it unchanged. By default, it logs a message at a frequency of 10Hz with the +Momentary loudness (identified by @code{M}), Short-term loudness (@code{S}), +Integrated loudness (@code{I}) and Loudness Range (@code{LRA}). + +The filter also has a video output (see the @var{video} option) with a real +time graph to observe the loudness evolution. The graphic contains the logged +message mentioned above, so it is not printed anymore when this option is set, +unless the verbose logging is set. The main graphing area contains the +short-term loudness (3 seconds of analysis), and the gauge on the right is for +the momentary loudness (400 milliseconds). + +More information about the Loudness Recommendation EBU R128 on +@url{http://tech.ebu.ch/loudness}. + +The filter accepts the following options: + +@table @option + +@item video +Activate the video output. The audio stream is passed unchanged whether this +option is set or no. The video stream will be the first output stream if +activated. Default is @code{0}. + +@item size +Set the video size. This option is for video only. For the syntax of this +option, check the +@ref{video size syntax,,"Video size" section in the ffmpeg-utils manual,ffmpeg-utils}. +Default and minimum resolution is @code{640x480}. + +@item meter +Set the EBU scale meter. Default is @code{9}. Common values are @code{9} and +@code{18}, respectively for EBU scale meter +9 and EBU scale meter +18. Any +other integer value between this range is allowed. + +@item metadata +Set metadata injection. If set to @code{1}, the audio input will be segmented +into 100ms output frames, each of them containing various loudness information +in metadata. All the metadata keys are prefixed with @code{lavfi.r128.}. + +Default is @code{0}. + +@item framelog +Force the frame logging level. + +Available values are: +@table @samp +@item info +information logging level +@item verbose +verbose logging level +@end table + +By default, the logging level is set to @var{info}. If the @option{video} or +the @option{metadata} options are set, it switches to @var{verbose}. + +@item peak +Set peak mode(s). + +Available modes can be cumulated (the option is a @code{flag} type). Possible +values are: +@table @samp +@item none +Disable any peak mode (default). +@item sample +Enable sample-peak mode. + +Simple peak mode looking for the higher sample value. It logs a message +for sample-peak (identified by @code{SPK}). +@item true +Enable true-peak mode. + +If enabled, the peak lookup is done on an over-sampled version of the input +stream for better peak accuracy. It logs a message for true-peak. +(identified by @code{TPK}) and true-peak per frame (identified by @code{FTPK}). +This mode requires a build with @code{libswresample}. +@end table + +@item dualmono +Treat mono input files as "dual mono". If a mono file is intended for playback +on a stereo system, its EBU R128 measurement will be perceptually incorrect. +If set to @code{true}, this option will compensate for this effect. +Multi-channel input files are not affected by this option. + +@item panlaw +Set a specific pan law to be used for the measurement of dual mono files. +This parameter is optional, and has a default value of -3.01dB. +@end table + +@subsection Examples + +@itemize +@item +Real-time graph using @command{ffplay}, with a EBU scale meter +18: +@example +ffplay -f lavfi -i "amovie=input.mp3,ebur128=video=1:meter=18 [out0][out1]" +@end example + +@item +Run an analysis with @command{ffmpeg}: +@example +ffmpeg -nostats -i input.mp3 -filter_complex ebur128 -f null - +@end example +@end itemize + +@section interleave, ainterleave + +Temporally interleave frames from several inputs. + +@code{interleave} works with video inputs, @code{ainterleave} with audio. + +These filters read frames from several inputs and send the oldest +queued frame to the output. + +Input streams must have a well defined, monotonically increasing frame +timestamp values. + +In order to submit one frame to output, these filters need to enqueue +at least one frame for each input, so they cannot work in case one +input is not yet terminated and will not receive incoming frames. + +For example consider the case when one input is a @code{select} filter +which always drop input frames. The @code{interleave} filter will keep +reading from that input, but it will never be able to send new frames +to output until the input will send an end-of-stream signal. + +Also, depending on inputs synchronization, the filters will drop +frames in case one input receives more frames than the other ones, and +the queue is already filled. + +These filters accept the following options: + +@table @option +@item nb_inputs, n +Set the number of different inputs, it is 2 by default. +@end table + +@subsection Examples + +@itemize +@item +Interleave frames belonging to different streams using @command{ffmpeg}: +@example +ffmpeg -i bambi.avi -i pr0n.mkv -filter_complex "[0:v][1:v] interleave" out.avi +@end example + +@item +Add flickering blur effect: +@example +select='if(gt(random(0), 0.2), 1, 2)':n=2 [tmp], boxblur=2:2, [tmp] interleave +@end example +@end itemize + +@section perms, aperms + +Set read/write permissions for the output frames. + +These filters are mainly aimed at developers to test direct path in the +following filter in the filtergraph. + +The filters accept the following options: + +@table @option +@item mode +Select the permissions mode. + +It accepts the following values: +@table @samp +@item none +Do nothing. This is the default. +@item ro +Set all the output frames read-only. +@item rw +Set all the output frames directly writable. +@item toggle +Make the frame read-only if writable, and writable if read-only. +@item random +Set each output frame read-only or writable randomly. +@end table + +@item seed +Set the seed for the @var{random} mode, must be an integer included between +@code{0} and @code{UINT32_MAX}. If not specified, or if explicitly set to +@code{-1}, the filter will try to use a good random seed on a best effort +basis. +@end table + +Note: in case of auto-inserted filter between the permission filter and the +following one, the permission might not be received as expected in that +following filter. Inserting a @ref{format} or @ref{aformat} filter before the +perms/aperms filter can avoid this problem. + +@section realtime, arealtime + +Slow down filtering to match real time approximatively. + +These filters will pause the filtering for a variable amount of time to +match the output rate with the input timestamps. +They are similar to the @option{re} option to @code{ffmpeg}. + +They accept the following options: + +@table @option +@item limit +Time limit for the pauses. Any pause longer than that will be considered +a timestamp discontinuity and reset the timer. Default is 2 seconds. +@end table + +@section select, aselect + +Select frames to pass in output. + +This filter accepts the following options: + +@table @option + +@item expr, e +Set expression, which is evaluated for each input frame. + +If the expression is evaluated to zero, the frame is discarded. + +If the evaluation result is negative or NaN, the frame is sent to the +first output; otherwise it is sent to the output with index +@code{ceil(val)-1}, assuming that the input index starts from 0. + +For example a value of @code{1.2} corresponds to the output with index +@code{ceil(1.2)-1 = 2-1 = 1}, that is the second output. + +@item outputs, n +Set the number of outputs. The output to which to send the selected +frame is based on the result of the evaluation. Default value is 1. +@end table + +The expression can contain the following constants: + +@table @option +@item n +The (sequential) number of the filtered frame, starting from 0. + +@item selected_n +The (sequential) number of the selected frame, starting from 0. + +@item prev_selected_n +The sequential number of the last selected frame. It's NAN if undefined. + +@item TB +The timebase of the input timestamps. + +@item pts +The PTS (Presentation TimeStamp) of the filtered video frame, +expressed in @var{TB} units. It's NAN if undefined. + +@item t +The PTS of the filtered video frame, +expressed in seconds. It's NAN if undefined. + +@item prev_pts +The PTS of the previously filtered video frame. It's NAN if undefined. + +@item prev_selected_pts +The PTS of the last previously filtered video frame. It's NAN if undefined. + +@item prev_selected_t +The PTS of the last previously selected video frame. It's NAN if undefined. + +@item start_pts +The PTS of the first video frame in the video. It's NAN if undefined. + +@item start_t +The time of the first video frame in the video. It's NAN if undefined. + +@item pict_type @emph{(video only)} +The type of the filtered frame. It can assume one of the following +values: +@table @option +@item I +@item P +@item B +@item S +@item SI +@item SP +@item BI +@end table + +@item interlace_type @emph{(video only)} +The frame interlace type. It can assume one of the following values: +@table @option +@item PROGRESSIVE +The frame is progressive (not interlaced). +@item TOPFIRST +The frame is top-field-first. +@item BOTTOMFIRST +The frame is bottom-field-first. +@end table + +@item consumed_sample_n @emph{(audio only)} +the number of selected samples before the current frame + +@item samples_n @emph{(audio only)} +the number of samples in the current frame + +@item sample_rate @emph{(audio only)} +the input sample rate + +@item key +This is 1 if the filtered frame is a key-frame, 0 otherwise. + +@item pos +the position in the file of the filtered frame, -1 if the information +is not available (e.g. for synthetic video) + +@item scene @emph{(video only)} +value between 0 and 1 to indicate a new scene; a low value reflects a low +probability for the current frame to introduce a new scene, while a higher +value means the current frame is more likely to be one (see the example below) + +@item concatdec_select +The concat demuxer can select only part of a concat input file by setting an +inpoint and an outpoint, but the output packets may not be entirely contained +in the selected interval. By using this variable, it is possible to skip frames +generated by the concat demuxer which are not exactly contained in the selected +interval. + +This works by comparing the frame pts against the @var{lavf.concat.start_time} +and the @var{lavf.concat.duration} packet metadata values which are also +present in the decoded frames. + +The @var{concatdec_select} variable is -1 if the frame pts is at least +start_time and either the duration metadata is missing or the frame pts is less +than start_time + duration, 0 otherwise, and NaN if the start_time metadata is +missing. + +That basically means that an input frame is selected if its pts is within the +interval set by the concat demuxer. + +@end table + +The default value of the select expression is "1". + +@subsection Examples + +@itemize +@item +Select all frames in input: +@example +select +@end example + +The example above is the same as: +@example +select=1 +@end example + +@item +Skip all frames: +@example +select=0 +@end example + +@item +Select only I-frames: +@example +select='eq(pict_type\,I)' +@end example + +@item +Select one frame every 100: +@example +select='not(mod(n\,100))' +@end example + +@item +Select only frames contained in the 10-20 time interval: +@example +select=between(t\,10\,20) +@end example + +@item +Select only I frames contained in the 10-20 time interval: +@example +select=between(t\,10\,20)*eq(pict_type\,I) +@end example + +@item +Select frames with a minimum distance of 10 seconds: +@example +select='isnan(prev_selected_t)+gte(t-prev_selected_t\,10)' +@end example + +@item +Use aselect to select only audio frames with samples number > 100: +@example +aselect='gt(samples_n\,100)' +@end example + +@item +Create a mosaic of the first scenes: +@example +ffmpeg -i video.avi -vf select='gt(scene\,0.4)',scale=160:120,tile -frames:v 1 preview.png +@end example + +Comparing @var{scene} against a value between 0.3 and 0.5 is generally a sane +choice. + +@item +Send even and odd frames to separate outputs, and compose them: +@example +select=n=2:e='mod(n, 2)+1' [odd][even]; [odd] pad=h=2*ih [tmp]; [tmp][even] overlay=y=h +@end example + +@item +Select useful frames from an ffconcat file which is using inpoints and +outpoints but where the source files are not intra frame only. +@example +ffmpeg -copyts -vsync 0 -segment_time_metadata 1 -i input.ffconcat -vf select=concatdec_select -af aselect=concatdec_select output.avi +@end example +@end itemize + +@section sendcmd, asendcmd + +Send commands to filters in the filtergraph. + +These filters read commands to be sent to other filters in the +filtergraph. + +@code{sendcmd} must be inserted between two video filters, +@code{asendcmd} must be inserted between two audio filters, but apart +from that they act the same way. + +The specification of commands can be provided in the filter arguments +with the @var{commands} option, or in a file specified by the +@var{filename} option. + +These filters accept the following options: +@table @option +@item commands, c +Set the commands to be read and sent to the other filters. +@item filename, f +Set the filename of the commands to be read and sent to the other +filters. +@end table + +@subsection Commands syntax + +A commands description consists of a sequence of interval +specifications, comprising a list of commands to be executed when a +particular event related to that interval occurs. The occurring event +is typically the current frame time entering or leaving a given time +interval. + +An interval is specified by the following syntax: +@example +@var{START}[-@var{END}] @var{COMMANDS}; +@end example + +The time interval is specified by the @var{START} and @var{END} times. +@var{END} is optional and defaults to the maximum time. + +The current frame time is considered within the specified interval if +it is included in the interval [@var{START}, @var{END}), that is when +the time is greater or equal to @var{START} and is lesser than +@var{END}. + +@var{COMMANDS} consists of a sequence of one or more command +specifications, separated by ",", relating to that interval. The +syntax of a command specification is given by: +@example +[@var{FLAGS}] @var{TARGET} @var{COMMAND} @var{ARG} +@end example + +@var{FLAGS} is optional and specifies the type of events relating to +the time interval which enable sending the specified command, and must +be a non-null sequence of identifier flags separated by "+" or "|" and +enclosed between "[" and "]". + +The following flags are recognized: +@table @option +@item enter +The command is sent when the current frame timestamp enters the +specified interval. In other words, the command is sent when the +previous frame timestamp was not in the given interval, and the +current is. + +@item leave +The command is sent when the current frame timestamp leaves the +specified interval. In other words, the command is sent when the +previous frame timestamp was in the given interval, and the +current is not. +@end table + +If @var{FLAGS} is not specified, a default value of @code{[enter]} is +assumed. + +@var{TARGET} specifies the target of the command, usually the name of +the filter class or a specific filter instance name. + +@var{COMMAND} specifies the name of the command for the target filter. + +@var{ARG} is optional and specifies the optional list of argument for +the given @var{COMMAND}. + +Between one interval specification and another, whitespaces, or +sequences of characters starting with @code{#} until the end of line, +are ignored and can be used to annotate comments. + +A simplified BNF description of the commands specification syntax +follows: +@example +@var{COMMAND_FLAG} ::= "enter" | "leave" +@var{COMMAND_FLAGS} ::= @var{COMMAND_FLAG} [(+|"|")@var{COMMAND_FLAG}] +@var{COMMAND} ::= ["[" @var{COMMAND_FLAGS} "]"] @var{TARGET} @var{COMMAND} [@var{ARG}] +@var{COMMANDS} ::= @var{COMMAND} [,@var{COMMANDS}] +@var{INTERVAL} ::= @var{START}[-@var{END}] @var{COMMANDS} +@var{INTERVALS} ::= @var{INTERVAL}[;@var{INTERVALS}] +@end example + +@subsection Examples + +@itemize +@item +Specify audio tempo change at second 4: +@example +asendcmd=c='4.0 atempo tempo 1.5',atempo +@end example + +@item +Specify a list of drawtext and hue commands in a file. +@example +# show text in the interval 5-10 +5.0-10.0 [enter] drawtext reinit 'fontfile=FreeSerif.ttf:text=hello world', + [leave] drawtext reinit 'fontfile=FreeSerif.ttf:text='; + +# desaturate the image in the interval 15-20 +15.0-20.0 [enter] hue s 0, + [enter] drawtext reinit 'fontfile=FreeSerif.ttf:text=nocolor', + [leave] hue s 1, + [leave] drawtext reinit 'fontfile=FreeSerif.ttf:text=color'; + +# apply an exponential saturation fade-out effect, starting from time 25 +25 [enter] hue s exp(25-t) +@end example + +A filtergraph allowing to read and process the above command list +stored in a file @file{test.cmd}, can be specified with: +@example +sendcmd=f=test.cmd,drawtext=fontfile=FreeSerif.ttf:text='',hue +@end example +@end itemize + +@anchor{setpts} +@section setpts, asetpts + +Change the PTS (presentation timestamp) of the input frames. + +@code{setpts} works on video frames, @code{asetpts} on audio frames. + +This filter accepts the following options: + +@table @option + +@item expr +The expression which is evaluated for each frame to construct its timestamp. + +@end table + +The expression is evaluated through the eval API and can contain the following +constants: + +@table @option +@item FRAME_RATE +frame rate, only defined for constant frame-rate video + +@item PTS +The presentation timestamp in input + +@item N +The count of the input frame for video or the number of consumed samples, +not including the current frame for audio, starting from 0. + +@item NB_CONSUMED_SAMPLES +The number of consumed samples, not including the current frame (only +audio) + +@item NB_SAMPLES, S +The number of samples in the current frame (only audio) + +@item SAMPLE_RATE, SR +The audio sample rate. + +@item STARTPTS +The PTS of the first frame. + +@item STARTT +the time in seconds of the first frame + +@item INTERLACED +State whether the current frame is interlaced. + +@item T +the time in seconds of the current frame + +@item POS +original position in the file of the frame, or undefined if undefined +for the current frame + +@item PREV_INPTS +The previous input PTS. + +@item PREV_INT +previous input time in seconds + +@item PREV_OUTPTS +The previous output PTS. + +@item PREV_OUTT +previous output time in seconds + +@item RTCTIME +The wallclock (RTC) time in microseconds. This is deprecated, use time(0) +instead. + +@item RTCSTART +The wallclock (RTC) time at the start of the movie in microseconds. + +@item TB +The timebase of the input timestamps. + +@end table + +@subsection Examples + +@itemize +@item +Start counting PTS from zero +@example +setpts=PTS-STARTPTS +@end example + +@item +Apply fast motion effect: +@example +setpts=0.5*PTS +@end example + +@item +Apply slow motion effect: +@example +setpts=2.0*PTS +@end example + +@item +Set fixed rate of 25 frames per second: +@example +setpts=N/(25*TB) +@end example + +@item +Set fixed rate 25 fps with some jitter: +@example +setpts='1/(25*TB) * (N + 0.05 * sin(N*2*PI/25))' +@end example + +@item +Apply an offset of 10 seconds to the input PTS: +@example +setpts=PTS+10/TB +@end example + +@item +Generate timestamps from a "live source" and rebase onto the current timebase: +@example +setpts='(RTCTIME - RTCSTART) / (TB * 1000000)' +@end example + +@item +Generate timestamps by counting samples: +@example +asetpts=N/SR/TB +@end example + +@end itemize + +@section settb, asettb + +Set the timebase to use for the output frames timestamps. +It is mainly useful for testing timebase configuration. + +It accepts the following parameters: + +@table @option + +@item expr, tb +The expression which is evaluated into the output timebase. + +@end table + +The value for @option{tb} is an arithmetic expression representing a +rational. The expression can contain the constants "AVTB" (the default +timebase), "intb" (the input timebase) and "sr" (the sample rate, +audio only). Default value is "intb". + +@subsection Examples + +@itemize +@item +Set the timebase to 1/25: +@example +settb=expr=1/25 +@end example + +@item +Set the timebase to 1/10: +@example +settb=expr=0.1 +@end example + +@item +Set the timebase to 1001/1000: +@example +settb=1+0.001 +@end example + +@item +Set the timebase to 2*intb: +@example +settb=2*intb +@end example + +@item +Set the default timebase value: +@example +settb=AVTB +@end example +@end itemize + +@section showcqt +Convert input audio to a video output representing frequency spectrum +logarithmically using Brown-Puckette constant Q transform algorithm with +direct frequency domain coefficient calculation (but the transform itself +is not really constant Q, instead the Q factor is actually variable/clamped), +with musical tone scale, from E0 to D#10. + +The filter accepts the following options: + +@table @option +@item size, s +Specify the video size for the output. It must be even. For the syntax of this option, +check the @ref{video size syntax,,"Video size" section in the ffmpeg-utils manual,ffmpeg-utils}. +Default value is @code{1920x1080}. + +@item fps, rate, r +Set the output frame rate. Default value is @code{25}. + +@item bar_h +Set the bargraph height. It must be even. Default value is @code{-1} which +computes the bargraph height automatically. + +@item axis_h +Set the axis height. It must be even. Default value is @code{-1} which computes +the axis height automatically. + +@item sono_h +Set the sonogram height. It must be even. Default value is @code{-1} which +computes the sonogram height automatically. + +@item fullhd +Set the fullhd resolution. This option is deprecated, use @var{size}, @var{s} +instead. Default value is @code{1}. + +@item sono_v, volume +Specify the sonogram volume expression. It can contain variables: +@table @option +@item bar_v +the @var{bar_v} evaluated expression +@item frequency, freq, f +the frequency where it is evaluated +@item timeclamp, tc +the value of @var{timeclamp} option +@end table +and functions: +@table @option +@item a_weighting(f) +A-weighting of equal loudness +@item b_weighting(f) +B-weighting of equal loudness +@item c_weighting(f) +C-weighting of equal loudness. +@end table +Default value is @code{16}. + +@item bar_v, volume2 +Specify the bargraph volume expression. It can contain variables: +@table @option +@item sono_v +the @var{sono_v} evaluated expression +@item frequency, freq, f +the frequency where it is evaluated +@item timeclamp, tc +the value of @var{timeclamp} option +@end table +and functions: +@table @option +@item a_weighting(f) +A-weighting of equal loudness +@item b_weighting(f) +B-weighting of equal loudness +@item c_weighting(f) +C-weighting of equal loudness. +@end table +Default value is @code{sono_v}. + +@item sono_g, gamma +Specify the sonogram gamma. Lower gamma makes the spectrum more contrast, +higher gamma makes the spectrum having more range. Default value is @code{3}. +Acceptable range is @code{[1, 7]}. + +@item bar_g, gamma2 +Specify the bargraph gamma. Default value is @code{1}. Acceptable range is +@code{[1, 7]}. + +@item timeclamp, tc +Specify the transform timeclamp. At low frequency, there is trade-off between +accuracy in time domain and frequency domain. If timeclamp is lower, +event in time domain is represented more accurately (such as fast bass drum), +otherwise event in frequency domain is represented more accurately +(such as bass guitar). Acceptable range is @code{[0.1, 1]}. Default value is @code{0.17}. + +@item basefreq +Specify the transform base frequency. Default value is @code{20.01523126408007475}, +which is frequency 50 cents below E0. Acceptable range is @code{[10, 100000]}. + +@item endfreq +Specify the transform end frequency. Default value is @code{20495.59681441799654}, +which is frequency 50 cents above D#10. Acceptable range is @code{[10, 100000]}. + +@item coeffclamp +This option is deprecated and ignored. + +@item tlength +Specify the transform length in time domain. Use this option to control accuracy +trade-off between time domain and frequency domain at every frequency sample. +It can contain variables: +@table @option +@item frequency, freq, f +the frequency where it is evaluated +@item timeclamp, tc +the value of @var{timeclamp} option. +@end table +Default value is @code{384*tc/(384+tc*f)}. + +@item count +Specify the transform count for every video frame. Default value is @code{6}. +Acceptable range is @code{[1, 30]}. + +@item fcount +Specify the transform count for every single pixel. Default value is @code{0}, +which makes it computed automatically. Acceptable range is @code{[0, 10]}. + +@item fontfile +Specify font file for use with freetype to draw the axis. If not specified, +use embedded font. Note that drawing with font file or embedded font is not +implemented with custom @var{basefreq} and @var{endfreq}, use @var{axisfile} +option instead. + +@item fontcolor +Specify font color expression. This is arithmetic expression that should return +integer value 0xRRGGBB. It can contain variables: +@table @option +@item frequency, freq, f +the frequency where it is evaluated +@item timeclamp, tc +the value of @var{timeclamp} option +@end table +and functions: +@table @option +@item midi(f) +midi number of frequency f, some midi numbers: E0(16), C1(24), C2(36), A4(69) +@item r(x), g(x), b(x) +red, green, and blue value of intensity x. +@end table +Default value is @code{st(0, (midi(f)-59.5)/12); +st(1, if(between(ld(0),0,1), 0.5-0.5*cos(2*PI*ld(0)), 0)); +r(1-ld(1)) + b(ld(1))}. + +@item axisfile +Specify image file to draw the axis. This option override @var{fontfile} and +@var{fontcolor} option. + +@item axis, text +Enable/disable drawing text to the axis. If it is set to @code{0}, drawing to +the axis is disabled, ignoring @var{fontfile} and @var{axisfile} option. +Default value is @code{1}. + +@end table + +@subsection Examples + +@itemize +@item +Playing audio while showing the spectrum: +@example +ffplay -f lavfi 'amovie=a.mp3, asplit [a][out1]; [a] showcqt [out0]' +@end example + +@item +Same as above, but with frame rate 30 fps: +@example +ffplay -f lavfi 'amovie=a.mp3, asplit [a][out1]; [a] showcqt=fps=30:count=5 [out0]' +@end example + +@item +Playing at 1280x720: +@example +ffplay -f lavfi 'amovie=a.mp3, asplit [a][out1]; [a] showcqt=s=1280x720:count=4 [out0]' +@end example + +@item +Disable sonogram display: +@example +sono_h=0 +@end example + +@item +A1 and its harmonics: A1, A2, (near)E3, A3: +@example +ffplay -f lavfi 'aevalsrc=0.1*sin(2*PI*55*t)+0.1*sin(4*PI*55*t)+0.1*sin(6*PI*55*t)+0.1*sin(8*PI*55*t), + asplit[a][out1]; [a] showcqt [out0]' +@end example + +@item +Same as above, but with more accuracy in frequency domain: +@example +ffplay -f lavfi 'aevalsrc=0.1*sin(2*PI*55*t)+0.1*sin(4*PI*55*t)+0.1*sin(6*PI*55*t)+0.1*sin(8*PI*55*t), + asplit[a][out1]; [a] showcqt=timeclamp=0.5 [out0]' +@end example + +@item +Custom volume: +@example +bar_v=10:sono_v=bar_v*a_weighting(f) +@end example + +@item +Custom gamma, now spectrum is linear to the amplitude. +@example +bar_g=2:sono_g=2 +@end example + +@item +Custom tlength equation: +@example +tc=0.33:tlength='st(0,0.17); 384*tc / (384 / ld(0) + tc*f /(1-ld(0))) + 384*tc / (tc*f / ld(0) + 384 /(1-ld(0)))' +@end example + +@item +Custom fontcolor and fontfile, C-note is colored green, others are colored blue: +@example +fontcolor='if(mod(floor(midi(f)+0.5),12), 0x0000FF, g(1))':fontfile=myfont.ttf +@end example + +@item +Custom frequency range with custom axis using image file: +@example +axisfile=myaxis.png:basefreq=40:endfreq=10000 +@end example +@end itemize + +@section showfreqs + +Convert input audio to video output representing the audio power spectrum. +Audio amplitude is on Y-axis while frequency is on X-axis. + +The filter accepts the following options: + +@table @option +@item size, s +Specify size of video. For the syntax of this option, check the +@ref{video size syntax,,"Video size" section in the ffmpeg-utils manual,ffmpeg-utils}. +Default is @code{1024x512}. + +@item mode +Set display mode. +This set how each frequency bin will be represented. + +It accepts the following values: +@table @samp +@item line +@item bar +@item dot +@end table +Default is @code{bar}. + +@item ascale +Set amplitude scale. + +It accepts the following values: +@table @samp +@item lin +Linear scale. + +@item sqrt +Square root scale. + +@item cbrt +Cubic root scale. + +@item log +Logarithmic scale. +@end table +Default is @code{log}. + +@item fscale +Set frequency scale. + +It accepts the following values: +@table @samp +@item lin +Linear scale. + +@item log +Logarithmic scale. + +@item rlog +Reverse logarithmic scale. +@end table +Default is @code{lin}. + +@item win_size +Set window size. + +It accepts the following values: +@table @samp +@item w16 +@item w32 +@item w64 +@item w128 +@item w256 +@item w512 +@item w1024 +@item w2048 +@item w4096 +@item w8192 +@item w16384 +@item w32768 +@item w65536 +@end table +Default is @code{w2048} + +@item win_func +Set windowing function. + +It accepts the following values: +@table @samp +@item rect +@item bartlett +@item hanning +@item hamming +@item blackman +@item welch +@item flattop +@item bharris +@item bnuttall +@item bhann +@item sine +@item nuttall +@item lanczos +@item gauss +@item tukey +@end table +Default is @code{hanning}. + +@item overlap +Set window overlap. In range @code{[0, 1]}. Default is @code{1}, +which means optimal overlap for selected window function will be picked. + +@item averaging +Set time averaging. Setting this to 0 will display current maximal peaks. +Default is @code{1}, which means time averaging is disabled. + +@item colors +Specify list of colors separated by space or by '|' which will be used to +draw channel frequencies. Unrecognized or missing colors will be replaced +by white color. + +@item cmode +Set channel display mode. + +It accepts the following values: +@table @samp +@item combined +@item separate +@end table +Default is @code{combined}. + +@end table + +@anchor{showspectrum} +@section showspectrum + +Convert input audio to a video output, representing the audio frequency +spectrum. + +The filter accepts the following options: + +@table @option +@item size, s +Specify the video size for the output. For the syntax of this option, check the +@ref{video size syntax,,"Video size" section in the ffmpeg-utils manual,ffmpeg-utils}. +Default value is @code{640x512}. + +@item slide +Specify how the spectrum should slide along the window. + +It accepts the following values: +@table @samp +@item replace +the samples start again on the left when they reach the right +@item scroll +the samples scroll from right to left +@item rscroll +the samples scroll from left to right +@item fullframe +frames are only produced when the samples reach the right +@end table + +Default value is @code{replace}. + +@item mode +Specify display mode. + +It accepts the following values: +@table @samp +@item combined +all channels are displayed in the same row +@item separate +all channels are displayed in separate rows +@end table + +Default value is @samp{combined}. + +@item color +Specify display color mode. + +It accepts the following values: +@table @samp +@item channel +each channel is displayed in a separate color +@item intensity +each channel is displayed using the same color scheme +@item rainbow +each channel is displayed using the rainbow color scheme +@item moreland +each channel is displayed using the moreland color scheme +@item nebulae +each channel is displayed using the nebulae color scheme +@item fire +each channel is displayed using the fire color scheme +@item fiery +each channel is displayed using the fiery color scheme +@item fruit +each channel is displayed using the fruit color scheme +@item cool +each channel is displayed using the cool color scheme +@end table + +Default value is @samp{channel}. + +@item scale +Specify scale used for calculating intensity color values. + +It accepts the following values: +@table @samp +@item lin +linear +@item sqrt +square root, default +@item cbrt +cubic root +@item 4thrt +4th root +@item 5thrt +5th root +@item log +logarithmic +@end table + +Default value is @samp{sqrt}. + +@item saturation +Set saturation modifier for displayed colors. Negative values provide +alternative color scheme. @code{0} is no saturation at all. +Saturation must be in [-10.0, 10.0] range. +Default value is @code{1}. + +@item win_func +Set window function. + +It accepts the following values: +@table @samp +@item rect +@item bartlett +@item hann +@item hanning +@item hamming +@item blackman +@item welch +@item flattop +@item bharris +@item bnuttall +@item bhann +@item sine +@item nuttall +@item lanczos +@item gauss +@item tukey +@end table + +Default value is @code{hann}. + +@item orientation +Set orientation of time vs frequency axis. Can be @code{vertical} or +@code{horizontal}. Default is @code{vertical}. + +@item overlap +Set ratio of overlap window. Default value is @code{0}. +When value is @code{1} overlap is set to recommended size for specific +window function currently used. + +@item gain +Set scale gain for calculating intensity color values. +Default value is @code{1}. + +@item data +Set which data to display. Can be @code{magnitude}, default or @code{phase}. +@end table + +The usage is very similar to the showwaves filter; see the examples in that +section. + +@subsection Examples + +@itemize +@item +Large window with logarithmic color scaling: +@example +showspectrum=s=1280x480:scale=log +@end example + +@item +Complete example for a colored and sliding spectrum per channel using @command{ffplay}: +@example +ffplay -f lavfi 'amovie=input.mp3, asplit [a][out1]; + [a] showspectrum=mode=separate:color=intensity:slide=1:scale=cbrt [out0]' +@end example +@end itemize + +@section showspectrumpic + +Convert input audio to a single video frame, representing the audio frequency +spectrum. + +The filter accepts the following options: + +@table @option +@item size, s +Specify the video size for the output. For the syntax of this option, check the +@ref{video size syntax,,"Video size" section in the ffmpeg-utils manual,ffmpeg-utils}. +Default value is @code{4096x2048}. + +@item mode +Specify display mode. + +It accepts the following values: +@table @samp +@item combined +all channels are displayed in the same row +@item separate +all channels are displayed in separate rows +@end table +Default value is @samp{combined}. + +@item color +Specify display color mode. + +It accepts the following values: +@table @samp +@item channel +each channel is displayed in a separate color +@item intensity +each channel is displayed using the same color scheme +@item rainbow +each channel is displayed using the rainbow color scheme +@item moreland +each channel is displayed using the moreland color scheme +@item nebulae +each channel is displayed using the nebulae color scheme +@item fire +each channel is displayed using the fire color scheme +@item fiery +each channel is displayed using the fiery color scheme +@item fruit +each channel is displayed using the fruit color scheme +@item cool +each channel is displayed using the cool color scheme +@end table +Default value is @samp{intensity}. + +@item scale +Specify scale used for calculating intensity color values. + +It accepts the following values: +@table @samp +@item lin +linear +@item sqrt +square root, default +@item cbrt +cubic root +@item 4thrt +4th root +@item 5thrt +5th root +@item log +logarithmic +@end table +Default value is @samp{log}. + +@item saturation +Set saturation modifier for displayed colors. Negative values provide +alternative color scheme. @code{0} is no saturation at all. +Saturation must be in [-10.0, 10.0] range. +Default value is @code{1}. + +@item win_func +Set window function. + +It accepts the following values: +@table @samp +@item rect +@item bartlett +@item hann +@item hanning +@item hamming +@item blackman +@item welch +@item flattop +@item bharris +@item bnuttall +@item bhann +@item sine +@item nuttall +@item lanczos +@item gauss +@item tukey +@end table +Default value is @code{hann}. + +@item orientation +Set orientation of time vs frequency axis. Can be @code{vertical} or +@code{horizontal}. Default is @code{vertical}. + +@item gain +Set scale gain for calculating intensity color values. +Default value is @code{1}. + +@item legend +Draw time and frequency axes and legends. Default is enabled. +@end table + +@subsection Examples + +@itemize +@item +Extract an audio spectrogram of a whole audio track +in a 1024x1024 picture using @command{ffmpeg}: +@example +ffmpeg -i audio.flac -lavfi showspectrumpic=s=1024x1024 spectrogram.png +@end example +@end itemize + +@section showvolume + +Convert input audio volume to a video output. + +The filter accepts the following options: + +@table @option +@item rate, r +Set video rate. + +@item b +Set border width, allowed range is [0, 5]. Default is 1. + +@item w +Set channel width, allowed range is [80, 1080]. Default is 400. + +@item h +Set channel height, allowed range is [1, 100]. Default is 20. + +@item f +Set fade, allowed range is [0.001, 1]. Default is 0.95. + +@item c +Set volume color expression. + +The expression can use the following variables: + +@table @option +@item VOLUME +Current max volume of channel in dB. + +@item CHANNEL +Current channel number, starting from 0. +@end table + +@item t +If set, displays channel names. Default is enabled. + +@item v +If set, displays volume values. Default is enabled. +@end table + +@section showwaves + +Convert input audio to a video output, representing the samples waves. + +The filter accepts the following options: + +@table @option +@item size, s +Specify the video size for the output. For the syntax of this option, check the +@ref{video size syntax,,"Video size" section in the ffmpeg-utils manual,ffmpeg-utils}. +Default value is @code{600x240}. + +@item mode +Set display mode. + +Available values are: +@table @samp +@item point +Draw a point for each sample. + +@item line +Draw a vertical line for each sample. + +@item p2p +Draw a point for each sample and a line between them. + +@item cline +Draw a centered vertical line for each sample. +@end table + +Default value is @code{point}. + +@item n +Set the number of samples which are printed on the same column. A +larger value will decrease the frame rate. Must be a positive +integer. This option can be set only if the value for @var{rate} +is not explicitly specified. + +@item rate, r +Set the (approximate) output frame rate. This is done by setting the +option @var{n}. Default value is "25". + +@item split_channels +Set if channels should be drawn separately or overlap. Default value is 0. + +@item colors +Set colors separated by '|' which are going to be used for drawing of each channel. + +@item scale +Set amplitude scale. Can be linear @code{lin} or logarithmic @code{log}. +Default is linear. + +@end table + +@subsection Examples + +@itemize +@item +Output the input file audio and the corresponding video representation +at the same time: +@example +amovie=a.mp3,asplit[out0],showwaves[out1] +@end example + +@item +Create a synthetic signal and show it with showwaves, forcing a +frame rate of 30 frames per second: +@example +aevalsrc=sin(1*2*PI*t)*sin(880*2*PI*t):cos(2*PI*200*t),asplit[out0],showwaves=r=30[out1] +@end example +@end itemize + +@section showwavespic + +Convert input audio to a single video frame, representing the samples waves. + +The filter accepts the following options: + +@table @option +@item size, s +Specify the video size for the output. For the syntax of this option, check the +@ref{video size syntax,,"Video size" section in the ffmpeg-utils manual,ffmpeg-utils}. +Default value is @code{600x240}. + +@item split_channels +Set if channels should be drawn separately or overlap. Default value is 0. + +@item colors +Set colors separated by '|' which are going to be used for drawing of each channel. + +@item scale +Set amplitude scale. Can be linear @code{lin} or logarithmic @code{log}. +Default is linear. +@end table + +@subsection Examples + +@itemize +@item +Extract a channel split representation of the wave form of a whole audio track +in a 1024x800 picture using @command{ffmpeg}: +@example +ffmpeg -i audio.flac -lavfi showwavespic=split_channels=1:s=1024x800 waveform.png +@end example + +@item +Colorize the waveform with colorchannelmixer. This example will make +the waveform a green color approximately RGB(66,217,150). Additional +channels will be shades of this color. +@example +ffmpeg -i audio.mp3 -filter_complex "showwavespic,colorchannelmixer=rr=66/255:gg=217/255:bb=150/255" waveform.png +@end example +@end itemize + +@section spectrumsynth + +Sythesize audio from 2 input video spectrums, first input stream represents +magnitude across time and second represents phase across time. +The filter will transform from frequency domain as displayed in videos back +to time domain as presented in audio output. + +This filter is primarly created for reversing processed @ref{showspectrum} +filter outputs, but can synthesize sound from other spectrograms too. +But in such case results are going to be poor if the phase data is not +available, because in such cases phase data need to be recreated, usually +its just recreated from random noise. +For best results use gray only output (@code{channel} color mode in +@ref{showspectrum} filter) and @code{log} scale for magnitude video and +@code{lin} scale for phase video. To produce phase, for 2nd video, use +@code{data} option. Inputs videos should generally use @code{fullframe} +slide mode as that saves resources needed for decoding video. + +The filter accepts the following options: + +@table @option +@item sample_rate +Specify sample rate of output audio, the sample rate of audio from which +spectrum was generated may differ. + +@item channels +Set number of channels represented in input video spectrums. + +@item scale +Set scale which was used when generating magnitude input spectrum. +Can be @code{lin} or @code{log}. Default is @code{log}. + +@item slide +Set slide which was used when generating inputs spectrums. +Can be @code{replace}, @code{scroll}, @code{fullframe} or @code{rscroll}. +Default is @code{fullframe}. + +@item win_func +Set window function used for resynthesis. + +@item overlap +Set window overlap. In range @code{[0, 1]}. Default is @code{1}, +which means optimal overlap for selected window function will be picked. + +@item orientation +Set orientation of input videos. Can be @code{vertical} or @code{horizontal}. +Default is @code{vertical}. +@end table + +@subsection Examples + +@itemize +@item +First create magnitude and phase videos from audio, assuming audio is stereo with 44100 sample rate, +then resynthesize videos back to audio with spectrumsynth: +@example +ffmpeg -i input.flac -lavfi showspectrum=mode=separate:scale=log:overlap=0.875:color=channel:slide=fullframe:data=magnitude -an -c:v rawvideo magnitude.nut +ffmpeg -i input.flac -lavfi showspectrum=mode=separate:scale=lin:overlap=0.875:color=channel:slide=fullframe:data=phase -an -c:v rawvideo phase.nut +ffmpeg -i magnitude.nut -i phase.nut -lavfi spectrumsynth=channels=2:sample_rate=44100:win_func=hann:overlap=0.875:slide=fullframe output.flac +@end example +@end itemize + +@section split, asplit + +Split input into several identical outputs. + +@code{asplit} works with audio input, @code{split} with video. + +The filter accepts a single parameter which specifies the number of outputs. If +unspecified, it defaults to 2. + +@subsection Examples + +@itemize +@item +Create two separate outputs from the same input: +@example +[in] split [out0][out1] +@end example + +@item +To create 3 or more outputs, you need to specify the number of +outputs, like in: +@example +[in] asplit=3 [out0][out1][out2] +@end example + +@item +Create two separate outputs from the same input, one cropped and +one padded: +@example +[in] split [splitout1][splitout2]; +[splitout1] crop=100:100:0:0 [cropout]; +[splitout2] pad=200:200:100:100 [padout]; +@end example + +@item +Create 5 copies of the input audio with @command{ffmpeg}: +@example +ffmpeg -i INPUT -filter_complex asplit=5 OUTPUT +@end example +@end itemize + +@section zmq, azmq + +Receive commands sent through a libzmq client, and forward them to +filters in the filtergraph. + +@code{zmq} and @code{azmq} work as a pass-through filters. @code{zmq} +must be inserted between two video filters, @code{azmq} between two +audio filters. + +To enable these filters you need to install the libzmq library and +headers and configure FFmpeg with @code{--enable-libzmq}. + +For more information about libzmq see: +@url{http://www.zeromq.org/} + +The @code{zmq} and @code{azmq} filters work as a libzmq server, which +receives messages sent through a network interface defined by the +@option{bind_address} option. + +The received message must be in the form: +@example +@var{TARGET} @var{COMMAND} [@var{ARG}] +@end example + +@var{TARGET} specifies the target of the command, usually the name of +the filter class or a specific filter instance name. + +@var{COMMAND} specifies the name of the command for the target filter. + +@var{ARG} is optional and specifies the optional argument list for the +given @var{COMMAND}. + +Upon reception, the message is processed and the corresponding command +is injected into the filtergraph. Depending on the result, the filter +will send a reply to the client, adopting the format: +@example +@var{ERROR_CODE} @var{ERROR_REASON} +@var{MESSAGE} +@end example + +@var{MESSAGE} is optional. + +@subsection Examples + +Look at @file{tools/zmqsend} for an example of a zmq client which can +be used to send commands processed by these filters. + +Consider the following filtergraph generated by @command{ffplay} +@example +ffplay -dumpgraph 1 -f lavfi " +color=s=100x100:c=red [l]; +color=s=100x100:c=blue [r]; +nullsrc=s=200x100, zmq [bg]; +[bg][l] overlay [bg+l]; +[bg+l][r] overlay=x=100 " +@end example + +To change the color of the left side of the video, the following +command can be used: +@example +echo Parsed_color_0 c yellow | tools/zmqsend +@end example + +To change the right side: +@example +echo Parsed_color_1 c pink | tools/zmqsend +@end example + +@c man end MULTIMEDIA FILTERS + +@chapter Multimedia Sources +@c man begin MULTIMEDIA SOURCES + +Below is a description of the currently available multimedia sources. + +@section amovie + +This is the same as @ref{movie} source, except it selects an audio +stream by default. + +@anchor{movie} +@section movie + +Read audio and/or video stream(s) from a movie container. + +It accepts the following parameters: + +@table @option +@item filename +The name of the resource to read (not necessarily a file; it can also be a +device or a stream accessed through some protocol). + +@item format_name, f +Specifies the format assumed for the movie to read, and can be either +the name of a container or an input device. If not specified, the +format is guessed from @var{movie_name} or by probing. + +@item seek_point, sp +Specifies the seek point in seconds. The frames will be output +starting from this seek point. The parameter is evaluated with +@code{av_strtod}, so the numerical value may be suffixed by an IS +postfix. The default value is "0". + +@item streams, s +Specifies the streams to read. Several streams can be specified, +separated by "+". The source will then have as many outputs, in the +same order. The syntax is explained in the ``Stream specifiers'' +section in the ffmpeg manual. Two special names, "dv" and "da" specify +respectively the default (best suited) video and audio stream. Default +is "dv", or "da" if the filter is called as "amovie". + +@item stream_index, si +Specifies the index of the video stream to read. If the value is -1, +the most suitable video stream will be automatically selected. The default +value is "-1". Deprecated. If the filter is called "amovie", it will select +audio instead of video. + +@item loop +Specifies how many times to read the stream in sequence. +If the value is less than 1, the stream will be read again and again. +Default value is "1". + +Note that when the movie is looped the source timestamps are not +changed, so it will generate non monotonically increasing timestamps. +@end table + +It allows overlaying a second video on top of the main input of +a filtergraph, as shown in this graph: +@example +input -----------> deltapts0 --> overlay --> output + ^ + | +movie --> scale--> deltapts1 -------+ +@end example +@subsection Examples + +@itemize +@item +Skip 3.2 seconds from the start of the AVI file in.avi, and overlay it +on top of the input labelled "in": +@example +movie=in.avi:seek_point=3.2, scale=180:-1, setpts=PTS-STARTPTS [over]; +[in] setpts=PTS-STARTPTS [main]; +[main][over] overlay=16:16 [out] +@end example + +@item +Read from a video4linux2 device, and overlay it on top of the input +labelled "in": +@example +movie=/dev/video0:f=video4linux2, scale=180:-1, setpts=PTS-STARTPTS [over]; +[in] setpts=PTS-STARTPTS [main]; +[main][over] overlay=16:16 [out] +@end example + +@item +Read the first video stream and the audio stream with id 0x81 from +dvd.vob; the video is connected to the pad named "video" and the audio is +connected to the pad named "audio": +@example +movie=dvd.vob:s=v:0+#0x81 [video] [audio] +@end example +@end itemize + +@c man