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author | Adrian Knoth <adi@drcomp.erfurt.thur.de> | 2011-01-26 19:32:14 +0100 |
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committer | Takashi Iwai <tiwai@suse.de> | 2011-01-27 12:09:18 +0100 |
commit | 0dca1793063c28dde8f6c49c9c72203fe5cb6efc (patch) | |
tree | 3743c433efe2a8fce31de9a6022d98ae493dabdf /include/sound/hdspm.h | |
parent | c6d43ba816d1cf1d125bfbfc938f2a28a87facf9 (diff) | |
download | kernel_samsung_smdk4412-0dca1793063c28dde8f6c49c9c72203fe5cb6efc.tar.gz kernel_samsung_smdk4412-0dca1793063c28dde8f6c49c9c72203fe5cb6efc.tar.bz2 kernel_samsung_smdk4412-0dca1793063c28dde8f6c49c9c72203fe5cb6efc.zip |
ALSA: hdspm - Add support for RME RayDAT and AIO
Incorporate changes by Florian Faber into hdspm.c. Code taken from
http://wiki.linuxproaudio.org/index.php/Driver:hdspe
Heavily reworked to mostly comply with the coding standard (whitespace
fixes, line width, C++ style comments)
The code was tested and confirmed to be working on RME RayDAT.
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Diffstat (limited to 'include/sound/hdspm.h')
-rw-r--r-- | include/sound/hdspm.h | 349 |
1 files changed, 307 insertions, 42 deletions
diff --git a/include/sound/hdspm.h b/include/sound/hdspm.h index 81990b2bcc9..c3f18194b08 100644 --- a/include/sound/hdspm.h +++ b/include/sound/hdspm.h @@ -3,8 +3,8 @@ /* * Copyright (C) 2003 Winfried Ritsch (IEM) * based on hdsp.h from Thomas Charbonnel (thomas@undata.org) - * - * + * + * * This program 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 2 of the License, or @@ -23,50 +23,41 @@ /* Maximum channels is 64 even on 56Mode you have 64playbacks to matrix */ #define HDSPM_MAX_CHANNELS 64 -/* -------------------- IOCTL Peak/RMS Meters -------------------- */ - -/* peam rms level structure like we get from hardware - - maybe in future we can memory map it so I just copy it - to user on ioctl call now an dont change anything - rms are made out of low and high values - where (long) ????_rms = (????_rms_l >> 8) + ((????_rms_h & 0xFFFFFF00)<<24) - (i asume so from the code) -*/ - -struct hdspm_peak_rms { - - unsigned int level_offset[1024]; +enum hdspm_io_type { + MADI, + MADIface, + AIO, + AES32, + RayDAT +}; - unsigned int input_peak[64]; - unsigned int playback_peak[64]; - unsigned int output_peak[64]; - unsigned int xxx_peak[64]; /* not used */ +enum hdspm_speed { + ss, + ds, + qs +}; - unsigned int reserved[256]; /* not used */ +/* -------------------- IOCTL Peak/RMS Meters -------------------- */ - unsigned int input_rms_l[64]; - unsigned int playback_rms_l[64]; - unsigned int output_rms_l[64]; - unsigned int xxx_rms_l[64