end MULTIMEDIA SOURCES diff --git a/doc/formats.texi b/doc/formats.texi new file mode 100644 index 0000000000..617cda54a9 --- /dev/null +++ b/doc/formats.texi @@ -0,0 +1,249 @@ +@chapter Format Options +@c man begin FORMAT OPTIONS + +The libavformat library provides some generic global options, which +can be set on all the muxers and demuxers. In addition each muxer or +demuxer may support so-called private options, which are specific for +that component. + +Options may be set by specifying -@var{option} @var{value} in the +FFmpeg tools, or by setting the value explicitly in the +@code{AVFormatContext} options or using the @file{libavutil/opt.h} API +for programmatic use. + +The list of supported options follows: + +@table @option +@item avioflags @var{flags} (@emph{input/output}) +Possible values: +@table @samp +@item direct +Reduce buffering. +@end table + +@item probesize @var{integer} (@emph{input}) +Set probing size in bytes, i.e. the size of the data to analyze to get +stream information. A higher value will enable detecting more +information in case it is dispersed into the stream, but will increase +latency. Must be an integer not lesser than 32. It is 5000000 by default. + +@item packetsize @var{integer} (@emph{output}) +Set packet size. + +@item fflags @var{flags} (@emph{input/output}) +Set format flags. + +Possible values: +@table @samp +@item ignidx +Ignore index. +@item fastseek +Enable fast, but inaccurate seeks for some formats. +@item genpts +Generate PTS. +@item nofillin +Do not fill in missing values that can be exactly calculated. +@item noparse +Disable AVParsers, this needs @code{+nofillin} too. +@item igndts +Ignore DTS. +@item discardcorrupt +Discard corrupted frames. +@item sortdts +Try to interleave output packets by DTS. +@item keepside +Do not merge side data. +@item latm +Enable RTP MP4A-LATM payload. +@item nobuffer +Reduce the latency introduced by optional buffering +@item bitexact +Only write platform-, build- and time-independent data. +This ensures that file and data checksums are reproducible and match between +platforms. Its primary use is for regression testing. +@end table + +@item seek2any @var{integer} (@emph{input}) +Allow seeking to non-keyframes on demuxer level when supported if set to 1. +Default is 0. + +@item analyzeduration @var{integer} (@emph{input}) +Specify how many microseconds are analyzed to probe the input. A +higher value will enable detecting more accurate information, but will +increase latency. It defaults to 5,000,000 microseconds = 5 seconds. + +@item cryptokey @var{hexadecimal string} (@emph{input}) +Set decryption key. + +@item indexmem @var{integer} (@emph{input}) +Set max memory used for timestamp index (per stream). + +@item rtbufsize @var{integer} (@emph{input}) +Set max memory used for buffering real-time frames. + +@item fdebug @var{flags} (@emph{input/output}) +Print specific debug info. + +Possible values: +@table @samp +@item ts +@end table + +@item max_delay @var{integer} (@emph{input/output}) +Set maximum muxing or demuxing delay in microseconds. + +@item fpsprobesize @var{integer} (@emph{input}) +Set number of frames used to probe fps. + +@item audio_preload @var{integer} (@emph{output}) +Set microseconds by which audio packets should be interleaved earlier. + +@item chunk_duration @var{integer} (@emph{output}) +Set microseconds for each chunk. + +@item chunk_size @var{integer} (@emph{output}) +Set size in bytes for each chunk. + +@item err_detect, f_err_detect @var{flags} (@emph{input}) +Set error detection flags. @code{f_err_detect} is deprecated and +should be used only via the @command{ffmpeg} tool. + +Possible values: +@table @samp +@item crccheck +Verify embedded CRCs. +@item bitstream +Detect bitstream specification deviations. +@item buffer +Detect improper bitstream length. +@item explode +Abort decoding on minor error detection. +@item careful +Consider things that violate the spec and have not been seen in the +wild as errors. +@item compliant +Consider all spec non compliancies as errors. +@item aggressive +Consider things that a sane encoder should not do as an error. +@end table + +@item max_interleave_delta @var{integer} (@emph{output}) +Set maximum buffering duration for interleaving. The duration is +expressed in microseconds, and defaults to 1000000 (1 second). + +To ensure all the streams are interleaved correctly, libavformat will +wait until it has at least one packet for each stream before actually +writing any packets to the output file. When some streams are +"sparse" (i.e. there are large gaps between successive packets), this +can result in excessive buffering. + +This field specifies the maximum difference between the timestamps of the +first and the last packet in the muxing queue, above which libavformat +will output a packet regardless of whether it has queued a packet for all +the streams. + +If set to 0, libavformat will continue buffering packets until it has +a packet for each stream, regardless of the maximum timestamp +difference between the buffered packets. + +@item use_wallclock_as_timestamps @var{integer} (@emph{input}) +Use wallclock as timestamps. + +@item avoid_negative_ts @var{integer} (@emph{output}) + +Possible values: +@table @samp +@item make_non_negative +Shift timestamps to make them non-negative. +Also note that this affects only leading negative timestamps, and not +non-monotonic negative timestamps. +@item make_zero +Shift timestamps so that the first timestamp is 0. +@item auto (default) +Enables shifting when required by the target format. +@item disabled +Disables shifting of timestamp. +@end table + +When shifting is enabled, all output timestamps are shifted by the +same amount. Audio, video, and subtitles desynching and relative +timestamp differences are preserved compared to how they would have +been without shifting. + +@item skip_initial_bytes @var{integer} (@emph{input}) +Set number of bytes to skip before reading header and frames if set to 1. +Default is 0. + +@item correct_ts_overflow @var{integer} (@emph{input}) +Correct single timestamp overflows if set to 1. Default is 1. + +@item flush_packets @var{integer} (@emph{output}) +Flush the underlying I/O stream after each packet. Default 1 enables it, and +has the effect of reducing the latency; 0 disables it and may slightly +increase performance in some cases. + +@item output_ts_offset @var{offset} (@emph{output}) +Set the output time offset. + +@var{offset} must be a time duration specification, +see @ref{time duration syntax,,the Time duration section in the ffmpeg-utils(1) manual,ffmpeg-utils}. + +The offset is added by the muxer to the output timestamps. + +Specifying a positive offset means that the corresponding streams are +delayed bt the time duration specified in @var{offset}. Default value +is @code{0} (meaning that no offset is applied). + +@item format_whitelist @var{list} (@emph{input}) +"," separated List of allowed demuxers. By default all are allowed. + +@item dump_separator @var{string} (@emph{input}) +Separator used to separate the fields printed on the command line about the +Stream parameters. +For example to separate the fields with newlines and indention: +@example +ffprobe -dump_separator " + " -i ~/videos/matrixbench_mpeg2.mpg +@end example +@end table + +@c man end FORMAT OPTIONS + +@anchor{Format stream specifiers} +@section Format stream specifiers + +Format stream specifiers allow selection of one or more streams that +match specific properties. + +Possible forms of stream specifiers are: +@table @option +@item @var{stream_index} +Matches the stream with this index. + +@item @var{stream_type}[:@var{stream_index}] +@var{stream_type} is one of following: 'v' for video, 'a' for audio, +'s' for subtitle, 'd' for data, and 't' for attachments. If +@var{stream_index} is given, then it matches the stream number +@var{stream_index} of this type. Otherwise, it matches all streams of +this type. + +@item p:@var{program_id}[:@var{stream_index}] +If @var{stream_index} is given, then it matches the stream with number +@var{stream_index} in the program with the id +@var{program_id}. Otherwise, it matches all streams in the program. + +@item #@var{stream_id} +Matches the stream by a format-specific ID. +@end table + +The exact semantics of stream specifiers is defined by the +@code{avformat_match_stream_specifier()} function declared in the +@file{libavformat/avformat.h} header. + +@ifclear config-writeonly +@include demuxers.texi +@end ifclear +@ifclear config-readonly +@include muxers.texi +@end ifclear +@include metadata.texi diff --git a/doc/general.texi b/doc/general.texi index 15e4a660d6..59ea4f44d9 100644 --- a/doc/general.texi +++ b/doc/general.texi @@ -1,4 +1,5 @@ \input texinfo @c -*- texinfo -*- +@documentencoding UTF-8 @settitle General Documentation @titlepage @@ -11,12 +12,20 @@ @chapter External libraries -Libav can be hooked up with a number of external libraries to add support +FFmpeg can be hooked up with a number of external libraries to add support for more formats. None of them are used by default, their use has to be explicitly requested by passing the appropriate flags to @command{./configure}. -@section OpenCORE and VisualOn libraries +@section OpenJPEG + +FFmpeg can use the OpenJPEG libraries for encoding/decoding J2K videos. Go to +@url{http://www.openjpeg.org/} to get the libraries and follow the installation +instructions. To enable using OpenJPEG in FFmpeg, pass @code{--enable-libopenjpeg} to +@file{./configure}. + + +@section OpenCORE, VisualOn, and Fraunhofer libraries Spun off Google Android sources, OpenCore, VisualOn and Fraunhofer libraries provide encoders for a number of audio codecs. @@ -24,14 +33,19 @@ libraries provide encoders for a number of audio codecs. @float NOTE OpenCORE and VisualOn libraries are under the Apache License 2.0 (see @url{http://www.apache.org/licenses/LICENSE-2.0} for details), which is -incompatible with the LGPL version 2.1 and GPL version 2. You have to -upgrade Libav's license to LGPL version 3 (or if you have enabled -GPL components, GPL version 3) to use it. +incompatible to the LGPL version 2.1 and GPL version 2. You have to +upgrade FFmpeg's license to LGPL version 3 (or if you have enabled +GPL components, GPL version 3) by passing @code{--enable-version3} to configure in +order to use it. + +The Fraunhofer AAC library is licensed under a license incompatible to the GPL +and is not known to be compatible to the LGPL. Therefore, you have to pass +@code{--enable-nonfree} to configure to use it. @end float @subsection OpenCORE AMR -Libav can make use of the OpenCORE libraries for AMR-NB +FFmpeg can make use of the OpenCORE libraries for AMR-NB decoding/encoding and AMR-WB decoding. Go to @url{http://sourceforge.net/projects/opencore-amr/} and follow the @@ -39,17 +53,9 @@ instructions for installing the libraries. Then pass @code{--enable-libopencore-amrnb} and/or @code{--enable-libopencore-amrwb} to configure to enable them. -@subsection VisualOn AAC encoder library - -Libav can make use of the VisualOn AACenc library for AAC encoding. - -Go to @url{http://sourceforge.net/projects/opencore-amr/} and follow the -instructions for installing the library. -Then pass @code{--enable-libvo-aacenc} to configure to enable it. - @subsection VisualOn AMR-WB encoder library -Libav can make use of the VisualOn AMR-WBenc library for AMR-WB encoding. +FFmpeg can make use of the VisualOn AMR-WBenc library for AMR-WB encoding. Go to @url{http://sourceforge.net/projects/opencore-amr/} and follow the instructions for installing the library. @@ -57,7 +63,7 @@ Then pass @code{--enable-libvo-amrwbenc} to configure to enable it. @subsection Fraunhofer AAC library -Libav can make use of the Fraunhofer AAC library for AAC encoding. +FFmpeg can make use of the Fraunhofer AAC library for AAC encoding. Go to @url{http://sourceforge.net/projects/opencore-amr/} and follow the instructions for installing the library. @@ -65,7 +71,7 @@ Then pass @code{--enable-libfdk-aac} to configure to enable it. @section LAME -Libav can make use of the LAME library for MP3 encoding. +FFmpeg can make use of the LAME library for MP3 encoding. Go to @url{http://lame.sourceforge.net/} and follow the instructions for installing the library. @@ -73,7 +79,7 @@ Then pass @code{--enable-libmp3lame} to configure to enable it. @section TwoLAME -Libav can make use of the TwoLAME library for MP2 encoding. +FFmpeg can make use of the TwoLAME library for MP2 encoding. Go to @url{http://www.twolame.org/} and follow the instructions for installing the library. @@ -81,7 +87,7 @@ Then pass @code{--enable-libtwolame} to configure to enable it. @section libvpx -Libav can make use of the libvpx library for VP8 encoding. +FFmpeg can make use of the libvpx library for VP8/VP9 encoding. Go to @url{http://www.webmproject.org/} and follow the instructions for installing the library. Then pass @code{--enable-libvpx} to configure to @@ -89,7 +95,7 @@ enable it. @section libwavpack -Libav can make use of the libwavpack library for WavPack encoding. +FFmpeg can make use of the libwavpack library for WavPack encoding. Go to @url{http://www.wavpack.com/} and follow the instructions for installing the library. Then pass @code{--enable-libwavpack} to configure to @@ -97,7 +103,7 @@ enable it. @section OpenH264 -Libav can make use of the OpenH264 library for H.264 encoding. +FFmpeg can make use of the OpenH264 library for H.264 encoding. Go to @url{http://www.openh264.org/} and follow the instructions for installing the library. Then pass @code{--enable-libopenh264} to configure to @@ -105,7 +111,7 @@ enable it. @section x264 -Libav can make use of the x264 library for H.264 encoding. +FFmpeg can make use of the x264 library for H.264 encoding. Go to @url{http://www.videolan.org/developers/x264.html} and follow the instructions for installing the library. Then pass @code{--enable-libx264} to @@ -114,26 +120,26 @@ configure to enable it. @float NOTE x264 is under the GNU Public License Version 2 or later (see @url{http://www.gnu.org/licenses/old-licenses/gpl-2.0.html} for -details), you must upgrade Libav's license to GPL in order to use it. +details), you must upgrade FFmpeg's license to GPL in order to use it. @end float @section x265 -Libav can make use of the x265 library for HEVC encoding. +FFmpeg can make use of the x265 library for HEVC encoding. Go to @url{http://x265.org/developers.html} and follow the instructions for installing the library. Then pass @code{--enable-libx265} to configure to enable it. -@float note +@float NOTE x265 is under the GNU Public License Version 2 or later (see @url{http://www.gnu.org/licenses/old-licenses/gpl-2.0.html} for -details), you must upgrade Libav's license to GPL in order to use it. +details), you must upgrade FFmpeg's license to GPL in order to use it. @end float @section kvazaar -Libav can make use of the kvazaar library for HEVC encoding. +FFmpeg can make use of the kvazaar library for HEVC encoding. Go to @url{https://github.com/ultravideo/kvazaar} and follow the instructions for installing the library. Then pass @@ -143,47 +149,38 @@ instructions for installing the library. Then pass iLBC is a narrowband speech codec that has been made freely available by Google as part of the WebRTC project. libilbc is a packaging friendly -copy of the iLBC codec. Libav can make use of the libilbc library for +copy of the iLBC codec. FFmpeg can make use of the libilbc library for iLBC encoding and decoding. -Go to @url{https://github.com/dekkers/libilbc} and follow the instructions for +Go to @url{https://github.com/TimothyGu/libilbc} and follow the instructions for installing the library. Then pass @code{--enable-libilbc} to configure to enable it. +@section libzvbi + +libzvbi is a VBI decoding library which can be used by FFmpeg to decode DVB +teletext pages and DVB teletext subtitles. + +Go to @url{http://sourceforge.net/projects/zapping/} and follow the instructions for +installing the library. Then pass @code{--enable-libzvbi} to configure to +enable it. + @section AviSynth -Libav can read AviSynth scripts as input. To enable support you need a -suitable @file{avisynth_c.h} header to compile against. The header as -provided by AviSynth+ is fully compatible. AviSynth 2.5 is not supported -by Libav. Once you have the appropriate header, pass -@code{--enable-avisynth} to configure to enable AviSynth support. +FFmpeg can read AviSynth scripts as input. To enable support, pass +@code{--enable-avisynth} to configure. The correct headers are +included in compat/avisynth/, which allows the user to enable support +without needing to search for these headers themselves. For Windows, supported AviSynth variants are @url{http://avisynth.nl, AviSynth 2.6 RC1 or higher} for 32-bit builds and @url{http://avs-plus.net, AviSynth+ r1718 or higher} for 32-bit and 64-bit builds. -@url{https://github.com/AviSynth/AviSynthPlus, AviSynth+'s git repository} -provides a GNU-style Makefile which can install just the headers using -@code{make install PREFIX=/install/prefix}. - -@float NOTE -When using AviSynth+'s installed headers, the user must also pass -the avisynth/ include directory to @code{--extra-cflags}. For example, -if the PREFIX given to AviSynth+'s Makefile was /usr/i686-w64-mingw32, -then the correct command would be -@code{--extra-cflags="-I/usr/i686-w64-mingw32/include/avisynth"}. -@end float For Linux and OS X, the supported AviSynth variant is @url{https://github.com/avxsynth/avxsynth, AvxSynth}. -@file{avxsynth_c.h} is installed as part of the normal -build routine, as illustrated on -@url{https://github.com/avxsynth/avxsynth/wiki/System-Setup, AvxSynth's wiki}. -(the instructions for compiling its prerequisites are outdated, as FFMS 2.18 -or higher is now needed; the list of dependencies to be downloaded from the -repositories is still the same, though). @float NOTE -AviSynth and AvxSynth are loaded dynamically. Distributors can build Libav +AviSynth and AvxSynth are loaded dynamically. Distributors can build FFmpeg with @code{--enable-avisynth}, and the binaries will work regardless of the end user having AviSynth or AvxSynth installed - they'll only need to be installed to use AviSynth scripts (obviously). @@ -191,29 +188,37 @@ installed to use AviSynth scripts (obviously). @section Intel QuickSync Video -Libav can use Intel QuickSync Video (QSV) for accelerated encoding and decoding -of multiple codecs. To use QSV, Libav must be linked against the @code{libmfx} +FFmpeg can use Intel QuickSync Video (QSV) for accelerated encoding and decoding +of multiple codecs. To use QSV, FFmpeg must be linked against the @code{libmfx} dispatcher, which loads the actual decoding libraries. The dispatcher is open source and can be downloaded from -@url{https://github.com/lu-zero/mfx_dispatch.git}. Libav needs to be configured +@url{https://github.com/lu-zero/mfx_dispatch.git}. FFmpeg needs to be configured with the @code{--enable-libmfx} option and @code{pkg-config} needs to be able to locate the dispatcher's @code{.pc} files. -@chapter Supported File Formats and Codecs + +@chapter Supported File Formats, Codecs or Features You can use the @code{-formats} and @code{-codecs} options to have an exhaustive list. @section File Formats -Libav supports the following file formats through the @code{libavformat} +FFmpeg supports the following file formats through the @code{libavformat} library: @multitable @columnfractions .4 .1 .1 .4 @item Name @tab Encoding @tab Decoding @tab Comments +@item 3dostr @tab @tab X @item 4xm @tab @tab X @tab 4X Technologies format, used in some games. @item 8088flex TMV @tab @tab X +@item AAX @tab @tab X + @tab Audible Enhanced Audio format, used in audiobooks. +@item AA @tab @tab X + @tab Audible Format 2, 3, and 4, used in audiobooks. +@item ACT Voice @tab @tab X + @tab contains G.729 audio @item Adobe Filmstrip @tab X @tab X @item Audio IFF (AIFF) @tab X @tab X @item American Laser Games MM @tab @tab X @@ -222,21 +227,40 @@ library: @item Amazing Studio Packed Animation File @tab @tab X @tab Multimedia format used in game Heart Of Darkness. @item Apple HTTP Live Streaming @tab @tab X +@item Artworx Data Format @tab @tab X +@item Interplay ACM @tab @tab X + @tab Audio only format used in some Interplay games. +@item ADP @tab @tab X + @tab Audio format used on the Nintendo Gamecube. +@item AFC @tab @tab X + @tab Audio format used on the Nintendo Gamecube. +@item ADS/SS2 @tab @tab X + @tab Audio format used on the PS2. +@item APNG @tab X @tab X @item ASF @tab X @tab X +@item AST @tab X @tab X + @tab Audio format used on the Nintendo Wii. @item AVI @tab X @tab X @item AviSynth @tab @tab X +@item AVR @tab @tab X + @tab Audio format used on Mac. @item AVS @tab @tab X @tab Multimedia format used by the Creature Shock game. @item Beam Software SIFF @tab @tab X @tab Audio and video format used in some games by Beam Software. @item Bethesda Softworks VID @tab @tab X @tab Used in some games from Bethesda Softworks. +@item Binary text @tab @tab X @item Bink @tab @tab X @tab Multimedia format used by many games. @item Bitmap Brothers JV @tab @tab X @tab Used in Z and Z95 games. @item Brute Force & Ignorance @tab @tab X @tab Used in the game Flash Traffic: City of Angels. +@item BFSTM @tab @tab X + @tab Audio format used on the Nintendo WiiU (based on BRSTM). +@item BRSTM @tab @tab X + @tab Audio format used on the Nintendo Wii. @item BWF @tab X @tab X @item CRI ADX @tab X @tab X @tab Audio-only format used in console video games. @@ -251,9 +275,11 @@ library: @item Canopus HQX @tab @tab X @item CD+G @tab @tab X @tab Video format used by CD+G karaoke disks +@item Phantom Cine @tab @tab X +@item Cineform HD @tab @tab X @item Commodore CDXL @tab @tab X @tab Amiga CD video format -@item Core Audio Format @tab @tab X +@item Core Audio Format @tab X @tab X @tab Apple Core Audio Format @item CRC testing format @tab X @tab @item Creative Voice @tab X @tab X @@ -262,9 +288,11 @@ library: @tab Audio format used in some games by CRYO Interactive Entertainment. @item D-Cinema audio @tab X @tab X @item Deluxe Paint Animation @tab @tab X +@item DCSTR @tab @tab X @item DFA @tab @tab X @tab This format is used in Chronomaster game @item DirectDraw Surface @tab @tab X +@item DSD Stream File (DSF) @tab @tab X @item DV video @tab X @tab X @item DXA @tab @tab X @tab This format is used in the non-Windows version of the Feeble Files @@ -272,6 +300,8 @@ library: @item Electronic Arts cdata @tab @tab X @item Electronic Arts Multimedia @tab @tab X @tab Used in various EA games; files have extensions like WVE and UV2. +@item Ensoniq Paris Audio File @tab @tab X +@item FFM (FFserver live feed) @tab X @tab X @item Flash (SWF) @tab X @tab X @item Flash 9 (AVM2) @tab X @tab X @tab Only embedded audio is decoded. @@ -282,15 +312,23 @@ library: @item framecrc testing format @tab X @tab @item FunCom ISS @tab @tab X @tab Audio format used in various games from FunCom like The Longest Journey. -@item GIF Animation @tab X @tab +@item G.723.1 @tab X @tab X +@item G.729 BIT @tab X @tab X +@item G.729 raw @tab @tab X +@item GENH @tab @tab X + @tab Audio format for various games. +@item GIF Animation @tab X @tab X @item GXF @tab X @tab X @tab General eXchange Format SMPTE 360M, used by Thomson Grass Valley playout servers. @item HNM @tab @tab X @tab Only version 4 supported, used in some games from Cryo Interactive +@item iCEDraw File @tab @tab X +@item ICO @tab X @tab X + @tab Microsoft Windows ICO @item id Quake II CIN video @tab @tab X @item id RoQ @tab X @tab X - @tab Used in Quake III, Jedi Knight 2, other computer games. + @tab Used in Quake III, Jedi Knight 2 and other computer games. @item IEC61937 encapsulation @tab X @tab X @item IFF @tab @tab X @tab Interchange File Format @@ -301,11 +339,18 @@ library: @tab A format generated by IndigoVision 8000 video server. @item IVF (On2) @tab X @tab X @tab A format used by libvpx +@item Internet Video Recording @tab @tab X +@item IRCAM @tab X @tab X @item LATM @tab X @tab X @item LMLM4 @tab @tab X @tab Used by Linux Media Labs MPEG-4 PCI boards +@item LOAS @tab @tab X + @tab contains LATM multiplexed AAC audio +@item LRC @tab X @tab X +@item LVF @tab @tab X @item LXF @tab @tab X @tab VR native stream format, used by Leitch/Harris' video servers. +@item Magic Lantern Video (MLV) @tab @tab X @item Matroska @tab X @tab X @item Matroska audio @tab X @tab @item FFmpeg metadata @tab X @tab X @@ -313,6 +358,9 @@ library: @item MAXIS XA @tab @tab X @tab Used in Sim City 3000; file extension .xa. @item MD Studio @tab @tab X +@item Metal Gear Solid: The Twin Snakes @tab @tab X +@item Megalux Frame @tab @tab X + @tab Used by Megalux Ultimate Paint @item Mobotix .mxg @tab @tab X @item Monkey's Audio @tab @tab X @item Motion Pixels MVI @tab @tab X @@ -328,6 +376,8 @@ library: @tab also known as DVB Transport Stream @item MPEG-4 @tab X @tab X @tab MPEG-4 is a variant of QuickTime. +@item MSF @tab @tab X + @tab Audio format used on the PS3. @item Mirillis FIC video @tab @tab X @tab No cursor rendering. @item MIME multipart JPEG @tab X @tab @@ -342,6 +392,7 @@ library: @tab SMPTE 386M, D-10/IMX Mapping. @item NC camera feed @tab @tab X @tab NC (AVIP NC4600) camera streams +@item NIST SPeech HEader REsources @tab @tab X @item NTT TwinVQ (VQF) @tab @tab X @tab Nippon Telegraph and Telephone Corporation TwinVQ. @item Nullsoft Streaming Video @tab @tab X @@ -350,6 +401,7 @@ library: @tab NUT Open Container Format @item Ogg @tab X @tab X @item Playstation Portable PMP @tab @tab X +@item Portable Voice Format @tab @tab X @item TechnoTrend PVA @tab @tab X @tab Used by TechnoTrend DVB PCI boards. @item QCP @tab @tab X @@ -360,6 +412,7 @@ library: @item raw Dirac @tab X @tab X @item raw DNxHD @tab X @tab X @item raw DTS @tab X @tab X +@item raw DTS-HD @tab @tab X @item raw E-AC-3 @tab X @tab X @item raw FLAC @tab X @tab X @item raw GSM @tab @tab X @@ -380,7 +433,7 @@ library: @item raw Shorten @tab @tab X @item raw TAK @tab @tab X @item raw TrueHD @tab X @tab X -@item raw VC-1 @tab @tab X +@item raw VC-1 @tab X @tab X @item raw PCM A-law @tab X @tab X @item raw PCM mu-law @tab X @tab X @item raw PCM signed 8 bit @tab X @tab X @@ -406,18 +459,20 @@ library: @tab File format used by RED Digital cameras, contains JPEG 2000 frames and PCM audio. @item RealMedia @tab X @tab X @item Redirector @tab @tab X +@item RedSpark @tab @tab X @item Renderware TeXture Dictionary @tab @tab X @item Resolume DXV @tab @tab X @item RL2 @tab @tab X @tab Audio and video format used in some games by Entertainment Software Partners. @item RPL/ARMovie @tab @tab X @item Lego Mindstorms RSO @tab X @tab X +@item RSD @tab @tab X @item RTMP @tab X @tab X @tab Output is performed by publishing stream to RTMP server @item RTP @tab X @tab X @item RTSP @tab X @tab X @item SAP @tab X @tab X -@item Screenpresso @tab @tab X +@item SBG @tab @tab X @item SDP @tab @tab X @item Sega FILM/CPK @tab @tab X @tab Used in many Sega Saturn console games. @@ -430,14 +485,17 @@ library: @tab Multimedia format used by many games. @item SMJPEG @tab X @tab X @tab Used in certain Loki game ports. -@item Smush +@item Smush @tab @tab X @tab Multimedia format used in some LucasArts games. @item Sony OpenMG (OMA) @tab X @tab X @tab Audio format used in Sony Sonic Stage and Sony Vegas. @item Sony PlayStation STR @tab @tab X -@item Sony Wave64 (W64) @tab @tab X +@item Sony Wave64 (W64) @tab X @tab X @item SoX native format @tab X @tab X @item SUN AU format @tab X @tab X +@item SUP raw PGS subtitles @tab @tab X +@item SVAG @tab @tab X + @tab Audio format used in Konami PS2 games. @item TDSC @tab @tab X @item Text files @tab @tab X @item THP @tab @tab X @@ -445,32 +503,40 @@ library: @item Tiertex Limited SEQ @tab @tab X @tab Tiertex .seq files used in the DOS CD-ROM version of the game Flashback. @item True Audio @tab @tab X +@item VAG @tab @tab X + @tab Audio format used in many Sony PS2 games. @item VC-1 test bitstream @tab X @tab X @item Vidvox Hap @tab X @tab X +@item Vivo @tab @tab X +@item VPK @tab @tab X + @tab Audio format used in Sony PS games. @item WAV @tab X @tab X -@item WavPack @tab @tab X +@item WavPack @tab X @tab X @item WebM @tab X @tab X -@item Windows Televison (WTV) @tab @tab X +@item Windows Televison (WTV) @tab X @tab X @item Wing Commander III movie @tab @tab X @tab Multimedia format used in Origin's Wing Commander III computer game. @item Westwood Studios audio @tab @tab X @tab Multimedia format used in Westwood Studios games. @item Westwood Studios VQA @tab @tab X @tab Multimedia format used in Westwood Studios games. +@item WVE @tab @tab X @item XMV @tab @tab X @tab Microsoft video container used in Xbox games. +@item XVAG @tab @tab X + @tab Audio format used on the PS3. @item xWMA @tab @tab X @tab Microsoft audio container used by XAudio 2. +@item eXtended BINary text (XBIN) @tab @tab X @item YUV4MPEG pipe @tab X @tab X @item Psygnosis YOP @tab @tab X -@item ZeroCodec Lossless Video @tab @tab X @end multitable @code{X} means that encoding (resp. decoding) is supported. @section Image Formats -Libav can read and write images for each frame of a video sequence. The +FFmpeg can read and write images for each frame of a video sequence. The following image formats are supported: @multitable @columnfractions .4 .1 .1 .4 @@ -480,7 +546,7 @@ following image formats are supported: @item Alias PIX @tab X @tab X @tab Alias/Wavefront PIX image format @item animated GIF @tab X @tab X - @tab Only uncompressed GIFs are generated. +@item APNG @tab X @tab X @item BMP @tab X @tab X @tab Microsoft BMP image @item BRender PIX @tab @tab X @@ -491,8 +557,7 @@ following image formats are supported: @tab OpenEXR @item JPEG @tab X @tab X @tab Progressive JPEG is not supported. -@item JPEG 2000 @tab E @tab X - @tab encoding supported through external library libopenjpeg +@item JPEG 2000 @tab X @tab X @item JPEG-LS @tab X @tab X @item LJPEG @tab X @tab @tab Lossless JPEG @@ -509,7 +574,6 @@ following image formats are supported: @item PIC @tab @tab X @tab Pictor/PC Paint @item PNG @tab X @tab X - @tab 2/4 bpp not supported yet @item PPM @tab X @tab X @tab Portable PixelMap image @item PTX @tab @tab X @@ -526,6 +590,8 @@ following image formats are supported: @tab WebP image format, encoding supported through external library libwebp @item XBM @tab X @tab X @tab X BitMap image format +@item XFace @tab X @tab X + @tab X-Face image format @item XWD @tab X @tab X @tab X Window Dump image format @end multitable @@ -541,14 +607,12 @@ following image formats are supported: @item 4X Movie @tab @tab X @tab Used in certain computer games. @item 8088flex TMV @tab @tab X -@item 8SVX exponential @tab @tab X -@item 8SVX fibonacci @tab @tab X @item A64 multicolor @tab X @tab @tab Creates video suitable to be played on a commodore 64 (multicolor mode). @item Amazing Studio PAF Video @tab @tab X @item American Laser Games MM @tab @tab X @tab Used in games like Mad Dog McCree. -@item AMV Video @tab @tab X +@item AMV Video @tab X @tab X @tab Used in Chinese MP3 players. @item ANSI/ASCII art @tab @tab X @item Apple Intermediate Codec @tab @tab X @@ -569,13 +633,18 @@ following image formats are supported: @item Autodesk Animator Flic video @tab @tab X @item Autodesk RLE @tab @tab X @tab fourcc: AASC +@item Avid 1:1 10-bit RGB Packer @tab X @tab X + @tab fourcc: AVrp @item AVS (Audio Video Standard) video @tab @tab X @tab Video encoding used by the Creature Shock game. +@item AYUV @tab X @tab X + @tab Microsoft uncompressed packed 4:4:4:4 @item Beam Software VB @tab @tab X @item Bethesda VID video @tab @tab X @tab Used in some games from Bethesda Softworks. @item Bink Video @tab @tab X @item Bitmap Brothers JV video @tab @tab X +@item y41p Brooktree uncompressed 4:1:1 12-bit @tab X @tab X @item Brute Force & Ignorance @tab @tab X @tab Used in the game Flash Traffic: City of Angels. @item C93 video @tab @tab X @@ -595,10 +664,11 @@ following image formats are supported: @item Cinepak @tab @tab X @item Cirrus Logic AccuPak @tab X @tab X @tab fourcc: CLJR +@item CPiA Video Format @tab @tab X @item Creative YUV (CYUV) @tab @tab X @item DFA @tab @tab X @tab Codec used in Chronomaster game. -@item Dirac @tab E @tab E +@item Dirac @tab E @tab X @tab supported through external library libschroedinger @item Deluxe Paint Animation @tab @tab X @item DNxHD @tab X @tab X @@ -620,10 +690,10 @@ following image formats are supported: @item Escape 124 @tab @tab X @item Escape 130 @tab @tab X @item FFmpeg video codec #1 @tab X @tab X - @tab experimental lossless codec (fourcc: FFV1) + @tab lossless codec (fourcc: FFV1) @item Flash Screen Video v1 @tab X @tab X @tab fourcc: FSV1 -@item Flash Screen Video v2 @tab @tab X +@item Flash Screen Video v2 @tab X @tab X @item Flash Video (FLV) @tab X @tab X @tab Sorenson H.263 used in Flash @item Forward Uncompressed @tab @tab X @@ -661,6 +731,7 @@ following image formats are supported: @tab Used in the game Cyberia from Interplay. @item Interplay MVE video @tab @tab X @tab Used in Interplay .MVE files. +@item J2K @tab X @tab X @item Karl Morton's video codec @tab @tab X @tab Codec used in Worms games. @item Kega Game Video (KGV1) @tab @tab X @@ -669,8 +740,8 @@ following image formats are supported: @item LCL (LossLess Codec Library) MSZH @tab @tab X @item LCL (LossLess Codec Library) ZLIB @tab E @tab E @item LOCO @tab @tab X -@item LucasArts SANM @tab @tab X - @tab Used in LucasArts SMUSH animations. +@item LucasArts SANM/Smush @tab @tab X + @tab Used in LucasArts games / SMUSH animations. @item lossless MJPEG @tab X @tab X @item Microsoft ATC Screen @tab @tab X @tab Also known as Microsoft Screen 3. @@ -710,9 +781,11 @@ following image formats are supported: @item VP8 @tab E @tab X @tab fourcc: VP80, encoding supported through external library libvpx @item VP9 @tab E @tab X - @tab Encoding supported through external library libvpx -@item planar RGB @tab @tab X - @tab fourcc: 8BPS + @tab encoding supported through external library libvpx +@item Pinnacle TARGA CineWave YUV16 @tab @tab X + @tab fourcc: Y216 +@item Prores @tab @tab X + @tab fourcc: apch,apcn,apcs,apco @item Q-team QPEG @tab @tab X @tab fourccs: QPEG, Q1.0, Q1.1 @item QuickTime 8BPS video @tab @tab X @@ -722,8 +795,8 @@ following image formats are supported: @tab fourcc: 'smc ' @item QuickTime video (RPZA) @tab @tab X @tab fourcc: rpza -@item R10K AJA Kona 10-bit RGB Codec @tab @tab X -@item R210 Quicktime Uncompressed RGB 10-bit @tab @tab X +@item R10K AJA Kona 10-bit RGB Codec @tab X @tab X +@item R210 Quicktime Uncompressed RGB 10-bit @tab X @tab X @item Raw Video @tab X @tab X @item RealVideo 1.0 @tab X @tab X @item RealVideo 2.0 @tab X @tab X @@ -734,6 +807,7 @@ following image formats are supported: @tab Texture dictionaries used by the Renderware Engine. @item RL2 video @tab @tab X @tab used in some games by Entertainment Software Partners +@item Screenpresso @tab @tab X @item Sierra VMD video @tab @tab X @tab Used in Sierra VMD files. @item Silicon Graphics Motion Video Compressor 1 (MVC1) @tab @tab X @@ -742,6 +816,8 @@ following image formats are supported: @item Smacker video @tab @tab X @tab Video encoding used in Smacker. @item SMPTE VC-1 @tab @tab X +@item Snow @tab X @tab X + @tab experimental wavelet codec (fourcc: SNOW) @item Sony PlayStation MDEC (Motion DECoder) @tab @tab X @item Sorenson Vector Quantizer 1 @tab X @tab X @tab fourcc: SVQ1 @@ -759,6 +835,8 @@ following image formats are supported: @tab Codec used in DOS CD-ROM FlashBack game. @item Ut Video @tab X @tab X @item v210 QuickTime uncompressed 4:2:2 10-bit @tab X @tab X +@item v308 QuickTime uncompressed 4:4:4 @tab X @tab X +@item v408 QuickTime uncompressed 4:4:4:4 @tab X @tab X @item v410 QuickTime uncompressed 4:4:4 10-bit @tab X @tab X @item VBLE Lossless Codec @tab @tab X @item VMware Screen Codec / VMware Video @tab @tab X @@ -777,6 +855,9 @@ following image formats are supported: @item WMV7 @tab X @tab X @item YAMAHA SMAF @tab X @tab X @item Psygnosis YOP Video @tab @tab X +@item yuv4 @tab X @tab X + @tab libquicktime uncompressed packed 4:2:0 +@item ZeroCodec Lossless Video @tab @tab X @item ZLIB @tab X @tab X @tab part of LCL, encoder experimental @item Zip Motion Blocks Video @tab X @tab X @@ -791,11 +872,15 @@ following image formats are supported: @multitable @columnfractions .4 .1 .1 .4 @item Name @tab Encoding @tab Decoding @tab Comments -@item 8SVX audio @tab @tab X -@item AAC @tab E @tab X - @tab encoding supported through external library libfaac and libvo-aacenc -@item AC-3 @tab IX @tab X +@item 8SVX exponential @tab @tab X +@item 8SVX fibonacci @tab @tab X +@item AAC @tab EX @tab X + @tab encoding supported through internal encoder and external libraries libfaac and libfdk-aac +@item AAC+ @tab E @tab IX + @tab encoding supported through external library libfdk-aac +@item AC-3 @tab IX @tab IX @item ADPCM 4X Movie @tab @tab X +@item APDCM Yamaha AICA @tab @tab X @item ADPCM CDROM XA @tab @tab X @item ADPCM Creative Technology @tab @tab X @tab 16 -> 4, 8 -> 4, 8 -> 3, 8 -> 2 @@ -820,23 +905,26 @@ following image formats are supported: @item ADPCM IMA Westwood @tab @tab X @item ADPCM ISS IMA @tab @tab X @tab Used in FunCom games. +@item ADPCM IMA Dialogic @tab @tab X @item ADPCM IMA Duck DK3 @tab @tab X @tab Used in some Sega Saturn console games. @item ADPCM IMA Duck DK4 @tab @tab X @tab Used in some Sega Saturn console games. +@item ADPCM IMA Radical @tab @tab X @item ADPCM Microsoft @tab X @tab X @item ADPCM MS IMA @tab X @tab X -@item ADPCM Nintendo Gamecube THP @tab @tab X +@item ADPCM Nintendo Gamecube AFC @tab @tab X +@item ADPCM Nintendo Gamecube DTK @tab @tab X +@item ADPCM Nintendo THP @tab @tab X +@item APDCM Playstation @tab @tab X @item ADPCM QT IMA @tab X @tab X @item ADPCM SEGA CRI ADX @tab X @tab X @tab Used in Sega Dreamcast games. @item ADPCM Shockwave Flash @tab X @tab X -@item ADPCM SMJPEG IMA @tab @tab X - @tab Used in certain Loki game ports. @item ADPCM Sound Blaster Pro 2-bit @tab @tab X @item ADPCM Sound Blaster Pro 2.6-bit @tab @tab X @item ADPCM Sound Blaster Pro 4-bit @tab @tab X -@item ADPCM VIMA +@item ADPCM VIMA @tab @tab X @tab Used in LucasArts SMUSH animations. @item ADPCM Westwood Studios IMA @tab @tab X @tab Used in Westwood Studios games like Command and Conquer. @@ -853,28 +941,38 @@ following image formats are supported: @item ATRAC3+ @tab @tab X @item Bink Audio @tab @tab X @tab Used in Bink and Smacker files in many games. +@item CELT @tab @tab E + @tab decoding supported through external library libcelt @item Delphine Software International CIN audio @tab @tab X @tab Codec used in Delphine Software International games. @item Digital Speech Standard - Standard Play mode (DSS SP) @tab @tab X @item Discworld II BMV Audio @tab @tab X @item COOK @tab @tab X @tab All versions except 5.1 are supported. -@item DCA (DTS Coherent Acoustics) @tab @tab X +@item DCA (DTS Coherent Acoustics) @tab X @tab X @tab supported extensions: XCh, XLL (partially) @item DPCM id RoQ @tab X @tab X - @tab Used in Quake III, Jedi Knight 2, other computer games. + @tab Used in Quake III, Jedi Knight 2 and other computer games. @item DPCM Interplay @tab @tab X @tab Used in various Interplay computer games. +@item DPCM Squareroot-Delta-Exact @tab @tab X + @tab Used in various games. @item DPCM Sierra Online @tab @tab X @tab Used in Sierra Online game audio files. @item DPCM Sol @tab @tab X @item DPCM Xan @tab @tab X @tab Used in Origin's Wing Commander IV AVI files. +@item DSD (Direct Stream Digitial), least significant bit first @tab @tab X +@item DSD (Direct Stream Digitial), most significant bit first @tab @tab X +@item DSD (Direct Stream Digitial), least significant bit first, planar @tab @tab X +@item DSD (Direct Stream Digitial), most significant bit first, planar @tab @tab X @item DSP Group TrueSpeech @tab @tab X @item DV audio @tab @tab X @item Enhanced AC-3 @tab X @tab X +@item EVRC (Enhanced Variable Rate Codec) @tab @tab X @item FLAC (Free Lossless Audio Codec) @tab X @tab IX -@item G.723.1 @tab X @tab X +@item G.723.1 @tab X @tab X +@item G.729 @tab @tab X @item GSM @tab E @tab X @tab encoding supported through external library libgsm @item GSM Microsoft variant @tab E @tab X @@ -883,14 +981,14 @@ following image formats are supported: @item iLBC (Internet Low Bitrate Codec) @tab E @tab E @tab encoding and decoding supported through external library libilbc @item IMC (Intel Music Coder) @tab @tab X +@item Interplay ACM @tab @tab X @item MACE (Macintosh Audio Compression/Expansion) 3:1 @tab @tab X @item MACE (Macintosh Audio Compression/Expansion) 6:1 @tab @tab X @item MLP (Meridian Lossless Packing) @tab @tab X @tab Used in DVD-Audio discs. @item Monkey's Audio @tab @tab X - @tab Only versions 3.97-3.99 are supported. @item MP1 (MPEG audio layer 1) @tab @tab IX -@item MP2 (MPEG audio layer 2) @tab IE @tab IX +@item MP2 (MPEG audio layer 2) @tab IX @tab IX @tab encoding supported also through external library TwoLAME @item MP3 (MPEG audio layer 3) @tab E @tab IX @tab encoding supported through external library LAME, ADU MP3 and MP3onMP4 also supported @@ -899,14 +997,15 @@ following image formats are supported: @item Musepack SV8 @tab @tab X @item Nellymoser Asao @tab X @tab X @item On2 AVC (Audio for Video Codec) @tab @tab X -@item Opus @tab E @tab E - @tab supported through external library libopus +@item Opus @tab E @tab X + @tab encoding supported through external library libopus @item PCM A-law @tab X @tab X @item PCM mu-law @tab X @tab X -@item PCM signed 16-bit big-endian planar @tab @tab X -@item PCM signed 16-bit little-endian planar @tab @tab X -@item PCM signed 24-bit little-endian planar @tab @tab X -@item PCM signed 32-bit little-endian planar @tab @tab X +@item PCM signed 8-bit planar @tab X @tab X +@item PCM signed 16-bit big-endian planar @tab X @tab X +@item PCM signed 16-bit little-endian planar @tab X @tab X +@item PCM signed 24-bit little-endian planar @tab X @tab X +@item PCM signed 32-bit little-endian planar @tab X @tab X @item PCM 32-bit floating point big-endian @tab X @tab X @item PCM 32-bit floating point little-endian @tab X @tab X @item PCM 64-bit floating point big-endian @tab X @tab X @@ -943,25 +1042,32 @@ following image formats are supported: @item Sierra VMD audio @tab @tab X @tab Used in Sierra VMD files. @item Smacker audio @tab @tab X -@item SMPTE 302M AES3 audio @tab @tab X +@item SMPTE 302M AES3 audio @tab X @tab X +@item Sonic @tab X @tab X + @tab experimental codec +@item Sonic lossless @tab X @tab X + @tab experimental codec @item Speex @tab E @tab E @tab supported through external library libspeex @item TAK (Tom's lossless Audio Kompressor) @tab @tab X -@item True Audio (TTA) @tab @tab X +@item True Audio (TTA) @tab X @tab X @item TrueHD @tab @tab X @tab Used in HD-DVD and Blu-Ray discs. @item TwinVQ (VQF flavor) @tab @tab X +@item VIMA @tab @tab X + @tab Used in LucasArts SMUSH animations. @item Vorbis @tab E @tab X @tab A native but very primitive encoder exists. @item Voxware MetaSound @tab @tab X -@item WavPack @tab E @tab X - @tab supported through external library libwavpack +@item WavPack @tab X @tab X @item Westwood Audio (SND1) @tab @tab X @item Windows Media Audio 1 @tab X @tab X @item Windows Media Audio 2 @tab X @tab X @item Windows Media Audio Lossless @tab @tab X @item Windows Media Audio Pro @tab @tab X @item Windows Media Audio Voice @tab @tab X +@item Xbox Media Audio 1 @tab @tab X +@item Xbox Media Audio 2 @tab @tab X @end multitable @code{X} means that encoding (resp. decoding) is supported. @@ -975,21 +1081,41 @@ performance on systems without hardware floating point support). @multitable @columnfractions .4 .1 .1 .1 .1 @item Name @tab Muxing @tab Demuxing @tab Encoding @tab Decoding -@item SSA/ASS @tab X @tab X @tab X @tab X -@item DVB @tab X @tab X @tab X @tab X -@item DVD @tab X @tab X @tab X @tab X -@item PGS @tab @tab @tab @tab X -@item SubRip (SRT) @tab X @tab X @tab @tab X -@item XSUB @tab @tab @tab X @tab X +@item 3GPP Timed Text @tab @tab @tab X @tab X +@item AQTitle @tab @tab X @tab @tab X +@item DVB @tab X @tab X @tab X @tab X +@item DVB teletext @tab @tab X @tab @tab E +@item DVD @tab X @tab X @tab X @tab X +@item JACOsub @tab X @tab X @tab @tab X +@item MicroDVD @tab X @tab X @tab @tab X +@item MPL2 @tab @tab X @tab @tab X +@item MPsub (MPlayer) @tab @tab X @tab @tab X +@item PGS @tab @tab @tab @tab X +@item PJS (Phoenix) @tab @tab X @tab @tab X +@item RealText @tab @tab X @tab @tab X +@item SAMI @tab @tab X @tab @tab X +@item Spruce format (STL) @tab @tab X @tab @tab X +@item SSA/ASS @tab X @tab X @tab X @tab X +@item SubRip (SRT) @tab X @tab X @tab X @tab X +@item SubViewer v1 @tab @tab X @tab @tab X +@item SubViewer @tab @tab X @tab @tab X +@item TED Talks captions @tab @tab X @tab @tab X +@item VobSub (IDX+SUB) @tab @tab X @tab @tab X +@item VPlayer @tab @tab X @tab @tab X +@item WebVTT @tab X @tab X @tab X @tab X +@item XSUB @tab @tab @tab X @tab X @end multitable @code{X} means that the feature is supported. +@code{E} means that support is provided through an external library. + @section Network Protocols @multitable @columnfractions .4 .1 @item Name @tab Support @item file @tab X +@item FTP @tab X @item Gopher @tab X @item HLS @tab X @item HTTP @tab X @@ -1005,7 +1131,9 @@ performance on systems without hardware floating point support). @item RTMPTE @tab X @item RTMPTS @tab X @item RTP @tab X +@item SAMBA @tab E @item SCTP @tab X +@item SFTP @tab E @item TCP @tab X @item TLS @tab X @item UDP @tab X @@ -1022,18 +1150,36 @@ performance on systems without hardware floating point support). @item Name @tab Input @tab Output @item ALSA @tab X @tab X @item BKTR @tab X @tab +@item caca @tab @tab X @item DV1394 @tab X @tab -@item Linux framebuffer @tab X @tab +@item Lavfi virtual device @tab X @tab +@item Linux framebuffer @tab X @tab X @item JACK @tab X @tab @item LIBCDIO @tab X @item LIBDC1394 @tab X @tab +@item OpenAL @tab X +@item OpenGL @tab @tab X @item OSS @tab X @tab X -@item Pulseaudio @tab X @tab -@item Video4Linux2 @tab X @tab +@item PulseAudio @tab X @tab X +@item SDL @tab @tab X +@item Video4Linux2 @tab X @tab X @item VfW capture @tab X @tab @item X11 grabbing @tab X @tab +@item Win32 grabbing @tab X @tab @end multitable @code{X} means that input/output is supported. +@section Timecode + +@multitable @columnfractions .4 .1 .1 +@item Codec/format @tab Read @tab Write +@item AVI @tab X @tab X +@item DV @tab X @tab X +@item GXF @tab X @tab X +@item MOV @tab X @tab X +@item MPEG1/2 @tab X @tab X +@item MXF @tab X @tab X +@end multitable + @bye diff --git a/doc/git-howto.texi b/doc/git-howto.texi index 5a8e2a3823..e5e3c81795 100644 --- a/doc/git-howto.texi +++ b/doc/git-howto.texi @@ -1,9 +1,10 @@ \input texinfo @c -*- texinfo -*- +@documentencoding UTF-8 -@settitle Using git to develop Libav +@settitle Using Git to develop FFmpeg @titlepage -@center @titlefont{Using git to develop Libav} +@center @titlefont{Using Git to develop FFmpeg} @end titlepage @top @@ -12,9 +13,9 @@ @chapter Introduction -This document aims in giving some quick references on a set of useful git +This document aims in giving some quick references on a set of useful Git commands. You should always use the extensive and detailed documentation -provided directly by git: +provided directly by Git: @example git --help @@ -31,40 +32,54 @@ man git-<command> shows information about the subcommand <command>. Additional information could be found on the -@url{http://gitref.org, Git Reference} website +@url{http://gitref.org, Git Reference} website. For more information about the Git project, visit the - -@url{http://git-scm.com/, Git website} +@url{http://git-scm.com/, Git website}. Consult these resources whenever you have problems, they are quite exhaustive. -What follows now is a basic introduction to Git and some Libav-specific -guidelines to ease the contribution to the project +What follows now is a basic introduction to Git and some FFmpeg-specific +guidelines to ease the contribution to the project. @chapter Basics Usage -@section Get GIT +@section Get Git -You can get git from @url{http://git-scm.com/} +You can get Git from @url{http://git-scm.com/} Most distribution and operating system provide a package for it. @section Cloning the source tree @example -git clone git://git.libav.org/libav.git <target> +git clone git://source.ffmpeg.org/ffmpeg <target> @end example -This will put the Libav sources into the directory @var{<target>}. +This will put the FFmpeg sources into the directory @var{<target>}. @example -git clone git@@git.libav.org:libav.git <target> +git clone git@@source.ffmpeg.org:ffmpeg <target> @end example -This will put the Libav sources into the directory @var{<target>} and let +This will put the FFmpeg sources into the directory @var{<target>} and let you push back your changes to the remote repository. +@example +git clone gil@@ffmpeg.org:ffmpeg-web <target> +@end example + +This will put the source of the FFmpeg website into the directory +@var{<target>} and let you push back your changes to the remote repository. +(Note that @var{gil} stands for GItoLite and is not a typo of @var{git}.) + +If you don't have write-access to the ffmpeg-web repository, you can +create patches after making a read-only ffmpeg-web clone: + +@example +git clone git://ffmpeg.org/ffmpeg-web <target> +@end example + Make sure that you do not have Windows line endings in your checkouts, otherwise you may experience spurious compilation failures. One way to achieve this is to run @@ -74,6 +89,7 @@ git config --global core.autocrlf false @end example +@anchor{Updating the source tree to the latest revision} @section Updating the source tree to the latest revision @example @@ -85,7 +101,7 @@ can be remote. By default the master branch tracks the branch master in the remote origin. @float IMPORTANT -Since merge commits are forbidden @command{--rebase} (see below) is recommended. +@command{--rebase} (see below) is recommended. @end float @section Rebasing your local branches @@ -96,7 +112,7 @@ git pull --rebase fetches the changes from the main repository and replays your local commits over it. This is required to keep all your local changes at the top of -Libav's master tree. The master tree will reject pushes with merge commits. +FFmpeg's master tree. The master tree will reject pushes with merge commits. @section Adding/removing files/directories @@ -106,7 +122,7 @@ git add [-A] <filename/dirname> git rm [-r] <filename/dirname> @end example -GIT needs to get notified of all changes you make to your working +Git needs to get notified of all changes you make to your working directory that makes files appear or disappear. Line moves across files are automatically tracked. @@ -126,8 +142,8 @@ will show all local modifications in your working directory as unified diff. git log <filename(s)> @end example -You may also use the graphical tools like gitview or gitk or the web -interface available at http://git.libav.org/ +You may also use the graphical tools like @command{gitview} or @command{gitk} +or the web interface available at @url{http://source.ffmpeg.org/}. @section Checking source tree status @@ -148,6 +164,7 @@ git diff --check to double check your changes before committing them to avoid trouble later on. All experienced developers do this on each and every commit, no matter how small. + Every one of them has been saved from looking like a fool by this many times. It's very easy for stray debug output or cosmetic modifications to slip in, please avoid problems through this extra level of scrutiny. @@ -170,14 +187,14 @@ to make sure you don't have untracked files or deletions. git add [-i|-p|-A] <filenames/dirnames> @end example -Make sure you have told git your name and email address +Make sure you have told Git your name and email address @example git config --global user.name "My Name" git config --global user.email my@@email.invalid @end example -Use @var{--global} to set the global configuration for all your git checkouts. +Use @option{--global} to set the global configuration for all your Git checkouts. Git will select the changes to the files for commit. Optionally you can use the interactive or the patch mode to select hunk by hunk what should be @@ -208,7 +225,7 @@ include filenames in log messages, Git provides that information. Possibly make the commit message have a terse, descriptive first line, an empty line and then a full description. The first line will be used to name -the patch by git format-patch. +the patch by @command{git format-patch}. @section Preparing a patchset @@ -261,7 +278,7 @@ git commit @chapter Git configuration In order to simplify a few workflows, it is advisable to configure both -your personal Git installation and your local Libav repository. +your personal Git installation and your local FFmpeg repository. @section Personal Git installation @@ -276,15 +293,15 @@ and @command{git format-patch} detect renames: @section Repository configuration In order to have @command{git send-email} automatically send patches -to the libav-devel mailing list, add the following stanza -to @file{/path/to/libav/repository/.git/config}: +to the ffmpeg-devel mailing list, add the following stanza +to @file{/path/to/ffmpeg/repository/.git/config}: @example [sendemail] - to = libav-devel@@libav.org + to = ffmpeg-devel@@ffmpeg.org @end example -@chapter Libav specific +@chapter FFmpeg specific @section Reverting broken commits @@ -299,7 +316,7 @@ the current branch history. git commit --amend @end example -allows to amend the last commit details quickly. +allows one to amend the last commit details quickly. @example git rebase -i origin/master @@ -324,12 +341,14 @@ faulty commit disappear from the history. @section Pushing changes to remote trees @example -git push +git push origin master --dry-run @end example -Will push the changes to the default remote (@var{origin}). +Will simulate a push of the local master branch to the default remote +(@var{origin}). And list which branches and ranges or commits would have been +pushed. Git will prevent you from pushing changes if the local and remote trees are -out of sync. Refer to and to sync the local tree. +out of sync. Refer to @ref{Updating the source tree to the latest revision}. @example git remote add <name> <url> @@ -348,23 +367,24 @@ branches matching the local ones. @section Finding a specific svn revision -Since version 1.7.1 git supports @var{:/foo} syntax for specifying commits +Since version 1.7.1 Git supports @samp{:/foo} syntax for specifying commits based on a regular expression. see man gitrevisions @example git show :/'as revision 23456' @end example -will show the svn changeset @var{r23456}. With older git versions searching in +will show the svn changeset @samp{r23456}. With older Git versions searching in the @command{git log} output is the easiest option (especially if a pager with search capabilities is used). + This commit can be checked out with @example git checkout -b svn_23456 :/'as revision 23456' @end example -or for git < 1.7.1 with +or for Git < 1.7.1 with @example git checkout -b svn_23456 $SHA1 @@ -373,7 +393,7 @@ git checkout -b svn_23456 $SHA1 where @var{$SHA1} is the commit hash from the @command{git log} output. -@chapter pre-push checklist +@chapter Pre-push checklist Once you have a set of commits that you feel are ready for pushing, work through the following checklist to doublecheck everything is in @@ -381,60 +401,35 @@ proper order. This list tries to be exhaustive. In case you are just pushing a typo in a comment, some of the steps may be unnecessary. Apply your common sense, but if in doubt, err on the side of caution. -First make sure your Git repository is on a branch that is a direct -descendant of the Libav master branch, which is the only one from which -pushing to Libav is possible. Then run the following command: - -@itemize -@item @command{git log --patch --stat origin/master..} +First, make sure that the commits and branches you are going to push +match what you want pushed and that nothing is missing, extraneous or +wrong. You can see what will be pushed by running the git push command +with @option{--dry-run} first. And then inspecting the commits listed with +@command{git log -p 1234567..987654}. The @command{git status} command +may help in finding local changes that have been forgotten to be added. -to make sure that only the commits you want to push are pending, that -the log messages of the commits are correct and descriptive and contain -no cruft from @command{git am} and to doublecheck that the commits you -want to push really only contain the changes they are supposed to contain. - -@item @command{git status} - -to ensure no local changes still need to be committed and that no local -changes may have thrown off the results of your testing. -@end itemize - -Next let the code pass through a full run of our testsuite. Before you do, -the command @command{make fate-rsync} will update the test samples. Changes -to the samples set are not very common and commits depending on samples -changes are delayed for at least 24 hours to allow the new samples to -propagate, so updating it once per day is sufficient. Now execute +Next let the code pass through a full run of our testsuite. @itemize @item @command{make distclean} -@item @command{/path/to/libav/configure} -@item @command{make check} +@item @command{/path/to/ffmpeg/configure} +@item @command{make fate} +@item if fate fails due to missing samples run @command{make fate-rsync} and retry @end itemize -While the test suite covers a wide range of possible problems, it is not -a panacea. Do not hesitate to perform any other tests necessary to convince -yourself that the changes you are about to push actually work as expected. +Make sure all your changes have been checked before pushing them, the +testsuite only checks against regressions and that only to some extend. It does +obviously not check newly added features/code to be working unless you have +added a test for that (which is recommended). Also note that every single commit should pass the test suite, not just -the result of a series of patches. So if you have a series of commits -to push, run the test suite on every single commit. - -Give other developers a reasonable amount of time to look at and review -patches before you push them. Not everybody is online 24/7, but may wish -to look at and comment on a patch nonetheless. The time you leave depends -on the urgency and complexity of the patch. Use your common sense to pick -a timeframe that allows everybody that you think may wish to comment -and/or should comment on the change an opportunity to see it. - -Finally, after pushing, mark all patches as committed on -@url{http://patches.libav.org/,patchwork}. -Sometimes this is not automatically done when a patch has been -slightly modified from the version on the mailing list. -Also update previous incarnations of the patches you push so that -patchwork is not cluttered with cruft. +the result of a series of patches. +Once everything passed, push the changes to your public ffmpeg clone and post a +merge request to ffmpeg-devel. You can also push them directly but this is not +recommended. @chapter Server Issues -Contact the project admins @email{git@@libav.org} if you have technical -problems with the GIT server. +Contact the project admins at @email{root@@ffmpeg.org} if you have technical +problems with the Git server. diff --git a/doc/git-howto.txt b/doc/git-howto.txt deleted file mode 100644 index 036b567084..0000000000 --- a/doc/git-howto.txt +++ /dev/null @@ -1,272 +0,0 @@ - -About Git write access: -~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ - -Before everything else, you should know how to use GIT properly. -Luckily Git comes with excellent documentation. - - git --help - man git - -shows you the available subcommands, - - git <command> --help - man git-<command> - -shows information about the subcommand <command>. - -The most comprehensive manual is the website Git Reference - -http://gitref.org/ - -For more information about the Git project, visit - -http://git-scm.com/ - -Consult these resources whenever you have problems, they are quite exhaustive. - -You do not need a special username or password. -All you need is to provide a ssh public key to the Git server admin. - -What follows now is a basic introduction to Git and some Libav-specific -guidelines. Read it at least once, if you are granted commit privileges to the -Libav project you are expected to be familiar with these rules. - - - -I. BASICS: -========== - -0. Get GIT: - - You can get git from http://git-scm.com/ - - -1. Cloning the source tree: - - git clone git://git.libav.org/libav.git <target> - - This will put the Libav sources into the directory <target>. - - git clone git@git.libav.org:libav.git <target> - - This will put the Libav sources into the directory <target> and let - you push back your changes to the remote repository. - - -2. Updating the source tree to the latest revision: - - git pull (--ff-only) - - pulls in the latest changes from the tracked branch. The tracked branch - can be remote. By default the master branch tracks the branch master in - the remote origin. - Caveat: Since merge commits are forbidden at least for the initial - months of git --ff-only or --rebase (see below) are recommended. - --ff-only will fail and not create merge commits if your branch - has diverged (has a different history) from the tracked branch. - -2.a Rebasing your local branches: - - git pull --rebase - - fetches the changes from the main repository and replays your local commits - over it. This is required to keep all your local changes at the top of - Libav's master tree. The master tree will reject pushes with merge commits. - - -3. Adding/removing files/directories: - - git add [-A] <filename/dirname> - git rm [-r] <filename/dirname> - - GIT needs to get notified of all changes you make to your working - directory that makes files appear or disappear. - Line moves across files are automatically tracked. - - -4. Showing modifications: - - git diff <filename(s)> - - will show all local modifications in your working directory as unified diff. - - -5. Inspecting the changelog: - - git log <filename(s)> - - You may also use the graphical tools like gitview or gitk or the web - interface available at http://git.libav.org/ - -6. Checking source tree status: - - git status - - detects all the changes you made and lists what actions will be taken in case - of a commit (additions, modifications, deletions, etc.). - - -7. Committing: - - git diff --check - - to double check your changes before committing them to avoid trouble later - on. All experienced developers do this on each and every commit, no matter - how small. - Every one of them has been saved from looking like a fool by this many times. - It's very easy for stray debug output or cosmetic modifications to slip in, - please avoid problems through this extra level of scrutiny. - - For cosmetics-only commits you should get (almost) empty output from - - git diff -w -b <filename(s)> - - Also check the output of - - git status - - to make sure you don't have untracked files or deletions. - - git add [-i|-p|-A] <filenames/dirnames> - - Make sure you have told git your name and email address, e.g. by running - git config --global user.name "My Name" - git config --global user.email my@email.invalid - (--global to set the global configuration for all your git checkouts). - - Git will select the changes to the files for commit. Optionally you can use - the interactive or the patch mode to select hunk by hunk what should be - added to the commit. - - git commit - - Git will commit the selected changes to your current local branch. - - You will be prompted for a log message in an editor, which is either - set in your personal configuration file through - - git config core.editor - - or set by one of the following environment variables: - GIT_EDITOR, VISUAL or EDITOR. - - Log messages should be concise but descriptive. Explain why you made a change, - what you did will be obvious from the changes themselves most of the time. - Saying just "bug fix" or "10l" is bad. Remember that people of varying skill - levels look at and educate themselves while reading through your code. Don't - include filenames in log messages, Git provides that information. - - Possibly make the commit message have a terse, descriptive first line, an - empty line and then a full description. The first line will be used to name - the patch by git format-patch. - - -8. Renaming/moving/copying files or contents of files: - - Git automatically tracks such changes, making those normal commits. - - mv/cp path/file otherpath/otherfile - - git add [-A] . - - git commit - - Do not move, rename or copy files of which you are not the maintainer without - discussing it on the mailing list first! - -9. Reverting broken commits - - git revert <commit> - - git revert will generate a revert commit. This will not make the faulty - commit disappear from the history. - - git reset <commit> - - git reset will uncommit the changes till <commit> rewriting the current - branch history. - - git commit --amend - - allows to amend the last commit details quickly. - - git rebase -i origin/master - - will replay local commits over the main repository allowing to edit, - merge or remove some of them in the process. - - Note that the reset, commit --amend and rebase rewrite history, so you - should use them ONLY on your local or topic branches. - - The main repository will reject those changes. - -10. Preparing a patchset. - - git format-patch <commit> [-o directory] - - will generate a set of patches for each commit between <commit> and - current HEAD. E.g. - - git format-patch origin/master - - will generate patches for all commits on current branch which are not - present in upstream. - A useful shortcut is also - - git format-patch -n - - which will generate patches from last n commits. - By default the patches are created in the current directory. - -11. Sending patches for review - - git send-email <commit list|directory> - - will send the patches created by git format-patch or directly generates - them. All the email fields can be configured in the global/local - configuration or overridden by command line. - Note that this tool must often be installed separately (e.g. git-email - package on Debian-based distros). - -12. Pushing changes to remote trees - - git push - - Will push the changes to the default remote (origin). - Git will prevent you from pushing changes if the local and remote trees are - out of sync. Refer to 2 and 2.a to sync the local tree. - - git remote add <name> <url> - - Will add additional remote with a name reference, it is useful if you want - to push your local branch for review on a remote host. - - git push <remote> <refspec> - - Will push the changes to the remote repository. Omitting refspec makes git - push update all the remote branches matching the local ones. - -13. Finding a specific svn revision - - Since version 1.7.1 git supports ':/foo' syntax for specifying commits - based on a regular expression. see man gitrevisions - - git show :/'as revision 23456' - - will show the svn changeset r23456. With older git versions searching in - the git log output is the easiest option (especially if a pager with - search capabilities is used). - This commit can be checked out with - - git checkout -b svn_23456 :/'as revision 23456' - - or for git < 1.7.1 with - - git checkout -b svn_23456 $SHA1 - - where $SHA1 is the commit SHA1 from the 'git log' output. - - -Contact the project admins <git at libav dot org> if you have technical -problems with the GIT server. diff --git a/doc/indevs.texi b/doc/indevs.texi index 30427905ff..3fb852b1f8 100644 --- a/doc/indevs.texi +++ b/doc/indevs.texi @@ -1,10 +1,10 @@ @chapter Input Devices @c man begin INPUT DEVICES -Input devices are configured elements in Libav which allow to access +Input devices are configured elements in FFmpeg which enable accessing the data coming from a multimedia device attached to your system. -When you configure your Libav build, all the supported input devices +When you configure your FFmpeg build, all the supported input devices are enabled by default. You can list all available ones using the configure option "--list-indevs". @@ -13,8 +13,8 @@ You can disable all the input devices using the configure option option "--enable-indev=@var{INDEV}", or you can disable a particular input device using the option "--disable-indev=@var{INDEV}". -The option "-formats" of the av* tools will display the list of -supported input devices (amongst the demuxers). +The option "-devices" of the ff* tools will display the list of +supported input devices. A description of the currently available input devices follows. @@ -42,23 +42,478 @@ specify card number or identifier, device number and subdevice number To see the list of cards currently recognized by your system check the files @file{/proc/asound/cards} and @file{/proc/asound/devices}. -For example to capture with @command{avconv} from an ALSA device with +For example to capture with @command{ffmpeg} from an ALSA device with card id 0, you may run the command: @example -avconv -f alsa -i hw:0 alsaout.wav +ffmpeg -f alsa -i hw:0 alsaout.wav @end example For more information see: @url{http://www.alsa-project.org/alsa-doc/alsa-lib/pcm.html} +@subsection Options + +@table @option + +@item sample_rate +Set the sample rate in Hz. Default is 48000. + +@item channels +Set the number of channels. Default is 2. + +@end table + +@section avfoundation + +AVFoundation input device. + +AVFoundation is the currently recommended framework by Apple for streamgrabbing on OSX >= 10.7 as well as on iOS. +The older QTKit framework has been marked deprecated since OSX version 10.7. + +The input filename has to be given in the following syntax: +@example +-i "[[VIDEO]:[AUDIO]]" +@end example +The first entry selects the video input while the latter selects the audio input. +The stream has to be specified by the device name or the device index as shown by the device list. +Alternatively, the video and/or audio input device can be chosen by index using the +@option{ + -video_device_index <INDEX> +} +and/or +@option{ + -audio_device_index <INDEX> +} +, overriding any +device name or index given in the input filename. + +All available devices can be enumerated by using @option{-list_devices true}, listing +all device names and corresponding indices. + +There are two device name aliases: +@table @code + +@item default +Select the AVFoundation default device of the corresponding type. + +@item none +Do not record the corresponding media type. +This is equivalent to specifying an empty device name or index. + +@end table + +@subsection Options + +AVFoundation supports the following options: + +@table @option + +@item -list_devices <TRUE|FALSE> +If set to true, a list of all available input devices is given showing all +device names and indices. + +@item -video_device_index <INDEX> +Specify the video device by its index. Overrides anything given in the input filename. + +@item -audio_device_index <INDEX> +Specify the audio device by its index. Overrides anything given in the input filename. + +@item -pixel_format <FORMAT> +Request the video device to use a specific pixel format. +If the specified format is not supported, a list of available formats is given +and the first one in this list is used instead. Available pixel formats are: +@code{monob, rgb555be, rgb555le, rgb565be, rgb565le, rgb24, bgr24, 0rgb, bgr0, 0bgr, rgb0, + bgr48be, uyvy422, yuva444p, yuva444p16le, yuv444p, yuv422p16, yuv422p10, yuv444p10, + yuv420p, nv12, yuyv422, gray} + +@item -framerate +Set the grabbing frame rate. Default is @code{ntsc}, corresponding to a +frame rate of @code{30000/1001}. + +@item -video_size +Set the video frame size. + +@item -capture_cursor +Capture the mouse pointer. Default is 0. + +@item -capture_mouse_clicks +Capture the screen mouse clicks. Default is 0. + +@end table + +@subsection Examples + +@itemize + +@item +Print the list of AVFoundation supported devices and exit: +@example +$ ffmpeg -f avfoundation -list_devices true -i "" +@end example + +@item +Record video from video device 0 and audio from audio device 0 into out.avi: +@example +$ ffmpeg -f avfoundation -i "0:0" out.avi +@end example + +@item +Record video from video device 2 and audio from audio device 1 into out.avi: +@example +$ ffmpeg -f avfoundation -video_device_index 2 -i ":1" out.avi +@end example + +@item +Record video from the system default video device using the pixel format bgr0 and do not record any audio into out.avi: +@example +$ ffmpeg -f avfoundation -pixel_format bgr0 -i "default:none" out.avi +@end example + +@end itemize + @section bktr BSD video input device. +@subsection Options + +@table @option + +@item framerate +Set the frame rate. + +@item video_size +Set the video frame size. Default is @code{vga}. + +@item standard + +Available values are: +@table @samp +@item pal + +@item ntsc + +@item secam + +@item paln + +@item palm + +@item ntscj + +@end table + +@end table + +@section decklink + +The decklink input device provides capture capabilities for Blackmagic +DeckLink devices. + +To enable this input device, you need the Blackmagic DeckLink SDK and you +need to configure with the appropriate @code{--extra-cflags} +and @code{--extra-ldflags}. +On Windows, you need to run the IDL files through @command{widl}. + +DeckLink is very picky about the formats it supports. Pixel format is +uyvy422 or v210, framerate and video size must be determined for your device with +@command{-list_formats 1}. Audio sample rate is always 48 kHz and the number +of channels can be 2, 8 or 16. Note that all audio channels are bundled in one single +audio track. + +@subsection Options + +@table @option + +@item list_devices +If set to @option{true}, print a list of devices and exit. +Defaults to @option{false}. + +@item list_formats +If set to @option{true}, print a list of supported formats and exit. +Defaults to @option{false}. + +@item bm_v210 +If set to @samp{1}, video is captured in 10 bit v210 instead +of uyvy422. Not all Blackmagic devices support this option. + +@item teletext_lines +If set to nonzero, an additional teletext stream will be captured from the +vertical ancillary data. This option is a bitmask of the VBI lines checked, +specifically lines 6 to 22, and lines 318 to 335. Line 6 is the LSB in the mask. +Selected lines which do not contain teletext information will be ignored. You +can use the special @option{all} constant to select all possible lines, or +@option{standard} to skip lines 6, 318 and 319, which are not compatible with all +receivers. Capturing teletext only works for SD PAL sources in 8 bit mode. +To use this option, ffmpeg needs to be compiled with @code{--enable-libzvbi}. + +@item channels +Defines number of audio channels to capture. Must be @samp{2}, @samp{8} or @samp{16}. +Defaults to @samp{2}. + +@end table + +@subsection Examples + +@itemize + +@item +List input devices: +@example +ffmpeg -f decklink -list_devices 1 -i dummy +@end example + +@item +List supported formats: +@example +ffmpeg -f decklink -list_formats 1 -i 'Intensity Pro' +@end example + +@item +Capture video clip at 1080i50 (format 11): +@example +ffmpeg -f decklink -i 'Intensity Pro@@11' -acodec copy -vcodec copy output.avi +@end example + +@item +Capture video clip at 1080i50 10 bit: +@example +ffmpeg -bm_v210 1 -f decklink -i 'UltraStudio Mini Recorder@@11' -acodec copy -vcodec copy output.avi +@end example + +@item +Capture video clip at 1080i50 with 16 audio channels: +@example +ffmpeg -channels 16 -f decklink -i 'UltraStudio Mini Recorder@@11' -acodec copy -vcodec copy output.avi +@end example + +@end itemize + +@section dshow + +Windows DirectShow input device. + +DirectShow support is enabled when FFmpeg is built with the mingw-w64 project. +Currently only audio and video devices are supported. + +Multiple devices may be opened as separate inputs, but they may also be +opened on the same input, which should improve synchronism between them. + +The input name should be in the format: + +@example +@var{TYPE}=@var{NAME}[:@var{TYPE}=@var{NAME}] +@end example + +where @var{TYPE} can be either @var{audio} or @var{video}, +and @var{NAME} is the device's name or alternative name.. + +@subsection Options + +If no options are specified, the device's defaults are used. +If the device does not support the requested options, it will +fail to open. + +@table @option + +@item video_size +Set the video size in the captured video. + +@item framerate +Set the frame rate in the captured video. + +@item sample_rate +Set the sample rate (in Hz) of the captured audio. + +@item sample_size +Set the sample size (in bits) of the captured audio. + +@item channels +Set the number of channels in the captured audio. + +@item list_devices +If set to @option{true}, print a list of devices and exit. + +@item list_options +If set to @option{true}, print a list of selected device's options +and exit. + +@item video_device_number +Set video device number for devices with the same name (starts at 0, +defaults to 0). + +@item audio_device_number +Set audio device number for devices with the same name (starts at 0, +defaults to 0). + +@item pixel_format +Select pixel format to be used by DirectShow. This may only be set when +the video codec is not set or set to rawvideo. + +@item audio_buffer_size +Set audio device buffer size in milliseconds (which can directly +impact latency, depending on the device). +Defaults to using the audio device's +default buffer size (typically some multiple of 500ms). +Setting this value too low can degrade performance. +See also +@url{http://msdn.microsoft.com/en-us/library/windows/desktop/dd377582(v=vs.85).aspx} + +@item video_pin_name +Select video capture pin to use by name or alternative name. + +@item audio_pin_name +Select audio capture pin to use by name or alternative name. + +@item crossbar_video_input_pin_number +Select video input pin number for crossbar device. This will be +routed to the crossbar device's Video Decoder output pin. +Note that changing this value can affect future invocations +(sets a new default) until system reboot occurs. + +@item crossbar_audio_input_pin_number +Select audio input pin number for crossbar device. This will be +routed to the crossbar device's Audio Decoder output pin. +Note that changing this value can affect future invocations +(sets a new default) until system reboot occurs. + +@item show_video_device_dialog +If set to @option{true}, before capture starts, popup a display dialog +to the end user, allowing them to change video filter properties +and configurations manually. +Note that for crossbar devices, adjusting values in this dialog +may be needed at times to toggle between PAL (25 fps) and NTSC (29.97) +input frame rates, sizes, interlacing, etc. Changing these values can +enable different scan rates/frame rates and avoiding green bars at +the bottom, flickering scan lines, etc. +Note that with some devices, changing these properties can also affect future +invocations (sets new defaults) until system reboot occurs. + +@item show_audio_device_dialog +If set to @option{true}, before capture starts, popup a display dialog +to the end user, allowing them to change audio filter properties +and configurations manually. + +@item show_video_crossbar_connection_dialog +If set to @option{true}, before capture starts, popup a display +dialog to the end user, allowing them to manually +modify crossbar pin routings, when it opens a video device. + +@item show_audio_crossbar_connection_dialog +If set to @option{true}, before capture starts, popup a display +dialog to the end user, allowing them to manually +modify crossbar pin routings, when it opens an audio device. + +@item show_analog_tv_tuner_dialog +If set to @option{true}, before capture starts, popup a display +dialog to the end user, allowing them to manually +modify TV channels and frequencies. + +@item show_analog_tv_tuner_audio_dialog +If set to @option{true}, before capture starts, popup a display +dialog to the end user, allowing them to manually +modify TV audio (like mono vs. stereo, Language A,B or C). + +@item audio_device_load +Load an audio capture filter device from file instead of searching +it by name. It may load additional parameters too, if the filter +supports the serialization of its properties to. +To use this an audio capture source has to be specified, but it can +be anything even fake one. + +@item audio_device_save +Save the currently used audio capture filter device and its +parameters (if the filter supports it) to a file. +If a file with the same name exists it will be overwritten. + +@item video_device_load +Load a video capture filter device from file instead of searching +it by name. It may load additional parameters too, if the filter +supports the serialization of its properties to. +To use this a video capture source has to be specified, but it can +be anything even fake one. + +@item video_device_save +Save the currently used video capture filter device and its +parameters (if the filter supports it) to a file. +If a file with the same name exists it will be overwritten. + +@end table + +@subsection Examples + +@itemize + +@item +Print the list of DirectShow supported devices and exit: +@example +$ ffmpeg -list_devices true -f dshow -i dummy +@end example + +@item +Open video device @var{Camera}: +@example +$ ffmpeg -f dshow -i video="Camera" +@end example + +@item +Open second video device with name @var{Camera}: +@example +$ ffmpeg -f dshow -video_device_number 1 -i video="Camera" +@end example + +@item +Open video device @var{Camera} and audio device @var{Microphone}: +@example +$ ffmpeg -f dshow -i video="Camera":audio="Microphone" +@end example + +@item +Print the list of supported options in selected device and exit: +@example +$ ffmpeg -list_options true -f dshow -i video="Camera" +@end example + +@item +Specify pin names to capture by name or alternative name, specify alternative device name: +@example +$ ffmpeg -f dshow -audio_pin_name "Audio Out" -video_pin_name 2 -i video=video="@@device_pnp_\\?\pci#ven_1a0a&dev_6200&subsys_62021461&rev_01#4&e2c7dd6&0&00e1#@{65e8773d-8f56-11d0-a3b9-00a0c9223196@}\@{ca465100-deb0-4d59-818f-8c477184adf6@}":audio="Microphone" +@end example + +@item +Configure a crossbar device, specifying crossbar pins, allow user to adjust video capture properties at startup: +@example +$ ffmpeg -f dshow -show_video_device_dialog true -crossbar_video_input_pin_number 0 + -crossbar_audio_input_pin_number 3 -i video="AVerMedia BDA Analog Capture":audio="AVerMedia BDA Analog Capture" +@end example + +@end itemize + @section dv1394 Linux DV 1394 input device. +@subsection Options + +@table @option + +@item framerate +Set the frame rate. Default is 25. + +@item standard + +Available values are: +@table @samp +@item pal + +@item ntsc + +@end table + +Default value is @code{ntsc}. + +@end table + @section fbdev Linux framebuffer input device. @@ -71,18 +526,162 @@ console. It is accessed through a file device node, usually For more detailed information read the file Documentation/fb/framebuffer.txt included in the Linux source tree. +See also @url{http://linux-fbdev.sourceforge.net/}, and fbset(1). + To record from the framebuffer device @file{/dev/fb0} with -@command{avconv}: +@command{ffmpeg}: @example -avconv -f fbdev -r 10 -i /dev/fb0 out.avi +ffmpeg -f fbdev -framerate 10 -i /dev/fb0 out.avi @end example You can take a single screenshot image with the command: @example -avconv -f fbdev -frames:v 1 -r 1 -i /dev/fb0 screenshot.jpeg +ffmpeg -f fbdev -framerate 1 -i /dev/fb0 -frames:v 1 screenshot.jpeg @end example -See also @url{http://linux-fbdev.sourceforge.net/}, and fbset(1). +@subsection Options + +@table @option + +@item framerate +Set the frame rate. Default is 25. + +@end table + +@section gdigrab + +Win32 GDI-based screen capture device. + +This device allows you to capture a region of the display on Windows. + +There are two options for the input filename: +@example +desktop +@end example +or +@example +title=@var{window_title} +@end example + +The first option will capture the entire desktop, or a fixed region of the +desktop. The second option will instead capture the contents of a single +window, regardless of its position on the screen. + +For example, to grab the entire desktop using @command{ffmpeg}: +@example +ffmpeg -f gdigrab -framerate 6 -i desktop out.mpg +@end example + +Grab a 640x480 region at position @code{10,20}: +@example +ffmpeg -f gdigrab -framerate 6 -offset_x 10 -offset_y 20 -video_size vga -i desktop out.mpg +@end example + +Grab the contents of the window named "Calculator" +@example +ffmpeg -f gdigrab -framerate 6 -i title=Calculator out.mpg +@end example + +@subsection Options + +@table @option +@item draw_mouse +Specify whether to draw the mouse pointer. Use the value @code{0} to +not draw the pointer. Default value is @code{1}. + +@item framerate +Set the grabbing frame rate. Default value is @code{ntsc}, +corresponding to a frame rate of @code{30000/1001}. + +@item show_region +Show grabbed region on screen. + +If @var{show_region} is specified with @code{1}, then the grabbing +region will be indicated on screen. With this option, it is easy to +know what is being grabbed if only a portion of the screen is grabbed. + +Note that @var{show_region} is incompatible with grabbing the contents +of a single window. + +For example: +@example +ffmpeg -f gdigrab -show_region 1 -framerate 6 -video_size cif -offset_x 10 -offset_y 20 -i desktop out.mpg +@end example + +@item video_size +Set the video frame size. The default is to capture the full screen if @file{desktop} is selected, or the full window size if @file{title=@var{window_title}} is selected. + +@item offset_x +When capturing a region with @var{video_size}, set the distance from the left edge of the screen or desktop. + +Note that the offset calculation is from the top left corner of the primary monitor on Windows. If you have a monitor positioned to the left of your primary monitor, you will need to use a negative @var{offset_x} value to move the region to that monitor. + +@item offset_y +When capturing a region with @var{video_size}, set the distance from the top edge of the screen or desktop. + +Note that the offset calculation is from the top left corner of the primary monitor on Windows. If you have a monitor positioned above your primary monitor, you will need to use a negative @var{offset_y} value to move the region to that monitor. + +@end table + +@section iec61883 + +FireWire DV/HDV input device using libiec61883. + +To enable this input device, you need libiec61883, libraw1394 and +libavc1394 installed on your system. Use the configure option +@code{--enable-libiec61883} to compile with the device enabled. + +The iec61883 capture device supports capturing from a video device +connected via IEEE1394 (FireWire), using libiec61883 and the new Linux +FireWire stack (juju). This is the default DV/HDV input method in Linux +Kernel 2.6.37 and later, since the old FireWire stack was removed. + +Specify the FireWire port to be used as input file, or "auto" +to choose the first port connected. + +@subsection Options + +@table @option + +@item dvtype +Override autodetection of DV/HDV. This should only be used if auto +detection does not work, or if usage of a different device type +should be prohibited. Treating a DV device as HDV (or vice versa) will +not work and result in undefined behavior. +The values @option{auto}, @option{dv} and @option{hdv} are supported. + +@item dvbuffer +Set maximum size of buffer for incoming data, in frames. For DV, this +is an exact value. For HDV, it is not frame exact, since HDV does +not have a fixed frame size. + +@item dvguid +Select the capture device by specifying it's GUID. Capturing will only +be performed from the specified device and fails if no device with the +given GUID is found. This is useful to select the input if multiple +devices are connected at the same time. +Look at /sys/bus/firewire/devices to find out the GUIDs. + +@end table + +@subsection Examples + +@itemize + +@item +Grab and show the input of a FireWire DV/HDV device. +@example +ffplay -f iec61883 -i auto +@end example + +@item +Grab and record the input of a FireWire DV/HDV device, +using a packet buffer of 100000 packets if the source is HDV. +@example +ffmpeg -f iec61883 -i auto -hdvbuffer 100000 out.mpg +@end example + +@end itemize @section jack @@ -95,24 +694,24 @@ A JACK input device creates one or more JACK writable clients, one for each audio channel, with name @var{client_name}:input_@var{N}, where @var{client_name} is the name provided by the application, and @var{N} is a number which identifies the channel. -Each writable client will send the acquired data to the Libav input +Each writable client will send the acquired data to the FFmpeg input device. Once you have created one or more JACK readable clients, you need to connect them to one or more JACK writable clients. -To connect or disconnect JACK clients you can use the -@file{jack_connect} and @file{jack_disconnect} programs, or do it -through a graphical interface, for example with @file{qjackctl}. +To connect or disconnect JACK clients you can use the @command{jack_connect} +and @command{jack_disconnect} programs, or do it through a graphical interface, +for example with @command{qjackctl}. To list the JACK clients and their properties you can invoke the command -@file{jack_lsp}. +@command{jack_lsp}. Follows an example which shows how to capture a JACK readable client -with @command{avconv}. +with @command{ffmpeg}. @example -# Create a JACK writable client with name "libav". -$ avconv -f jack -i libav -y out.wav +# Create a JACK writable client with name "ffmpeg". +$ ffmpeg -f jack -i ffmpeg -y out.wav # Start the sample jack_metro readable client. $ jack_metro -b 120 -d 0.2 -f 4000 @@ -123,20 +722,252 @@ system:capture_1 system:capture_2 system:playback_1 system:playback_2 -libav:input_1 +ffmpeg:input_1 metro:120_bpm -# Connect metro to the avconv writable client. -$ jack_connect metro:120_bpm libav:input_1 +# Connect metro to the ffmpeg writable client. +$ jack_connect metro:120_bpm ffmpeg:input_1 @end example For more information read: @url{http://jackaudio.org/} +@subsection Options + +@table @option + +@item channels +Set the number of channels. Default is 2. + +@end table + +@section lavfi + +Libavfilter input virtual device. + +This input device reads data from the open output pads of a libavfilter +filtergraph. + +For each filtergraph open output, the input device will create a +corresponding stream which is mapped to the generated output. Currently +only video data is supported. The filtergraph is specified through the +option @option{graph}. + +@subsection Options + +@table @option + +@item graph +Specify the filtergraph to use as input. Each video open output must be +labelled by a unique string of the form "out@var{N}", where @var{N} is a +number starting from 0 corresponding to the mapped input stream +generated by the device. +The first unlabelled output is automatically assigned to the "out0" +label, but all the others need to be specified explicitly. + +The suffix "+subcc" can be appended to the output label to create an extra +stream with the closed captions packets attached to that output +(experimental; only for EIA-608 / CEA-708 for now). +The subcc streams are created after all the normal streams, in the order of +the corresponding stream. +For example, if there is "out19+subcc", "out7+subcc" and up to "out42", the +stream #43 is subcc for stream #7 and stream #44 is subcc for stream #19. + +If not specified defaults to the filename specified for the input +device. + +@item graph_file +Set the filename of the filtergraph to be read and sent to the other +filters. Syntax of the filtergraph is the same as the one specified by +the option @var{graph}. + +@item dumpgraph +Dump graph to stderr. + +@end table + +@subsection Examples + +@itemize +@item +Create a color video stream and play it back with @command{ffplay}: +@example +ffplay -f lavfi -graph "color=c=pink [out0]" dummy +@end example + +@item +As the previous example, but use filename for specifying the graph +description, and omit the "out0" label: +@example +ffplay -f lavfi color=c=pink +@end example + +@item +Create three different video test filtered sources and play them: +@example +ffplay -f lavfi -graph "testsrc [out0]; testsrc,hflip [out1]; testsrc,negate [out2]" test3 +@end example + +@item +Read an audio stream from a file using the amovie source and play it +back with @command{ffplay}: +@example +ffplay -f lavfi "amovie=test.wav" +@end example + +@item +Read an audio stream and a video stream and play it back with +@command{ffplay}: +@example +ffplay -f lavfi "movie=test.avi[out0];amovie=test.wav[out1]" +@end example + +@item +Dump decoded frames to images and closed captions to a file (experimental): +@example +ffmpeg -f lavfi -i "movie=test.ts[out0+subcc]" -map v frame%08d.png -map s -c copy -f rawvideo subcc.bin +@end example + +@end itemize + +@section libcdio + +Audio-CD input device based on libcdio. + +To enable this input device during configuration you need libcdio +installed on your system. It requires the configure option +@code{--enable-libcdio}. + +This device allows playing and grabbing from an Audio-CD. + +For example to copy with @command{ffmpeg} the entire Audio-CD in @file{/dev/sr0}, +you may run the command: +@example +ffmpeg -f libcdio -i /dev/sr0 cd.wav +@end example + +@subsection Options +@table @option +@item speed +Set drive reading speed. Default value is 0. + +The speed is specified CD-ROM speed units. The speed is set through +the libcdio @code{cdio_cddap_speed_set} function. On many CD-ROM +drives, specifying a value too large will result in using the fastest +speed. + +@item paranoia_mode +Set paranoia recovery mode flags. It accepts one of the following values: + +@table @samp +@item disable +@item verify +@item overlap +@item neverskip +@item full +@end table + +Default value is @samp{disable}. + +For more information about the available recovery modes, consult the +paranoia project documentation. +@end table + @section libdc1394 IIDC1394 input device, based on libdc1394 and libraw1394. +Requires the configure option @code{--enable-libdc1394}. + +@section openal + +The OpenAL input device provides audio capture on all systems with a +working OpenAL 1.1 implementation. + +To enable this input device during configuration, you need OpenAL +headers and libraries installed on your system, and need to configure +FFmpeg with @code{--enable-openal}. + +OpenAL headers and libraries should be provided as part of your OpenAL +implementation, or as an additional download (an SDK). Depending on your +installation you may need to specify additional flags via the +@code{--extra-cflags} and @code{--extra-ldflags} for allowing the build +system to locate the OpenAL headers and libraries. + +An incomplete list of OpenAL implementations follows: + +@table @strong +@item Creative +The official Windows implementation, providing hardware acceleration +with supported devices and software fallback. +See @url{http://openal.org/}. +@item OpenAL Soft +Portable, open source (LGPL) software implementation. Includes +backends for the most common sound APIs on the Windows, Linux, +Solaris, and BSD operating systems. +See @url{http://kcat.strangesoft.net/openal.html}. +@item Apple +OpenAL is part of Core Audio, the official Mac OS X Audio interface. +See @url{http://developer.apple.com/technologies/mac/audio-and-video.html} +@end table + +This device allows one to capture from an audio input device handled +through OpenAL. + +You need to specify the name of the device to capture in the provided +filename. If the empty string is provided, the device will +automatically select the default device. You can get the list of the +supported devices by using the option @var{list_devices}. + +@subsection Options + +@table @option + +@item channels +Set the number of channels in the captured audio. Only the values +@option{1} (monaural) and @option{2} (stereo) are currently supported. +Defaults to @option{2}. + +@item sample_size +Set the sample size (in bits) of the captured audio. Only the values +@option{8} and @option{16} are currently supported. Defaults to +@option{16}. + +@item sample_rate +Set the sample rate (in Hz) of the captured audio. +Defaults to @option{44.1k}. + +@item list_devices +If set to @option{true}, print a list of devices and exit. +Defaults to @option{false}. + +@end table + +@subsection Examples + +Print the list of OpenAL supported devices and exit: +@example +$ ffmpeg -list_devices true -f openal -i dummy out.ogg +@end example + +Capture from the OpenAL device @file{DR-BT101 via PulseAudio}: +@example +$ ffmpeg -f openal -i 'DR-BT101 via PulseAudio' out.ogg +@end example + +Capture from the default device (note the empty string '' as filename): +@example +$ ffmpeg -f openal -i '' out.ogg +@end example + +Capture from two devices simultaneously, writing to two different files, +within the same @command{ffmpeg} command: +@example +$ ffmpeg -f openal -i 'DR-BT101 via PulseAudio' out1.ogg -f openal -i 'ALSA Default' out2.ogg +@end example +Note: not all OpenAL implementations support multiple simultaneous capture - +try the latest OpenAL Soft if the above does not work. + @section oss Open Sound System input device. @@ -145,97 +976,122 @@ The filename to provide to the input device is the device node representing the OSS input device, and is usually set to @file{/dev/dsp}. -For example to grab from @file{/dev/dsp} using @command{avconv} use the +For example to grab from @file{/dev/dsp} using @command{ffmpeg} use the command: @example -avconv -f oss -i /dev/dsp /tmp/oss.wav +ffmpeg -f oss -i /dev/dsp /tmp/oss.wav @end example For more information about OSS see: @url{http://manuals.opensound.com/usersguide/dsp.html} +@subsection Options + +@table @option + +@item sample_rate +Set the sample rate in Hz. Default is 48000. + +@item channels +Set the number of channels. Default is 2. + +@end table + + @section pulse -pulseaudio input device. +PulseAudio input device. -To enable this input device during configuration you need libpulse-simple -installed in your system. +To enable this output device you need to configure FFmpeg with @code{--enable-libpulse}. The filename to provide to the input device is a source device or the string "default" -To list the pulse source devices and their properties you can invoke -the command @file{pactl list sources}. +To list the PulseAudio source devices and their properties you can invoke +the command @command{pactl list sources}. -@example -avconv -f pulse -i default /tmp/pulse.wav -@end example - -@subsection @var{server} AVOption +More information about PulseAudio can be found on @url{http://www.pulseaudio.org}. -The syntax is: -@example --server @var{server name} -@end example +@subsection Options +@table @option +@item server +Connect to a specific PulseAudio server, specified by an IP address. +Default server is used when not provided. -Connects to a specific server. +@item name +Specify the application name PulseAudio will use when showing active clients, +by default it is the @code{LIBAVFORMAT_IDENT} string. -@subsection @var{name} AVOption +@item stream_name +Specify the stream name PulseAudio will use when showing active streams, +by default it is "record". -The syntax is: -@example --name @var{application name} -@end example +@item sample_rate +Specify the samplerate in Hz, by default 48kHz is used. -Specify the application name pulse will use when showing active clients, -by default it is "libav" +@item channels +Specify the channels in use, by default 2 (stereo) is set. -@subsection @var{stream_name} AVOption +@item frame_size +Specify the number of bytes per frame, by default it is set to 1024. -The syntax is: -@example --stream_name @var{stream name} -@end example +@item fragment_size +Specify the minimal buffering fragment in PulseAudio, it will affect the +audio latency. By default it is unset. -Specify the stream name pulse will use when showing active streams, -by default it is "record" +@item wallclock +Set the initial PTS using the current time. Default is 1. -@subsection @var{sample_rate} AVOption +@end table -The syntax is: +@subsection Examples +Record a stream from default device: @example --sample_rate @var{samplerate} +ffmpeg -f pulse -i default /tmp/pulse.wav @end example -Specify the samplerate in Hz, by default 48kHz is used. +@section qtkit + +QTKit input device. -@subsection @var{channels} AVOption +The filename passed as input is parsed to contain either a device name or index. +The device index can also be given by using -video_device_index. +A given device index will override any given device name. +If the desired device consists of numbers only, use -video_device_index to identify it. +The default device will be chosen if an empty string or the device name "default" is given. +The available devices can be enumerated by using -list_devices. -The syntax is: @example --channels @var{N} +ffmpeg -f qtkit -i "0" out.mpg @end example -Specify the channels in use, by default 2 (stereo) is set. +@example +ffmpeg -f qtkit -video_device_index 0 -i "" out.mpg +@end example -@subsection @var{frame_size} AVOption +@example +ffmpeg -f qtkit -i "default" out.mpg +@end example -The syntax is: @example --frame_size @var{bytes} +ffmpeg -f qtkit -list_devices true -i "" @end example -Specify the number of byte per frame, by default it is set to 1024. +@subsection Options -@subsection @var{fragment_size} AVOption +@table @option -The syntax is: -@example --fragment_size @var{bytes} -@end example +@item frame_rate +Set frame rate. Default is 30. -Specify the minimal buffering fragment in pulseaudio, it will affect the -audio latency. By default it is unset. +@item list_devices +If set to @code{true}, print a list of devices and exit. Default is +@code{false}. + +@item video_device_index +Select the video device by index for devices with the same name (starts at 0). + +@end table @section sndio @@ -248,16 +1104,34 @@ The filename to provide to the input device is the device node representing the sndio input device, and is usually set to @file{/dev/audio0}. -For example to grab from @file{/dev/audio0} using @command{avconv} use the +For example to grab from @file{/dev/audio0} using @command{ffmpeg} use the command: @example -avconv -f sndio -i /dev/audio0 /tmp/oss.wav +ffmpeg -f sndio -i /dev/audio0 /tmp/oss.wav @end example -@section video4linux2 +@subsection Options + +@table @option + +@item sample_rate +Set the sample rate in Hz. Default is 48000. + +@item channels +Set the number of channels. Default is 2. + +@end table + +@section video4linux2, v4l2 Video4Linux2 input video device. +"v4l2" can be used as alias for "video4linux2". + +If FFmpeg is built with v4l-utils support (by using the +@code{--enable-libv4l2} configure option), it is possible to use it with the +@code{-use_libv4l2} input device option. + The name of the device to grab is a file device node, usually Linux systems tend to automatically create such nodes when the device (e.g. an USB webcam) is plugged into the system, and has a name of the @@ -265,22 +1139,116 @@ kind @file{/dev/video@var{N}}, where @var{N} is a number associated to the device. Video4Linux2 devices usually support a limited set of -@var{width}x@var{height} sizes and framerates. You can check which are +@var{width}x@var{height} sizes and frame rates. You can check which are supported using @command{-list_formats all} for Video4Linux2 devices. +Some devices, like TV cards, support one or more standards. It is possible +to list all the supported standards using @command{-list_standards all}. + +The time base for the timestamps is 1 microsecond. Depending on the kernel +version and configuration, the timestamps may be derived from the real time +clock (origin at the Unix Epoch) or the monotonic clock (origin usually at +boot time, unaffected by NTP or manual changes to the clock). The +@option{-timestamps abs} or @option{-ts abs} option can be used to force +conversion into the real time clock. + +Some usage examples of the video4linux2 device with @command{ffmpeg} +and @command{ffplay}: +@itemize +@item +List supported formats for a video4linux2 device: +@example +ffplay -f video4linux2 -list_formats all /dev/video0 +@end example -Some usage examples of the video4linux2 devices with avconv and avplay: +@item +Grab and show the input of a video4linux2 device: +@example +ffplay -f video4linux2 -framerate 30 -video_size hd720 /dev/video0 +@end example +@item +Grab and record the input of a video4linux2 device, leave the +frame rate and size as previously set: @example -# List supported formats for a video4linux2 device. -avplay -f video4linux2 -list_formats all /dev/video0 +ffmpeg -f video4linux2 -input_format mjpeg -i /dev/video0 out.mpeg +@end example +@end itemize -# Grab and show the input of a video4linux2 device. -avplay -f video4linux2 -framerate 30 -video_size hd720 /dev/video0 +For more information about Video4Linux, check @url{http://linuxtv.org/}. -# Grab and record the input of a video4linux2 device, leave the -framerate and size as previously set. -avconv -f video4linux2 -input_format mjpeg -i /dev/video0 out.mpeg -@end example +@subsection Options + +@table @option +@item standard +Set the standard. Must be the name of a supported standard. To get a +list of the supported standards, use the @option{list_standards} +option. + +@item channel +Set the input channel number. Default to -1, which means using the +previously selected channel. + +@item video_size +Set the video frame size. The argument must be a string in the form +@var{WIDTH}x@var{HEIGHT} or a valid size abbreviation. + +@item pixel_format +Select the pixel format (only valid for raw video input). + +@item input_format +Set the preferred pixel format (for raw video) or a codec name. +This option allows one to select the input format, when several are +available. + +@item framerate +Set the preferred video frame rate. + +@item list_formats +List available formats (supported pixel formats, codecs, and frame +sizes) and exit. + +Available values are: +@table @samp +@item all +Show all available (compressed and non-compressed) formats. + +@item raw +Show only raw video (non-compressed) formats. + +@item compressed +Show only compressed formats. +@end table + +@item list_standards +List supported standards and exit. + +Available values are: +@table @samp +@item all +Show all supported standards. +@end table + +@item timestamps, ts +Set type of timestamps for grabbed frames. + +Available values are: +@table @samp +@item default +Use timestamps from the kernel. + +@item abs +Use absolute timestamps (wall clock). + +@item mono2abs +Force conversion from monotonic to absolute timestamps. +@end table + +Default value is @code{default}. + +@item use_libv4l2 +Use libv4l2 (v4l-utils) conversion functions. Default is 0. + +@end table @section vfwcap @@ -290,11 +1258,31 @@ The filename passed as input is the capture driver number, ranging from 0 to 9. You may use "list" as filename to print a list of drivers. Any other filename will be interpreted as device number 0. +@subsection Options + +@table @option + +@item video_size +Set the video frame size. + +@item framerate +Set the grabbing frame rate. Default value is @code{ntsc}, +corresponding to a frame rate of @code{30000/1001}. + +@end table + @section x11grab X11 video input device. -This device allows to capture a region of an X11 display. +To enable this input device during configuration you need libxcb +installed on your system. It will be automatically detected during +configuration. + +Alternatively, the configure option @option{--enable-x11grab} exists +for legacy Xlib users. + +This device allows one to capture a region of an X11 display. The filename passed as input has the syntax: @example @@ -310,26 +1298,34 @@ omitted, and defaults to "localhost". The environment variable area with respect to the top-left border of the X11 screen. They default to 0. -Check the X11 documentation (e.g. man X) for more detailed information. +Check the X11 documentation (e.g. @command{man X}) for more detailed +information. -Use the @file{dpyinfo} program for getting basic information about the -properties of your X11 display (e.g. grep for "name" or "dimensions"). +Use the @command{xdpyinfo} program for getting basic information about +the properties of your X11 display (e.g. grep for "name" or +"dimensions"). -For example to grab from @file{:0.0} using @command{avconv}: +For example to grab from @file{:0.0} using @command{ffmpeg}: @example -avconv -f x11grab -r 25 -s cif -i :0.0 out.mpg - -# Grab at position 10,20. -avconv -f x11grab -r 25 -s cif -i :0.0+10,20 out.mpg +ffmpeg -f x11grab -framerate 25 -video_size cif -i :0.0 out.mpg @end example -@subsection @var{follow_mouse} AVOption - -The syntax is: +Grab at position @code{10,20}: @example --follow_mouse centered|@var{PIXELS} +ffmpeg -f x11grab -framerate 25 -video_size cif -i :0.0+10,20 out.mpg @end example +@subsection Options + +@table @option +@item draw_mouse +Specify whether to draw the mouse pointer. A value of @code{0} specify +not to draw the pointer. Default value is @code{1}. + +@item follow_mouse +Make the grabbed area follow the mouse. The argument can be +@code{centered} or a number of pixels @var{PIXELS}. + When it is specified with "centered", the grabbing region follows the mouse pointer and keeps the pointer at the center of region; otherwise, the region follows only when the mouse pointer reaches within @var{PIXELS} (greater than @@ -337,39 +1333,53 @@ zero) to the edge of region. For example: @example -avconv -f x11grab -follow_mouse centered -r 25 -s cif -i :0.0 out.mpg - -# Follows only when the mouse pointer reaches within 100 pixels to edge -avconv -f x11grab -follow_mouse 100 -r 25 -s cif -i :0.0 out.mpg +ffmpeg -f x11grab -follow_mouse centered -framerate 25 -video_size cif -i :0.0 out.mpg @end example -@subsection @var{show_region} AVOption - -The syntax is: +To follow only when the mouse pointer reaches within 100 pixels to edge: @example --show_region 1 +ffmpeg -f x11grab -follow_mouse 100 -framerate 25 -video_size cif -i :0.0 out.mpg @end example -If @var{show_region} AVOption is specified with @var{1}, then the grabbing -region will be indicated on screen. With this option, it's easy to know what is -being grabbed if only a portion of the screen is grabbed. +@item framerate +Set the grabbing frame rate. Default value is @code{ntsc}, +corresponding to a frame rate of @code{30000/1001}. + +@item show_region +Show grabbed region on screen. + +If @var{show_region} is specified with @code{1}, then the grabbing +region will be indicated on screen. With this option, it is easy to +know what is being grabbed if only a portion of the screen is grabbed. + +@item region_border +Set the region border thickness if @option{-show_region 1} is used. +Range is 1 to 128 and default is 3 (XCB-based x11grab only). For example: @example -avconv -f x11grab -show_region 1 -r 25 -s cif -i :0.0+10,20 out.mpg - -# With follow_mouse -avconv -f x11grab -follow_mouse centered -show_region 1 -r 25 -s cif -i :0.0 out.mpg +ffmpeg -f x11grab -show_region 1 -framerate 25 -video_size cif -i :0.0+10,20 out.mpg @end example -@subsection @var{grab_x} @var{grab_y} AVOption - -The syntax is: +With @var{follow_mouse}: @example --grab_x @var{x_offset} -grab_y @var{y_offset} +ffmpeg -f x11grab -follow_mouse centered -show_region 1 -framerate 25 -video_size cif -i :0.0 out.mpg @end example -Set the grabing region coordinates. The are expressed as offset from the top left -corner of the X11 window. The default value is 0. +@item video_size +Set the video frame size. Default value is @code{vga}. + +@item use_shm +Use the MIT-SHM extension for shared memory. Default value is @code{1}. +It may be necessary to disable it for remote displays (legacy x11grab +only). + +@item grab_x +@item grab_y +Set the grabbing region coordinates. They are expressed as offset from +the top left corner of the X11 window and correspond to the +@var{x_offset} and @var{y_offset} parameters in the device name. The +default value for both options is 0. +@end table @c man end INPUT DEVICES diff --git a/doc/issue_tracker.txt b/doc/issue_tracker.txt new file mode 100644 index 0000000000..e8e85304b3 --- /dev/null +++ b/doc/issue_tracker.txt @@ -0,0 +1,218 @@ +FFmpeg's bug/feature request tracker manual +================================================= + +Overview: +--------- + +FFmpeg uses Trac for tracking issues, new issues and changes to +existing issues can be done through a web interface. + +Issues can be different kinds of things we want to keep track of +but that do not belong into the source tree itself. This includes +bug reports, feature requests and license violations. We +might add more items to this list in the future, so feel free to +propose a new `type of issue' on the ffmpeg-devel mailing list if +you feel it is worth tracking. + +It is possible to subscribe to individual issues by adding yourself to the +Cc list or to subscribe to the ffmpeg-trac mailing list which receives +a mail for every change to every issue. +(the above does all work already after light testing) + +The subscription URL for the ffmpeg-trac list is: +https://lists.ffmpeg.org/mailman/listinfo/ffmpeg-trac +The URL of the webinterface of the tracker is: +https://trac.ffmpeg.org + +Type: +----- +art + Artwork such as photos, music, banners, and logos. + +bug / defect + An error, flaw, mistake, failure, or fault in FFmpeg or libav* that + prevents it from behaving as intended. + +feature request / enhancement + Request of support for encoding or decoding of a new codec, container + or variant. + Request of support for more, less or plain different output or behavior + where the current implementation cannot be considered wrong. + +license violation + Ticket to keep track of (L)GPL violations of ffmpeg by others. + +sponsoring request + Developer requests for hardware, software, specifications, money, + refunds, etc. + +task + A task/reminder such as setting up a FATE client, adding filters to + Trac, etc. + +Priority: +--------- +critical + Bugs about data loss and security issues. + No feature request can be critical. + +important + Bugs which make FFmpeg unusable for a significant number of users. + Examples here might be completely broken MPEG-4 decoding or a build issue + on Linux. + While broken 4xm decoding or a broken OS/2 build would not be important, + the separation to normal is somewhat fuzzy. + For feature requests this priority would be used for things many people + want. + Regressions also should be marked as important, regressions are bugs that + don't exist in a past revision or another branch. + +normal + Default setting. Use this if the bug does not match the other + priorities or if you are unsure of what priority to choose. + +minor + Bugs about things like spelling errors, "mp2" instead of + "mp3" being shown and such. + Feature requests about things few people want or which do not make a big + difference. + +wish + Something that is desirable to have but that there is no urgency at + all to implement, e.g. something completely cosmetic like a website + restyle or a personalized doxy template or the FFmpeg logo. + This priority is not valid for bugs. + + +Status: +------- +new + initial state + +open + intermediate states + +closed + final state + + +Analyzed flag: +-------------- +Bugs which have been analyzed and where it is understood what causes them +and which exact chain of events triggers them. This analysis should be +available as a message in the bug report. +Note, do not change the status to analyzed without also providing a clear +and understandable analysis. +This state implicates that the bug either has been reproduced or that +reproduction is not needed as the bug is already understood. + + +Type/Status: +---------- +*/new + Initial state of new bugs and feature requests submitted by + users. + +*/open + Issues which have been briefly looked at and which did not look outright + invalid. + This implicates that no real more detailed state applies yet. Conversely, + the more detailed states below implicate that the issue has been briefly + looked at. + +*/closed/duplicate + Bugs or feature requests which are duplicates. + Note, if you mark something as duplicate, do not forget setting the + superseder so bug reports are properly linked. + +*/closed/invalid + Bugs caused by user errors, random ineligible or otherwise nonsense stuff. + +*/closed/needs_more_info + Issues for which some information has been requested by the developers, + but which has not been provided by anyone within reasonable time. + + +bug/closed/fixed + Bugs which have to the best of our knowledge been fixed. + +bug/closed/wontfix + Bugs which we will not fix. Possible reasons include legality, high + complexity for the sake of supporting obscure corner cases, speed loss + for similarly esoteric purposes, et cetera. + This also means that we would reject a patch. + If we are just too lazy to fix a bug then the correct state is open + and unassigned. Closed means that the case is closed which is not + the case if we are just waiting for a patch. + +bug/closed/works_for_me + Bugs for which sufficient information was provided to reproduce but + reproduction failed - that is the code seems to work correctly to the + best of our knowledge. + +feature_request/closed/fixed + Feature requests which have been implemented. + +feature_request/closed/wontfix + Feature requests which will not be implemented. The reasons here could + be legal, philosophical or others. + +Note2, if you provide the requested info do not forget to remove the +needs_more_info resolution. + +Component: +---------- + +avcodec + issues in libavcodec/* + +avdevice + issues in libavdevice/* + +avfilter + issues in libavfilter/* + +avformat + issues in libavformat/* + +avutil + issues in libavutil/* + +build system + issues in or related to configure/Makefile + +documentation + issues in or related to doc/* + +ffmpeg + issues in or related to ffmpeg.c + +ffplay + issues in or related to ffplay.c + +ffprobe + issues in or related to ffprobe.c + +ffserver + issues in or related to ffserver.c + +postproc + issues in libpostproc/* + +swresample + issues in libswresample/* + +swscale + issues in libswscale/* + +trac + issues related to our issue tracker + +undetermined + default component; choose this if unsure + +website + issues related to the website + +wiki + issues related to the wiki diff --git a/doc/libavcodec.texi b/doc/libavcodec.texi new file mode 100644 index 0000000000..87b90db48c --- /dev/null +++ b/doc/libavcodec.texi @@ -0,0 +1,49 @@ +\input texinfo @c -*- texinfo -*- +@documentencoding UTF-8 + +@settitle Libavcodec Documentation +@titlepage +@center @titlefont{Libavcodec Documentation} +@end titlepage + +@top + +@contents + +@chapter Description +@c man begin DESCRIPTION + +The libavcodec library provides a generic encoding/decoding framework +and contains multiple decoders and encoders for audio, video and +subtitle streams, and several bitstream filters. + +The shared architecture provides various services ranging from bit +stream I/O to DSP optimizations, and makes it suitable for +implementing robust and fast codecs as well as for experimentation. + +@c man end DESCRIPTION + +@chapter See Also + +@ifhtml +@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver}, +@url{ffmpeg-codecs.html,ffmpeg-codecs}, @url{ffmpeg-bitstream-filters.html,bitstream-filters}, +@url{libavutil.html,libavutil} +@end ifhtml + +@ifnothtml +ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), +ffmpeg-codecs(1), ffmpeg-bitstream-filters(1), +libavutil(3) +@end ifnothtml + +@include authors.texi + +@ignore + +@setfilename libavcodec +@settitle media streams decoding and encoding library + +@end ignore + +@bye diff --git a/doc/libavdevice.texi b/doc/libavdevice.texi new file mode 100644 index 0000000000..9b10282cde --- /dev/null +++ b/doc/libavdevice.texi @@ -0,0 +1,46 @@ +\input texinfo @c -*- texinfo -*- +@documentencoding UTF-8 + +@settitle Libavdevice Documentation +@titlepage +@center @titlefont{Libavdevice Documentation} +@end titlepage + +@top + +@contents + +@chapter Description +@c man begin DESCRIPTION + +The libavdevice library provides a generic framework for grabbing from +and rendering to many common multimedia input/output devices, and +supports several input and output devices, including Video4Linux2, +VfW, DShow, and ALSA. + +@c man end DESCRIPTION + +@chapter See Also + +@ifhtml +@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver}, +@url{ffmpeg-devices.html,ffmpeg-devices}, +@url{libavutil.html,libavutil}, @url{libavcodec.html,libavcodec}, @url{libavformat.html,libavformat} +@end ifhtml + +@ifnothtml +ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), +ffmpeg-devices(1), +libavutil(3), libavcodec(3), libavformat(3) +@end ifnothtml + +@include authors.texi + +@ignore + +@setfilename libavdevice +@settitle multimedia device handling library + +@end ignore + +@bye diff --git a/doc/libavfilter.texi b/doc/libavfilter.texi index 84bad29dab..52e075369c 100644 --- a/doc/libavfilter.texi +++ b/doc/libavfilter.texi @@ -1,4 +1,5 @@ \input texinfo @c -*- texinfo -*- +@documentencoding UTF-8 @settitle Libavfilter Documentation @titlepage @@ -9,84 +10,36 @@ @contents -@chapter Introduction +@chapter Description +@c man begin DESCRIPTION -Libavfilter is the filtering API of Libav. It replaces 'vhooks', and -started as a Google Summer of Code project. +The libavfilter library provides a generic audio/video filtering +framework containing several filters, sources and sinks. -Note that there may still be serious bugs in the code and its API -and ABI should not be considered stable yet! +@c man end DESCRIPTION -@chapter Tutorial +@chapter See Also -In libavfilter, it is possible for filters to have multiple inputs and -multiple outputs. -To illustrate the sorts of things that are possible, we can -use a complex filter graph. For example, the following one: +@ifhtml +@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver}, +@url{ffmpeg-filters.html,ffmpeg-filters}, +@url{libavutil.html,libavutil}, @url{libswscale.html,libswscale}, @url{libswresample.html,libswresample}, +@url{libavcodec.html,libavcodec}, @url{libavformat.html,libavformat}, @url{libavdevice.html,libavdevice} +@end ifhtml -@example -input --> split --> fifo -----------------------> overlay --> output - | ^ - | | - +------> fifo --> crop --> vflip --------+ -@end example +@ifnothtml +ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), +ffmpeg-filters(1), +libavutil(3), libswscale(3), libswresample(3), libavcodec(3), libavformat(3), libavdevice(3) +@end ifnothtml -splits the stream in two streams, then sends one stream through the crop filter -and the vflip filter, before merging it back with the other stream by -overlaying it on top. You can use the following command to achieve this: +@include authors.texi -@example -./avconv -i input -vf "[in] split [T1], fifo, [T2] overlay=0:H/2 [out]; [T1] fifo, crop=iw:ih/2:0:ih/2, vflip [T2]" output -@end example +@ignore -The result will be that the top half of the video is mirrored -onto the bottom half of the output video. +@setfilename libavfilter +@settitle multimedia filtering library -Video filters are loaded using the @var{-vf} option passed to -avconv or to avplay. Filters in the same linear chain are separated by -commas. In our example, @var{split}, @var{fifo}, and @var{overlay} are in one -linear chain, and @var{fifo}, @var{crop}, and @var{vflip} are in another. The -points where the linear chains join are labeled by names enclosed in square -brackets. In our example, they join at @var{[T1]} and @var{[T2]}. The magic -labels @var{[in]} and @var{[out]} are the points where video is input -and output. - -Some filters take a list of parameters: they are specified -after the filter name and an equal sign, and are separated -by a semicolon. - -There are so-called @var{source filters} that do not take video -input, and we expect that some @var{sink filters} will -not have video output, at some point in the future. - -@chapter graph2dot - -The @file{graph2dot} program included in the Libav @file{tools} -directory can be used to parse a filter graph description and issue a -corresponding textual representation in the dot language. - -Invoke the command: -@example -graph2dot -h -@end example - -to see how to use @file{graph2dot}. - -You can then pass the dot description to the @file{dot} program (from -the graphviz suite of programs) and obtain a graphical representation -of the filter graph. - -For example the sequence of commands: -@example -echo @var{GRAPH_DESCRIPTION} | \ -tools/graph2dot -o graph.tmp && \ -dot -Tpng graph.tmp -o graph.png && \ -display graph.png -@end example - -can be used to create and display an image representing the graph -described by the @var{GRAPH_DESCRIPTION} string. - -@include filters.texi +@end ignore @bye diff --git a/doc/libavformat.texi b/doc/libavformat.texi new file mode 100644 index 0000000000..d505d644f6 --- /dev/null +++ b/doc/libavformat.texi @@ -0,0 +1,49 @@ +\input texinfo @c -*- texinfo -*- +@documentencoding UTF-8 + +@settitle Libavformat Documentation +@titlepage +@center @titlefont{Libavformat Documentation} +@end titlepage + +@top + +@contents + +@chapter Description +@c man begin DESCRIPTION + +The libavformat library provides a generic framework for multiplexing +and demultiplexing (muxing and demuxing) audio, video and subtitle +streams. It encompasses multiple muxers and demuxers for multimedia +container formats. + +It also supports several input and output protocols to access a media +resource. + +@c man end DESCRIPTION + +@chapter See Also + +@ifhtml +@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver}, +@url{ffmpeg-formats.html,ffmpeg-formats}, @url{ffmpeg-protocols.html,ffmpeg-protocols}, +@url{libavutil.html,libavutil}, @url{libavcodec.html,libavcodec} +@end ifhtml + +@ifnothtml +ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), +ffmpeg-formats(1), ffmpeg-protocols(1), +libavutil(3), libavcodec(3) +@end ifnothtml + +@include authors.texi + +@ignore + +@setfilename libavformat +@settitle multimedia muxing and demuxing library + +@end ignore + +@bye diff --git a/doc/libavutil.texi b/doc/libavutil.texi new file mode 100644 index 0000000000..7a1c332b81 --- /dev/null +++ b/doc/libavutil.texi @@ -0,0 +1,63 @@ +\input texinfo @c -*- texinfo -*- +@documentencoding UTF-8 + +@settitle Libavutil Documentation +@titlepage +@center @titlefont{Libavutil Documentation} +@end titlepage + +@top + +@contents + +@chapter Description +@c man begin DESCRIPTION + +The libavutil library is a utility library to aid portable +multimedia programming. It contains safe portable string functions, +random number generators, data structures, additional mathematics +functions, cryptography and multimedia related functionality (like +enumerations for pixel and sample formats). It is not a library for +code needed by both libavcodec and libavformat. + +The goals for this library is to be: + +@table @strong +@item Modular +It should have few interdependencies and the possibility of disabling individual +parts during @command{./configure}. + +@item Small +Both sources and objects should be small. + +@item Efficient +It should have low CPU and memory usage. + +@item Useful +It should avoid useless features that almost no one needs. +@end table + +@c man end DESCRIPTION + +@chapter See Also + +@ifhtml +@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver}, +@url{ffmpeg-utils.html,ffmpeg-utils} +@end ifhtml + +@ifnothtml +ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), +ffmpeg-utils(1) +@end ifnothtml + +@include authors.texi + +@ignore + +@setfilename libavutil +@settitle multimedia-biased utility library + +@end ignore + +@bye diff --git a/doc/libswresample.texi b/doc/libswresample.texi new file mode 100644 index 0000000000..bb57278314 --- /dev/null +++ b/doc/libswresample.texi @@ -0,0 +1,71 @@ +\input texinfo @c -*- texinfo -*- +@documentencoding UTF-8 + +@settitle Libswresample Documentation +@titlepage +@center @titlefont{Libswresample Documentation} +@end titlepage + +@top + +@contents + +@chapter Description +@c man begin DESCRIPTION + +The libswresample library performs highly optimized audio resampling, +rematrixing and sample format conversion operations. + +Specifically, this library performs the following conversions: + +@itemize +@item +@emph{Resampling}: is the process of changing the audio rate, for +example from a high sample rate of 44100Hz to 8000Hz. Audio +conversion from high to low sample rate is a lossy process. Several +resampling options and algorithms are available. + +@item +@emph{Format conversion}: is the process of converting the type of +samples, for example from 16-bit signed samples to unsigned 8-bit or +float samples. It also handles packing conversion, when passing from +packed layout (all samples belonging to distinct channels interleaved +in the same buffer), to planar layout (all samples belonging to the +same channel stored in a dedicated buffer or "plane"). + +@item +@emph{Rematrixing}: is the process of changing the channel layout, for +example from stereo to mono. When the input channels cannot be mapped +to the output streams, the process is lossy, since it involves +different gain factors and mixing. +@end itemize + +Various other audio conversions (e.g. stretching and padding) are +enabled through dedicated options. + +@c man end DESCRIPTION + +@chapter See Also + +@ifhtml +@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver}, +@url{ffmpeg-resampler.html,ffmpeg-resampler}, +@url{libavutil.html,libavutil} +@end ifhtml + +@ifnothtml +ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), +ffmpeg-resampler(1), +libavutil(3) +@end ifnothtml + +@include authors.texi + +@ignore + +@setfilename libswresample +@settitle audio resampling library + +@end ignore + +@bye diff --git a/doc/libswscale.texi b/doc/libswscale.texi new file mode 100644 index 0000000000..757fd24139 --- /dev/null +++ b/doc/libswscale.texi @@ -0,0 +1,64 @@ +\input texinfo @c -*- texinfo -*- +@documentencoding UTF-8 + +@settitle Libswscale Documentation +@titlepage +@center @titlefont{Libswscale Documentation} +@end titlepage + +@top + +@contents + +@chapter Description +@c man begin DESCRIPTION + +The libswscale library performs highly optimized image scaling and +colorspace and pixel format conversion operations. + +Specifically, this library performs the following conversions: + +@itemize +@item +@emph{Rescaling}: is the process of changing the video size. Several +rescaling options and algorithms are available. This is usually a +lossy process. + +@item +@emph{Pixel format conversion}: is the process of converting the image +format and colorspace of the image, for example from planar YUV420P to +RGB24 packed. It also handles packing conversion, that is converts +from packed layout (all pixels belonging to distinct planes +interleaved in the same buffer), to planar layout (all samples +belonging to the same plane stored in a dedicated buffer or "plane"). + +This is usually a lossy process in case the source and destination +colorspaces differ. +@end itemize + +@c man end DESCRIPTION + +@chapter See Also + +@ifhtml +@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver}, +@url{ffmpeg-scaler.html,ffmpeg-scaler}, +@url{libavutil.html,libavutil} +@end ifhtml + +@ifnothtml +ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), +ffmpeg-scaler(1), +libavutil(3) +@end ifnothtml + +@include authors.texi + +@ignore + +@setfilename libswscale +@settitle video scaling and pixel format conversion library + +@end ignore + +@bye diff --git a/doc/metadata.texi b/doc/metadata.texi index cfaf491c2d..bddcc99470 100644 --- a/doc/metadata.texi +++ b/doc/metadata.texi @@ -1,7 +1,7 @@ @chapter Metadata @c man begin METADATA -Libav is able to dump metadata from media files into a simple UTF-8-encoded +FFmpeg is able to dump metadata from media files into a simple UTF-8-encoded INI-like text file and then load it back using the metadata muxer/demuxer. The file format is as follows: @@ -12,10 +12,10 @@ A file consists of a header and a number of metadata tags divided into sections, each on its own line. @item -The header is a ';FFMETADATA' string, followed by a version number (now 1). +The header is a @samp{;FFMETADATA} string, followed by a version number (now 1). @item -Metadata tags are of the form 'key=value' +Metadata tags are of the form @samp{key=value} @item Immediately after header follows global metadata @@ -26,26 +26,30 @@ metadata. @item A section starts with the section name in uppercase (i.e. STREAM or CHAPTER) in -brackets ('[', ']') and ends with next section or end of file. +brackets (@samp{[}, @samp{]}) and ends with next section or end of file. @item At the beginning of a chapter section there may be an optional timebase to be -used for start/end values. It must be in form 'TIMEBASE=num/den', where num and -den are integers. If the timebase is missing then start/end times are assumed to +used for start/end values. It must be in form +@samp{TIMEBASE=@var{num}/@var{den}}, where @var{num} and @var{den} are +integers. If the timebase is missing then start/end times are assumed to be in milliseconds. + Next a chapter section must contain chapter start and end times in form -'START=num', 'END=num', where num is a positive integer. +@samp{START=@var{num}}, @samp{END=@var{num}}, where @var{num} is a positive +integer. @item -Empty lines and lines starting with ';' or '#' are ignored. +Empty lines and lines starting with @samp{;} or @samp{#} are ignored. @item -Metadata keys or values containing special characters ('=', ';', '#', '\' and a -newline) must be escaped with a backslash '\'. +Metadata keys or values containing special characters (@samp{=}, @samp{;}, +@samp{#}, @samp{\} and a newline) must be escaped with a backslash @samp{\}. @item -Note that whitespace in metadata (e.g. foo = bar) is considered to be a part of -the tag (in the example above key is 'foo ', value is ' bar'). +Note that whitespace in metadata (e.g. @samp{foo = bar}) is considered to be +a part of the tag (in the example above key is @samp{foo }, value is +@samp{ bar}). @end enumerate A ffmetadata file might look like this: @@ -53,7 +57,7 @@ A ffmetadata file might look like this: ;FFMETADATA1 title=bike\\shed ;this is a comment -artist=Libav troll team +artist=FFmpeg troll team [CHAPTER] TIMEBASE=1/1000 @@ -65,4 +69,20 @@ title=chapter \#1 title=multi\ line @end example + +By using the ffmetadata muxer and demuxer it is possible to extract +metadata from an input file to an ffmetadata file, and then transcode +the file into an output file with the edited ffmetadata file. + +Extracting an ffmetadata file with @file{ffmpeg} goes as follows: +@example +ffmpeg -i INPUT -f ffmetadata FFMETADATAFILE +@end example + +Reinserting edited metadata information from the FFMETADATAFILE file can +be done as: +@example +ffmpeg -i INPUT -i FFMETADATAFILE -map_metadata 1 -codec copy OUTPUT +@end example + @c man end METADATA diff --git a/doc/mips.txt b/doc/mips.txt new file mode 100644 index 0000000000..a84e89ae79 --- /dev/null +++ b/doc/mips.txt @@ -0,0 +1,79 @@ +MIPS optimizations info +=============================================== + +MIPS optimizations of codecs are targeting MIPS 74k family of +CPUs. Some of these optimizations are relying more on properties of +this architecture and some are relying less (and can be used on most +MIPS architectures without degradation in performance). + +Along with FFMPEG copyright notice, there is MIPS copyright notice in +all the files that are created by people from MIPS Technologies. + +Example of copyright notice: +=============================================== +/* + * Copyright (c) 2012 + * MIPS Technologies, Inc., California. + * + * Redistribution and use in source and binary forms, with or without + * modification, are permitted provided that the following conditions + * are met: + * 1. Redistributions of source code must retain the above copyright + * notice, this list of conditions and the following disclaimer. + * 2. Redistributions in binary form must reproduce the above copyright + * notice, this list of conditions and the following disclaimer in the + * documentation and/or other materials provided with the distribution. + * 3. Neither the name of the MIPS Technologies, Inc., nor the names of its + * contributors may be used to endorse or promote products derived from + * this software without specific prior written permission. + * + * THIS SOFTWARE IS PROVIDED BY THE MIPS TECHNOLOGIES, INC. ``AS IS'' AND + * ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE + * IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE + * ARE DISCLAIMED. IN NO EVENT SHALL THE MIPS TECHNOLOGIES, INC. BE LIABLE + * FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL + * DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS + * OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) + * HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT + * LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY + * OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF + * SUCH DAMAGE. + * + * Author: Author Name (author_name@@mips.com) + */ + +Files that have MIPS copyright notice in them: +=============================================== +* libavutil/mips/ + float_dsp_mips.c + libm_mips.h + softfloat_tables.h +* libavcodec/ + fft_fixed_32.c + fft_init_table.c + fft_table.h + mdct_fixed_32.c +* libavcodec/mips/ + aacdec_fixed.c + aacsbr_fixed.c + aacsbr_template.c + aaccoder_mips.c + aacpsy_mips.h + ac3dsp_mips.c + acelp_filters_mips.c + acelp_vectors_mips.c + amrwbdec_mips.c + amrwbdec_mips.h + celp_filters_mips.c + celp_math_mips.c + compute_antialias_fixed.h + compute_antialias_float.h + lsp_mips.h + dsputil_mips.c + fft_mips.c + fft_table.h + fft_init_table.c + fmtconvert_mips.c + iirfilter_mips.c + mpegaudiodsp_mips_fixed.c + mpegaudiodsp_mips_float.c diff --git a/doc/multithreading.txt b/doc/multithreading.txt index 9b27b108c6..83849deacc 100644 --- a/doc/multithreading.txt +++ b/doc/multithreading.txt @@ -1,7 +1,7 @@ -Libav multithreading methods +FFmpeg multithreading methods ============================================== -Libav provides two methods for multithreading codecs. +FFmpeg provides two methods for multithreading codecs. Slice threading decodes multiple parts of a frame at the same time, using AVCodecContext execute() and execute2(). @@ -54,7 +54,7 @@ thread. If the codec allocates writable tables in its init(), add an init_thread_copy() which re-allocates them for other threads. -Add CODEC_CAP_FRAME_THREADS to the codec capabilities. There will be very little +Add AV_CODEC_CAP_FRAME_THREADS to the codec capabilities. There will be very little speed gain at this point but it should work. If there are inter-frame dependencies, so the codec calls diff --git a/doc/muxers.texi b/doc/muxers.texi index 58801aa1ad..2e6bb4ca2a 100644 --- a/doc/muxers.texi +++ b/doc/muxers.texi @@ -1,10 +1,10 @@ @chapter Muxers @c man begin MUXERS -Muxers are configured elements in Libav which allow writing +Muxers are configured elements in FFmpeg which allow writing multimedia streams to a particular type of file. -When you configure your Libav build, all the supported muxers +When you configure your FFmpeg build, all the supported muxers are enabled by default. You can list all available muxers using the configure option @code{--list-muxers}. @@ -13,11 +13,85 @@ You can disable all the muxers with the configure option with the options @code{--enable-muxer=@var{MUXER}} / @code{--disable-muxer=@var{MUXER}}. -The option @code{-formats} of the av* tools will display the list of +The option @code{-formats} of the ff* tools will display the list of enabled muxers. A description of some of the currently available muxers follows. +@anchor{aiff} +@section aiff + +Audio Interchange File Format muxer. + +@subsection Options + +It accepts the following options: + +@table @option +@item write_id3v2 +Enable ID3v2 tags writing when set to 1. Default is 0 (disabled). + +@item id3v2_version +Select ID3v2 version to write. Currently only version 3 and 4 (aka. +ID3v2.3 and ID3v2.4) are supported. The default is version 4. + +@end table + +@anchor{asf} +@section asf + +Advanced Systems Format muxer. + +Note that Windows Media Audio (wma) and Windows Media Video (wmv) use this +muxer too. + +@subsection Options + +It accepts the following options: + +@table @option +@item packet_size +Set the muxer packet size. By tuning this setting you may reduce data +fragmentation or muxer overhead depending on your source. Default value is +3200, minimum is 100, maximum is 64k. + +@end table + +@anchor{chromaprint} +@section chromaprint + +Chromaprint fingerprinter + +This muxer feeds audio data to the Chromaprint library, which generates +a fingerprint for the provided audio data. It takes a single signed +native-endian 16-bit raw audio stream. + +@subsection Options + +@table @option +@item silence_threshold +Threshold for detecting silence, ranges from 0 to 32767. -1 for default +(required for use with the AcoustID service). + +@item algorithm +Algorithm index to fingerprint with. + +@item fp_format +Format to output the fingerprint as. Accepts the following options: +@table @samp +@item raw +Binary raw fingerprint + +@item compressed +Binary compressed fingerprint + +@item base64 +Base64 compressed fingerprint + +@end table + +@end table + @anchor{crc} @section crc @@ -32,98 +106,323 @@ The output of the muxer consists of a single line of the form: CRC=0x@var{CRC}, where @var{CRC} is a hexadecimal number 0-padded to 8 digits containing the CRC for all the decoded input frames. +See also the @ref{framecrc} muxer. + +@subsection Examples + For example to compute the CRC of the input, and store it in the file @file{out.crc}: @example -avconv -i INPUT -f crc out.crc +ffmpeg -i INPUT -f crc out.crc @end example You can print the CRC to stdout with the command: @example -avconv -i INPUT -f crc - +ffmpeg -i INPUT -f crc - @end example -You can select the output format of each frame with @command{avconv} by +You can select the output format of each frame with @command{ffmpeg} by specifying the audio and video codec and format. For example to compute the CRC of the input audio converted to PCM unsigned 8-bit and the input video converted to MPEG-2 video, use the command: @example -avconv -i INPUT -c:a pcm_u8 -c:v mpeg2video -f crc - +ffmpeg -i INPUT -c:a pcm_u8 -c:v mpeg2video -f crc - @end example -See also the @ref{framecrc} muxer. - @anchor{framecrc} @section framecrc -Per-frame CRC (Cyclic Redundancy Check) testing format. +Per-packet CRC (Cyclic Redundancy Check) testing format. -This muxer computes and prints the Adler-32 CRC for each decoded audio -and video frame. By default audio frames are converted to signed +This muxer computes and prints the Adler-32 CRC for each audio +and video packet. By default audio frames are converted to signed 16-bit raw audio and video frames to raw video before computing the CRC. The output of the muxer consists of a line for each audio and video -frame of the form: @var{stream_index}, @var{frame_dts}, -@var{frame_size}, 0x@var{CRC}, where @var{CRC} is a hexadecimal -number 0-padded to 8 digits containing the CRC of the decoded frame. +packet of the form: +@example +@var{stream_index}, @var{packet_dts}, @var{packet_pts}, @var{packet_duration}, @var{packet_size}, 0x@var{CRC} +@end example + +@var{CRC} is a hexadecimal number 0-padded to 8 digits containing the +CRC of the packet. + +@subsection Examples -For example to compute the CRC of each decoded frame in the input, and -store it in the file @file{out.crc}: +For example to compute the CRC of the audio and video frames in +@file{INPUT}, converted to raw audio and video packets, and store it +in the file @file{out.crc}: @example -avconv -i INPUT -f framecrc out.crc +ffmpeg -i INPUT -f framecrc out.crc @end example -You can print the CRC of each decoded frame to stdout with the command: +To print the information to stdout, use the command: @example -avconv -i INPUT -f framecrc - +ffmpeg -i INPUT -f framecrc - @end example -You can select the output format of each frame with @command{avconv} by -specifying the audio and video codec and format. For example, to +With @command{ffmpeg}, you can select the output format to which the +audio and video frames are encoded before computing the CRC for each +packet by specifying the audio and video codec. For example, to compute the CRC of each decoded input audio frame converted to PCM unsigned 8-bit and of each decoded input video frame converted to MPEG-2 video, use the command: @example -avconv -i INPUT -c:a pcm_u8 -c:v mpeg2video -f framecrc - +ffmpeg -i INPUT -c:a pcm_u8 -c:v mpeg2video -f framecrc - @end example See also the @ref{crc} muxer. +@anchor{framemd5} +@section framemd5 + +Per-packet MD5 testing format. + +This muxer computes and prints the MD5 hash for each audio +and video packet. By default audio frames are converted to signed +16-bit raw audio and video frames to raw video before computing the +hash. + +The output of the muxer consists of a line for each audio and video +packet of the form: +@example +@var{stream_index}, @var{packet_dts}, @var{packet_pts}, @var{packet_duration}, @var{packet_size}, @var{MD5} +@end example + +@var{MD5} is a hexadecimal number representing the computed MD5 hash +for the packet. + +@subsection Examples + +For example to compute the MD5 of the audio and video frames in +@file{INPUT}, converted to raw audio and video packets, and store it +in the file @file{out.md5}: +@example +ffmpeg -i INPUT -f framemd5 out.md5 +@end example + +To print the information to stdout, use the command: +@example +ffmpeg -i INPUT -f framemd5 - +@end example + +See also the @ref{md5} muxer. + +@anchor{gif} +@section gif + +Animated GIF muxer. + +It accepts the following options: + +@table @option +@item loop +Set the number of times to loop the output. Use @code{-1} for no loop, @code{0} +for looping indefinitely (default). + +@item final_delay +Force the delay (expressed in centiseconds) after the last frame. Each frame +ends with a delay until the next frame. The default is @code{-1}, which is a +special value to tell the muxer to re-use the previous delay. In case of a +loop, you might want to customize this value to mark a pause for instance. +@end table + +For example, to encode a gif looping 10 times, with a 5 seconds delay between +the loops: +@example +ffmpeg -i INPUT -loop 10 -final_delay 500 out.gif +@end example + +Note 1: if you wish to extract the frames in separate GIF files, you need to +force the @ref{image2} muxer: +@example +ffmpeg -i INPUT -c:v gif -f image2 "out%d.gif" +@end example + +Note 2: the GIF format has a very small time base: the delay between two frames +can not be smaller than one centi second. + @anchor{hls} @section hls Apple HTTP Live Streaming muxer that segments MPEG-TS according to -the HTTP Live Streaming specification. +the HTTP Live Streaming (HLS) specification. -It creates a playlist file and numbered segment files. The output -filename specifies the playlist filename; the segment filenames -receive the same basename as the playlist, a sequential number and -a .ts extension. +It creates a playlist file, and one or more segment files. The output filename +specifies the playlist filename. +By default, the muxer creates a file for each segment produced. These files +have the same name as the playlist, followed by a sequential number and a +.ts extension. + +For example, to convert an input file with @command{ffmpeg}: @example -avconv -i in.nut out.m3u8 +ffmpeg -i in.nut out.m3u8 @end example +This example will produce the playlist, @file{out.m3u8}, and segment files: +@file{out0.ts}, @file{out1.ts}, @file{out2.ts}, etc. + +See also the @ref{segment} muxer, which provides a more generic and +flexible implementation of a segmenter, and can be used to perform HLS +segmentation. + +@subsection Options + +This muxer supports the following options: @table @option -@item -hls_time @var{seconds} -Set the segment length in seconds. -@item -hls_list_size @var{size} -Set the maximum number of playlist entries. -@item -hls_wrap @var{wrap} -Set the number after which index wraps. -@item -start_number @var{number} -Start the sequence from @var{number}. -@item -hls_base_url @var{baseurl} +@item hls_time @var{seconds} +Set the segment length in seconds. Default value is 2. + +@item hls_list_size @var{size} +Set the maximum number of playlist entries. If set to 0 the list file +will contain all the segments. Default value is 5. + +@item hls_ts_options @var{options_list} +Set output format options using a :-separated list of key=value +parameters. Values containing @code{:} special characters must be +escaped. + +@item hls_wrap @var{wrap} +Set the number after which the segment filename number (the number +specified in each segment file) wraps. If set to 0 the number will be +never wrapped. Default value is 0. + +This option is useful to avoid to fill the disk with many segment +files, and limits the maximum number of segment files written to disk +to @var{wrap}. + +@item start_number @var{number} +Start the playlist sequence number from @var{number}. Default value is +0. + +@item hls_allow_cache @var{allowcache} +Explicitly set whether the client MAY (1) or MUST NOT (0) cache media segments. + +@item hls_base_url @var{baseurl} Append @var{baseurl} to every entry in the playlist. Useful to generate playlists with absolute paths. -@item -hls_allow_cache @var{allowcache} -Explicitly set whether the client MAY (1) or MUST NOT (0) cache media segments -@item -hls_version @var{version} -Set the protocol version. Enables or disables version-specific features -such as the integer (version 2) or decimal EXTINF values (version 3). + +Note that the playlist sequence number must be unique for each segment +and it is not to be confused with the segment filename sequence number +which can be cyclic, for example if the @option{wrap} option is +specified. + +@item hls_segment_filename @var{filename} +Set the segment filename. Unless hls_flags single_file is set @var{filename} +is used as a string format with the segment number: +@example +ffmpeg in.nut -hls_segment_filename 'file%03d.ts' out.m3u8 +@end example +This example will produce the playlist, @file{out.m3u8}, and segment files: +@file{file000.ts}, @file{file001.ts}, @file{file002.ts}, etc. + +@item hls_key_info_file @var{key_info_file} +Use the information in @var{key_info_file} for segment encryption. The first +line of @var{key_info_file} specifies the key URI written to the playlist. The +key URL is used to access the encryption key during playback. The second line +specifies the path to the key file used to obtain the key during the encryption +process. The key file is read as a single packed array of 16 octets in binary +format. The optional third line specifies the initialization vector (IV) as a +hexadecimal string to be used instead of the segment sequence number (default) +for encryption. Changes to @var{key_info_file} will result in segment +encryption with the new key/IV and an entry in the playlist for the new key +URI/IV. + +Key info file format: +@example +@var{key URI} +@var{key file path} +@var{IV} (optional) +@end example + +Example key URIs: +@example +http://server/file.key +/path/to/file.key +file.key +@end example + +Example key file paths: +@example +file.key +/path/to/file.key +@end example + +Example IV: +@example +0123456789ABCDEF0123456789ABCDEF +@end example + +Key info file example: +@example +http://server/file.key +/path/to/file.key +0123456789ABCDEF0123456789ABCDEF +@end example + +Example shell script: +@example +#!/bin/sh +BASE_URL=$@{1:-'.'@} +openssl rand 16 > file.key +echo $BASE_URL/file.key > file.keyinfo +echo file.key >> file.keyinfo +echo $(openssl rand -hex 16) >> file.keyinfo +ffmpeg -f lavfi -re -i testsrc -c:v h264 -hls_flags delete_segments \ + -hls_key_info_file file.keyinfo out.m3u8 +@end example + +@item hls_flags single_file +If this flag is set, the muxer will store all segments in a single MPEG-TS +file, and will use byte ranges in the playlist. HLS playlists generated with +this way will have the version number 4. +For example: +@example +ffmpeg -i in.nut -hls_flags single_file out.m3u8 +@end example +Will produce the playlist, @file{out.m3u8}, and a single segment file, +@file{out.ts}. + +@item hls_flags delete_segments +Segment files removed from the playlist are deleted after a period of time +equal to the duration of the segment plus the duration of the playlist. @end table +@anchor{ico} +@section ico + +ICO file muxer. + +Microsoft's icon file format (ICO) has some strict limitations that should be noted: + +@itemize +@item +Size cannot exceed 256 pixels in any dimension + +@item +Only BMP and PNG images can be stored + +@item +If a BMP image is used, it must be one of the following pixel formats: +@example +BMP Bit Depth FFmpeg Pixel Format +1bit pal8 +4bit pal8 +8bit pal8 +16bit rgb555le +24bit bgr24 +32bit bgra +@end example + +@item +If a BMP image is used, it must use the BITMAPINFOHEADER DIB header + +@item +If a PNG image is used, it must use the rgba pixel format +@end itemize + @anchor{image2} @section image2 @@ -154,64 +453,90 @@ The pattern "img%%-%d.jpg" will specify a sequence of filenames of the form @file{img%-1.jpg}, @file{img%-2.jpg}, ..., @file{img%-10.jpg}, etc. -The following example shows how to use @command{avconv} for creating a +@subsection Examples + +The following example shows how to use @command{ffmpeg} for creating a sequence of files @file{img-001.jpeg}, @file{img-002.jpeg}, ..., taking one image every second from the input video: @example -avconv -i in.avi -vsync 1 -r 1 -f image2 'img-%03d.jpeg' +ffmpeg -i in.avi -vsync 1 -r 1 -f image2 'img-%03d.jpeg' @end example -Note that with @command{avconv}, if the format is not specified with the +Note that with @command{ffmpeg}, if the format is not specified with the @code{-f} option and the output filename specifies an image file format, the image2 muxer is automatically selected, so the previous command can be written as: @example -avconv -i in.avi -vsync 1 -r 1 'img-%03d.jpeg' +ffmpeg -i in.avi -vsync 1 -r 1 'img-%03d.jpeg' @end example Note also that the pattern must not necessarily contain "%d" or "%0@var{N}d", for example to create a single image file @file{img.jpeg} from the input video you can employ the command: @example -avconv -i in.avi -f image2 -frames:v 1 img.jpeg +ffmpeg -i in.avi -f image2 -frames:v 1 img.jpeg +@end example + +The @option{strftime} option allows you to expand the filename with +date and time information. Check the documentation of +the @code{strftime()} function for the syntax. + +For example to generate image files from the @code{strftime()} +"%Y-%m-%d_%H-%M-%S" pattern, the following @command{ffmpeg} command +can be used: +@example +ffmpeg -f v4l2 -r 1 -i /dev/video0 -f image2 -strftime 1 "%Y-%m-%d_%H-%M-%S.jpg" @end example +@subsection Options + @table @option -@item -start_number @var{number} -Start the sequence from @var{number}. +@item start_number +Start the sequence from the specified number. Default value is 0. -@item -update @var{number} -If @var{number} is nonzero, the filename will always be interpreted as just a -filename, not a pattern, and this file will be continuously overwritten with new -images. +@item update +If set to 1, the filename will always be interpreted as just a +filename, not a pattern, and the corresponding file will be continuously +overwritten with new images. Default value is 0. +@item strftime +If set to 1, expand the filename with date and time information from +@code{strftime()}. Default value is 0. @end table +The image muxer supports the .Y.U.V image file format. This format is +special in that that each image frame consists of three files, for +each of the YUV420P components. To read or write this image file format, +specify the name of the '.Y' file. The muxer will automatically open the +'.U' and '.V' files as required. + @section matroska Matroska container muxer. This muxer implements the matroska and webm container specs. +@subsection Metadata + The recognized metadata settings in this muxer are: @table @option +@item title +Set title name provided to a single track. -@item title=@var{title name} -Name provided to a single track -@end table +@item language +Specify the language of the track in the Matroska languages form. -@table @option +The language can be either the 3 letters bibliographic ISO-639-2 (ISO +639-2/B) form (like "fre" for French), or a language code mixed with a +country code for specialities in languages (like "fre-ca" for Canadian +French). -@item language=@var{language name} -Specifies the language of the track in the Matroska languages form -@end table +@item stereo_mode +Set stereo 3D video layout of two views in a single video track. -@table @option - -@item STEREO_MODE=@var{mode} -Stereo 3D video layout of two views in a single video track -@table @option +The following values are recognized: +@table @samp @item mono video is not stereo @item left_right @@ -247,13 +572,14 @@ Both eyes laced in one Block, Right-eye view is first For example a 3D WebM clip can be created using the following command line: @example -avconv -i sample_left_right_clip.mpg -an -c:v libvpx -metadata STEREO_MODE=left_right -y stereo_clip.webm +ffmpeg -i sample_left_right_clip.mpg -an -c:v libvpx -metadata stereo_mode=left_right -y stereo_clip.webm @end example +@subsection Options + This muxer supports the following options: @table @option - @item reserve_index_space By default, this muxer writes the index for seeking (called cues in Matroska terms) at the end of the file, because it cannot know in advance how much space @@ -268,15 +594,44 @@ for most use cases should be about 50kB per hour of video. Note that cues are only written if the output is seekable and this option will have no effect if it is not. - @end table +@anchor{md5} +@section md5 + +MD5 testing format. + +This muxer computes and prints the MD5 hash of all the input audio +and video frames. By default audio frames are converted to signed +16-bit raw audio and video frames to raw video before computing the +hash. Timestamps are ignored. + +The output of the muxer consists of a single line of the form: +MD5=@var{MD5}, where @var{MD5} is a hexadecimal number representing +the computed MD5 hash. + +For example to compute the MD5 hash of the input converted to raw +audio and video, and store it in the file @file{out.md5}: +@example +ffmpeg -i INPUT -f md5 out.md5 +@end example + +You can print the MD5 to stdout with the command: +@example +ffmpeg -i INPUT -f md5 - +@end example + +See also the @ref{framemd5} muxer. + @section mov, mp4, ismv +MOV/MP4/ISMV (Smooth Streaming) muxer. + The mov/mp4/ismv muxer supports fragmentation. Normally, a MOV/MP4 file has all the metadata about all packets stored in one location (written at the end of the file, it can be moved to the start for -better playback using the @command{qt-faststart} tool). A fragmented +better playback by adding @var{faststart} to the @var{movflags}, or +using the @command{qt-faststart} tool). A fragmented file consists of a number of fragments, where packets and metadata about these packets are stored together. Writing a fragmented file has the advantage that the file is decodable even if the @@ -286,10 +641,15 @@ very long files (since writing normal MOV/MP4 files stores info about every single packet in memory until the file is closed). The downside is that it is less compatible with other applications. +@subsection Options + Fragmentation is enabled by setting one of the AVOptions that define how to cut the file into fragments: @table @option +@item -moov_size @var{bytes} +Reserves space for the moov atom at the beginning of the file instead of placing the +moov atom at the end. If the space reserved is insufficient, muxing will fail. @item -movflags frag_keyframe Start a new fragment at each video keyframe. @item -frag_duration @var{duration} @@ -300,7 +660,7 @@ Create fragments that contain up to @var{size} bytes of payload data. Allow the caller to manually choose when to cut fragments, by calling @code{av_write_frame(ctx, NULL)} to write a fragment with the packets written so far. (This is only useful with other -applications integrating libavformat, not from @command{avconv}.) +applications integrating libavformat, not from @command{ffmpeg}.) @item -min_frag_duration @var{duration} Don't create fragments that are shorter than @var{duration} microseconds long. @end table @@ -334,11 +694,14 @@ This option is implicitly set when writing ismv (Smooth Streaming) files. Run a second pass moving the index (moov atom) to the beginning of the file. This operation can take a while, and will not work in various situations such as fragmented output, thus it is not enabled by default. +@item -movflags rtphint +Add RTP hinting tracks to the output file. @item -movflags disable_chpl Disable Nero chapter markers (chpl atom). Normally, both Nero chapters and a QuickTime chapter track are written to the file. With this option set, only the QuickTime chapter track will be written. Nero chapters can -cause failures when the file is reprocessed with certain tagging programs. +cause failures when the file is reprocessed with certain tagging programs, like +mp3Tag 2.61a and iTunes 11.3, most likely other versions are affected as well. @item -movflags omit_tfhd_offset Do not write any absolute base_data_offset in tfhd atoms. This avoids tying fragments to absolute byte positions in the file/streams. @@ -351,10 +714,19 @@ circumstances (avoiding basing track fragment location calculations on the implicit end of the previous track fragment). @end table +@subsection Example + Smooth Streaming content can be pushed in real time to a publishing point on IIS with this muxer. Example: @example -avconv -re @var{<normal input/transcoding options>} -movflags isml+frag_keyframe -f ismv http://server/publishingpoint.isml/Streams(Encoder1) +ffmpeg -re @var{<normal input/transcoding options>} -movflags isml+frag_keyframe -f ismv http://server/publishingpoint.isml/Streams(Encoder1) +@end example + +@subsection Audible AAX + +Audible AAX files are encrypted M4B files, and they can be decrypted by specifying a 4 byte activation secret. +@example +ffmpeg -activation_bytes 1CEB00DA -i test.aax -vn -c:a copy output.mp4 @end example @section mp3 @@ -395,18 +767,19 @@ Examples: Write an mp3 with an ID3v2.3 header and an ID3v1 footer: @example -avconv -i INPUT -id3v2_version 3 -write_id3v1 1 out.mp3 +ffmpeg -i INPUT -id3v2_version 3 -write_id3v1 1 out.mp3 @end example -Attach a picture to an mp3: +To attach a picture to an mp3 file select both the audio and the picture stream +with @code{map}: @example -avconv -i input.mp3 -i cover.png -c copy -metadata:s:v title="Album cover" --metadata:s:v comment="Cover (Front)" out.mp3 +ffmpeg -i input.mp3 -i cover.png -c copy -map 0 -map 1 +-metadata:s:v title="Album cover" -metadata:s:v comment="Cover (Front)" out.mp3 @end example Write a "clean" MP3 without any extra features: @example -avconv -i input.wav -write_xing 0 -id3v2_version 0 out.mp3 +ffmpeg -i input.wav -write_xing 0 -id3v2_version 0 out.mp3 @end example @section mpegts @@ -415,6 +788,13 @@ MPEG transport stream muxer. This muxer implements ISO 13818-1 and part of ETSI EN 300 468. +The recognized metadata settings in mpegts muxer are @code{service_provider} +and @code{service_name}. If they are not set the default for +@code{service_provider} is "FFmpeg" and the default for +@code{service_name} is "Service01". + +@subsection Options + The muxer options are: @table @option @@ -427,24 +807,85 @@ Set the transport_stream_id (default 0x0001). This identifies a transponder in DVB. @item -mpegts_service_id @var{number} Set the service_id (default 0x0001) also known as program in DVB. +@item -mpegts_service_type @var{number} +Set the program service_type (default @var{digital_tv}), see below +a list of pre defined values. @item -mpegts_pmt_start_pid @var{number} Set the first PID for PMT (default 0x1000, max 0x1f00). @item -mpegts_start_pid @var{number} Set the first PID for data packets (default 0x0100, max 0x0f00). +@item -mpegts_m2ts_mode @var{number} +Enable m2ts mode if set to 1. Default value is -1 which disables m2ts mode. @item -muxrate @var{number} Set a constant muxrate (default VBR). @item -pcr_period @var{numer} Override the default PCR retransmission time (default 20ms), ignored if variable muxrate is selected. +@item pat_period @var{number} +Maximal time in seconds between PAT/PMT tables. +@item sdt_period @var{number} +Maximal time in seconds between SDT tables. +@item -pes_payload_size @var{number} +Set minimum PES packet payload in bytes. +@item -mpegts_flags @var{flags} +Set flags (see below). +@item -mpegts_copyts @var{number} +Preserve original timestamps, if value is set to 1. Default value is -1, which +results in shifting timestamps so that they start from 0. +@item -tables_version @var{number} +Set PAT, PMT and SDT version (default 0, valid values are from 0 to 31, inclusively). +This option allows updating stream structure so that standard consumer may +detect the change. To do so, reopen output AVFormatContext (in case of API +usage) or restart ffmpeg instance, cyclically changing tables_version value: +@example +ffmpeg -i source1.ts -codec copy -f mpegts -tables_version 0 udp://1.1.1.1:1111 +ffmpeg -i source2.ts -codec copy -f mpegts -tables_version 1 udp://1.1.1.1:1111 +... +ffmpeg -i source3.ts -codec copy -f mpegts -tables_version 31 udp://1.1.1.1:1111 +ffmpeg -i source1.ts -codec copy -f mpegts -tables_version 0 udp://1.1.1.1:1111 +ffmpeg -i source2.ts -codec copy -f mpegts -tables_version 1 udp://1.1.1.1:1111 +... +@end example @end table -The recognized metadata settings in mpegts muxer are @code{service_provider} -and @code{service_name}. If they are not set the default for -@code{service_provider} is "Libav" and the default for -@code{service_name} is "Service01". +Option mpegts_service_type accepts the following values: + +@table @option +@item hex_value +Any hexdecimal value between 0x01 to 0xff as defined in ETSI 300 468. +@item digital_tv +Digital TV service. +@item digital_radio +Digital Radio service. +@item teletext +Teletext service. +@item advanced_codec_digital_radio +Advanced Codec Digital Radio service. +@item mpeg2_digital_hdtv +MPEG2 Digital HDTV service. +@item advanced_codec_digital_sdtv +Advanced Codec Digital SDTV service. +@item advanced_codec_digital_hdtv +Advanced Codec Digital HDTV service. +@end table + +Option mpegts_flags may take a set of such flags: + +@table @option +@item resend_headers +Reemit PAT/PMT before writing the next packet. +@item latm +Use LATM packetization for AAC. +@item pat_pmt_at_frames +Reemit PAT and PMT at each video frame. +@item system_b +Conform to System B (DVB) instead of System A (ATSC). +@end table + +@subsection Example @example -avconv -i file.mpg -c copy \ +ffmpeg -i file.mpg -c copy \ -mpegts_original_network_id 0x1122 \ -mpegts_transport_stream_id 0x3344 \ -mpegts_service_id 0x5566 \ @@ -455,6 +896,21 @@ avconv -i file.mpg -c copy \ -y out.ts @end example +@section mxf, mxf_d10 + +MXF muxer. + +@subsection Options + +The muxer options are: + +@table @option +@item store_user_comments @var{bool} +Set if user comments should be stored if available or never. +IRT D-10 does not allow user comments. The default is thus to write them for +mxf but not for mxf_d10 +@end table + @section null Null muxer. @@ -462,19 +918,19 @@ Null muxer. This muxer does not generate any output file, it is mainly useful for testing or benchmarking purposes. -For example to benchmark decoding with @command{avconv} you can use the +For example to benchmark decoding with @command{ffmpeg} you can use the command: @example -avconv -benchmark -i INPUT -f null out.null +ffmpeg -benchmark -i INPUT -f null out.null @end example Note that the above command does not read or write the @file{out.null} -file, but specifying the output file is required by the @command{avconv} +file, but specifying the output file is required by the @command{ffmpeg} syntax. Alternatively you can write the command as: @example -avconv -benchmark -i INPUT -f null - +ffmpeg -benchmark -i INPUT -f null - @end example @section nut @@ -485,13 +941,20 @@ Change the syncpoint usage in nut: @table @option @item @var{default} use the normal low-overhead seeking aids. @item @var{none} do not use the syncpoints at all, reducing the overhead but making the stream non-seekable; + Use of this option is not recommended, as the resulting files are very damage + sensitive and seeking is not possible. Also in general the overhead from + syncpoints is negligible. Note, -@code{write_index} 0 can be used to disable + all growing data tables, allowing to mux endless streams with limited memory + and without these disadvantages. @item @var{timestamped} extend the syncpoint with a wallclock field. @end table The @var{none} and @var{timestamped} flags are experimental. +@item -write_index @var{bool} +Write index at the end, the default is to write an index. @end table @example -avconv -i INPUT -f_strict experimental -syncpoints none - | processor +ffmpeg -i INPUT -f_strict experimental -syncpoints none - | processor @end example @section ogg @@ -514,44 +977,508 @@ ogg files can be safely chained. @end table -@section segment +@anchor{segment} +@section segment, stream_segment, ssegment Basic stream segmenter. -The segmenter muxer outputs streams to a number of separate files of nearly -fixed duration. Output filename pattern can be set in a fashion similar to -@ref{image2}. +This muxer outputs streams to a number of separate files of nearly +fixed duration. Output filename pattern can be set in a fashion +similar to @ref{image2}, or by using a @code{strftime} template if +the @option{strftime} option is enabled. + +@code{stream_segment} is a variant of the muxer used to write to +streaming output formats, i.e. which do not require global headers, +and is recommended for outputting e.g. to MPEG transport stream segments. +@code{ssegment} is a shorter alias for @code{stream_segment}. + +Every segment starts with a keyframe of the selected reference stream, +which is set through the @option{reference_stream} option. + +Note that if you want accurate splitting for a video file, you need to +make the input key frames correspond to the exact splitting times +expected by the segmenter, or the segment muxer will start the new +segment with the key frame found next after the specified start +time. -Every segment starts with a video keyframe, if a video stream is present. The segment muxer works best with a single constant frame rate video. -Optionally it can generate a flat list of the created segments, one segment -per line. +Optionally it can generate a list of the created segments, by setting +the option @var{segment_list}. The list type is specified by the +@var{segment_list_type} option. The entry filenames in the segment +list are set by default to the basename of the corresponding segment +files. + +See also the @ref{hls} muxer, which provides a more specific +implementation for HLS segmentation. + +@subsection Options + +The segment muxer supports the following options: @table @option +@item reference_stream @var{specifier} +Set the reference stream, as specified by the string @var{specifier}. +If @var{specifier} is set to @code{auto}, the reference is chosen +automatically. Otherwise it must be a stream specifier (see the ``Stream +specifiers'' chapter in the ffmpeg manual) which specifies the +reference stream. The default value is @code{auto}. + @item segment_format @var{format} Override the inner container format, by default it is guessed by the filename extension. -@item segment_time @var{t} -Set segment duration to @var{t} seconds. + +@item segment_format_options @var{options_list} +Set output format options using a :-separated list of key=value +parameters. Values containing the @code{:} special character must be +escaped. + @item segment_list @var{name} -Generate also a listfile named @var{name}. -@item segment_list_type @var{type} -Select the listing format. -@table @option -@item @var{flat} use a simple flat list of entries. -@item @var{hls} use a m3u8-like structure. +Generate also a listfile named @var{name}. If not specified no +listfile is generated. + +@item segment_list_flags @var{flags} +Set flags affecting the segment list generation. + +It currently supports the following flags: +@table @samp +@item cache +Allow caching (only affects M3U8 list files). + +@item live +Allow live-friendly file generation. @end table + @item segment_list_size @var{size} -Overwrite the listfile once it reaches @var{size} entries. +Update the list file so that it contains at most @var{size} +segments. If 0 the list file will contain all the segments. Default +value is 0. + @item segment_list_entry_prefix @var{prefix} Prepend @var{prefix} to each entry. Useful to generate absolute paths. +By default no prefix is applied. + +@item segment_list_type @var{type} +Select the listing format. + +The following values are recognized: +@table @samp +@item flat +Generate a flat list for the created segments, one segment per line. + +@item csv, ext +Generate a list for the created segments, one segment per line, +each line matching the format (comma-separated values): +@example +@var{segment_filename},@var{segment_start_time},@var{segment_end_time} +@end example + +@var{segment_filename} is the name of the output file generated by the +muxer according to the provided pattern. CSV escaping (according to +RFC4180) is applied if required. + +@var{segment_start_time} and @var{segment_end_time} specify +the segment start and end time expressed in seconds. + +A list file with the suffix @code{".csv"} or @code{".ext"} will +auto-select this format. + +@samp{ext} is deprecated in favor or @samp{csv}. + +@item ffconcat +Generate an ffconcat file for the created segments. The resulting file +can be read using the FFmpeg @ref{concat} demuxer. + +A list file with the suffix @code{".ffcat"} or @code{".ffconcat"} will +auto-select this format. + +@item m3u8 +Generate an extended M3U8 file, version 3, compliant with +@url{http://tools.ietf.org/id/draft-pantos-http-live-streaming}. + +A list file with the suffix @code{".m3u8"} will auto-select this format. +@end table + +If not specified the type is guessed from the list file name suffix. + +@item segment_time @var{time} +Set segment duration to @var{time}, the value must be a duration +specification. Default value is "2". See also the +@option{segment_times} option. + +Note that splitting may not be accurate, unless you force the +reference stream key-frames at the given time. See the introductory +notice and the examples below. + +@item segment_atclocktime @var{1|0} +If set to "1" split at regular clock time intervals starting from 00:00 +o'clock. The @var{time} value specified in @option{segment_time} is +used for setting the length of the splitting interval. + +For example with @option{segment_time} set to "900" this makes it possible +to create files at 12:00 o'clock, 12:15, 12:30, etc. + +Default value is "0". + +@item segment_clocktime_offset @var{duration} +Delay the segment splitting times with the specified duration when using +@option{segment_atclocktime}. + +For example with @option{segment_time} set to "900" and +@option{segment_clocktime_offset} set to "300" this makes it possible to +create files at 12:05, 12:20, 12:35, etc. + +Default value is "0". + +@item segment_clocktime_wrap_duration @var{duration} +Force the segmenter to only start a new segment if a packet reaches the muxer +within the specified duration after the segmenting clock time. This way you +can make the segmenter more resilient to backward local time jumps, such as +leap seconds or transition to standard time from daylight savings time. + +Assuming that the delay between the packets of your source is less than 0.5 +second you can detect a leap second by specifying 0.5 as the duration. + +Default is the maximum possible duration which means starting a new segment +regardless of the elapsed time since the last clock time. + +@item segment_time_delta @var{delta} +Specify the accuracy time when selecting the start time for a +segment, expressed as a duration specification. Default value is "0". + +When delta is specified a key-frame will start a new segment if its +PTS satisfies the relation: +@example +PTS >= start_time - time_delta +@end example + +This option is useful when splitting video content, which is always +split at GOP boundaries, in case a key frame is found just before the +specified split time. + +In particular may be used in combination with the @file{ffmpeg} option +@var{force_key_frames}. The key frame times specified by +@var{force_key_frames} may not be set accurately because of rounding +issues, with the consequence that a key frame time may result set just +before the specified time. For constant frame rate videos a value of +1/(2*@var{frame_rate}) should address the worst case mismatch between +the specified time and the time set by @var{force_key_frames}. + +@item segment_times @var{times} +Specify a list of split points. @var{times} contains a list of comma +separated duration specifications, in increasing order. See also +the @option{segment_time} option. + +@item segment_frames @var{frames} +Specify a list of split video frame numbers. @var{frames} contains a +list of comma separated integer numbers, in increasing order. + +This option specifies to start a new segment whenever a reference +stream key frame is found and the sequential number (starting from 0) +of the frame is greater or equal to the next value in the list. + @item segment_wrap @var{limit} Wrap around segment index once it reaches @var{limit}. + +@item segment_start_number @var{number} +Set the sequence number of the first segment. Defaults to @code{0}. + +@item strftime @var{1|0} +Use the @code{strftime} function to define the name of the new +segments to write. If this is selected, the output segment name must +contain a @code{strftime} function template. Default value is +@code{0}. + +@item break_non_keyframes @var{1|0} +If enabled, allow segments to start on frames other than keyframes. This +improves behavior on some players when the time between keyframes is +inconsistent, but may make things worse on others, and can cause some oddities +during seeking. Defaults to @code{0}. + +@item reset_timestamps @var{1|0} +Reset timestamps at the begin of each segment, so that each segment +will start with near-zero timestamps. It is meant to ease the playback +of the generated segments. May not work with some combinations of +muxers/codecs. It is set to @code{0} by default. + +@item initial_offset @var{offset} +Specify timestamp offset to apply to the output packet timestamps. The +argument must be a time duration specification, and defaults to 0. +@end table + +@subsection Examples + +@itemize +@item +Remux the content of file @file{in.mkv} to a list of segments +@file{out-000.nut}, @file{out-001.nut}, etc., and write the list of +generated segments to @file{out.list}: +@example +ffmpeg -i in.mkv -codec copy -map 0 -f segment -segment_list out.list out%03d.nut +@end example + +@item +Segment input and set output format options for the output segments: +@example +ffmpeg -i in.mkv -f segment -segment_time 10 -segment_format_options movflags=+faststart out%03d.mp4 +@end example + +@item +Segment the input file according to the split points specified by the +@var{segment_times} option: +@example +ffmpeg -i in.mkv -codec copy -map 0 -f segment -segment_list out.csv -segment_times 1,2,3,5,8,13,21 out%03d.nut +@end example + +@item +Use the @command{ffmpeg} @option{force_key_frames} +option to force key frames in the input at the specified location, together +with the segment option @option{segment_time_delta} to account for +possible roundings operated when setting key frame times. +@example +ffmpeg -i in.mkv -force_key_frames 1,2,3,5,8,13,21 -codec:v mpeg4 -codec:a pcm_s16le -map 0 \ +-f segment -segment_list out.csv -segment_times 1,2,3,5,8,13,21 -segment_time_delta 0.05 out%03d.nut +@end example +In order to force key frames on the input file, transcoding is +required. + +@item +Segment the input file by splitting the input file according to the +frame numbers sequence specified with the @option{segment_frames} option: +@example +ffmpeg -i in.mkv -codec copy -map 0 -f segment -segment_list out.csv -segment_frames 100,200,300,500,800 out%03d.nut +@end example + +@item +Convert the @file{in.mkv} to TS segments using the @code{libx264} +and @code{libfaac} encoders: +@example +ffmpeg -i in.mkv -map 0 -codec:v libx264 -codec:a libfaac -f ssegment -segment_list out.list out%03d.ts +@end example + +@item +Segment the input file, and create an M3U8 live playlist (can be used +as live HLS source): +@example +ffmpeg -re -i in.mkv -codec copy -map 0 -f segment -segment_list playlist.m3u8 \ +-segment_list_flags +live -segment_time 10 out%03d.mkv +@end example +@end itemize + +@section smoothstreaming + +Smooth Streaming muxer generates a set of files (Manifest, chunks) suitable for serving with conventional web server. + +@table @option +@item window_size +Specify the number of fragments kept in the manifest. Default 0 (keep all). + +@item extra_window_size +Specify the number of fragments kept outside of the manifest before removing from disk. Default 5. + +@item lookahead_count +Specify the number of lookahead fragments. Default 2. + +@item min_frag_duration +Specify the minimum fragment duration (in microseconds). Default 5000000. + +@item remove_at_exit +Specify whether to remove all fragments when finished. Default 0 (do not remove). + +@end table + +@section tee + +The tee muxer can be used to write the same data to several files or any +other kind of muxer. It can be used, for example, to both stream a video to +the network and save it to disk at the same time. + +It is different from specifying several outputs to the @command{ffmpeg} +command-line tool because the audio and video data will be encoded only once +with the tee muxer; encoding can be a very expensive process. It is not +useful when using the libavformat API directly because it is then possible +to feed the same packets to several muxers directly. + +The slave outputs are specified in the file name given to the muxer, +separated by '|'. If any of the slave name contains the '|' separator, +leading or trailing spaces or any special character, it must be +escaped (see @ref{quoting_and_escaping,,the "Quoting and escaping" +section in the ffmpeg-utils(1) manual,ffmpeg-utils}). + +Muxer options can be specified for each slave by prepending them as a list of +@var{key}=@var{value} pairs separated by ':', between square brackets. If +the options values contain a special character or the ':' separator, they +must be escaped; note that this is a second level escaping. + +The following special options are also recognized: +@table @option +@item f +Specify the format name. Useful if it cannot be guessed from the +output name suffix. + +@item bsfs[/@var{spec}] +Specify a list of bitstream filters to apply to the specified +output. + +It is possible to specify to which streams a given bitstream filter +applies, by appending a stream specifier to the option separated by +@code{/}. @var{spec} must be a stream specifier (see @ref{Format +stream specifiers}). If the stream specifier is not specified, the +bitstream filters will be applied to all streams in the output. + +Several bitstream filters can be specified, separated by ",". + +@item select +Select the streams that should be mapped to the slave output, +specified by a stream specifier. If not specified, this defaults to +all the input streams. You may use multiple stream specifiers +separated by commas (@code{,}) e.g.: @code{a:0,v} +@end table + +@subsection Examples + +@itemize +@item +Encode something and both archive it in a WebM file and stream it +as MPEG-TS over UDP (the streams need to be explicitly mapped): +@example +ffmpeg -i ... -c:v libx264 -c:a mp2 -f tee -map 0:v -map 0:a + "archive-20121107.mkv|[f=mpegts]udp://10.0.1.255:1234/" +@end example + +@item +Use @command{ffmpeg} to encode the input, and send the output +to three different destinations. The @code{dump_extra} bitstream +filter is used to add extradata information to all the output video +keyframes packets, as requested by the MPEG-TS format. The select +option is applied to @file{out.aac} in order to make it contain only +audio packets. +@example +ffmpeg -i ... -map 0 -flags +global_header -c:v libx264 -c:a aac -strict experimental + -f tee "[bsfs/v=dump_extra]out.ts|[movflags=+faststart]out.mp4|[select=a]out.aac" +@end example + +@item +As below, but select only stream @code{a:1} for the audio output. Note +that a second level escaping must be performed, as ":" is a special +character used to separate options. +@example +ffmpeg -i ... -map 0 -flags +global_header -c:v libx264 -c:a aac -strict experimental + -f tee "[bsfs/v=dump_extra]out.ts|[movflags=+faststart]out.mp4|[select=\'a:1\']out.aac" +@end example +@end itemize + +Note: some codecs may need different options depending on the output format; +the auto-detection of this can not work with the tee muxer. The main example +is the @option{global_header} flag. + +@section webm_dash_manifest + +WebM DASH Manifest muxer. + +This muxer implements the WebM DASH Manifest specification to generate the DASH +manifest XML. It also supports manifest generation for DASH live streams. + +For more information see: + +@itemize @bullet +@item +WebM DASH Specification: @url{https://sites.google.com/a/webmproject.org/wiki/adaptive-streaming/webm-dash-specification} +@item +ISO DASH Specification: @url{http://standards.iso.org/ittf/PubliclyAvailableStandards/c065274_ISO_IEC_23009-1_2014.zip} +@end itemize + +@subsection Options + +This muxer supports the following options: + +@table @option +@item adaptation_sets +This option has the following syntax: "id=x,streams=a,b,c id=y,streams=d,e" where x and y are the +unique identifiers of the adaptation sets and a,b,c,d and e are the indices of the corresponding +audio and video streams. Any number of adaptation sets can be added using this option. + +@item live +Set this to 1 to create a live stream DASH Manifest. Default: 0. + +@item chunk_start_index +Start index of the first chunk. This will go in the @samp{startNumber} attribute +of the @samp{SegmentTemplate} element in the manifest. Default: 0. + +@item chunk_duration_ms +Duration of each chunk in milliseconds. This will go in the @samp{duration} +attribute of the @samp{SegmentTemplate} element in the manifest. Default: 1000. + +@item utc_timing_url +URL of the page that will return the UTC timestamp in ISO format. This will go +in the @samp{value} attribute of the @samp{UTCTiming} element in the manifest. +Default: None. + +@item time_shift_buffer_depth +Smallest time (in seconds) shifting buffer for which any Representation is +guaranteed to be available. This will go in the @samp{timeShiftBufferDepth} +attribute of the @samp{MPD} element. Default: 60. + +@item minimum_update_period +Minimum update period (in seconds) of the manifest. This will go in the +@samp{minimumUpdatePeriod} attribute of the @samp{MPD} element. Default: 0. + +@end table + +@subsection Example +@example +ffmpeg -f webm_dash_manifest -i video1.webm \ + -f webm_dash_manifest -i video2.webm \ + -f webm_dash_manifest -i audio1.webm \ + -f webm_dash_manifest -i audio2.webm \ + -map 0 -map 1 -map 2 -map 3 \ + -c copy \ + -f webm_dash_manifest \ + -adaptation_sets "id=0,streams=0,1 id=1,streams=2,3" \ + manifest.xml +@end example + +@section webm_chunk + +WebM Live Chunk Muxer. + +This muxer writes out WebM headers and chunks as separate files which can be +consumed by clients that support WebM Live streams via DASH. + +@subsection Options + +This muxer supports the following options: + +@table @option +@item chunk_start_index +Index of the first chunk (defaults to 0). + +@item header +Filename of the header where the initialization data will be written. + +@item audio_chunk_duration +Duration of each audio chunk in milliseconds (defaults to 5000). @end table +@subsection Example @example -avconv -i in.mkv -c copy -map 0 -f segment -list out.list out%03d.nut +ffmpeg -f v4l2 -i /dev/video0 \ + -f alsa -i hw:0 \ + -map 0:0 \ + -c:v libvpx-vp9 \ + -s 640x360 -keyint_min 30 -g 30 \ + -f webm_chunk \ + -header webm_live_video_360.hdr \ + -chunk_start_index 1 \ + webm_live_video_360_%d.chk \ + -map 1:0 \ + -c:a libvorbis \ + -b:a 128k \ + -f webm_chunk \ + -header webm_live_audio_128.hdr \ + -chunk_start_index 1 \ + -audio_chunk_duration 1000 \ + webm_live_audio_128_%d.chk @end example @c man end MUXERS diff --git a/doc/nut.texi b/doc/nut.texi index 042c88a3ab..a02f86ace0 100644 --- a/doc/nut.texi +++ b/doc/nut.texi @@ -1,4 +1,5 @@ \input texinfo @c -*- texinfo -*- +@documentencoding UTF-8 @settitle NUT @@ -17,6 +18,10 @@ subtitle and user-defined streams in a simple, yet efficient, way. It was created by a group of FFmpeg and MPlayer developers in 2003 and was finalized in 2008. +The official nut specification is at svn://svn.mplayerhq.hu/nut +In case of any differences between this text and the official specification, +the official specification shall prevail. + @chapter Modes NUT has some variants signaled by using the flags field in its main header. @@ -130,6 +135,7 @@ PFD[32] would for example be signed 32 bit little-endian IEEE float @item RV20 @tab RealVideo 2.0 @item RV30 @tab RealVideo 3.0 @item RV40 @tab RealVideo 4.0 +@item SNOW @tab FFmpeg Snow @item SVQ1 @tab Sorenson Video 1 @item SVQ3 @tab Sorenson Video 3 @item theo @tab Xiph Theora diff --git a/doc/optimization.txt b/doc/optimization.txt index b3dca645a8..1a0b98cd0e 100644 --- a/doc/optimization.txt +++ b/doc/optimization.txt @@ -17,15 +17,15 @@ Understanding these overoptimized functions: As many functions tend to be a bit difficult to understand because of optimizations, it can be hard to optimize them further, or write architecture-specific versions. It is recommended to look at older -revisions of the interesting files (web frontends for the various Libav -branches are listed at http://libav.org/download.html). +revisions of the interesting files (web frontends for the various FFmpeg +branches are listed at http://ffmpeg.org/download.html). Alternatively, look into the other architecture-specific versions in the x86/, ppc/, alpha/ subdirectories. Even if you don't exactly comprehend the instructions, it could help understanding the functions and how they can be optimized. NOTE: If you still don't understand some function, ask at our mailing list!!! -(https://lists.libav.org/mailman/listinfo/libav-devel) +(http://lists.ffmpeg.org/mailman/listinfo/ffmpeg-devel) When is an optimization justified? @@ -191,11 +191,16 @@ __asm__() block. Use external asm (nasm/yasm) or inline asm (__asm__()), do not use intrinsics. The latter requires a good optimizing compiler which gcc is not. +When debugging a x86 external asm compilation issue, if lost in the macro +expansions, add DBG=1 to your make command-line: the input file will be +preprocessed, stripped of the debug/empty lines, then compiled, showing the +actual lines causing issues. + Inline asm vs. external asm --------------------------- Both inline asm (__asm__("..") in a .c file, handled by a compiler such as gcc) and external asm (.s or .asm files, handled by an assembler such as yasm/nasm) -are accepted in Libav. Which one to use differs per specific case. +are accepted in FFmpeg. Which one to use differs per specific case. - if your code is intended to be inlined in a C function, inline asm is always better, because external asm cannot be inlined diff --git a/doc/outdevs.texi b/doc/outdevs.texi index dd7bd6475d..e68653fd7a 100644 --- a/doc/outdevs.texi +++ b/doc/outdevs.texi @@ -1,10 +1,10 @@ @chapter Output Devices @c man begin OUTPUT DEVICES -Output devices are configured elements in Libav which allow to write +Output devices are configured elements in FFmpeg that can write multimedia data to an output device attached to your system. -When you configure your Libav build, all the supported output devices +When you configure your FFmpeg build, all the supported output devices are enabled by default. You can list all available ones using the configure option "--list-outdevs". @@ -13,8 +13,8 @@ You can disable all the output devices using the configure option option "--enable-outdev=@var{OUTDEV}", or you can disable a particular input device using the option "--disable-outdev=@var{OUTDEV}". -The option "-formats" of the av* tools will display the list of -enabled output devices (amongst the muxers). +The option "-devices" of the ff* tools will display the list of +enabled output devices. A description of the currently available output devices follows. @@ -22,12 +22,429 @@ A description of the currently available output devices follows. ALSA (Advanced Linux Sound Architecture) output device. +@subsection Examples + +@itemize +@item +Play a file on default ALSA device: +@example +ffmpeg -i INPUT -f alsa default +@end example + +@item +Play a file on soundcard 1, audio device 7: +@example +ffmpeg -i INPUT -f alsa hw:1,7 +@end example +@end itemize + +@section caca + +CACA output device. + +This output device allows one to show a video stream in CACA window. +Only one CACA window is allowed per application, so you can +have only one instance of this output device in an application. + +To enable this output device you need to configure FFmpeg with +@code{--enable-libcaca}. +libcaca is a graphics library that outputs text instead of pixels. + +For more information about libcaca, check: +@url{http://caca.zoy.org/wiki/libcaca} + +@subsection Options + +@table @option + +@item window_title +Set the CACA window title, if not specified default to the filename +specified for the output device. + +@item window_size +Set the CACA window size, can be a string of the form +@var{width}x@var{height} or a video size abbreviation. +If not specified it defaults to the size of the input video. + +@item driver +Set display driver. + +@item algorithm +Set dithering algorithm. Dithering is necessary +because the picture being rendered has usually far more colours than +the available palette. +The accepted values are listed with @code{-list_dither algorithms}. + +@item antialias +Set antialias method. Antialiasing smoothens the rendered +image and avoids the commonly seen staircase effect. +The accepted values are listed with @code{-list_dither antialiases}. + +@item charset +Set which characters are going to be used when rendering text. +The accepted values are listed with @code{-list_dither charsets}. + +@item color +Set color to be used when rendering text. +The accepted values are listed with @code{-list_dither colors}. + +@item list_drivers +If set to @option{true}, print a list of available drivers and exit. + +@item list_dither +List available dither options related to the argument. +The argument must be one of @code{algorithms}, @code{antialiases}, +@code{charsets}, @code{colors}. +@end table + +@subsection Examples + +@itemize +@item +The following command shows the @command{ffmpeg} output is an +CACA window, forcing its size to 80x25: +@example +ffmpeg -i INPUT -vcodec rawvideo -pix_fmt rgb24 -window_size 80x25 -f caca - +@end example + +@item +Show the list of available drivers and exit: +@example +ffmpeg -i INPUT -pix_fmt rgb24 -f caca -list_drivers true - +@end example + +@item +Show the list of available dither colors and exit: +@example +ffmpeg -i INPUT -pix_fmt rgb24 -f caca -list_dither colors - +@end example +@end itemize + +@section decklink + +The decklink output device provides playback capabilities for Blackmagic +DeckLink devices. + +To enable this output device, you need the Blackmagic DeckLink SDK and you +need to configure with the appropriate @code{--extra-cflags} +and @code{--extra-ldflags}. +On Windows, you need to run the IDL files through @command{widl}. + +DeckLink is very picky about the formats it supports. Pixel format is always +uyvy422, framerate and video size must be determined for your device with +@command{-list_formats 1}. Audio sample rate is always 48 kHz. + +@subsection Options + +@table @option + +@item list_devices +If set to @option{true}, print a list of devices and exit. +Defaults to @option{false}. + +@item list_formats +If set to @option{true}, print a list of supported formats and exit. +Defaults to @option{false}. + +@item preroll +Amount of time to preroll video in seconds. +Defaults to @option{0.5}. + +@end table + +@subsection Examples + +@itemize + +@item +List output devices: +@example +ffmpeg -i test.avi -f decklink -list_devices 1 dummy +@end example + +@item +List supported formats: +@example +ffmpeg -i test.avi -f decklink -list_formats 1 'DeckLink Mini Monitor' +@end example + +@item +Play video clip: +@example +ffmpeg -i test.avi -f decklink -pix_fmt uyvy422 'DeckLink Mini Monitor' +@end example + +@item +Play video clip with non-standard framerate or video size: +@example +ffmpeg -i test.avi -f decklink -pix_fmt uyvy422 -s 720x486 -r 24000/1001 'DeckLink Mini Monitor' +@end example + +@end itemize + +@section fbdev + +Linux framebuffer output device. + +The Linux framebuffer is a graphic hardware-independent abstraction +layer to show graphics on a computer monitor, typically on the +console. It is accessed through a file device node, usually +@file{/dev/fb0}. + +For more detailed information read the file +@file{Documentation/fb/framebuffer.txt} included in the Linux source tree. + +@subsection Options +@table @option + +@item xoffset +@item yoffset +Set x/y coordinate of top left corner. Default is 0. +@end table + +@subsection Examples +Play a file on framebuffer device @file{/dev/fb0}. +Required pixel format depends on current framebuffer settings. +@example +ffmpeg -re -i INPUT -vcodec rawvideo -pix_fmt bgra -f fbdev /dev/fb0 +@end example + +See also @url{http://linux-fbdev.sourceforge.net/}, and fbset(1). + +@section opengl +OpenGL output device. + +To enable this output device you need to configure FFmpeg with @code{--enable-opengl}. + +This output device allows one to render to OpenGL context. +Context may be provided by application or default SDL window is created. + +When device renders to external context, application must implement handlers for following messages: +@code{AV_DEV_TO_APP_CREATE_WINDOW_BUFFER} - create OpenGL context on current thread. +@code{AV_DEV_TO_APP_PREPARE_WINDOW_BUFFER} - make OpenGL context current. +@code{AV_DEV_TO_APP_DISPLAY_WINDOW_BUFFER} - swap buffers. +@code{AV_DEV_TO_APP_DESTROY_WINDOW_BUFFER} - destroy OpenGL context. +Application is also required to inform a device about current resolution by sending @code{AV_APP_TO_DEV_WINDOW_SIZE} message. + +@subsection Options +@table @option + +@item background +Set background color. Black is a default. +@item no_window +Disables default SDL window when set to non-zero value. +Application must provide OpenGL context and both @code{window_size_cb} and @code{window_swap_buffers_cb} callbacks when set. +@item window_title +Set the SDL window title, if not specified default to the filename specified for the output device. +Ignored when @option{no_window} is set. +@item window_size +Set preferred window size, can be a string of the form widthxheight or a video size abbreviation. +If not specified it defaults to the size of the input video, downscaled according to the aspect ratio. +Mostly usable when @option{no_window} is not set. + +@end table + +@subsection Examples +Play a file on SDL window using OpenGL rendering: +@example +ffmpeg -i INPUT -f opengl "window title" +@end example + @section oss OSS (Open Sound System) output device. +@section pulse + +PulseAudio output device. + +To enable this output device you need to configure FFmpeg with @code{--enable-libpulse}. + +More information about PulseAudio can be found on @url{http://www.pulseaudio.org} + +@subsection Options +@table @option + +@item server +Connect to a specific PulseAudio server, specified by an IP address. +Default server is used when not provided. + +@item name +Specify the application name PulseAudio will use when showing active clients, +by default it is the @code{LIBAVFORMAT_IDENT} string. + +@item stream_name +Specify the stream name PulseAudio will use when showing active streams, +by default it is set to the specified output name. + +@item device +Specify the device to use. Default device is used when not provided. +List of output devices can be obtained with command @command{pactl list sinks}. + +@item buffer_size +@item buffer_duration +Control the size and duration of the PulseAudio buffer. A small buffer +gives more control, but requires more frequent updates. + +@option{buffer_size} specifies size in bytes while +@option{buffer_duration} specifies duration in milliseconds. + +When both options are provided then the highest value is used +(duration is recalculated to bytes using stream parameters). If they +are set to 0 (which is default), the device will use the default +PulseAudio duration value. By default PulseAudio set buffer duration +to around 2 seconds. + +@item prebuf +Specify pre-buffering size in bytes. The server does not start with +playback before at least @option{prebuf} bytes are available in the +buffer. By default this option is initialized to the same value as +@option{buffer_size} or @option{buffer_duration} (whichever is bigger). + +@item minreq +Specify minimum request size in bytes. The server does not request less +than @option{minreq} bytes from the client, instead waits until the buffer +is free enough to request more bytes at once. It is recommended to not set +this option, which will initialize this to a value that is deemed sensible +by the server. + +@end table + +@subsection Examples +Play a file on default device on default server: +@example +ffmpeg -i INPUT -f pulse "stream name" +@end example + +@section sdl + +SDL (Simple DirectMedia Layer) output device. + +This output device allows one to show a video stream in an SDL +window. Only one SDL window is allowed per application, so you can +have only one instance of this output device in an application. + +To enable this output device you need libsdl installed on your system +when configuring your build. + +For more information about SDL, check: +@url{http://www.libsdl.org/} + +@subsection Options + +@table @option + +@item window_title +Set the SDL window title, if not specified default to the filename +specified for the output device. + +@item icon_title +Set the name of the iconified SDL window, if not specified it is set +to the same value of @var{window_title}. + +@item window_size +Set the SDL window size, can be a string of the form +@var{width}x@var{height} or a video size abbreviation. +If not specified it defaults to the size of the input video, +downscaled according to the aspect ratio. + +@item window_fullscreen +Set fullscreen mode when non-zero value is provided. +Default value is zero. +@end table + +@subsection Interactive commands + +The window created by the device can be controlled through the +following interactive commands. + +@table @key +@item q, ESC +Quit the device immediately. +@end table + +@subsection Examples + +The following command shows the @command{ffmpeg} output is an +SDL window, forcing its size to the qcif format: +@example +ffmpeg -i INPUT -vcodec rawvideo -pix_fmt yuv420p -window_size qcif -f sdl "SDL output" +@end example + @section sndio sndio audio output device. +@section xv + +XV (XVideo) output device. + +This output device allows one to show a video stream in a X Window System +window. + +@subsection Options + +@table @option +@item display_name +Specify the hardware display name, which determines the display and +communications domain to be used. + +The display name or DISPLAY environment variable can be a string in +the format @var{hostname}[:@var{number}[.@var{screen_number}]]. + +@var{hostname} specifies the name of the host machine on which the +display is physically attached. @var{number} specifies the number of +the display server on that host machine. @var{screen_number} specifies +the screen to be used on that server. + +If unspecified, it defaults to the value of the DISPLAY environment +variable. + +For example, @code{dual-headed:0.1} would specify screen 1 of display +0 on the machine named ``dual-headed''. + +Check the X11 specification for more detailed information about the +display name format. + +@item window_id +When set to non-zero value then device doesn't create new window, +but uses existing one with provided @var{window_id}. By default +this options is set to zero and device creates its own window. + +@item window_size +Set the created window size, can be a string of the form +@var{width}x@var{height} or a video size abbreviation. If not +specified it defaults to the size of the input video. +Ignored when @var{window_id} is set. + +@item window_x +@item window_y +Set the X and Y window offsets for the created window. They are both +set to 0 by default. The values may be ignored by the window manager. +Ignored when @var{window_id} is set. + +@item window_title +Set the window title, if not specified default to the filename +specified for the output device. Ignored when @var{window_id} is set. +@end table + +For more information about XVideo see @url{http://www.x.org/}. + +@subsection Examples + +@itemize +@item +Decode, display and encode video input with @command{ffmpeg} at the +same time: +@example +ffmpeg -i INPUT OUTPUT -f xv display +@end example + +@item +Decode and display the input video to multiple X11 windows: +@example +ffmpeg -i INPUT -f xv normal -vf negate -f xv negated +@end example +@end itemize + @c man end OUTPUT DEVICES diff --git a/doc/platform.texi b/doc/platform.texi index 9c2b9bb067..f7ee456483 100644 --- a/doc/platform.texi +++ b/doc/platform.texi @@ -1,8 +1,9 @@ \input texinfo @c -*- texinfo -*- +@documentencoding UTF-8 -@settitle Platform Specific information +@settitle Platform Specific Information @titlepage -@center @titlefont{Platform Specific information} +@center @titlefont{Platform Specific Information} @end titlepage @top @@ -11,7 +12,7 @@ @chapter Unix-like -Some parts of Libav cannot be built with version 2.15 of the GNU +Some parts of FFmpeg cannot be built with version 2.15 of the GNU assembler which is still provided by a few AMD64 distributions. To make sure your compiler really uses the required version of gas after a binutils upgrade, run: @@ -26,9 +27,9 @@ to configure. @section Advanced linking configuration -If you compiled Libav libraries statically and you want to use them to +If you compiled FFmpeg libraries statically and you want to use them to build your own shared library, you may need to force PIC support (with -@code{--enable-pic} during Libav configure) and add the following option +@code{--enable-pic} during FFmpeg configure) and add the following option to your project LDFLAGS: @example @@ -40,12 +41,12 @@ pass the correct linking flag (e.g. @code{-pie}) to @code{--extra-ldexeflags}. @section BSD -BSD make will not build Libav, you need to install and use GNU Make +BSD make will not build FFmpeg, you need to install and use GNU Make (@command{gmake}). @section (Open)Solaris -GNU Make is required to build Libav, so you have to invoke (@command{gmake}), +GNU Make is required to build FFmpeg, so you have to invoke (@command{gmake}), standard Solaris Make will not work. When building with a non-c99 front-end (gcc, generic suncc) add either @code{--extra-libs=/usr/lib/values-xpg6.o} or @code{--extra-libs=/usr/lib/64/values-xpg6.o} to the configure options @@ -59,20 +60,22 @@ bash ./configure @end example @anchor{Darwin} -@section Darwin (OS X, iPhone) +@section Darwin (Mac OS X, iPhone) The toolchain provided with Xcode is sufficient to build the basic unacelerated code. -OS X on PowerPC or ARM (iPhone) requires a preprocessor from -@url{git://git.libav.org/gas-preprocessor.git} to build the optimized +Mac OS X on PowerPC or ARM (iPhone) requires a preprocessor from +@url{https://github.com/FFmpeg/gas-preprocessor} or +@url{https://github.com/yuvi/gas-preprocessor}(currently outdated) to build the optimized assembly functions. Put the Perl script somewhere -in your PATH, Libav's configure will pick it up automatically. +in your PATH, FFmpeg's configure will pick it up automatically. -OS X on AMD64 and x86 requires @command{yasm} to build most of the -optimized assembly functions @url{http://mxcl.github.com/homebrew/, Homebrew}, -@url{http://www.gentoo.org/proj/en/gentoo-alt/prefix/bootstrap-macos.xml, Gentoo Prefix} -or @url{http://www.macports.org, MacPorts} can easily provide it. +Mac OS X on amd64 and x86 requires @command{yasm} to build most of the +optimized assembly functions. @uref{http://www.finkproject.org/, Fink}, +@uref{http://www.gentoo.org/proj/en/gentoo-alt/prefix/bootstrap-macos.xml, Gentoo Prefix}, +@uref{https://mxcl.github.com/homebrew/, Homebrew} +or @uref{http://www.macports.org, MacPorts} can easily provide it. @chapter DOS @@ -83,15 +86,18 @@ Using a cross-compiler is preferred for various reasons. @chapter OS/2 -For information about compiling Libav on OS/2 see +For information about compiling FFmpeg on OS/2 see @url{http://www.edm2.com/index.php/FFmpeg}. @chapter Windows +To get help and instructions for building FFmpeg under Windows, check out +the FFmpeg Windows Help Forum at @url{http://ffmpeg.zeranoe.com/forum/}. + @section Native Windows compilation using MinGW or MinGW-w64 -Libav can be built to run natively on Windows using the MinGW-w64 +FFmpeg can be built to run natively on Windows using the MinGW-w64 toolchain. Install the latest versions of MSYS2 and MinGW-w64 from @url{http://msys2.github.io/} and/or @url{http://mingw-w64.sourceforge.net/}. You can find detailed installation instructions in the download section and @@ -112,11 +118,12 @@ speed up is close to non-existent for normal one-off builds and is only noticeable when running make for a second time (for example during @code{make install}). -@item In order to compile @command{avplay}, you must have the MinGW development -library of @uref{http://www.libsdl.org/, SDL} and @code{pkg-config} installed. +@item In order to compile FFplay, you must have the MinGW development library +of @uref{http://www.libsdl.org/, SDL} and @code{pkg-config} installed. -@item By using @code{./configure --enable-shared} when configuring Libav, -you can build all libraries as DLLs. +@item By using @code{./configure --enable-shared} when configuring FFmpeg, +you can build the FFmpeg libraries (e.g. libavutil, libavcodec, +libavformat) as DLLs. @end itemize @@ -141,7 +148,7 @@ To target 32bit replace the @code{x86_64} with @code{i686} in the command above. @section Microsoft Visual C++ or Intel C++ Compiler for Windows -Libav can be built with MSVC 2012 or earlier using a C99-to-C89 conversion utility +FFmpeg can be built with MSVC 2012 or earlier using a C99-to-C89 conversion utility and wrapper, or with MSVC 2013 and ICL natively. You will need the following prerequisites: @@ -200,15 +207,15 @@ follow step 3, or compilation will fail. @enumerate @item Grab the @uref{http://zlib.net/, zlib sources}. @item Edit @code{win32/Makefile.msc} so that it uses -MT instead of -MD, since -this is how Libav is built as well. +this is how FFmpeg is built as well. @item Edit @code{zconf.h} and remove its inclusion of @code{unistd.h}. This gets -erroneously included when building Libav. +erroneously included when building FFmpeg. @item Run @code{nmake -f win32/Makefile.msc}. @item Move @code{zlib.lib}, @code{zconf.h}, and @code{zlib.h} to somewhere MSVC can see. @end enumerate -@item Libav has been tested with the following on i686 and x86_64: +@item FFmpeg has been tested with the following on i686 and x86_64: @itemize @item Visual Studio 2010 Pro and Express @item Visual Studio 2012 Pro and Express @@ -220,7 +227,7 @@ Anything else is not officially supported. @end itemize -@subsection Linking to Libav with Microsoft Visual C++ +@subsection Linking to FFmpeg with Microsoft Visual C++ If you plan to link with MSVC-built static libraries, you will need to make sure you have @code{Runtime Library} set to @@ -272,14 +279,14 @@ Replace @code{foo-version} and @code{foo} with the respective library names. You must use the MinGW cross compilation tools available at @url{http://www.mingw.org/}. -Then configure Libav with the following options: +Then configure FFmpeg with the following options: @example ./configure --target-os=mingw32 --cross-prefix=i386-mingw32msvc- @end example (you can change the cross-prefix according to the prefix chosen for the MinGW tools). -Then you can easily test Libav with @uref{http://www.winehq.com/, Wine}. +Then you can easily test FFmpeg with @uref{http://www.winehq.com/, Wine}. @section Compilation under Cygwin @@ -289,7 +296,7 @@ llrint() in its C library. Install your Cygwin with all the "Base" packages, plus the following "Devel" ones: @example -binutils, gcc4-core, make, git, mingw-runtime, texi2html +binutils, gcc4-core, make, git, mingw-runtime, texinfo @end example In order to run FATE you will also need the following "Utils" packages: @@ -297,7 +304,7 @@ In order to run FATE you will also need the following "Utils" packages: diffutils @end example -If you want to build Libav with additional libraries, download Cygwin +If you want to build FFmpeg with additional libraries, download Cygwin "Devel" packages for Ogg and Vorbis from any Cygwin packages repository: @example libogg-devel, libvorbis-devel @@ -339,7 +346,7 @@ and for a build with shared libraries @chapter Plan 9 The native @uref{http://plan9.bell-labs.com/plan9/, Plan 9} compiler -does not implement all the C99 features needed by Libav so the gcc +does not implement all the C99 features needed by FFmpeg so the gcc port must be used. Furthermore, a few items missing from the C library and shell environment need to be fixed. @@ -355,7 +362,7 @@ utility by setting @code{pkgpath} to @item Missing/broken @code{head} and @code{printf} commands -Replacements adequate for building Libav can be found in the +Replacements adequate for building FFmpeg can be found in the @code{compat/plan9} directory. Place these somewhere they will be found by the shell. These are not full implementations of the commands and are @emph{not} suitable for general use. @@ -393,7 +400,7 @@ build system of this library. @item FPU exceptions enabled by default Unlike most other systems, Plan 9 enables FPU exceptions by default. -These must be disabled before calling any Libav functions. While the +These must be disabled before calling any FFmpeg functions. While the included tools will do this automatically, other users of the libraries must do it themselves. diff --git a/doc/print_options.c b/doc/print_options.c index aa75a00ba2..9fd66ca380 100644 --- a/doc/print_options.c +++ b/doc/print_options.c @@ -1,20 +1,20 @@ /* * Copyright (c) 2012 Anton Khirnov * - * This file is part of Libav. + * This file is part of FFmpeg. * - * Libav is free software; you can redistribute it and/or + * FFmpeg is free software; you can redistribute it and/or * modify it under the terms of the GNU Lesser General Public * License as published by the Free Software Foundation; either * version 2.1 of the License, or (at your option) any later version. * - * Libav is distributed in the hope that it will be useful, + * FFmpeg is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Lesser General Public License for more details. * * You should have received a copy of the GNU Lesser General Public - * License along with Libav; if not, write to the Free Software + * License along with FFmpeg; if not, write to the Free Software * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ @@ -26,6 +26,10 @@ #include <string.h> #include <float.h> +// print_options is build for the host, os_support.h isn't needed and is setup +// for the target. without this build breaks on mingw +#define AVFORMAT_OS_SUPPORT_H + #include "libavformat/avformat.h" #include "libavformat/options_table.h" #include "libavcodec/avcodec.h" @@ -113,6 +117,8 @@ int main(int argc, char **argv) if (argc < 2) print_usage(); + printf("@c DO NOT EDIT THIS FILE!\n" + "@c It was generated by print_options.\n\n"); if (!strcmp(argv[1], "format")) show_format_opts(); else if (!strcmp(argv[1], "codec")) diff --git a/doc/protocols.texi b/doc/protocols.texi index f30567d839..05c4bdbfc9 100644 --- a/doc/protocols.texi +++ b/doc/protocols.texi @@ -1,10 +1,10 @@ @chapter Protocols @c man begin PROTOCOLS -Protocols are configured elements in Libav which allow to access -resources which require the use of a particular protocol. +Protocols are configured elements in FFmpeg that enable access to +resources that require specific protocols. -When you configure your Libav build, all the supported protocols are +When you configure your FFmpeg build, all the supported protocols are enabled by default. You can list all available ones using the configure option "--list-protocols". @@ -14,16 +14,68 @@ option "--enable-protocol=@var{PROTOCOL}", or you can disable a particular protocol using the option "--disable-protocol=@var{PROTOCOL}". -The option "-protocols" of the av* tools will display the list of +The option "-protocols" of the ff* tools will display the list of supported protocols. A description of the currently available protocols follows. +@section async + +Asynchronous data filling wrapper for input stream. + +Fill data in a background thread, to decouple I/O operation from demux thread. + +@example +async:@var{URL} +async:http://host/resource +async:cache:http://host/resource +@end example + +@section bluray + +Read BluRay playlist. + +The accepted options are: +@table @option + +@item angle +BluRay angle + +@item chapter +Start chapter (1...N) + +@item playlist +Playlist to read (BDMV/PLAYLIST/?????.mpls) + +@end table + +Examples: + +Read longest playlist from BluRay mounted to /mnt/bluray: +@example +bluray:/mnt/bluray +@end example + +Read angle 2 of playlist 4 from BluRay mounted to /mnt/bluray, start from chapter 2: +@example +-playlist 4 -angle 2 -chapter 2 bluray:/mnt/bluray +@end example + +@section cache + +Caching wrapper for input stream. + +Cache the input stream to temporary file. It brings seeking capability to live streams. + +@example +cache:@var{URL} +@end example + @section concat Physical concatenation protocol. -Allow to read and seek from many resource in sequence as if they were +Read and seek from many resources in sequence as if they were a unique resource. A URL accepted by this protocol has the syntax: @@ -36,30 +88,114 @@ resource to be concatenated, each one possibly specifying a distinct protocol. For example to read a sequence of files @file{split1.mpeg}, -@file{split2.mpeg}, @file{split3.mpeg} with @command{avplay} use the +@file{split2.mpeg}, @file{split3.mpeg} with @command{ffplay} use the command: @example -avplay concat:split1.mpeg\|split2.mpeg\|split3.mpeg +ffplay concat:split1.mpeg\|split2.mpeg\|split3.mpeg @end example Note that you may need to escape the character "|" which is special for many shells. +@section crypto + +AES-encrypted stream reading protocol. + +The accepted options are: +@table @option +@item key +Set the AES decryption key binary block from given hexadecimal representation. + +@item iv +Set the AES decryption initialization vector binary block from given hexadecimal representation. +@end table + +Accepted URL formats: +@example +crypto:@var{URL} +crypto+@var{URL} +@end example + +@section data + +Data in-line in the URI. See @url{http://en.wikipedia.org/wiki/Data_URI_scheme}. + +For example, to convert a GIF file given inline with @command{ffmpeg}: +@example +ffmpeg -i "data:image/gif;base64,R0lGODdhCAAIAMIEAAAAAAAA//8AAP//AP///////////////ywAAAAACAAIAAADF0gEDLojDgdGiJdJqUX02iB4E8Q9jUMkADs=" smiley.png +@end example + @section file File access protocol. -Allow to read from or read to a file. +Read from or write to a file. + +A file URL can have the form: +@example +file:@var{filename} +@end example + +where @var{filename} is the path of the file to read. + +An URL that does not have a protocol prefix will be assumed to be a +file URL. Depending on the build, an URL that looks like a Windows +path with the drive letter at the beginning will also be assumed to be +a file URL (usually not the case in builds for unix-like systems). -For example to read from a file @file{input.mpeg} with @command{avconv} +For example to read from a file @file{input.mpeg} with @command{ffmpeg} use the command: @example -avconv -i file:input.mpeg output.mpeg +ffmpeg -i file:input.mpeg output.mpeg @end example -The av* tools default to the file protocol, that is a resource -specified with the name "FILE.mpeg" is interpreted as the URL -"file:FILE.mpeg". +This protocol accepts the following options: + +@table @option +@item truncate +Truncate existing files on write, if set to 1. A value of 0 prevents +truncating. Default value is 1. + +@item blocksize +Set I/O operation maximum block size, in bytes. Default value is +@code{INT_MAX}, which results in not limiting the requested block size. +Setting this value reasonably low improves user termination request reaction +time, which is valuable for files on slow medium. +@end table + +@section ftp + +FTP (File Transfer Protocol). + +Read from or write to remote resources using FTP protocol. + +Following syntax is required. +@example +ftp://[user[:password]@@]server[:port]/path/to/remote/resource.mpeg +@end example + +This protocol accepts the following options. + +@table @option +@item timeout +Set timeout in microseconds of socket I/O operations used by the underlying low level +operation. By default it is set to -1, which means that the timeout is +not specified. + +@item ftp-anonymous-password +Password used when login as anonymous user. Typically an e-mail address +should be used. + +@item ftp-write-seekable +Control seekability of connection during encoding. If set to 1 the +resource is supposed to be seekable, if set to 0 it is assumed not +to be seekable. Default value is 0. +@end table + +NOTE: Protocol can be used as output, but it is recommended to not do +it, unless special care is taken (tests, customized server configuration +etc.). Different FTP servers behave in different way during seek +operation. ff* tools may produce incomplete content due to server limitations. @section gopher @@ -92,12 +228,21 @@ HTTP (Hyper Text Transfer Protocol). This protocol accepts the following options: @table @option +@item seekable +Control seekability of connection. If set to 1 the resource is +supposed to be seekable, if set to 0 it is assumed not to be seekable, +if set to -1 it will try to autodetect if it is seekable. Default +value is -1. + @item chunked_post If set to 1 use chunked Transfer-Encoding for posts, default is 1. @item content_type Set a specific content type for the POST messages. +@item http_proxy +set HTTP proxy to tunnel through e.g. http://example.com:1234 + @item headers Set custom HTTP headers, can override built in default headers. The value must be a string encoding the headers. @@ -108,9 +253,25 @@ Use persistent connections if set to 1, default is 0. @item post_data Set custom HTTP post data. +@item user-agent @item user_agent -Override the User-Agent header. If not specified a string of the form -"Lavf/<version>" will be used. +Override the User-Agent header. If not specified the protocol will use a +string describing the libavformat build. ("Lavf/<version>") + +@item timeout +Set timeout in microseconds of socket I/O operations used by the underlying low level +operation. By default it is set to -1, which means that the timeout is +not specified. + +@item reconnect_at_eof +If set then eof is treated like an error and causes reconnection, this is useful +for live / endless streams. + +@item reconnect_streamed +If set then even streamed/non seekable streams will be reconnected on errors. + +@item reconnect_delay_max +Sets the maximum delay in seconds after which to give up reconnecting @item mime_type Export the MIME type. @@ -131,16 +292,72 @@ contains the last non-empty metadata packet sent by the server. It should be polled in regular intervals by applications interested in mid-stream metadata updates. +@item cookies +Set the cookies to be sent in future requests. The format of each cookie is the +same as the value of a Set-Cookie HTTP response field. Multiple cookies can be +delimited by a newline character. + @item offset Set initial byte offset. @item end_offset Try to limit the request to bytes preceding this offset. + +@item method +When used as a client option it sets the HTTP method for the request. + +When used as a server option it sets the HTTP method that is going to be +expected from the client(s). +If the expected and the received HTTP method do not match the client will +be given a Bad Request response. +When unset the HTTP method is not checked for now. This will be replaced by +autodetection in the future. + +@item listen +If set to 1 enables experimental HTTP server. This can be used to send data when +used as an output option, or read data from a client with HTTP POST when used as +an input option. +If set to 2 enables experimental mutli-client HTTP server. This is not yet implemented +in ffmpeg.c or ffserver.c and thus must not be used as a command line option. +@example +# Server side (sending): +ffmpeg -i somefile.ogg -c copy -listen 1 -f ogg http://@var{server}:@var{port} + +# Client side (receiving): +ffmpeg -i http://@var{server}:@var{port} -c copy somefile.ogg + +# Client can also be done with wget: +wget http://@var{server}:@var{port} -O somefile.ogg + +# Server side (receiving): +ffmpeg -listen 1 -i http://@var{server}:@var{port} -c copy somefile.ogg + +# Client side (sending): +ffmpeg -i somefile.ogg -chunked_post 0 -c copy -f ogg http://@var{server}:@var{port} + +# Client can also be done with wget: +wget --post-file=somefile.ogg http://@var{server}:@var{port} +@end example + @end table +@subsection HTTP Cookies + +Some HTTP requests will be denied unless cookie values are passed in with the +request. The @option{cookies} option allows these cookies to be specified. At +the very least, each cookie must specify a value along with a path and domain. +HTTP requests that match both the domain and path will automatically include the +cookie value in the HTTP Cookie header field. Multiple cookies can be delimited +by a newline. + +The required syntax to play a stream specifying a cookie is: +@example +ffplay -cookies "nlqptid=nltid=tsn; path=/; domain=somedomain.com;" http://somedomain.com/somestream.m3u8 +@end example + @section Icecast -Icecast (stream to Icecast servers) +Icecast protocol (stream to Icecast servers) This protocol accepts the following options: @@ -158,7 +375,7 @@ Set the stream description. Set the stream website URL. @item ice_public -Set if the stream should be public or not. +Set if the stream should be public. The default is 0 (not public). @item user_agent @@ -178,6 +395,10 @@ HTTP PUT method but the SOURCE method. @end table +@example +icecast://[@var{username}[:@var{password}]@@]@var{server}:@var{port}/@var{mountpoint} +@end example + @section mmst MMS (Microsoft Media Server) protocol over TCP. @@ -202,10 +423,10 @@ be used to test muxers without writing an actual file. Some examples follow. @example # Write the MD5 hash of the encoded AVI file to the file output.avi.md5. -avconv -i input.flv -f avi -y md5:output.avi.md5 +ffmpeg -i input.flv -f avi -y md5:output.avi.md5 # Write the MD5 hash of the encoded AVI file to stdout. -avconv -i input.flv -f avi -y md5: +ffmpeg -i input.flv -f avi -y md5: @end example Note that some formats (typically MOV) require the output protocol to @@ -215,7 +436,7 @@ be seekable, so they will fail with the MD5 output protocol. UNIX pipe access protocol. -Allow to read and write from UNIX pipes. +Read and write from UNIX pipes. The accepted syntax is: @example @@ -227,20 +448,30 @@ pipe (e.g. 0 for stdin, 1 for stdout, 2 for stderr). If @var{number} is not specified, by default the stdout file descriptor will be used for writing, stdin for reading. -For example to read from stdin with @command{avconv}: +For example to read from stdin with @command{ffmpeg}: @example -cat test.wav | avconv -i pipe:0 +cat test.wav | ffmpeg -i pipe:0 # ...this is the same as... -cat test.wav | avconv -i pipe: +cat test.wav | ffmpeg -i pipe: @end example -For writing to stdout with @command{avconv}: +For writing to stdout with @command{ffmpeg}: @example -avconv -i test.wav -f avi pipe:1 | cat > test.avi +ffmpeg -i test.wav -f avi pipe:1 | cat > test.avi # ...this is the same as... -avconv -i test.wav -f avi pipe: | cat > test.avi +ffmpeg -i test.wav -f avi pipe: | cat > test.avi @end example +This protocol accepts the following options: + +@table @option +@item blocksize +Set I/O operation maximum block size, in bytes. Default value is +@code{INT_MAX}, which results in not limiting the requested block size. +Setting this value reasonably low improves user termination request reaction +time, which is valuable if data transmission is slow. +@end table + Note that some formats (typically MOV), require the output protocol to be seekable, so they will fail with the pipe output protocol. @@ -360,16 +591,16 @@ URL of the target stream. Defaults to proto://host[:port]/app. @end table -For example to read with @command{avplay} a multimedia resource named +For example to read with @command{ffplay} a multimedia resource named "sample" from the application "vod" from an RTMP server "myserver": @example -avplay rtmp://myserver/vod/sample +ffplay rtmp://myserver/vod/sample @end example To publish to a password protected server, passing the playpath and app names separately: @example -avconv -re -i <input> -f flv -rtmp_playpath some/long/path -rtmp_app long/app/name rtmp://username:password@@myserver/ +ffmpeg -re -i <input> -f flv -rtmp_playpath some/long/path -rtmp_app long/app/name rtmp://username:password@@myserver/ @end example @section rtmpe @@ -412,6 +643,71 @@ The Real-Time Messaging Protocol tunneled through HTTPS (RTMPTS) is used for streaming multimedia content within HTTPS requests to traverse firewalls. +@section libsmbclient + +libsmbclient permits one to manipulate CIFS/SMB network resources. + +Following syntax is required. + +@example +smb://[[domain:]user[:password@@]]server[/share[/path[/file]]] +@end example + +This protocol accepts the following options. + +@table @option +@item timeout +Set timeout in miliseconds of socket I/O operations used by the underlying +low level operation. By default it is set to -1, which means that the timeout +is not specified. + +@item truncate +Truncate existing files on write, if set to 1. A value of 0 prevents +truncating. Default value is 1. + +@item workgroup +Set the workgroup used for making connections. By default workgroup is not specified. + +@end table + +For more information see: @url{http://www.samba.org/}. + +@section libssh + +Secure File Transfer Protocol via libssh + +Read from or write to remote resources using SFTP protocol. + +Following syntax is required. + +@example +sftp://[user[:password]@@]server[:port]/path/to/remote/resource.mpeg +@end example + +This protocol accepts the following options. + +@table @option +@item timeout +Set timeout of socket I/O operations used by the underlying low level +operation. By default it is set to -1, which means that the timeout +is not specified. + +@item truncate +Truncate existing files on write, if set to 1. A value of 0 prevents +truncating. Default value is 1. + +@item private_key +Specify the path of the file containing private key to use during authorization. +By default libssh searches for keys in the @file{~/.ssh/} directory. + +@end table + +Example: Play a file stored on remote server. + +@example +ffplay sftp://user:password@@server_address:22/home/user/resource.mpeg +@end example + @section librtmp rtmp, rtmpe, rtmps, rtmpt, rtmpte Real-Time Messaging Protocol and its variants supported through @@ -442,22 +738,87 @@ meaning as specified for the RTMP native protocol. See the librtmp manual page (man 3 librtmp) for more information. For example, to stream a file in real-time to an RTMP server using -@command{avconv}: +@command{ffmpeg}: @example -avconv -re -i myfile -f flv rtmp://myserver/live/mystream +ffmpeg -re -i myfile -f flv rtmp://myserver/live/mystream @end example -To play the same stream using @command{avplay}: +To play the same stream using @command{ffplay}: @example -avplay "rtmp://myserver/live/mystream live=1" +ffplay "rtmp://myserver/live/mystream live=1" @end example @section rtp -Real-Time Protocol. +Real-time Transport Protocol. + +The required syntax for an RTP URL is: +rtp://@var{hostname}[:@var{port}][?@var{option}=@var{val}...] + +@var{port} specifies the RTP port to use. + +The following URL options are supported: + +@table @option + +@item ttl=@var{n} +Set the TTL (Time-To-Live) value (for multicast only). + +@item rtcpport=@var{n} +Set the remote RTCP port to @var{n}. + +@item localrtpport=@var{n} +Set the local RTP port to @var{n}. + +@item localrtcpport=@var{n}' +Set the local RTCP port to @var{n}. + +@item pkt_size=@var{n} +Set max packet size (in bytes) to @var{n}. + +@item connect=0|1 +Do a @code{connect()} on the UDP socket (if set to 1) or not (if set +to 0). + +@item sources=@var{ip}[,@var{ip}] +List allowed source IP addresses. + +@item block=@var{ip}[,@var{ip}] +List disallowed (blocked) source IP addresses. + +@item write_to_source=0|1 +Send packets to the source address of the latest received packet (if +set to 1) or to a default remote address (if set to 0). + +@item localport=@var{n} +Set the local RTP port to @var{n}. + +This is a deprecated option. Instead, @option{localrtpport} should be +used. + +@end table + +Important notes: + +@enumerate + +@item +If @option{rtcpport} is not set the RTCP port will be set to the RTP +port value plus 1. + +@item +If @option{localrtpport} (the local RTP port) is not set any available +port will be used for the local RTP and RTCP ports. + +@item +If @option{localrtcpport} (the local RTCP port) is not set it will be +set to the local RTP port value plus 1. +@end enumerate @section rtsp +Real-Time Streaming Protocol. + RTSP is not technically a protocol handler in libavformat, it is a demuxer and muxer. The demuxer supports both normal RTSP (with data transferred over RTP; this is used by e.g. Apple and Microsoft) and Real-RTSP (with @@ -465,21 +826,29 @@ data transferred over RDT). The muxer can be used to send a stream using RTSP ANNOUNCE to a server supporting it (currently Darwin Streaming Server and Mischa Spiegelmock's -@uref{http://github.com/revmischa/rtsp-server, RTSP server}). +@uref{https://github.com/revmischa/rtsp-server, RTSP server}). The required syntax for a RTSP url is: @example rtsp://@var{hostname}[:@var{port}]/@var{path} @end example -The following options (set on the @command{avconv}/@command{avplay} command -line, or set in code via @code{AVOption}s or in @code{avformat_open_input}), -are supported: +Options can be set on the @command{ffmpeg}/@command{ffplay} command +line, or set in code via @code{AVOption}s or in +@code{avformat_open_input}. -Flags for @code{rtsp_transport}: +The following options are supported. @table @option +@item initial_pause +Do not start playing the stream immediately if set to 1. Default value +is 0. +@item rtsp_transport +Set RTSP transport protocols. + +It accepts the following values: +@table @samp @item udp Use UDP as lower transport protocol. @@ -497,15 +866,56 @@ passing proxies. Multiple lower transport protocols may be specified, in that case they are tried one at a time (if the setup of one fails, the next one is tried). -For the muxer, only the @code{tcp} and @code{udp} options are supported. +For the muxer, only the @samp{tcp} and @samp{udp} options are supported. -Flags for @code{rtsp_flags}: +@item rtsp_flags +Set RTSP flags. -@table @option +The following values are accepted: +@table @samp @item filter_src Accept packets only from negotiated peer address and port. @item listen Act as a server, listening for an incoming connection. +@item prefer_tcp +Try TCP for RTP transport first, if TCP is available as RTSP RTP transport. +@end table + +Default value is @samp{none}. + +@item allowed_media_types +Set media types to accept from the server. + +The following flags are accepted: +@table @samp +@item video +@item audio +@item data +@end table + +By default it accepts all media types. + +@item min_port +Set minimum local UDP port. Default value is 5000. + +@item max_port +Set maximum local UDP port. Default value is 65000. + +@item timeout +Set maximum timeout (in seconds) to wait for incoming connections. + +A value of -1 means infinite (default). This option implies the +@option{rtsp_flags} set to @samp{listen}. + +@item reorder_queue_size +Set number of packets to buffer for handling of reordered packets. + +@item stimeout +Set socket TCP I/O timeout in microseconds. + +@item user-agent +Override User-Agent header. If not specified, it defaults to the +libavformat identifier string. @end table When receiving data over UDP, the demuxer tries to reorder received packets @@ -513,36 +923,41 @@ When receiving data over UDP, the demuxer tries to reorder received packets can be disabled by setting the maximum demuxing delay to zero (via the @code{max_delay} field of AVFormatContext). -When watching multi-bitrate Real-RTSP streams with @command{avplay}, the +When watching multi-bitrate Real-RTSP streams with @command{ffplay}, the streams to display can be chosen with @code{-vst} @var{n} and @code{-ast} @var{n} for video and audio respectively, and can be switched on the fly by pressing @code{v} and @code{a}. -Example command lines: +@subsection Examples -To watch a stream over UDP, with a max reordering delay of 0.5 seconds: +The following examples all make use of the @command{ffplay} and +@command{ffmpeg} tools. +@itemize +@item +Watch a stream over UDP, with a max reordering delay of 0.5 seconds: @example -avplay -max_delay 500000 -rtsp_transport udp rtsp://server/video.mp4 +ffplay -max_delay 500000 -rtsp_transport udp rtsp://server/video.mp4 @end example -To watch a stream tunneled over HTTP: - +@item +Watch a stream tunneled over HTTP: @example -avplay -rtsp_transport http rtsp://server/video.mp4 +ffplay -rtsp_transport http rtsp://server/video.mp4 @end example -To send a stream in realtime to a RTSP server, for others to watch: - +@item +Send a stream in realtime to a RTSP server, for others to watch: @example -avconv -re -i @var{input} -f rtsp -muxdelay 0.1 rtsp://server/live.sdp +ffmpeg -re -i @var{input} -f rtsp -muxdelay 0.1 rtsp://server/live.sdp @end example -To receive a stream in realtime: - +@item +Receive a stream in realtime: @example -avconv -rtsp_flags listen -i rtsp://ownaddress/live.sdp @var{output} +ffmpeg -rtsp_flags listen -i rtsp://ownaddress/live.sdp @var{output} @end example +@end itemize @section sap @@ -593,19 +1008,19 @@ Example command lines follow. To broadcast a stream on the local subnet, for watching in VLC: @example -avconv -re -i @var{input} -f sap sap://224.0.0.255?same_port=1 +ffmpeg -re -i @var{input} -f sap sap://224.0.0.255?same_port=1 @end example -Similarly, for watching in avplay: +Similarly, for watching in @command{ffplay}: @example -avconv -re -i @var{input} -f sap sap://224.0.0.255 +ffmpeg -re -i @var{input} -f sap sap://224.0.0.255 @end example -And for watching in avplay, over IPv6: +And for watching in @command{ffplay}, over IPv6: @example -avconv -re -i @var{input} -f sap sap://[ff0e::1:2:3:4] +ffmpeg -re -i @var{input} -f sap sap://[ff0e::1:2:3:4] @end example @subsection Demuxer @@ -627,13 +1042,83 @@ Example command lines follow. To play back the first stream announced on the normal SAP multicast address: @example -avplay sap:// +ffplay sap:// @end example To play back the first stream announced on one the default IPv6 SAP multicast address: @example -avplay sap://[ff0e::2:7ffe] +ffplay sap://[ff0e::2:7ffe] +@end example + +@section sctp + +Stream Control Transmission Protocol. + +The accepted URL syntax is: +@example +sctp://@var{host}:@var{port}[?@var{options}] +@end example + +The protocol accepts the following options: +@table @option +@item listen +If set to any value, listen for an incoming connection. Outgoing connection is done by default. + +@item max_streams +Set the maximum number of streams. By default no limit is set. +@end table + +@section srtp + +Secure Real-time Transport Protocol. + +The accepted options are: +@table @option +@item srtp_in_suite +@item srtp_out_suite +Select input and output encoding suites. + +Supported values: +@table @samp +@item AES_CM_128_HMAC_SHA1_80 +@item SRTP_AES128_CM_HMAC_SHA1_80 +@item AES_CM_128_HMAC_SHA1_32 +@item SRTP_AES128_CM_HMAC_SHA1_32 +@end table + +@item srtp_in_params +@item srtp_out_params +Set input and output encoding parameters, which are expressed by a +base64-encoded representation of a binary block. The first 16 bytes of +this binary block are used as master key, the following 14 bytes are +used as master salt. +@end table + +@section subfile + +Virtually extract a segment of a file or another stream. +The underlying stream must be seekable. + +Accepted options: +@table @option +@item start +Start offset of the extracted segment, in bytes. +@item end +End offset of the extracted segment, in bytes. +@end table + +Examples: + +Extract a chapter from a DVD VOB file (start and end sectors obtained +externally and multiplied by 2048): +@example +subfile,,start,153391104,end,268142592,,:/media/dvd/VIDEO_TS/VTS_08_1.VOB +@end example + +Play an AVI file directly from a TAR archive: +@example +subfile,,start,183241728,end,366490624,,:archive.tar @end example @section tcp @@ -645,25 +1130,45 @@ The required syntax for a TCP url is: tcp://@var{hostname}:@var{port}[?@var{options}] @end example +@var{options} contains a list of &-separated options of the form +@var{key}=@var{val}. + +The list of supported options follows. + @table @option +@item listen=@var{1|0} +Listen for an incoming connection. Default value is 0. -@item listen -Listen for an incoming connection +@item timeout=@var{microseconds} +Set raise error timeout, expressed in microseconds. -@example -avconv -i @var{input} -f @var{format} tcp://@var{hostname}:@var{port}?listen -avplay tcp://@var{hostname}:@var{port} -@end example +This option is only relevant in read mode: if no data arrived in more +than this time interval, raise error. + +@item listen_timeout=@var{milliseconds} +Set listen timeout, expressed in milliseconds. +@item recv_buffer_size=@var{bytes} +Set receive buffer size, expressed bytes. + +@item send_buffer_size=@var{bytes} +Set send buffer size, expressed bytes. @end table +The following example shows how to setup a listening TCP connection +with @command{ffmpeg}, which is then accessed with @command{ffplay}: +@example +ffmpeg -i @var{input} -f @var{format} tcp://@var{hostname}:@var{port}?listen +ffplay tcp://@var{hostname}:@var{port} +@end example + @section tls Transport Layer Security (TLS) / Secure Sockets Layer (SSL) -The required syntax for a TLS url is: +The required syntax for a TLS/SSL url is: @example -tls://@var{hostname}:@var{port} +tls://@var{hostname}:@var{port}[?@var{options}] @end example The following parameters can be set via command line options @@ -671,11 +1176,12 @@ The following parameters can be set via command line options @table @option -@item ca_file +@item ca_file, cafile=@var{filename} A file containing certificate authority (CA) root certificates to treat as trusted. If the linked TLS library contains a default this might not need to be specified for verification to work, but not all libraries and setups have defaults built in. +The file must be in OpenSSL PEM format. @item tls_verify=@var{1|0} If enabled, try to verify the peer that we are communicating with. @@ -688,13 +1194,13 @@ the host name is validated as well.) This is disabled by default since it requires a CA database to be provided by the caller in many cases. -@item cert_file +@item cert_file, cert=@var{filename} A file containing a certificate to use in the handshake with the peer. (When operating as server, in listen mode, this is more often required by the peer, while client certificates only are mandated in certain setups.) -@item key_file +@item key_file, key=@var{filename} A file containing the private key for the certificate. @item listen=@var{1|0} @@ -703,25 +1209,46 @@ the server role in the handshake instead of the client role. @end table +Example command lines: + +To create a TLS/SSL server that serves an input stream. + +@example +ffmpeg -i @var{input} -f @var{format} tls://@var{hostname}:@var{port}?listen&cert=@var{server.crt}&key=@var{server.key} +@end example + +To play back a stream from the TLS/SSL server using @command{ffplay}: + +@example +ffplay tls://@var{hostname}:@var{port} +@end example + @section udp User Datagram Protocol. -The required syntax for a UDP url is: +The required syntax for an UDP URL is: @example udp://@var{hostname}:@var{port}[?@var{options}] @end example @var{options} contains a list of &-separated options of the form @var{key}=@var{val}. -Follow the list of supported options. -@table @option +In case threading is enabled on the system, a circular buffer is used +to store the incoming data, which allows one to reduce loss of data due to +UDP socket buffer overruns. The @var{fifo_size} and +@var{overrun_nonfatal} options are related to this buffer. + +The list of supported options follows. +@table @option @item buffer_size=@var{size} -set the UDP buffer size in bytes +Set the UDP maximum socket buffer size in bytes. This is used to set either +the receive or send buffer size, depending on what the socket is used for. +Default is 64KB. See also @var{fifo_size}. @item localport=@var{port} -override the local UDP port to bind with +Override the local UDP port to bind with. @item localaddr=@var{addr} Choose the local IP address. This is useful e.g. if sending multicast @@ -729,13 +1256,13 @@ and the host has multiple interfaces, where the user can choose which interface to send on by specifying the IP address of that interface. @item pkt_size=@var{size} -set the size in bytes of UDP packets +Set the size in bytes of UDP packets. @item reuse=@var{1|0} -explicitly allow or disallow reusing UDP sockets +Explicitly allow or disallow reusing UDP sockets. @item ttl=@var{ttl} -set the time to live value (for multicast only) +Set the time to live value (for multicast only). @item connect=@var{1|0} Initialize the UDP socket with @code{connect()}. In this case, the @@ -755,24 +1282,50 @@ specified sender IP addresses. @item block=@var{address}[,@var{address}] Ignore packets sent to the multicast group from the specified sender IP addresses. + +@item fifo_size=@var{units} +Set the UDP receiving circular buffer size, expressed as a number of +packets with size of 188 bytes. If not specified defaults to 7*4096. + +@item overrun_nonfatal=@var{1|0} +Survive in case of UDP receiving circular buffer overrun. Default +value is 0. + +@item timeout=@var{microseconds} +Set raise error timeout, expressed in microseconds. + +This option is only relevant in read mode: if no data arrived in more +than this time interval, raise error. + +@item broadcast=@var{1|0} +Explicitly allow or disallow UDP broadcasting. + +Note that broadcasting may not work properly on networks having +a broadcast storm protection. @end table -Some usage examples of the udp protocol with @command{avconv} follow. +@subsection Examples -To stream over UDP to a remote endpoint: +@itemize +@item +Use @command{ffmpeg} to stream over UDP to a remote endpoint: @example -avconv -i @var{input} -f @var{format} udp://@var{hostname}:@var{port} +ffmpeg -i @var{input} -f @var{format} udp://@var{hostname}:@var{port} @end example -To stream in mpegts format over UDP using 188 sized UDP packets, using a large input buffer: +@item +Use @command{ffmpeg} to stream in mpegts format over UDP using 188 +sized UDP packets, using a large input buffer: @example -avconv -i @var{input} -f mpegts udp://@var{hostname}:@var{port}?pkt_size=188&buffer_size=65535 +ffmpeg -i @var{input} -f mpegts udp://@var{hostname}:@var{port}?pkt_size=188&buffer_size=65535 @end example -To receive over UDP from a remote endpoint: +@item +Use @command{ffmpeg} to receive over UDP from a remote endpoint: @example -avconv -i udp://[@var{multicast-address}]:@var{port} +ffmpeg -i udp://[@var{multicast-address}]:@var{port} ... @end example +@end itemize @section unix diff --git a/doc/resampler.texi b/doc/resampler.texi new file mode 100644 index 0000000000..cb7d536cfb --- /dev/null +++ b/doc/resampler.texi @@ -0,0 +1,232 @@ +@chapter Resampler Options +@c man begin RESAMPLER OPTIONS + +The audio resampler supports the following named options. + +Options may be set by specifying -@var{option} @var{value} in the +FFmpeg tools, @var{option}=@var{value} for the aresample filter, +by setting the value explicitly in the +@code{SwrContext} options or using the @file{libavutil/opt.h} API for +programmatic use. + +@table @option + +@item ich, in_channel_count +Set the number of input channels. Default value is 0. Setting this +value is not mandatory if the corresponding channel layout +@option{in_channel_layout} is set. + +@item och, out_channel_count +Set the number of output channels. Default value is 0. Setting this +value is not mandatory if the corresponding channel layout +@option{out_channel_layout} is set. + +@item uch, used_channel_count +Set the number of used input channels. Default value is 0. This option is +only used for special remapping. + +@item isr, in_sample_rate +Set the input sample rate. Default value is 0. + +@item osr, out_sample_rate +Set the output sample rate. Default value is 0. + +@item isf, in_sample_fmt +Specify the input sample format. It is set by default to @code{none}. + +@item osf, out_sample_fmt +Specify the output sample format. It is set by default to @code{none}. + +@item tsf, internal_sample_fmt +Set the internal sample format. Default value is @code{none}. +This will automatically be chosen when it is not explicitly set. + +@item icl, in_channel_layout +@item ocl, out_channel_layout +Set the input/output channel layout. + +See @ref{channel layout syntax,,the Channel Layout section in the ffmpeg-utils(1) manual,ffmpeg-utils} +for the required syntax. + +@item clev, center_mix_level +Set the center mix level. It is a value expressed in deciBel, and must be +in the interval [-32,32]. + +@item slev, surround_mix_level +Set the surround mix level. It is a value expressed in deciBel, and must +be in the interval [-32,32]. + +@item lfe_mix_level +Set LFE mix into non LFE level. It is used when there is a LFE input but no +LFE output. It is a value expressed in deciBel, and must +be in the interval [-32,32]. + +@item rmvol, rematrix_volume +Set rematrix volume. Default value is 1.0. + +@item rematrix_maxval +Set maximum output value for rematrixing. +This can be used to prevent clipping vs. preventing volume reduction. +A value of 1.0 prevents clipping. + +@item flags, swr_flags +Set flags used by the converter. Default value is 0. + +It supports the following individual flags: +@table @option +@item res +force resampling, this flag forces resampling to be used even when the +input and output sample rates match. +@end table + +@item dither_scale +Set the dither scale. Default value is 1. + +@item dither_method +Set dither method. Default value is 0. + +Supported values: +@table @samp +@item rectangular +select rectangular dither +@item triangular +select triangular dither +@item triangular_hp +select triangular dither with high pass +@item lipshitz +select Lipshitz noise shaping dither. +@item shibata +select Shibata noise shaping dither. +@item low_shibata +select low Shibata noise shaping dither. +@item high_shibata +select high Shibata noise shaping dither. +@item f_weighted +select f-weighted noise shaping dither +@item modified_e_weighted +select modified-e-weighted noise shaping dither +@item improved_e_weighted +select improved-e-weighted noise shaping dither + +@end table + +@item resampler +Set resampling engine. Default value is swr. + +Supported values: +@table @samp +@item swr +select the native SW Resampler; filter options precision and cheby are not +applicable in this case. +@item soxr +select the SoX Resampler (where available); compensation, and filter options +filter_size, phase_shift, filter_type & kaiser_beta, are not applicable in this +case. +@end table + +@item filter_size +For swr only, set resampling filter size, default value is 32. + +@item phase_shift +For swr only, set resampling phase shift, default value is 10, and must be in +the interval [0,30]. + +@item linear_interp +Use linear interpolation if set to 1, default value is 0. + +@item cutoff +Set cutoff frequency (swr: 6dB point; soxr: 0dB point) ratio; must be a float +value between 0 and 1. Default value is 0.97 with swr, and 0.91 with soxr +(which, with a sample-rate of 44100, preserves the entire audio band to 20kHz). + +@item precision +For soxr only, the precision in bits to which the resampled signal will be +calculated. The default value of 20 (which, with suitable dithering, is +appropriate for a destination bit-depth of 16) gives SoX's 'High Quality'; a +value of 28 gives SoX's 'Very High Quality'. + +@item cheby +For soxr only, selects passband rolloff none (Chebyshev) & higher-precision +approximation for 'irrational' ratios. Default value is 0. + +@item async +For swr only, simple 1 parameter audio sync to timestamps using stretching, +squeezing, filling and trimming. Setting this to 1 will enable filling and +trimming, larger values represent the maximum amount in samples that the data +may be stretched or squeezed for each second. +Default value is 0, thus no compensation is applied to make the samples match +the audio timestamps. + +@item first_pts +For swr only, assume the first pts should be this value. The time unit is 1 / sample rate. +This allows for padding/trimming at the start of stream. By default, no +assumption is made about the first frame's expected pts, so no padding or +trimming is done. For example, this could be set to 0 to pad the beginning with +silence if an audio stream starts after the video stream or to trim any samples +with a negative pts due to encoder delay. + +@item min_comp +For swr only, set the minimum difference between timestamps and audio data (in +seconds) to trigger stretching/squeezing/filling or trimming of the +data to make it match the timestamps. The default is that +stretching/squeezing/filling and trimming is disabled +(@option{min_comp} = @code{FLT_MAX}). + +@item min_hard_comp +For swr only, set the minimum difference between timestamps and audio data (in +seconds) to trigger adding/dropping samples to make it match the +timestamps. This option effectively is a threshold to select between +hard (trim/fill) and soft (squeeze/stretch) compensation. Note that +all compensation is by default disabled through @option{min_comp}. +The default is 0.1. + +@item comp_duration +For swr only, set duration (in seconds) over which data is stretched/squeezed +to make it match the timestamps. Must be a non-negative double float value, +default value is 1.0. + +@item max_soft_comp +For swr only, set maximum factor by which data is stretched/squeezed to make it +match the timestamps. Must be a non-negative double float value, default value +is 0. + +@item matrix_encoding +Select matrixed stereo encoding. + +It accepts the following values: +@table @samp +@item none +select none +@item dolby +select Dolby +@item dplii +select Dolby Pro Logic II +@end table + +Default value is @code{none}. + +@item filter_type +For swr only, select resampling filter type. This only affects resampling +operations. + +It accepts the following values: +@table @samp +@item cubic +select cubic +@item blackman_nuttall +select Blackman Nuttall windowed sinc +@item kaiser +select Kaiser windowed sinc +@end table + +@item kaiser_beta +For swr only, set Kaiser window beta value. Must be a double float value in the +interval [2,16], default value is 9. + +@item output_sample_bits +For swr only, set number of used output sample bits for dithering. Must be an integer in the +interval [0,64], default value is 0, which means it's not used. + +@end table + +@c man end RESAMPLER OPTIONS diff --git a/doc/scaler.texi b/doc/scaler.texi new file mode 100644 index 0000000000..3e115cdda5 --- /dev/null +++ b/doc/scaler.texi @@ -0,0 +1,144 @@ +@anchor{scaler_options} +@chapter Scaler Options +@c man begin SCALER OPTIONS + +The video scaler supports the following named options. + +Options may be set by specifying -@var{option} @var{value} in the +FFmpeg tools. For programmatic use, they can be set explicitly in the +@code{SwsContext} options or through the @file{libavutil/opt.h} API. + +@table @option + +@anchor{sws_flags} +@item sws_flags +Set the scaler flags. This is also used to set the scaling +algorithm. Only a single algorithm should be selected. + +It accepts the following values: +@table @samp +@item fast_bilinear +Select fast bilinear scaling algorithm. + +@item bilinear +Select bilinear scaling algorithm. + +@item bicubic +Select bicubic scaling algorithm. + +@item experimental +Select experimental scaling algorithm. + +@item neighbor +Select nearest neighbor rescaling algorithm. + +@item area +Select averaging area rescaling algorithm. + +@item bicublin +Select bicubic scaling algorithm for the luma component, bilinear for +chroma components. + +@item gauss +Select Gaussian rescaling algorithm. + +@item sinc +Select sinc rescaling algorithm. + +@item lanczos +Select Lanczos rescaling algorithm. + +@item spline +Select natural bicubic spline rescaling algorithm. + +@item print_info +Enable printing/debug logging. + +@item accurate_rnd +Enable accurate rounding. + +@item full_chroma_int +Enable full chroma interpolation. + +@item full_chroma_inp +Select full chroma input. + +@item bitexact +Enable bitexact output. +@end table + +@item srcw +Set source width. + +@item srch +Set source height. + +@item dstw +Set destination width. + +@item dsth +Set destination height. + +@item src_format +Set source pixel format (must be expressed as an integer). + +@item dst_format +Set destination pixel format (must be expressed as an integer). + +@item src_range +Select source range. + +@item dst_range +Select destination range. + +@anchor{sws_params} +@item param0, param1 +Set scaling algorithm parameters. The specified values are specific of +some scaling algorithms and ignored by others. The specified values +are floating point number values. + +@item sws_dither +Set the dithering algorithm. Accepts one of the following +values. Default value is @samp{auto}. + +@table @samp +@item auto +automatic choice + +@item none +no dithering + +@item bayer +bayer dither + +@item ed +error diffusion dither + +@item a_dither +arithmetic dither, based using addition + +@item x_dither +arithmetic dither, based using xor (more random/less apparent patterning that +a_dither). + +@end table + +@item alphablend +Set the alpha blending to use when the input has alpha but the output does not. +Default value is @samp{none}. + +@table @samp +@item uniform_color +Blend onto a uniform background color + +@item checkerboard +Blend onto a checkerboard + +@item none +No blending + +@end table + +@end table + +@c man end SCALER OPTIONS diff --git a/doc/snow.txt b/doc/snow.txt new file mode 100644 index 0000000000..9d5778d55d --- /dev/null +++ b/doc/snow.txt @@ -0,0 +1,637 @@ +============================================= +Snow Video Codec Specification Draft 20080110 +============================================= + +Introduction: +============= +This specification describes the Snow bitstream syntax and semantics as +well as the formal Snow decoding process. + +The decoding process is described precisely and any compliant decoder +MUST produce the exact same output for a spec-conformant Snow stream. +For encoding, though, any process which generates a stream compliant to +the syntactical and semantic requirements and which is decodable by +the process described in this spec shall be considered a conformant +Snow encoder. + +Definitions: +============ + +MUST the specific part must be done to conform to this standard +SHOULD it is recommended to be done that way, but not strictly required + +ilog2(x) is the rounded down logarithm of x with basis 2 +ilog2(0) = 0 + +Type definitions: +================= + +b 1-bit range coded +u unsigned scalar value range coded +s signed scalar value range coded + + +Bitstream syntax: +================= + +frame: + header + prediction + residual + +header: + keyframe b MID_STATE + if(keyframe || always_reset) + reset_contexts + if(keyframe){ + version u header_state + always_reset b header_state + temporal_decomposition_type u header_state + temporal_decomposition_count u header_state + spatial_decomposition_count u header_state + colorspace_type u header_state + if (nb_planes > 2) { + chroma_h_shift u header_state + chroma_v_shift u header_state + } + spatial_scalability b header_state + max_ref_frames-1 u header_state + qlogs + } + if(!keyframe){ + update_mc b header_state + if(update_mc){ + for(plane=0; plane<nb_plane_types; plane++){ + diag_mc b header_state + htaps/2-1 u header_state + for(i= p->htaps/2; i; i--) + |hcoeff[i]| u header_state + } + } + update_qlogs b header_state + if(update_qlogs){ + spatial_decomposition_count u header_state + qlogs + } + } + + spatial_decomposition_type s header_state + qlog s header_state + mv_scale s header_state + qbias s header_state + block_max_depth s header_state + +qlogs: + for(plane=0; plane<nb_plane_types; plane++){ + quant_table[plane][0][0] s header_state + for(level=0; level < spatial_decomposition_count; level++){ + quant_table[plane][level][1]s header_state + quant_table[plane][level][3]s header_state + } + } + +reset_contexts + *_state[*]= MID_STATE + +prediction: + for(y=0; y<block_count_vertical; y++) + for(x=0; x<block_count_horizontal; x++) + block(0) + +block(level): + mvx_diff=mvy_diff=y_diff=cb_diff=cr_diff=0 + if(keyframe){ + intra=1 + }else{ + if(level!=max_block_depth){ + s_context= 2*left->level + 2*top->level + topleft->level + topright->level + leaf b block_state[4 + s_context] + } + if(level==max_block_depth || leaf){ + intra b block_state[1 + left->intra + top->intra] + if(intra){ + y_diff s block_state[32] + cb_diff s block_state[64] + cr_diff s block_state[96] + }else{ + ref_context= ilog2(2*left->ref) + ilog2(2*top->ref) + if(ref_frames > 1) + ref u block_state[128 + 1024 + 32*ref_context] + mx_context= ilog2(2*abs(left->mx - top->mx)) + my_context= ilog2(2*abs(left->my - top->my)) + mvx_diff s block_state[128 + 32*(mx_context + 16*!!ref)] + mvy_diff s block_state[128 + 32*(my_context + 16*!!ref)] + } + }else{ + block(level+1) + block(level+1) + block(level+1) + block(level+1) + } + } + + +residual: + residual2(luma) + if (nb_planes > 2) { + residual2(chroma_cr) + residual2(chroma_cb) + } + +residual2: + for(level=0; level<spatial_decomposition_count; level++){ + if(level==0) + subband(LL, 0) + subband(HL, level) + subband(LH, level) + subband(HH, level) + } + +subband: + FIXME + +nb_plane_types = gray ? 1 : 2; + +Tag description: +---------------- + +version + 0 + this MUST NOT change within a bitstream + +always_reset + if 1 then the range coder contexts will be reset after each frame + +temporal_decomposition_type + 0 + +temporal_decomposition_count + 0 + +spatial_decomposition_count + FIXME + +colorspace_type + 0 unspecified YcbCr + 1 Gray + 2 Gray + Alpha + 3 GBR + 4 GBRA + this MUST NOT change within a bitstream + +chroma_h_shift + log2(luma.width / chroma.width) + this MUST NOT change within a bitstream + +chroma_v_shift + log2(luma.height / chroma.height) + this MUST NOT change within a bitstream + +spatial_scalability + 0 + +max_ref_frames + maximum number of reference frames + this MUST NOT change within a bitstream + +update_mc + indicates that motion compensation filter parameters are stored in the + header + +diag_mc + flag to enable faster diagonal interpolation + this SHOULD be 1 unless it turns out to be covered by a valid patent + +htaps + number of half pel interpolation filter taps, MUST be even, >0 and <10 + +hcoeff + half pel interpolation filter coefficients, hcoeff[0] are the 2 middle + coefficients [1] are the next outer ones and so on, resulting in a filter + like: ...eff[2], hcoeff[1], hcoeff[0], hcoeff[0], hcoeff[1], hcoeff[2] ... + the sign of the coefficients is not explicitly stored but alternates + after each coeff and coeff[0] is positive, so ...,+,-,+,-,+,+,-,+,-,+,... + hcoeff[0] is not explicitly stored but found by subtracting the sum + of all stored coefficients with signs from 32 + hcoeff[0]= 32 - hcoeff[1] - hcoeff[2] - ... + a good choice for hcoeff and htaps is + htaps= 6 + hcoeff={40,-10,2} + an alternative which requires more computations at both encoder and + decoder side and may or may not be better is + htaps= 8 + hcoeff={42,-14,6,-2} + + +ref_frames + minimum of the number of available reference frames and max_ref_frames + for example the first frame after a key frame always has ref_frames=1 + +spatial_decomposition_type + wavelet type + 0 is a 9/7 symmetric compact integer wavelet + 1 is a 5/3 symmetric compact integer wavelet + others are reserved + stored as delta from last, last is reset to 0 if always_reset || keyframe + +qlog + quality (logarthmic quantizer scale) + stored as delta from last, last is reset to 0 if always_reset || keyframe + +mv_scale + stored as delta from last, last is reset to 0 if always_reset || keyframe + FIXME check that everything works fine if this changes between frames + +qbias + dequantization bias + stored as delta from last, last is reset to 0 if always_reset || keyframe + +block_max_depth + maximum depth of the block tree + stored as delta from last, last is reset to 0 if always_reset || keyframe + +quant_table + quantiztation table + + +Highlevel bitstream structure: +============================= + -------------------------------------------- +| Header | + -------------------------------------------- +| ------------------------------------ | +| | Block0 | | +| | split? | | +| | yes no | | +| | ......... intra? | | +| | : Block01 : yes no | | +| | : Block02 : ....... .......... | | +| | : Block03 : : y DC : : ref index: | | +| | : Block04 : : cb DC : : motion x : | | +| | ......... : cr DC : : motion y : | | +| | ....... .......... | | +| ------------------------------------ | +| ------------------------------------ | +| | Block1 | | +| ... | + -------------------------------------------- +| ------------ ------------ ------------ | +|| Y subbands | | Cb subbands| | Cr subbands|| +|| --- --- | | --- --- | | --- --- || +|| |LL0||HL0| | | |LL0||HL0| | | |LL0||HL0| || +|| --- --- | | --- --- | | --- --- || +|| --- --- | | --- --- | | --- --- || +|| |LH0||HH0| | | |LH0||HH0| | | |LH0||HH0| || +|| --- --- | | --- --- | | --- --- || +|| --- --- | | --- --- | | --- --- || +|| |HL1||LH1| | | |HL1||LH1| | | |HL1||LH1| || +|| --- --- | | --- --- | | --- --- || +|| --- --- | | --- --- | | --- --- || +|| |HH1||HL2| | | |HH1||HL2| | | |HH1||HL2| || +|| ... | | ... | | ... || +| ------------ ------------ ------------ | + -------------------------------------------- + +Decoding process: +================= + + ------------ + | | + | Subbands | + ------------ | | + | | ------------ + | Intra DC | | + | | LL0 subband prediction + ------------ | + \ Dequantizaton + ------------------- \ | +| Reference frames | \ IDWT +| ------- ------- | Motion \ | +||Frame 0| |Frame 1|| Compensation . OBMC v ------- +| ------- ------- | --------------. \------> + --->|Frame n|-->output +| ------- ------- | ------- +||Frame 2| |Frame 3||<----------------------------------/ +| ... | + ------------------- + + +Range Coder: +============ + +Binary Range Coder: +------------------- +The implemented range coder is an adapted version based upon "Range encoding: +an algorithm for removing redundancy from a digitised message." by G. N. N. +Martin. +The symbols encoded by the Snow range coder are bits (0|1). The +associated probabilities are not fix but change depending on the symbol mix +seen so far. + + +bit seen | new state +---------+----------------------------------------------- + 0 | 256 - state_transition_table[256 - old_state]; + 1 | state_transition_table[ old_state]; + +state_transition_table = { + 0, 0, 0, 0, 0, 0, 0, 0, 20, 21, 22, 23, 24, 25, 26, 27, + 28, 29, 30, 31, 32, 33, 34, 35, 36, 37, 37, 38, 39, 40, 41, 42, + 43, 44, 45, 46, 47, 48, 49, 50, 51, 52, 53, 54, 55, 56, 56, 57, + 58, 59, 60, 61, 62, 63, 64, 65, 66, 67, 68, 69, 70, 71, 72, 73, + 74, 75, 75, 76, 77, 78, 79, 80, 81, 82, 83, 84, 85, 86, 87, 88, + 89, 90, 91, 92, 93, 94, 94, 95, 96, 97, 98, 99, 100, 101, 102, 103, +104, 105, 106, 107, 108, 109, 110, 111, 112, 113, 114, 114, 115, 116, 117, 118, +119, 120, 121, 122, 123, 124, 125, 126, 127, 128, 129, 130, 131, 132, 133, 133, +134, 135, 136, 137, 138, 139, 140, 141, 142, 143, 144, 145, 146, 147, 148, 149, +150, 151, 152, 152, 153, 154, 155, 156, 157, 158, 159, 160, 161, 162, 163, 164, +165, 166, 167, 168, 169, 170, 171, 171, 172, 173, 174, 175, 176, 177, 178, 179, +180, 181, 182, 183, 184, 185, 186, 187, 188, 189, 190, 190, 191, 192, 194, 194, +195, 196, 197, 198, 199, 200, 201, 202, 202, 204, 205, 206, 207, 208, 209, 209, +210, 211, 212, 213, 215, 215, 216, 217, 218, 219, 220, 220, 222, 223, 224, 225, +226, 227, 227, 229, 229, 230, 231, 232, 234, 234, 235, 236, 237, 238, 239, 240, +241, 242, 243, 244, 245, 246, 247, 248, 248, 0, 0, 0, 0, 0, 0, 0}; + +FIXME + + +Range Coding of integers: +------------------------- +FIXME + + +Neighboring Blocks: +=================== +left and top are set to the respective blocks unless they are outside of +the image in which case they are set to the Null block + +top-left is set to the top left block unless it is outside of the image in +which case it is set to the left block + +if this block has no larger parent block or it is at the left side of its +parent block and the top right block is not outside of the image then the +top right block is used for top-right else the top-left block is used + +Null block +y,cb,cr are 128 +level, ref, mx and my are 0 + + +Motion Vector Prediction: +========================= +1. the motion vectors of all the neighboring blocks are scaled to +compensate for the difference of reference frames + +scaled_mv= (mv * (256 * (current_reference+1) / (mv.reference+1)) + 128)>>8 + +2. the median of the scaled left, top and top-right vectors is used as +motion vector prediction + +3. the used motion vector is the sum of the predictor and + (mvx_diff, mvy_diff)*mv_scale + + +Intra DC Predicton: +====================== +the luma and chroma values of the left block are used as predictors + +the used luma and chroma is the sum of the predictor and y_diff, cb_diff, cr_diff +to reverse this in the decoder apply the following: +block[y][x].dc[0] = block[y][x-1].dc[0] + y_diff; +block[y][x].dc[1] = block[y][x-1].dc[1] + cb_diff; +block[y][x].dc[2] = block[y][x-1].dc[2] + cr_diff; +block[*][-1].dc[*]= 128; + + +Motion Compensation: +==================== + +Halfpel interpolation: +---------------------- +halfpel interpolation is done by convolution with the halfpel filter stored +in the header: + +horizontal halfpel samples are found by +H1[y][x] = hcoeff[0]*(F[y][x ] + F[y][x+1]) + + hcoeff[1]*(F[y][x-1] + F[y][x+2]) + + hcoeff[2]*(F[y][x-2] + F[y][x+3]) + + ... +h1[y][x] = (H1[y][x] + 32)>>6; + +vertical halfpel samples are found by +H2[y][x] = hcoeff[0]*(F[y ][x] + F[y+1][x]) + + hcoeff[1]*(F[y-1][x] + F[y+2][x]) + + ... +h2[y][x] = (H2[y][x] + 32)>>6; + +vertical+horizontal halfpel samples are found by +H3[y][x] = hcoeff[0]*(H2[y][x ] + H2[y][x+1]) + + hcoeff[1]*(H2[y][x-1] + H2[y][x+2]) + + ... +H3[y][x] = hcoeff[0]*(H1[y ][x] + H1[y+1][x]) + + hcoeff[1]*(H1[y+1][x] + H1[y+2][x]) + + ... +h3[y][x] = (H3[y][x] + 2048)>>12; + + + F H1 F + | | | + | | | + | | | + F H1 F + | | | + | | | + | | | + F-------F-------F-> H1<-F-------F-------F + v v v + H2 H3 H2 + ^ ^ ^ + F-------F-------F-> H1<-F-------F-------F + | | | + | | | + | | | + F H1 F + | | | + | | | + | | | + F H1 F + + +unavailable fullpel samples (outside the picture for example) shall be equal +to the closest available fullpel sample + + +Smaller pel interpolation: +-------------------------- +if diag_mc is set then points which lie on a line between 2 vertically, +horiziontally or diagonally adjacent halfpel points shall be interpolated +linearls with rounding to nearest and halfway values rounded up. +points which lie on 2 diagonals at the same time should only use the one +diagonal not containing the fullpel point + + + + F-->O---q---O<--h1->O---q---O<--F + v \ / v \ / v + O O O O O O O + | / | \ | + q q q q q + | / | \ | + O O O O O O O + ^ / \ ^ / \ ^ + h2-->O---q---O<--h3->O---q---O<--h2 + v \ / v \ / v + O O O O O O O + | \ | / | + q q q q q + | \ | / | + O O O O O O O + ^ / \ ^ / \ ^ + F-->O---q---O<--h1->O---q---O<--F + + + +the remaining points shall be bilinearly interpolated from the +up to 4 surrounding halfpel and fullpel points, again rounding should be to +nearest and halfway values rounded up + +compliant Snow decoders MUST support 1-1/8 pel luma and 1/2-1/16 pel chroma +interpolation at least + + +Overlapped block motion compensation: +------------------------------------- +FIXME + +LL band prediction: +=================== +Each sample in the LL0 subband is predicted by the median of the left, top and +left+top-topleft samples, samples outside the subband shall be considered to +be 0. To reverse this prediction in the decoder apply the following. +for(y=0; y<height; y++){ + for(x=0; x<width; x++){ + sample[y][x] += median(sample[y-1][x], + sample[y][x-1], + sample[y-1][x]+sample[y][x-1]-sample[y-1][x-1]); + } +} +sample[-1][*]=sample[*][-1]= 0; +width,height here are the width and height of the LL0 subband not of the final +video + + +Dequantizaton: +============== +FIXME + +Wavelet Transform: +================== + +Snow supports 2 wavelet transforms, the symmetric biorthogonal 5/3 integer +transform and a integer approximation of the symmetric biorthogonal 9/7 +daubechies wavelet. + +2D IDWT (inverse discrete wavelet transform) +-------------------------------------------- +The 2D IDWT applies a 2D filter recursively, each time combining the +4 lowest frequency subbands into a single subband until only 1 subband +remains. +The 2D filter is done by first applying a 1D filter in the vertical direction +and then applying it in the horizontal one. + --------------- --------------- --------------- --------------- +|LL0|HL0| | | | | | | | | | | | +|---+---| HL1 | | L0|H0 | HL1 | | LL1 | HL1 | | | | +|LH0|HH0| | | | | | | | | | | | +|-------+-------|->|-------+-------|->|-------+-------|->| L1 | H1 |->... +| | | | | | | | | | | | +| LH1 | HH1 | | LH1 | HH1 | | LH1 | HH1 | | | | +| | | | | | | | | | | | + --------------- --------------- --------------- --------------- + + +1D Filter: +---------- +1. interleave the samples of the low and high frequency subbands like +s={L0, H0, L1, H1, L2, H2, L3, H3, ... } +note, this can end with a L or a H, the number of elements shall be w +s[-1] shall be considered equivalent to s[1 ] +s[w ] shall be considered equivalent to s[w-2] + +2. perform the lifting steps in order as described below + +5/3 Integer filter: +1. s[i] -= (s[i-1] + s[i+1] + 2)>>2; for all even i < w +2. s[i] += (s[i-1] + s[i+1] )>>1; for all odd i < w + +\ | /|\ | /|\ | /|\ | /|\ + \|/ | \|/ | \|/ | \|/ | + + | + | + | + | -1/4 + /|\ | /|\ | /|\ | /|\ | +/ | \|/ | \|/ | \|/ | \|/ + | + | + | + | + +1/2 + + +Snow's 9/7 Integer filter: +1. s[i] -= (3*(s[i-1] + s[i+1]) + 4)>>3; for all even i < w +2. s[i] -= s[i-1] + s[i+1] ; for all odd i < w +3. s[i] += ( s[i-1] + s[i+1] + 4*s[i] + 8)>>4; for all even i < w +4. s[i] += (3*(s[i-1] + s[i+1]) )>>1; for all odd i < w + +\ | /|\ | /|\ | /|\ | /|\ + \|/ | \|/ | \|/ | \|/ | + + | + | + | + | -3/8 + /|\ | /|\ | /|\ | /|\ | +/ | \|/ | \|/ | \|/ | \|/ + (| + (| + (| + (| + -1 +\ + /|\ + /|\ + /|\ + /|\ +1/4 + \|/ | \|/ | \|/ | \|/ | + + | + | + | + | +1/16 + /|\ | /|\ | /|\ | /|\ | +/ | \|/ | \|/ | \|/ | \|/ + | + | + | + | + +3/2 + +optimization tips: +following are exactly identical +(3a)>>1 == a + (a>>1) +(a + 4b + 8)>>4 == ((a>>2) + b + 2)>>2 + +16bit implementation note: +The IDWT can be implemented with 16bits, but this requires some care to +prevent overflows, the following list, lists the minimum number of bits needed +for some terms +1. lifting step +A= s[i-1] + s[i+1] 16bit +3*A + 4 18bit +A + (A>>1) + 2 17bit + +3. lifting step +s[i-1] + s[i+1] 17bit + +4. lifiting step +3*(s[i-1] + s[i+1]) 17bit + + +TODO: +===== +Important: +finetune initial contexts +flip wavelet? +try to use the wavelet transformed predicted image (motion compensated image) as context for coding the residual coefficients +try the MV length as context for coding the residual coefficients +use extradata for stuff which is in the keyframes now? +implement per picture halfpel interpolation +try different range coder state transition tables for different contexts + +Not Important: +compare the 6 tap and 8 tap hpel filters (psnr/bitrate and subjective quality) +spatial_scalability b vs u (!= 0 breaks syntax anyway so we can add a u later) + + +Credits: +======== +Michael Niedermayer +Loren Merritt + + +Copyright: +========== +GPL + GFDL + whatever is needed to make this a RFC diff --git a/doc/soc.txt b/doc/soc.txt deleted file mode 100644 index 89728b5201..0000000000 --- a/doc/soc.txt +++ /dev/null @@ -1,24 +0,0 @@ -Google Summer of Code and similar project guidelines - -Summer of Code is a project by Google in which students are paid to implement -some nice new features for various participating open source projects ... - -This text is a collection of things to take care of for the next soc as -it's a little late for this year's soc (2006). - -The Goal: -Our goal in respect to soc is and must be of course exactly one thing and -that is to improve Libav, to reach this goal, code must -* conform to the development policy and patch submission guidelines -* must improve Libav somehow (faster, smaller, "better", - more codecs supported, fewer bugs, cleaner, ...) - -for mentors and other developers to help students to reach that goal it is -essential that changes to their codebase are publicly visible, clean and -easy reviewable that again leads us to: -* use of a revision control system like git -* separation of cosmetic from non-cosmetic changes (this is almost entirely - ignored by mentors and students in soc 2006 which might lead to a surprise - when the code will be reviewed at the end before a possible inclusion in - Libav, individual changes were generally not reviewable due to cosmetics). -* frequent commits, so that comments can be provided early diff --git a/doc/style.min.css b/doc/style.min.css new file mode 100644 index 0000000000..6843fda57d --- /dev/null +++ b/doc/style.min.css @@ -0,0 +1,23 @@ +/*! +The MIT License (MIT) + +Copyright (c) 2014 Barbara Lepage <db0company@gmail.com> + +Permission is hereby granted, free of charge, to any person obtaining a copy +of this software and associated documentation files (the "Software"), to deal +in the Software without restriction, including without limitation the rights +to use, copy, modify, merge, publish, distribute, sublicense, and/or sell +copies of the Software, and to permit persons to whom the Software is +furnished to do so, subject to the following conditions: + +The above copyright notice and this permission notice shall be included in all +copies or substantial portions of the Software. + +THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR +IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, +FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE +AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER +LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, +OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE +SOFTWARE. + */body{background-color:#313131;color:#e6e6e6;text-align:justify}body, h1, h2, h3, h4, h5, h6{font-family:"Lucida Grande","Lucida Sans Unicode","Lucida Sans","Helvetica Neue",Helvetica,Verdana,Tahoma,sans-serif}a{color:#4cae4c}a strong{color:#e6e6e6}a:hover{color:#7fc77f}a:hover strong{color:#4cae4c}main{width:100% ! important;min-height:600px;margin:auto}h1, h2, h3, h4{font-weight:bold;text-align:left}h1, h2, h3{color:#bebebe}h1 strong, h2 strong, h3 strong{color:#e6e6e6}h4, h5, h6{color:#3c8b3c}h1{border-bottom:4px #bebebe solid;padding:20px 2%}h3{border-bottom:2px #bebebe solid;padding:15px 1%}h4{border-bottom:1px solid #e6e6e6;padding:10px 0;margin:20px 0;color:#e6e6e6}.list-group 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___________/|\___________ _____________ + | / | \ | | + | Rematrix | resample <---->| Buffers | + | \___________ | ___________/ |_____________| + v \|/ +Special Converter v + v ___________/|\___________ _____________ + | / | \ | | + | Rematrix | resample <---->| Buffers | + | \___________ | ___________/ |_____________| + | \|/ + | v + | |\___________ + \ | \ + \ output sample format convert v + \_________________ | ___________/ + \|/ + v + Output + +Planar/Packed conversion is done when needed during sample format conversion. +Every step can be skipped without memcpy when it is not needed. +Either Resampling and Rematrixing can be performed first depending on which +way it is faster. +The Buffers are needed for resampling due to resamplng being a process that +requires future and past data, it thus also introduces inevitably a delay when +used. +Internally 32bit float and 16bit int is supported currently, other formats can +easily be added. +Externally all sample formats in packed and planar configuration are supported +It's also trivial to add special converters for common cases. +If only sample format and/or packed/planar conversion is needed, it +is performed from input to output directly in a single pass with no intermediates. diff --git a/doc/t2h.init b/doc/t2h.init index a42637ae0a..c41be2ef37 100644 --- a/doc/t2h.init +++ b/doc/t2h.init @@ -1,160 +1,53 @@ -# no horiz rules between sections -$end_section = \&Libav_end_section; -sub Libav_end_section($$) -{ -} - -$EXTRA_HEAD = -'<link rel="icon" href="favicon.png" type="image/png" /> -'; - -$CSS_LINES = $ENV{"LIBAV_CSS"} || <<EOT; -<style type="text/css"> -<!-- -.container { - margin-right: auto; - margin-left: auto; - width: 1070px; -} -body { - font-size: 14px; - line-height: 20px; - color: #333333; - background-color: #ffffff; -} -a { - color: #0088cc; - text-decoration: none; -} -a:hover { - color: #005580; - text-decoration: underline; -} -p { - margin: 0 0 10px; -} -h2, -h3, -h4 { - margin: 10px 0; - font-family: inherit; - font-weight: bold; - line-height: 1; - border-color: #D6E9C6; - color: #468847; - border-style: solid; - border-width: 0 0 1px; - padding-left: 0.5em; -} +# Init file for texi2html. -h1 a, -h2 a, -h3 a, -h4 a { - color: inherit; -} -h1 { - font-size: 30px; - line-height: 40px; -} -h2 { - font-size: 20px; - line-height: 40px; -} -h3 { - font-size: 18px; - line-height: 40px; -} -code, -pre { - padding: 0 3px 2px; - font-family: monospace; - font-size: 12px; - color: #333333; - border-radius: 3px; -} -pre { - display: block; - padding: 9.5px; - margin: 0 0 10px; - font-size: 13px; - line-height: 20px; - word-break: break-all; - word-wrap: break-word; - white-space: pre; - white-space: pre-wrap; - background-color: #f5f5f5; - border: 1px solid #ccc; - border-radius: 4px; -} - -code { - padding: 2px 4px; - color: #d14; - background-color: #f7f7f9; - border: 1px solid #e1e1e8; -} -pre code { - padding: 0; - color: inherit; - background-color: transparent; - border: 0; -} -.alert { - padding: 8px 35px 8px 14px; - margin-bottom: 20px; - text-shadow: 0 1px 0 rgba(255, 255, 255, 0.5); - background-color: #fcf8e3; - border: 1px solid #fbeed5; - border-radius: 4px; - color: #c09853; -} +# This is deprecated, and the makeinfo/texi2any version is doc/t2h.pm -.alert-danger, -.alert-error { - background-color: #f2dede; - border-color: #eed3d7; - color: #b94a48; -} -.alert-info { - background-color: #d9edf7; - border-color: #bce8f1; - color: #3a87ad; +# no horiz rules between sections +$end_section = \&FFmpeg_end_section; +sub FFmpeg_end_section($$) +{ } -ul.toc { - list-style-type: none; -} ---> -</style> +my $TEMPLATE_HEADER1 = $ENV{"FFMPEG_HEADER1"} || <<EOT; +<!DOCTYPE html> +<html lang="en"> + <head> + <meta charset="utf-8" /> + <meta http-equiv="X-UA-Compatible" content="IE=edge" /> + <title>FFmpeg documentation</title> + <link rel="stylesheet" href="bootstrap.min.css" /> + <link rel="stylesheet" href="style.min.css" /> EOT -my $TEMPLATE_HEADER = $ENV{"LIBAV_HEADER"} || <<EOT; -<link rel="icon" href="favicon.png" type="image/png" /> -</head> -<body> -<div class="container"> +my $TEMPLATE_HEADER2 = $ENV{"FFMPEG_HEADER2"} || <<EOT; + </head> + <body> + <div style="width: 95%; margin: auto"> EOT -$PRE_BODY_CLOSE = '</div></div>'; +my $TEMPLATE_FOOTER = $ENV{"FFMPEG_FOOTER"} || <<EOT; + </div> + </body> +</html> +EOT $SMALL_RULE = ''; $BODYTEXT = ''; -$print_page_foot = \&Libav_print_page_foot; -sub Libav_print_page_foot($$) +$print_page_foot = \&FFmpeg_print_page_foot; +sub FFmpeg_print_page_foot($$) { my $fh = shift; my $program_string = defined &T2H_DEFAULT_program_string ? T2H_DEFAULT_program_string() : program_string(); print $fh '<footer class="footer pagination-right">' . "\n"; print $fh '<span class="label label-info">' . $program_string; - print $fh "</span></footer></div>\n"; + print $fh "</span></footer></div></div></body>\n"; } -$float = \&Libav_float; +$float = \&FFmpeg_float; -sub Libav_float($$$$) +sub FFmpeg_float($$$$) { my $text = shift; my $float = shift; @@ -181,8 +74,8 @@ sub Libav_float($$$$) return '<div class="float ' . $class . '">' . "$label\n" . $text . '</div>'; } -$print_page_head = \&Libav_print_page_head; -sub Libav_print_page_head($$) +$print_page_head = \&FFmpeg_print_page_head; +sub FFmpeg_print_page_head($$) { my $fh = shift; my $longtitle = "$Texi2HTML::THISDOC{'fulltitle_no_texi'}"; @@ -195,29 +88,34 @@ sub Libav_print_page_head($$) my $encoding = ''; $encoding = "<meta http-equiv=\"Content-Type\" content=\"text/html; charset=$ENCODING\">" if (defined($ENCODING) and ($ENCODING ne '')); $longtitle =~ s/Documentation.*//g; - $longtitle = "Libav documentation : " . $longtitle; + $longtitle = "FFmpeg documentation : " . $longtitle; print $fh <<EOT; -<!DOCTYPE html> -<html> +$TEMPLATE_HEADER1 +$description +<meta name="keywords" content="$longtitle"> +<meta name="Generator" content="$Texi2HTML::THISDOC{program}"> $Texi2HTML::THISDOC{'copying'}<!-- Created on $Texi2HTML::THISDOC{today} by $Texi2HTML::THISDOC{program} --> <!-- $Texi2HTML::THISDOC{program_authors} --> -<head> -<title>$longtitle</title> - -$description -<meta name="keywords" content="$longtitle"> -<meta name="resource-type" content="document"> -<meta name="distribution" content="global"> -<meta name="Generator" content="$Texi2HTML::THISDOC{program}"> $encoding -$CSS_LINES -$TEMPLATE_HEADER +$TEMPLATE_HEADER2 EOT } +$print_page_foot = \&FFmpeg_print_page_foot; +sub FFmpeg_print_page_foot($$) +{ + my $fh = shift; + print $fh <<EOT; +$TEMPLATE_FOOTER +EOT +} + +# declare encoding in header +$IN_ENCODING = $ENCODING = "utf-8"; + # no navigation elements $SECTION_NAVIGATION = 0; # the same for texi2html 5.0 diff --git a/doc/t2h.pm b/doc/t2h.pm new file mode 100644 index 0000000000..5efb2da483 --- /dev/null +++ b/doc/t2h.pm @@ -0,0 +1,339 @@ +# makeinfo HTML output init file +# +# Copyright (c) 2011, 2012 Free Software Foundation, Inc. +# Copyright (c) 2014 Andreas Cadhalpun +# Copyright (c) 2014 Tiancheng "Timothy" Gu +# +# This file is part of FFmpeg. +# +# FFmpeg is free software; you can redistribute it and/or modify +# it under the terms of the GNU General Public License as published by +# the Free Software Foundation; either version 3 of the License, or +# (at your option) any later version. +# +# FFmpeg is distributed in the hope that it will be useful, +# but WITHOUT ANY WARRANTY; without even the implied warranty of +# MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU +# General Public License for more details. +# +# You should have received a copy of the GNU General Public +# License along with FFmpeg; if not, write to the Free Software +# Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA + +# no navigation elements +set_from_init_file('HEADERS', 0); + +sub ffmpeg_heading_command($$$$$) +{ + my $self = shift; + my $cmdname = shift; + my $command = shift; + my $args = shift; + my $content = shift; + + my $result = ''; + + # not clear that it may really happen + if ($self->in_string) { + $result .= $self->command_string($command) ."\n" if ($cmdname ne 'node'); + $result .= $content if (defined($content)); + return $result; + } + + my $element_id = $self->command_id($command); + $result .= "<a name=\"$element_id\"></a>\n" + if (defined($element_id) and $element_id ne ''); + + print STDERR "Process $command " + .Texinfo::Structuring::_print_root_command_texi($command)."\n" + if ($self->get_conf('DEBUG')); + my $element; + if ($Texinfo::Common::root_commands{$command->{'cmdname'}} + and $command->{'parent'} + and $command->{'parent'}->{'type'} + and $command->{'parent'}->{'type'} eq 'element') { + $element = $command->{'parent'}; + } + if ($element) { + $result .= &{$self->{'format_element_header'}}($self, $cmdname, + $command, $element); + } + + my $heading_level; + # node is used as heading if there is nothing else. + if ($cmdname eq 'node') { + if (!$element or (!$element->{'extra'}->{'section'} + and $element->{'extra'}->{'node'} + and $element->{'extra'}->{'node'} eq $command + # bogus node may not have been normalized + and defined($command->{'extra'}->{'normalized'}))) { + if ($command->{'extra'}->{'normalized'} eq 'Top') { + $heading_level = 0; + } else { + $heading_level = 3; + } + } + } else { + $heading_level = $command->{'level'}; + } + + my $heading = $self->command_text($command); + # $heading not defined may happen if the command is a @node, for example + # if there is an error in the node. + if (defined($heading) and $heading ne '' and defined($heading_level)) { + + if ($Texinfo::Common::root_commands{$cmdname} + and $Texinfo::Common::sectioning_commands{$cmdname}) { + my $content_href = $self->command_contents_href($command, 'contents', + $self->{'current_filename'}); + if ($content_href) { + my $this_href = $content_href =~ s/^\#toc-/\#/r; + $heading .= '<span class="pull-right">'. + '<a class="anchor hidden-xs" '. + "href=\"$this_href\" aria-hidden=\"true\">". + ($ENV{"FA_ICONS"} ? '<i class="fa fa-link"></i>' + : '#'). + '</a> '. + '<a class="anchor hidden-xs"'. + "href=\"$content_href\" aria-hidden=\"true\">". + ($ENV{"FA_ICONS"} ? '<i class="fa fa-navicon"></i>' + : 'TOC'). + '</a>'. + '</span>'; + } + } + + if ($self->in_preformatted()) { + $result .= $heading."\n"; + } else { + # if the level was changed, set the command name right + if ($cmdname ne 'node' + and $heading_level ne $Texinfo::Common::command_structuring_level{$cmdname}) { + $cmdname + = $Texinfo::Common::level_to_structuring_command{$cmdname}->[$heading_level]; + } + $result .= &{$self->{'format_heading_text'}}( + $self, $cmdname, $heading, + $heading_level + + $self->get_conf('CHAPTER_HEADER_LEVEL') - 1, $command); + } + } + $result .= $content if (defined($content)); + return $result; +} + +foreach my $command (keys(%Texinfo::Common::sectioning_commands), 'node') { + texinfo_register_command_formatting($command, \&ffmpeg_heading_command); +} + +# print the TOC where @contents is used +set_from_init_file('INLINE_CONTENTS', 1); + +# make chapters <h2> +set_from_init_file('CHAPTER_HEADER_LEVEL', 2); + +# Do not add <hr> +set_from_init_file('DEFAULT_RULE', ''); +set_from_init_file('BIG_RULE', ''); + +# Customized file beginning +sub ffmpeg_begin_file($$$) +{ + my $self = shift; + my $filename = shift; + my $element = shift; + + my $command; + if ($element and $self->get_conf('SPLIT')) { + $command = $self->element_command($element); + } + + my ($title, $description, $encoding, $date, $css_lines, + $doctype, $bodytext, $copying_comment, $after_body_open, + $extra_head, $program_and_version, $program_homepage, + $program, $generator) = $self->_file_header_informations($command); + + my $links = $self->_get_links ($filename, $element); + + my $head1 = $ENV{"FFMPEG_HEADER1"} || <<EOT; +<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN" "http://www.w3.org/TR/html4/loose.dtd"> +<html> +<!-- Created by $program_and_version, $program_homepage --> + <head> + <meta charset="utf-8"> + <title> +EOT + my $head_title = <<EOT; + $title +EOT + + my $head2 = $ENV{"FFMPEG_HEADER2"} || <<EOT; + </title> + <meta name="viewport" content="width=device-width,initial-scale=1.0"> + <link rel="stylesheet" type="text/css" href="bootstrap.min.css"> + <link rel="stylesheet" type="text/css" href="style.min.css"> + </head> + <body> + <div style="width: 95%; margin: auto"> + <h1> +EOT + + my $head3 = $ENV{"FFMPEG_HEADER3"} || <<EOT; + </h1> +EOT + + return $head1 . $head_title . $head2 . $head_title . $head3; +} +texinfo_register_formatting_function('begin_file', \&ffmpeg_begin_file); + +sub ffmpeg_program_string($) +{ + my $self = shift; + if (defined($self->get_conf('PROGRAM')) + and $self->get_conf('PROGRAM') ne '' + and defined($self->get_conf('PACKAGE_URL'))) { + return $self->convert_tree( + $self->gdt('This document was generated using @uref{{program_homepage}, @emph{{program}}}.', + { 'program_homepage' => $self->get_conf('PACKAGE_URL'), + 'program' => $self->get_conf('PROGRAM') })); + } else { + return $self->convert_tree( + $self->gdt('This document was generated automatically.')); + } +} +texinfo_register_formatting_function('program_string', \&ffmpeg_program_string); + +# Customized file ending +sub ffmpeg_end_file($) +{ + my $self = shift; + my $program_string = &{$self->{'format_program_string'}}($self); + my $program_text = <<EOT; + <p style="font-size: small;"> + $program_string + </p> +EOT + my $footer = $ENV{FFMPEG_FOOTER} || <<EOT; + </div> + </body> +</html> +EOT + return $program_text . $footer; +} +texinfo_register_formatting_function('end_file', \&ffmpeg_end_file); + +# Dummy title command +# Ignore title. Title is handled through ffmpeg_begin_file(). +set_from_init_file('USE_TITLEPAGE_FOR_TITLE', 1); +sub ffmpeg_title($$$$) +{ + return ''; +} + +texinfo_register_command_formatting('titlefont', + \&ffmpeg_title); + +# Customized float command. Part of code borrowed from GNU Texinfo. +sub ffmpeg_float($$$$$) +{ + my $self = shift; + my $cmdname = shift; + my $command = shift; + my $args = shift; + my $content = shift; + + my ($caption, $prepended) = Texinfo::Common::float_name_caption($self, + $command); + my $caption_text = ''; + my $prepended_text; + my $prepended_save = ''; + + if ($self->in_string()) { + if ($prepended) { + $prepended_text = $self->convert_tree_new_formatting_context( + $prepended, 'float prepended'); + } else { + $prepended_text = ''; + } + if ($caption) { + $caption_text = $self->convert_tree_new_formatting_context( + {'contents' => $caption->{'args'}->[0]->{'contents'}}, + 'float caption'); + } + return $prepended.$content.$caption_text; + } + + my $id = $self->command_id($command); + my $label; + if (defined($id) and $id ne '') { + $label = "<a name=\"$id\"></a>"; + } else { + $label = ''; + } + + if ($prepended) { + if ($caption) { + # prepend the prepended tree to the first paragraph + my @caption_original_contents = @{$caption->{'args'}->[0]->{'contents'}}; + my @caption_contents; + my $new_paragraph; + while (@caption_original_contents) { + my $content = shift @caption_original_contents; + if ($content->{'type'} and $content->{'type'} eq 'paragraph') { + %{$new_paragraph} = %{$content}; + $new_paragraph->{'contents'} = [@{$content->{'contents'}}]; + unshift (@{$new_paragraph->{'contents'}}, {'cmdname' => 'strong', + 'args' => [{'type' => 'brace_command_arg', + 'contents' => [$prepended]}]}); + push @caption_contents, $new_paragraph; + last; + } else { + push @caption_contents, $content; + } + } + push @caption_contents, @caption_original_contents; + if ($new_paragraph) { + $caption_text = $self->convert_tree_new_formatting_context( + {'contents' => \@caption_contents}, 'float caption'); + $prepended_text = ''; + } + } + if ($caption_text eq '') { + $prepended_text = $self->convert_tree_new_formatting_context( + $prepended, 'float prepended'); + if ($prepended_text ne '') { + $prepended_save = $prepended_text; + $prepended_text = '<p><strong>'.$prepended_text.'</strong></p>'; + } + } + } else { + $prepended_text = ''; + } + + if ($caption and $caption_text eq '') { + $caption_text = $self->convert_tree_new_formatting_context( + $caption->{'args'}->[0], 'float caption'); + } + if ($prepended_text.$caption_text ne '') { + $prepended_text = $self->_attribute_class('div','float-caption'). '>' + . $prepended_text; + $caption_text .= '</div>'; + } + my $html_class = ''; + if ($prepended_save =~ /NOTE/) { + $html_class = 'info'; + $prepended_text = ''; + $caption_text = ''; + } elsif ($prepended_save =~ /IMPORTANT/) { + $html_class = 'warning'; + $prepended_text = ''; + $caption_text = ''; + } + return $self->_attribute_class('div', $html_class). '>' . "\n" . + $prepended_text . $caption_text . $content . '</div>'; +} + +texinfo_register_command_formatting('float', + \&ffmpeg_float); + +1; diff --git a/doc/texi2pod.pl b/doc/texi2pod.pl index e4eb61c26c..9a9b34fc15 100755..100644 --- a/doc/texi2pod.pl +++ b/doc/texi2pod.pl @@ -27,9 +27,9 @@ use warnings; $output = 0; $skipping = 0; -%sects = (); -@sects_sequence = (); -$section = ""; +%chapters = (); +@chapters_sequence = (); +$chapter = ""; @icstack = (); @endwstack = (); @skstack = (); @@ -116,18 +116,24 @@ INF: while(<$inf>) { die "cannot open $1: $!\n"; }; - # Look for blocks surrounded by @c man begin SECTION ... @c man end. - # This really oughta be @ifman ... @end ifman and the like, but such - # would require rev'ing all other Texinfo translators. - /^\@c\s+man\s+begin\s+([A-Za-z ]+)/ and $sect = $1, push (@sects_sequence, $sect), $output = 1, next; - /^\@c\s+man\s+end/ and do { - $sects{$sect} = "" unless exists $sects{$sect}; - $sects{$sect} .= postprocess($section); - $section = ""; - $output = 0; + /^\@chapter\s+([A-Za-z ]+)/ and do { + # close old chapter + $chapters{$chapter_name} .= postprocess($chapter) if ($chapter_name); + + # start new chapter + $chapter_name = $1, push (@chapters_sequence, $chapter_name) unless $skipping; + $chapters{$chapter_name} = "" unless exists $chapters{$chapter_name}; + $chapter = ""; + $output = 1; next; }; + /^\@bye/ and do { + # close old chapter + $chapters{$chapter_name} .= postprocess($chapter) if ($chapter_name); + last INF; + }; + # handle variables /^\@set\s+([a-zA-Z0-9_-]+)\s*(.*)$/ and do { $defs{$1} = $2; @@ -150,17 +156,17 @@ INF: while(<$inf>) { # Ignore @end foo, where foo is not an operation which may # cause us to skip, if we are presently skipping. my $ended = $1; - next if $skipping && $ended !~ /^(?:ifset|ifclear|ignore|menu|iftex)$/; + next if $skipping && $ended !~ /^(?:ifset|ifclear|ignore|menu|iftex|ifhtml|ifnothtml)$/; die "\@end $ended without \@$ended at line $.\n" unless defined $endw; die "\@$endw ended by \@end $ended at line $.\n" unless $ended eq $endw; $endw = pop @endwstack; - if ($ended =~ /^(?:ifset|ifclear|ignore|menu|iftex)$/) { + if ($ended =~ /^(?:ifset|ifclear|ignore|menu|iftex|ifhtml|ifnothtml)$/) { $skipping = pop @skstack; next; - } elsif ($ended =~ /^(?:example|smallexample|display)$/) { + } elsif ($ended =~ /^(?:example|smallexample|verbatim|display)$/) { $shift = ""; $_ = ""; # need a paragraph break } elsif ($ended =~ /^(?:itemize|enumerate|(?:multi|[fv])?table)$/) { @@ -190,11 +196,11 @@ INF: while(<$inf>) { next; }; - /^\@(ignore|menu|iftex)\b/ and do { + /^\@(ignore|menu|iftex|ifhtml|ifnothtml)\b/ and do { push @endwstack, $endw; push @skstack, $skipping; $endw = $1; - $skipping = 1; + $skipping = $endw !~ /ifnothtml/; next; }; @@ -211,7 +217,6 @@ INF: while(<$inf>) { s/\@TeX\{\}/TeX/g; s/\@pounds\{\}/\#/g; s/\@minus(?:\{\})?/-/g; - s/\\,/,/g; # Now the ones that have to be replaced by special escapes # (which will be turned back into text by unmunge()) @@ -269,7 +274,7 @@ INF: while(<$inf>) { push @icstack, $ic; $endw = $1; $ic = $2; - $ic =~ s/\@(?:samp|strong|key|gcctabopt|option|env)/B/; + $ic =~ s/\@(?:samp|strong|key|gcctabopt|option|env|command)/B/; $ic =~ s/\@(?:code|kbd)/C/; $ic =~ s/\@(?:dfn|var|emph|cite|i)/I/; $ic =~ s/\@(?:file)/F/; @@ -285,7 +290,7 @@ INF: while(<$inf>) { $_ = "\n=over 4\n"; }; - /^\@((?:small)?example|display)/ and do { + /^\@((?:small)?example|verbatim|display)/ and do { push @endwstack, $endw; $endw = $1; $shift = "\t"; @@ -304,7 +309,7 @@ INF: while(<$inf>) { $columns =~ s/\@tab//; $_ = $columns; - $section =~ s/$//; + $chapter =~ s/$//; }; /^\@itemx?\s*(.+)?$/ and do { @@ -318,7 +323,7 @@ INF: while(<$inf>) { } }; - $section .= $shift.$_."\n"; + $chapter .= $shift.$_."\n"; } # End of current file. close($inf); @@ -330,16 +335,15 @@ die "No filename or title\n" unless defined $fn && defined $tl; # always use utf8 print "=encoding utf8\n\n"; -$sects{NAME} = "$fn \- $tl\n"; -$sects{FOOTNOTES} .= "=back\n" if exists $sects{FOOTNOTES}; +$chapters{NAME} = "$fn \- $tl\n"; +$chapters{FOOTNOTES} .= "=back\n" if exists $chapters{FOOTNOTES}; -unshift @sects_sequence, "NAME"; -for $sect (@sects_sequence) { - if(exists $sects{$sect}) { - $head = $sect; - $head =~ s/SEEALSO/SEE ALSO/; +unshift @chapters_sequence, "NAME"; +for $chapter (@chapters_sequence) { + if (exists $chapters{$chapter}) { + $head = uc($chapter); print "=head1 $head\n\n"; - print scalar unmunge ($sects{$sect}); + print scalar unmunge ($chapters{$chapter}); print "\n"; } } @@ -380,11 +384,12 @@ sub postprocess # @* is also impossible in .pod; we discard it and any newline that # follows it. Similarly, our macro @gol must be discarded. - s/\@anchor{(?:[^\}]*)\}//g; + s/\@anchor\{(?:[^\}]*)\}//g; s/\(?\@xref\{(?:[^\}]*)\}(?:[^.<]|(?:<[^<>]*>))*\.\)?//g; s/\s+\(\@pxref\{(?:[^\}]*)\}\)//g; s/;\s+\@pxref\{(?:[^\}]*)\}//g; - s/\@ref\{([^\}]*)\}/$1/g; + s/\@ref\{(?:[^,\}]*,)(?:[^,\}]*,)([^,\}]*).*\}/B<$1>/g; + s/\@ref\{([^\}]*)\}/B<$1>/g; s/\@noindent\s*//g; s/\@refill//g; s/\@gol//g; @@ -393,7 +398,7 @@ sub postprocess # @uref can take one, two, or three arguments, with different # semantics each time. @url and @email are just like @uref with # one argument, for our purposes. - s/\@(?:uref|url|email)\{([^\},]*)\}/<B<$1>>/g; + s/\@(?:uref|url|email)\{([^\},]*),?[^\}]*\}/<B<$1>>/g; s/\@uref\{([^\},]*),([^\},]*)\}/$2 (C<$1>)/g; s/\@uref\{([^\},]*),([^\},]*),([^\},]*)\}/$3/g; @@ -437,13 +442,13 @@ sub unmunge sub add_footnote { - unless (exists $sects{FOOTNOTES}) { - $sects{FOOTNOTES} = "\n=over 4\n\n"; + unless (exists $chapters{FOOTNOTES}) { + $chapters{FOOTNOTES} = "\n=over 4\n\n"; } - $sects{FOOTNOTES} .= "=item $fnno.\n\n"; $fnno++; - $sects{FOOTNOTES} .= $_[0]; - $sects{FOOTNOTES} .= "\n\n"; + $chapters{FOOTNOTES} .= "=item $fnno.\n\n"; $fnno++; + $chapters{FOOTNOTES} .= $_[0]; + $chapters{FOOTNOTES} .= "\n\n"; } # stolen from Symbol.pm diff --git a/doc/texidep.pl b/doc/texidep.pl new file mode 100644 index 0000000000..099690378e --- /dev/null +++ b/doc/texidep.pl @@ -0,0 +1,32 @@ +#! /usr/bin/env perl + +# This script will print the dependency of a Texinfo file to stdout. +# texidep.pl <src-path> <input.texi> <output.ext> + +use warnings; +use strict; + +die unless @ARGV == 3; + +my ($src_path, $root, $target) = @ARGV; + +sub print_deps { + my ($file, $deps) = @_; + $deps->{$file} = 1; + + open(my $fh, "<", "$file") or die "Cannot open file '$file': $!"; + while (<$fh>) { + if (my ($i) = /^\@(?:verbatim)?include\s+(\S+)/) { + die "Circular dependency found in file $root\n" if exists $deps->{"doc/$1"}; + print "$target: doc/$1\n"; + + # skip looking for config.texi dependencies, since it has + # none, and is not located in the source tree + if ("$1" ne "config.texi") { + print_deps("$src_path/doc/$1", {%$deps}); + } + } + } +} + +print_deps($root, {}); diff --git a/doc/utils.texi b/doc/utils.texi new file mode 100644 index 0000000000..756c609072 --- /dev/null +++ b/doc/utils.texi @@ -0,0 +1,1075 @@ +@chapter Syntax +@c man begin SYNTAX + +This section documents the syntax and formats employed by the FFmpeg +libraries and tools. + +@anchor{quoting_and_escaping} +@section Quoting and escaping + +FFmpeg adopts the following quoting and escaping mechanism, unless +explicitly specified. The following rules are applied: + +@itemize +@item +@samp{'} and @samp{\} are special characters (respectively used for +quoting and escaping). In addition to them, there might be other +special characters depending on the specific syntax where the escaping +and quoting are employed. + +@item +A special character is escaped by prefixing it with a @samp{\}. + +@item +All characters enclosed between @samp{''} are included literally in the +parsed string. The quote character @samp{'} itself cannot be quoted, +so you may need to close the quote and escape it. + +@item +Leading and trailing whitespaces, unless escaped or quoted, are +removed from the parsed string. +@end itemize + +Note that you may need to add a second level of escaping when using +the command line or a script, which depends on the syntax of the +adopted shell language. + +The function @code{av_get_token} defined in +@file{libavutil/avstring.h} can be used to parse a token quoted or +escaped according to the rules defined above. + +The tool @file{tools/ffescape} in the FFmpeg source tree can be used +to automatically quote or escape a string in a script. + +@subsection Examples + +@itemize +@item +Escape the string @code{Crime d'Amour} containing the @code{'} special +character: +@example +Crime d\'Amour +@end example + +@item +The string above contains a quote, so the @code{'} needs to be escaped +when quoting it: +@example +'Crime d'\''Amour' +@end example + +@item +Include leading or trailing whitespaces using quoting: +@example +' this string starts and ends with whitespaces ' +@end example + +@item +Escaping and quoting can be mixed together: +@example +' The string '\'string\'' is a string ' +@end example + +@item +To include a literal @samp{\} you can use either escaping or quoting: +@example +'c:\foo' can be written as c:\\foo +@end example +@end itemize + +@anchor{date syntax} +@section Date + +The accepted syntax is: +@example +[(YYYY-MM-DD|YYYYMMDD)[T|t| ]]((HH:MM:SS[.m...]]])|(HHMMSS[.m...]]]))[Z] +now +@end example + +If the value is "now" it takes the current time. + +Time is local time unless Z is appended, in which case it is +interpreted as UTC. +If the year-month-day part is not specified it takes the current +year-month-day. + +@anchor{time duration syntax} +@section Time duration + +There are two accepted syntaxes for expressing time duration. + +@example +[-][@var{HH}:]@var{MM}:@var{SS}[.@var{m}...] +@end example + +@var{HH} expresses the number of hours, @var{MM} the number of minutes +for a maximum of 2 digits, and @var{SS} the number of seconds for a +maximum of 2 digits. The @var{m} at the end expresses decimal value for +@var{SS}. + +@emph{or} + +@example +[-]@var{S}+[.@var{m}...] +@end example + +@var{S} expresses the number of seconds, with the optional decimal part +@var{m}. + +In both expressions, the optional @samp{-} indicates negative duration. + +@subsection Examples + +The following examples are all valid time duration: + +@table @samp +@item 55 +55 seconds + +@item 12:03:45 +12 hours, 03 minutes and 45 seconds + +@item 23.189 +23.189 seconds +@end table + +@anchor{video size syntax} +@section Video size +Specify the size of the sourced video, it may be a string of the form +@var{width}x@var{height}, or the name of a size abbreviation. + +The following abbreviations are recognized: +@table @samp +@item ntsc +720x480 +@item pal +720x576 +@item qntsc +352x240 +@item qpal +352x288 +@item sntsc +640x480 +@item spal +768x576 +@item film +352x240 +@item ntsc-film +352x240 +@item sqcif +128x96 +@item qcif +176x144 +@item cif +352x288 +@item 4cif +704x576 +@item 16cif +1408x1152 +@item qqvga +160x120 +@item qvga +320x240 +@item vga +640x480 +@item svga +800x600 +@item xga +1024x768 +@item uxga +1600x1200 +@item qxga +2048x1536 +@item sxga +1280x1024 +@item qsxga +2560x2048 +@item hsxga +5120x4096 +@item wvga +852x480 +@item wxga +1366x768 +@item wsxga +1600x1024 +@item wuxga +1920x1200 +@item woxga +2560x1600 +@item wqsxga +3200x2048 +@item wquxga +3840x2400 +@item whsxga +6400x4096 +@item whuxga +7680x4800 +@item cga +320x200 +@item ega +640x350 +@item hd480 +852x480 +@item hd720 +1280x720 +@item hd1080 +1920x1080 +@item 2k +2048x1080 +@item 2kflat +1998x1080 +@item 2kscope +2048x858 +@item 4k +4096x2160 +@item 4kflat +3996x2160 +@item 4kscope +4096x1716 +@item nhd +640x360 +@item hqvga +240x160 +@item wqvga +400x240 +@item fwqvga +432x240 +@item hvga +480x320 +@item qhd +960x540 +@item 2kdci +2048x1080 +@item 4kdci +4096x2160 +@item uhd2160 +3840x2160 +@item uhd4320 +7680x4320 +@end table + +@anchor{video rate syntax} +@section Video rate + +Specify the frame rate of a video, expressed as the number of frames +generated per second. It has to be a string in the format +@var{frame_rate_num}/@var{frame_rate_den}, an integer number, a float +number or a valid video frame rate abbreviation. + +The following abbreviations are recognized: +@table @samp +@item ntsc +30000/1001 +@item pal +25/1 +@item qntsc +30000/1001 +@item qpal +25/1 +@item sntsc +30000/1001 +@item spal +25/1 +@item film +24/1 +@item ntsc-film +24000/1001 +@end table + +@anchor{ratio syntax} +@section Ratio + +A ratio can be expressed as an expression, or in the form +@var{numerator}:@var{denominator}. + +Note that a ratio with infinite (1/0) or negative value is +considered valid, so you should check on the returned value if you +want to exclude those values. + +The undefined value can be expressed using the "0:0" string. + +@anchor{color syntax} +@section Color + +It can be the name of a color as defined below (case insensitive match) or a +@code{[0x|#]RRGGBB[AA]} sequence, possibly followed by @@ and a string +representing the alpha component. + +The alpha component may be a string composed by "0x" followed by an +hexadecimal number or a decimal number between 0.0 and 1.0, which +represents the opacity value (@samp{0x00} or @samp{0.0} means completely +transparent, @samp{0xff} or @samp{1.0} completely opaque). If the alpha +component is not specified then @samp{0xff} is assumed. + +The string @samp{random} will result in a random color. + +The following names of colors are recognized: +@table @samp +@item AliceBlue +0xF0F8FF +@item AntiqueWhite +0xFAEBD7 +@item Aqua +0x00FFFF +@item Aquamarine +0x7FFFD4 +@item Azure +0xF0FFFF +@item Beige +0xF5F5DC +@item Bisque +0xFFE4C4 +@item Black +0x000000 +@item BlanchedAlmond +0xFFEBCD +@item Blue +0x0000FF +@item BlueViolet +0x8A2BE2 +@item Brown +0xA52A2A +@item BurlyWood +0xDEB887 +@item CadetBlue +0x5F9EA0 +@item Chartreuse +0x7FFF00 +@item Chocolate +0xD2691E +@item Coral +0xFF7F50 +@item CornflowerBlue +0x6495ED +@item Cornsilk +0xFFF8DC +@item Crimson +0xDC143C +@item Cyan +0x00FFFF +@item DarkBlue +0x00008B +@item DarkCyan +0x008B8B +@item DarkGoldenRod +0xB8860B +@item DarkGray +0xA9A9A9 +@item DarkGreen +0x006400 +@item DarkKhaki +0xBDB76B +@item DarkMagenta +0x8B008B +@item DarkOliveGreen +0x556B2F +@item Darkorange +0xFF8C00 +@item DarkOrchid +0x9932CC +@item DarkRed +0x8B0000 +@item DarkSalmon +0xE9967A +@item DarkSeaGreen +0x8FBC8F +@item DarkSlateBlue +0x483D8B +@item DarkSlateGray +0x2F4F4F +@item DarkTurquoise +0x00CED1 +@item DarkViolet +0x9400D3 +@item DeepPink +0xFF1493 +@item DeepSkyBlue +0x00BFFF +@item DimGray +0x696969 +@item DodgerBlue +0x1E90FF +@item FireBrick +0xB22222 +@item FloralWhite +0xFFFAF0 +@item ForestGreen +0x228B22 +@item Fuchsia +0xFF00FF +@item Gainsboro +0xDCDCDC +@item GhostWhite +0xF8F8FF +@item Gold +0xFFD700 +@item GoldenRod +0xDAA520 +@item Gray +0x808080 +@item Green +0x008000 +@item GreenYellow +0xADFF2F +@item HoneyDew +0xF0FFF0 +@item HotPink +0xFF69B4 +@item IndianRed +0xCD5C5C +@item Indigo +0x4B0082 +@item Ivory +0xFFFFF0 +@item Khaki +0xF0E68C +@item Lavender +0xE6E6FA +@item LavenderBlush +0xFFF0F5 +@item LawnGreen +0x7CFC00 +@item LemonChiffon +0xFFFACD +@item LightBlue +0xADD8E6 +@item LightCoral +0xF08080 +@item LightCyan +0xE0FFFF +@item LightGoldenRodYellow +0xFAFAD2 +@item LightGreen +0x90EE90 +@item LightGrey +0xD3D3D3 +@item LightPink +0xFFB6C1 +@item LightSalmon +0xFFA07A +@item LightSeaGreen +0x20B2AA +@item LightSkyBlue +0x87CEFA +@item LightSlateGray +0x778899 +@item LightSteelBlue +0xB0C4DE +@item LightYellow +0xFFFFE0 +@item Lime +0x00FF00 +@item LimeGreen +0x32CD32 +@item Linen +0xFAF0E6 +@item Magenta +0xFF00FF +@item Maroon +0x800000 +@item MediumAquaMarine +0x66CDAA +@item MediumBlue +0x0000CD +@item MediumOrchid +0xBA55D3 +@item MediumPurple +0x9370D8 +@item MediumSeaGreen +0x3CB371 +@item MediumSlateBlue +0x7B68EE +@item MediumSpringGreen +0x00FA9A +@item MediumTurquoise +0x48D1CC +@item MediumVioletRed +0xC71585 +@item MidnightBlue +0x191970 +@item MintCream +0xF5FFFA +@item MistyRose +0xFFE4E1 +@item Moccasin +0xFFE4B5 +@item NavajoWhite +0xFFDEAD +@item Navy +0x000080 +@item OldLace +0xFDF5E6 +@item Olive +0x808000 +@item OliveDrab +0x6B8E23 +@item Orange +0xFFA500 +@item OrangeRed +0xFF4500 +@item Orchid +0xDA70D6 +@item PaleGoldenRod +0xEEE8AA +@item PaleGreen +0x98FB98 +@item PaleTurquoise +0xAFEEEE +@item PaleVioletRed +0xD87093 +@item PapayaWhip +0xFFEFD5 +@item PeachPuff +0xFFDAB9 +@item Peru +0xCD853F +@item Pink +0xFFC0CB +@item Plum +0xDDA0DD +@item PowderBlue +0xB0E0E6 +@item Purple +0x800080 +@item Red +0xFF0000 +@item RosyBrown +0xBC8F8F +@item RoyalBlue +0x4169E1 +@item SaddleBrown +0x8B4513 +@item Salmon +0xFA8072 +@item SandyBrown +0xF4A460 +@item SeaGreen +0x2E8B57 +@item SeaShell +0xFFF5EE +@item Sienna +0xA0522D +@item Silver +0xC0C0C0 +@item SkyBlue +0x87CEEB +@item SlateBlue +0x6A5ACD +@item SlateGray +0x708090 +@item Snow +0xFFFAFA +@item SpringGreen +0x00FF7F +@item SteelBlue +0x4682B4 +@item Tan +0xD2B48C +@item Teal +0x008080 +@item Thistle +0xD8BFD8 +@item Tomato +0xFF6347 +@item Turquoise +0x40E0D0 +@item Violet +0xEE82EE +@item Wheat +0xF5DEB3 +@item White +0xFFFFFF +@item WhiteSmoke +0xF5F5F5 +@item Yellow +0xFFFF00 +@item YellowGreen +0x9ACD32 +@end table + +@anchor{channel layout syntax} +@section Channel Layout + +A channel layout specifies the spatial disposition of the channels in +a multi-channel audio stream. To specify a channel layout, FFmpeg +makes use of a special syntax. + +Individual channels are identified by an id, as given by the table +below: +@table @samp +@item FL +front left +@item FR +front right +@item FC +front center +@item LFE +low frequency +@item BL +back left +@item BR +back right +@item FLC +front left-of-center +@item FRC +front right-of-center +@item BC +back center +@item SL +side left +@item SR +side right +@item TC +top center +@item TFL +top front left +@item TFC +top front center +@item TFR +top front right +@item TBL +top back left +@item TBC +top back center +@item TBR +top back right +@item DL +downmix left +@item DR +downmix right +@item WL +wide left +@item WR +wide right +@item SDL +surround direct left +@item SDR +surround direct right +@item LFE2 +low frequency 2 +@end table + +Standard channel layout compositions can be specified by using the +following identifiers: +@table @samp +@item mono +FC +@item stereo +FL+FR +@item 2.1 +FL+FR+LFE +@item 3.0 +FL+FR+FC +@item 3.0(back) +FL+FR+BC +@item 4.0 +FL+FR+FC+BC +@item quad +FL+FR+BL+BR +@item quad(side) +FL+FR+SL+SR +@item 3.1 +FL+FR+FC+LFE +@item 5.0 +FL+FR+FC+BL+BR +@item 5.0(side) +FL+FR+FC+SL+SR +@item 4.1 +FL+FR+FC+LFE+BC +@item 5.1 +FL+FR+FC+LFE+BL+BR +@item 5.1(side) +FL+FR+FC+LFE+SL+SR +@item 6.0 +FL+FR+FC+BC+SL+SR +@item 6.0(front) +FL+FR+FLC+FRC+SL+SR +@item hexagonal +FL+FR+FC+BL+BR+BC +@item 6.1 +FL+FR+FC+LFE+BC+SL+SR +@item 6.1 +FL+FR+FC+LFE+BL+BR+BC +@item 6.1(front) +FL+FR+LFE+FLC+FRC+SL+SR +@item 7.0 +FL+FR+FC+BL+BR+SL+SR +@item 7.0(front) +FL+FR+FC+FLC+FRC+SL+SR +@item 7.1 +FL+FR+FC+LFE+BL+BR+SL+SR +@item 7.1(wide) +FL+FR+FC+LFE+BL+BR+FLC+FRC +@item 7.1(wide-side) +FL+FR+FC+LFE+FLC+FRC+SL+SR +@item octagonal +FL+FR+FC+BL+BR+BC+SL+SR +@item downmix +DL+DR +@end table + +A custom channel layout can be specified as a sequence of terms, separated by +'+' or '|'. Each term can be: +@itemize +@item +the name of a standard channel layout (e.g. @samp{mono}, +@samp{stereo}, @samp{4.0}, @samp{quad}, @samp{5.0}, etc.) + +@item +the name of a single channel (e.g. @samp{FL}, @samp{FR}, @samp{FC}, @samp{LFE}, etc.) + +@item +a number of channels, in decimal, optionally followed by 'c', yielding +the default channel layout for that number of channels (see the +function @code{av_get_default_channel_layout}) + +@item +a channel layout mask, in hexadecimal starting with "0x" (see the +@code{AV_CH_*} macros in @file{libavutil/channel_layout.h}. +@end itemize + +Starting from libavutil version 53 the trailing character "c" to +specify a number of channels will be required, while a channel layout +mask could also be specified as a decimal number (if and only if not +followed by "c"). + +See also the function @code{av_get_channel_layout} defined in +@file{libavutil/channel_layout.h}. +@c man end SYNTAX + +@chapter Expression Evaluation +@c man begin EXPRESSION EVALUATION + +When evaluating an arithmetic expression, FFmpeg uses an internal +formula evaluator, implemented through the @file{libavutil/eval.h} +interface. + +An expression may contain unary, binary operators, constants, and +functions. + +Two expressions @var{expr1} and @var{expr2} can be combined to form +another expression "@var{expr1};@var{expr2}". +@var{expr1} and @var{expr2} are evaluated in turn, and the new +expression evaluates to the value of @var{expr2}. + +The following binary operators are available: @code{+}, @code{-}, +@code{*}, @code{/}, @code{^}. + +The following unary operators are available: @code{+}, @code{-}. + +The following functions are available: +@table @option +@item abs(x) +Compute absolute value of @var{x}. + +@item acos(x) +Compute arccosine of @var{x}. + +@item asin(x) +Compute arcsine of @var{x}. + +@item atan(x) +Compute arctangent of @var{x}. + +@item between(x, min, max) +Return 1 if @var{x} is greater than or equal to @var{min} and lesser than or +equal to @var{max}, 0 otherwise. + +@item bitand(x, y) +@item bitor(x, y) +Compute bitwise and/or operation on @var{x} and @var{y}. + +The results of the evaluation of @var{x} and @var{y} are converted to +integers before executing the bitwise operation. + +Note that both the conversion to integer and the conversion back to +floating point can lose precision. Beware of unexpected results for +large numbers (usually 2^53 and larger). + +@item ceil(expr) +Round the value of expression @var{expr} upwards to the nearest +integer. For example, "ceil(1.5)" is "2.0". + +@item clip(x, min, max) +Return the value of @var{x} clipped between @var{min} and @var{max}. + +@item cos(x) +Compute cosine of @var{x}. + +@item cosh(x) +Compute hyperbolic cosine of @var{x}. + +@item eq(x, y) +Return 1 if @var{x} and @var{y} are equivalent, 0 otherwise. + +@item exp(x) +Compute exponential of @var{x} (with base @code{e}, the Euler's number). + +@item floor(expr) +Round the value of expression @var{expr} downwards to the nearest +integer. For example, "floor(-1.5)" is "-2.0". + +@item gauss(x) +Compute Gauss function of @var{x}, corresponding to +@code{exp(-x*x/2) / sqrt(2*PI)}. + +@item gcd(x, y) +Return the greatest common divisor of @var{x} and @var{y}. If both @var{x} and +@var{y} are 0 or either or both are less than zero then behavior is undefined. + +@item gt(x, y) +Return 1 if @var{x} is greater than @var{y}, 0 otherwise. + +@item gte(x, y) +Return 1 if @var{x} is greater than or equal to @var{y}, 0 otherwise. + +@item hypot(x, y) +This function is similar to the C function with the same name; it returns +"sqrt(@var{x}*@var{x} + @var{y}*@var{y})", the length of the hypotenuse of a +right triangle with sides of length @var{x} and @var{y}, or the distance of the +point (@var{x}, @var{y}) from the origin. + +@item if(x, y) +Evaluate @var{x}, and if the result is non-zero return the result of +the evaluation of @var{y}, return 0 otherwise. + +@item if(x, y, z) +Evaluate @var{x}, and if the result is non-zero return the evaluation +result of @var{y}, otherwise the evaluation result of @var{z}. + +@item ifnot(x, y) +Evaluate @var{x}, and if the result is zero return the result of the +evaluation of @var{y}, return 0 otherwise. + +@item ifnot(x, y, z) +Evaluate @var{x}, and if the result is zero return the evaluation +result of @var{y}, otherwise the evaluation result of @var{z}. + +@item isinf(x) +Return 1.0 if @var{x} is +/-INFINITY, 0.0 otherwise. + +@item isnan(x) +Return 1.0 if @var{x} is NAN, 0.0 otherwise. + +@item ld(var) +Load the value of the internal variable with number +@var{var}, which was previously stored with st(@var{var}, @var{expr}). +The function returns the loaded value. + +@item log(x) +Compute natural logarithm of @var{x}. + +@item lt(x, y) +Return 1 if @var{x} is lesser than @var{y}, 0 otherwise. + +@item lte(x, y) +Return 1 if @var{x} is lesser than or equal to @var{y}, 0 otherwise. + +@item max(x, y) +Return the maximum between @var{x} and @var{y}. + +@item min(x, y) +Return the maximum between @var{x} and @var{y}. + +@item mod(x, y) +Compute the remainder of division of @var{x} by @var{y}. + +@item not(expr) +Return 1.0 if @var{expr} is zero, 0.0 otherwise. + +@item pow(x, y) +Compute the power of @var{x} elevated @var{y}, it is equivalent to +"(@var{x})^(@var{y})". + +@item print(t) +@item print(t, l) +Print the value of expression @var{t} with loglevel @var{l}. If +@var{l} is not specified then a default log level is used. +Returns the value of the expression printed. + +Prints t with loglevel l + +@item random(x) +Return a pseudo random value between 0.0 and 1.0. @var{x} is the index of the +internal variable which will be used to save the seed/state. + +@item root(expr, max) +Find an input value for which the function represented by @var{expr} +with argument @var{ld(0)} is 0 in the interval 0..@var{max}. + +The expression in @var{expr} must denote a continuous function or the +result is undefined. + +@var{ld(0)} is used to represent the function input value, which means +that the given expression will be evaluated multiple times with +various input values that the expression can access through +@code{ld(0)}. When the expression evaluates to 0 then the +corresponding input value will be returned. + +@item sin(x) +Compute sine of @var{x}. + +@item sinh(x) +Compute hyperbolic sine of @var{x}. + +@item sqrt(expr) +Compute the square root of @var{expr}. This is equivalent to +"(@var{expr})^.5". + +@item squish(x) +Compute expression @code{1/(1 + exp(4*x))}. + +@item st(var, expr) +Store the value of the expression @var{expr} in an internal +variable. @var{var} specifies the number of the variable where to +store the value, and it is a value ranging from 0 to 9. The function +returns the value stored in the internal variable. +Note, Variables are currently not shared between expressions. + +@item tan(x) +Compute tangent of @var{x}. + +@item tanh(x) +Compute hyperbolic tangent of @var{x}. + +@item taylor(expr, x) +@item taylor(expr, x, id) +Evaluate a Taylor series at @var{x}, given an expression representing +the @code{ld(id)}-th derivative of a function at 0. + +When the series does not converge the result is undefined. + +@var{ld(id)} is used to represent the derivative order in @var{expr}, +which means that the given expression will be evaluated multiple times +with various input values that the expression can access through +@code{ld(id)}. If @var{id} is not specified then 0 is assumed. + +Note, when you have the derivatives at y instead of 0, +@code{taylor(expr, x-y)} can be used. + +@item time(0) +Return the current (wallclock) time in seconds. + +@item trunc(expr) +Round the value of expression @var{expr} towards zero to the nearest +integer. For example, "trunc(-1.5)" is "-1.0". + +@item while(cond, expr) +Evaluate expression @var{expr} while the expression @var{cond} is +non-zero, and returns the value of the last @var{expr} evaluation, or +NAN if @var{cond} was always false. +@end table + +The following constants are available: +@table @option +@item PI +area of the unit disc, approximately 3.14 +@item E +exp(1) (Euler's number), approximately 2.718 +@item PHI +golden ratio (1+sqrt(5))/2, approximately 1.618 +@end table + +Assuming that an expression is considered "true" if it has a non-zero +value, note that: + +@code{*} works like AND + +@code{+} works like OR + +For example the construct: +@example +if (A AND B) then C +@end example +is equivalent to: +@example +if(A*B, C) +@end example + +In your C code, you can extend the list of unary and binary functions, +and define recognized constants, so that they are available for your +expressions. + +The evaluator also recognizes the International System unit prefixes. +If 'i' is appended after the prefix, binary prefixes are used, which +are based on powers of 1024 instead of powers of 1000. +The 'B' postfix multiplies the value by 8, and can be appended after a +unit prefix or used alone. This allows using for example 'KB', 'MiB', +'G' and 'B' as number postfix. + +The list of available International System prefixes follows, with +indication of the corresponding powers of 10 and of 2. +@table @option +@item y +10^-24 / 2^-80 +@item z +10^-21 / 2^-70 +@item a +10^-18 / 2^-60 +@item f +10^-15 / 2^-50 +@item p +10^-12 / 2^-40 +@item n +10^-9 / 2^-30 +@item u +10^-6 / 2^-20 +@item m +10^-3 / 2^-10 +@item c +10^-2 +@item d +10^-1 +@item h +10^2 +@item k +10^3 / 2^10 +@item K +10^3 / 2^10 +@item M +10^6 / 2^20 +@item G +10^9 / 2^30 +@item T +10^12 / 2^40 +@item P +10^15 / 2^40 +@item E +10^18 / 2^50 +@item Z +10^21 / 2^60 +@item Y +10^24 / 2^70 +@end table + +@c man end EXPRESSION EVALUATION + +@chapter OpenCL Options +@c man begin OPENCL OPTIONS + +When FFmpeg is configured with @code{--enable-opencl}, it is possible +to set the options for the global OpenCL context. + +The list of supported options follows: + +@table @option +@item build_options +Set build options used to compile the registered kernels. + +See reference "OpenCL Specification Version: 1.2 chapter 5.6.4". + +@item platform_idx +Select the index of the platform to run OpenCL code. + +The specified index must be one of the indexes in the device list +which can be obtained with @code{ffmpeg -opencl_bench} or @code{av_opencl_get_device_list()}. + +@item device_idx +Select the index of the device used to run OpenCL code. + +The specified index must be one of the indexes in the device list which +can be obtained with @code{ffmpeg -opencl_bench} or @code{av_opencl_get_device_list()}. + +@end table + +@c man end OPENCL OPTIONS diff --git a/doc/viterbi.txt b/doc/viterbi.txt deleted file mode 100644 index 97825462cc..0000000000 --- a/doc/viterbi.txt +++ /dev/null @@ -1,109 +0,0 @@ -This is a quick description of the viterbi aka dynamic programing -algorthm. - -Its reason for existence is that wikipedia has become very poor on -describing algorithms in a way that makes it useable for understanding -them or anything else actually. It tends now to describe the very same -algorithm under 50 different names and pages with few understandable -by even people who fully understand the algorithm and the theory behind. - -Problem description: (that is what it can solve) -assume we have a 2d table, or you could call it a graph or matrix if you -prefer - - O O O O O O O - - O O O O O O O - - O O O O O O O - - O O O O O O O - - -That table has edges connecting points from each column to the next column -and each edge has a score like: (only some edge and scores shown to keep it -readable) - - - O--5--O-----O-----O-----O-----O - 2 / 7 / \ / \ / \ / - \ / \ / \ / \ / \ / - O7-/--O--/--O--/--O--/--O--/--O - \/ \/ 1/ \/ \/ \/ \/ \/ \/ \/ - /\ /\ 2\ /\ /\ /\ /\ /\ /\ /\ - O3-/--O--/--O--/--O--/--O--/--O - / \ / \ / \ / \ / \ - 1 \ 9 \ / \ / \ / \ - O--2--O--1--O--5--O--3--O--8--O - - - -Our goal is to find a path from left to right through it which -minimizes the sum of the score of all edges. -(and of course left/right is just a convention here it could be top down too) -Similarly the minimum could be the maximum by just fliping the sign, -Example of a path with scores: - - O O O O O O O - ->---O. O O .O-2-O O O - 5. .7 . - O O-1-O O O 8 O O - . - O O O O O O-1-O---> (sum here is 24) - - -The viterbi algorthm now solves this simply column by column -For the previous column each point has a best path and a associated -score: - - O-----5 O - \ - \ - O \ 1 O - \/ - /\ - O / 2 O - / - / - O-----2 O - - -To move one column forward we just need to find the best path and associated -scores for the next column -here are some edges we could choose from: - - - O-----5--3--O - \ \8 - \ \ - O \ 1--9--O - \/ \3 - /\ \ - O / 2--1--O - / \2 - / \ - O-----2--4--O - -Finding the new best paths and scores for each point of our new column is -trivial given we know the previous column best paths and scores: - - O-----0-----8 - \ - \ - O \ 0----10 - \/ - /\ - O / 0-----3 - / \ - / \ - O 0 4 - - -the viterbi algorthm continues exactly like this column for column until the -end and then just picks the path with the best score (above that would be the -one with score 3) - - -Author: Michael niedermayer -Copyright LGPL diff --git a/doc/writing_filters.txt b/doc/writing_filters.txt new file mode 100644 index 0000000000..66ebb53243 --- /dev/null +++ b/doc/writing_filters.txt @@ -0,0 +1,423 @@ +This document is a tutorial/initiation for writing simple filters in +libavfilter. + +Foreword: just like everything else in FFmpeg, libavfilter is monolithic, which +means that it is highly recommended that you submit your filters to the FFmpeg +development mailing-list and make sure that they are applied. Otherwise, your filters +are likely to have a very short lifetime due to more or less regular internal API +changes, and a limited distribution, review, and testing. + +Bootstrap +========= + +Let's say you want to write a new simple video filter called "foobar" which +takes one frame in input, changes the pixels in whatever fashion you fancy, and +outputs the modified frame. The most simple way of doing this is to take a +similar filter. We'll pick edgedetect, but any other should do. You can look +for others using the `./ffmpeg -v 0 -filters|grep ' V->V '` command. + + - sed 's/edgedetect/foobar/g;s/EdgeDetect/Foobar/g' libavfilter/vf_edgedetect.c > libavfilter/vf_foobar.c + - edit libavfilter/Makefile, and add an entry for "foobar" following the + pattern of the other filters. + - edit libavfilter/allfilters.c, and add an entry for "foobar" following the + pattern of the other filters. + - ./configure ... + - make -j<whatever> ffmpeg + - ./ffmpeg -i http://samples.ffmpeg.org/image-samples/lena.pnm -vf foobar foobar.png + Note here: you can obviously use a random local image instead of a remote URL. + +If everything went right, you should get a foobar.png with Lena edge-detected. + +That's it, your new playground is ready. + +Some little details about what's going on: +libavfilter/allfilters.c:avfilter_register_all() is called at runtime to create +a list of the available filters, but it's important to know that this file is +also parsed by the configure script, which in turn will define variables for +the build system and the C: + + --- after running configure --- + + $ grep FOOBAR config.mak + CONFIG_FOOBAR_FILTER=yes + $ grep FOOBAR config.h + #define CONFIG_FOOBAR_FILTER 1 + +CONFIG_FOOBAR_FILTER=yes from the config.mak is later used to enable the filter in +libavfilter/Makefile and CONFIG_FOOBAR_FILTER=1 from the config.h will be used +for registering the filter in libavfilter/allfilters.c. + +Filter code layout +================== + +You now need some theory about the general code layout of a filter. Open your +libavfilter/vf_foobar.c. This section will detail the important parts of the +code you need to understand before messing with it. + +Copyright +--------- + +First chunk is the copyright. Most filters are LGPL, and we are assuming +vf_foobar is as well. We are also assuming vf_foobar is not an edge detector +filter, so you can update the boilerplate with your credits. + +Doxy +---- + +Next chunk is the Doxygen about the file. See https://ffmpeg.org/doxygen/trunk/. +Detail here what the filter is, does, and add some references if you feel like +it. + +Context +------- + +Skip the headers and scroll down to the definition of FoobarContext. This is +your local state context. It is already filled with 0 when you get it so do not +worry about uninitialized reads into this context. This is where you put all +"global" information that you need; typically the variables storing the user options. +You'll notice the first field "const AVClass *class"; it's the only field you +need to keep assuming you have a context. There is some magic you don't need to +care about around this field, just let it be (in the first position) for now. + +Options +------- + +Then comes the options array. This is what will define the user accessible +options. For example, -vf foobar=mode=colormix:high=0.4:low=0.1. Most options +have the following pattern: + name, description, offset, type, default value, minimum value, maximum value, flags + + - name is the option name, keep it simple and lowercase + - description are short, in lowercase, without period, and describe what they + do, for example "set the foo of the bar" + - offset is the offset of the field in your local context, see the OFFSET() + macro; the option parser will use that information to fill the fields + according to the user input + - type is any of AV_OPT_TYPE_* defined in libavutil/opt.h + - default value is an union where you pick the appropriate type; "{.dbl=0.3}", + "{.i64=0x234}", "{.str=NULL}", ... + - min and max values define the range of available values, inclusive + - flags are AVOption generic flags. See AV_OPT_FLAG_* definitions + +When in doubt, just look at the other AVOption definitions all around the codebase, +there are tons of examples. + +Class +----- + +AVFILTER_DEFINE_CLASS(foobar) will define a unique foobar_class with some kind +of signature referencing the options, etc. which will be referenced in the +definition of the AVFilter. + +Filter definition +----------------- + +At the end of the file, you will find foobar_inputs, foobar_outputs and +the AVFilter ff_vf_foobar. Don't forget to update the AVFilter.description with +a description of what the filter does, starting with a capitalized letter and +ending with a period. You'd better drop the AVFilter.flags entry for now, and +re-add them later depending on the capabilities of your filter. + +Callbacks +--------- + +Let's now study the common callbacks. Before going into details, note that all +these callbacks are explained in details in libavfilter/avfilter.h, so in +doubt, refer to the doxy in that file. + +init() +~~~~~~ + +First one to be called is init(). It's flagged as cold because not called +often. Look for "cold" on +http://gcc.gnu.org/onlinedocs/gcc/Function-Attributes.html for more +information. + +As the name suggests, init() is where you eventually initialize and allocate +your buffers, pre-compute your data, etc. Note that at this point, your local +context already has the user options initialized, but you still haven't any +clue about the kind of data input you will get, so this function is often +mainly used to sanitize the user options. + +Some init()s will also define the number of inputs or outputs dynamically +according to the user options. A good example of this is the split filter, but +we won't cover this here since vf_foobar is just a simple 1:1 filter. + +uninit() +~~~~~~~~ + +Similarly, there is the uninit() callback, doing what the name suggests. Free +everything you allocated here. + +query_formats() +~~~~~~~~~~~~~~~ + +This follows the init() and is used for the format negotiation. Basically +you specify here what pixel format(s) (gray, rgb 32, yuv 4:2:0, ...) you accept +for your inputs, and what you can output. All pixel formats are defined in +libavutil/pixfmt.h. If you don't change the pixel format between the input and +the output, you just have to define a pixel formats array and call +ff_set_common_formats(). For more complex negotiation, you can refer to other +filters such as vf_scale. + +config_props() +~~~~~~~~~~~~~~ + +This callback is not necessary, but you will probably have one or more +config_props() anyway. It's not a callback for the filter itself but for its +inputs or outputs (they're called "pads" - AVFilterPad - in libavfilter's +lexicon). + +Inside the input config_props(), you are at a point where you know which pixel +format has been picked after query_formats(), and more information such as the +video width and height (inlink->{w,h}). So if you need to update your internal +context state depending on your input you can do it here. In edgedetect you can +see that this callback is used to allocate buffers depending on these +information. They will be destroyed in uninit(). + +Inside the output config_props(), you can define what you want to change in the +output. Typically, if your filter is going to double the size of the video, you +will update outlink->w and outlink->h. + +filter_frame() +~~~~~~~~~~~~~~ + +This is the callback you are waiting for from the beginning: it is where you +process the received frames. Along with the frame, you get the input link from +where the frame comes from. + + static int filter_frame(AVFilterLink *inlink, AVFrame *in) { ... } + +You can get the filter context through that input link: + + AVFilterContext *ctx = inlink->dst; + +Then access your internal state context: + + FoobarContext *foobar = ctx->priv; + +And also the output link where you will send your frame when you are done: + + AVFilterLink *outlink = ctx->outputs[0]; + +Here, we are picking the first output. You can have several, but in our case we +only have one since we are in a 1:1 input-output situation. + +If you want to define a simple pass-through filter, you can just do: + + return ff_filter_frame(outlink, in); + +But of course, you probably want to change the data of that frame. + +This can be done by accessing frame->data[] and frame->linesize[]. Important +note here: the width does NOT match the linesize. The linesize is always +greater or equal to the width. The padding created should not be changed or +even read. Typically, keep in mind that a previous filter in your chain might +have altered the frame dimension but not the linesize. Imagine a crop filter +that halves the video size: the linesizes won't be changed, just the width. + + <-------------- linesize ------------------------> + +-------------------------------+----------------+ ^ + | | | | + | | | | + | picture | padding | | height + | | | | + | | | | + +-------------------------------+----------------+ v + <----------- width -------------> + +Before modifying the "in" frame, you have to make sure it is writable, or get a +new one. Multiple scenarios are possible here depending on the kind of +processing you are doing. + +Let's say you want to change one pixel depending on multiple pixels (typically +the surrounding ones) of the input. In that case, you can't do an in-place +processing of the input so you will need to allocate a new frame, with the same +properties as the input one, and send that new frame to the next filter: + + AVFrame *out = ff_get_video_buffer(outlink, outlink->w, outlink->h); + if (!out) { + av_frame_free(&in); + return AVERROR(ENOMEM); + } + av_frame_copy_props(out, in); + + // out->data[...] = foobar(in->data[...]) + + av_frame_free(&in); + return ff_filter_frame(outlink, out); + +In-place processing +~~~~~~~~~~~~~~~~~~~ + +If you can just alter the input frame, you probably just want to do that +instead: + + av_frame_make_writable(in); + // in->data[...] = foobar(in->data[...]) + return ff_filter_frame(outlink, in); + +You may wonder why a frame might not be writable. The answer is that for +example a previous filter might still own the frame data: imagine a filter +prior to yours in the filtergraph that needs to cache the frame. You must not +alter that frame, otherwise it will make that previous filter buggy. This is +where av_frame_make_writable() helps (it won't have any effect if the frame +already is writable). + +The problem with using av_frame_make_writable() is that in the worst case it +will copy the whole input frame before you change it all over again with your +filter: if the frame is not writable, av_frame_make_writable() will allocate +new buffers, and copy the input frame data. You don't want that, and you can +avoid it by just allocating a new buffer if necessary, and process from in to +out in your filter, saving the memcpy. Generally, this is done following this +scheme: + + int direct = 0; + AVFrame *out; + + if (av_frame_is_writable(in)) { + direct = 1; + out = in; + } else { + out = ff_get_video_buffer(outlink, outlink->w, outlink->h); + if (!out) { + av_frame_free(&in); + return AVERROR(ENOMEM); + } + av_frame_copy_props(out, in); + } + + // out->data[...] = foobar(in->data[...]) + + if (!direct) + av_frame_free(&in); + return ff_filter_frame(outlink, out); + +Of course, this will only work if you can do in-place processing. To test if +your filter handles well the permissions, you can use the perms filter. For +example with: + + -vf perms=random,foobar + +Make sure no automatic pixel conversion is inserted between perms and foobar, +otherwise the frames permissions might change again and the test will be +meaningless: add av_log(0,0,"direct=%d\n",direct) in your code to check that. +You can avoid the issue with something like: + + -vf format=rgb24,perms=random,foobar + +...assuming your filter accepts rgb24 of course. This will make sure the +necessary conversion is inserted before the perms filter. + +Timeline +~~~~~~~~ + +Adding timeline support +(http://ffmpeg.org/ffmpeg-filters.html#Timeline-editing) is often an easy +feature to add. In the most simple case, you just have to add +AVFILTER_FLAG_SUPPORT_TIMELINE_GENERIC to the AVFilter.flags. You can typically +do this when your filter does not need to save the previous context frames, or +basically if your filter just alters whatever goes in and doesn't need +previous/future information. See for instance commit 86cb986ce that adds +timeline support to the fieldorder filter. + +In some cases, you might need to reset your context somehow. This is handled by +the AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL flag which is used if the filter +must not process the frames but still wants to keep track of the frames going +through (to keep them in cache for when it's enabled again). See for example +commit 69d72140a that adds timeline support to the phase filter. + +Threading +~~~~~~~~~ + +libavfilter does not yet support frame threading, but you can add slice +threading to your filters. + +Let's say the foobar filter has the following frame processing function: + + dst = out->data[0]; + src = in ->data[0]; + + for (y = 0; y < inlink->h; y++) { + for (x = 0; x < inlink->w; x++) + dst[x] = foobar(src[x]); + dst += out->linesize[0]; + src += in ->linesize[0]; + } + +The first thing is to make this function work into slices. The new code will +look like this: + + for (y = slice_start; y < slice_end; y++) { + for (x = 0; x < inlink->w; x++) + dst[x] = foobar(src[x]); + dst += out->linesize[0]; + src += in ->linesize[0]; + } + +The source and destination pointers, and slice_start/slice_end will be defined +according to the number of jobs. Generally, it looks like this: + + const int slice_start = (in->height * jobnr ) / nb_jobs; + const int slice_end = (in->height * (jobnr+1)) / nb_jobs; + uint8_t *dst = out->data[0] + slice_start * out->linesize[0]; + const uint8_t *src = in->data[0] + slice_start * in->linesize[0]; + +This new code will be isolated in a new filter_slice(): + + static int filter_slice(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs) { ... } + +Note that we need our input and output frame to define slice_{start,end} and +dst/src, which are not available in that callback. They will be transmitted +through the opaque void *arg. You have to define a structure which contains +everything you need: + + typedef struct ThreadData { + AVFrame *in, *out; + } ThreadData; + +If you need some more information from your local context, put them here. + +In you filter_slice function, you access it like that: + + const ThreadData *td = arg; + +Then in your filter_frame() callback, you need to call the threading +distributor with something like this: + + ThreadData td; + + // ... + + td.in = in; + td.out = out; + ctx->internal->execute(ctx, filter_slice, &td, NULL, FFMIN(outlink->h, ctx->graph->nb_threads)); + + // ... + + return ff_filter_frame(outlink, out); + +Last step is to add AVFILTER_FLAG_SLICE_THREADS flag to AVFilter.flags. + +For more example of slice threading additions, you can try to run git log -p +--grep 'slice threading' libavfilter/ + +Finalization +~~~~~~~~~~~~ + +When your awesome filter is finished, you have a few more steps before you're +done: + + - write its documentation in doc/filters.texi, and test the output with make + doc/ffmpeg-filters.html. + - add a FATE test, generally by adding an entry in + tests/fate/filter-video.mak, add running make fate-filter-foobar GEN=1 to + generate the data. + - add an entry in the Changelog + - edit libavfilter/version.h and increase LIBAVFILTER_VERSION_MINOR by one + (and reset LIBAVFILTER_VERSION_MICRO to 100) + - git add ... && git commit -m "avfilter: add foobar filter." && git format-patch -1 + +When all of this is done, you can submit your patch to the ffmpeg-devel +mailing-list for review. If you need any help, feel free to come on our IRC +channel, #ffmpeg-devel on irc.freenode.net. |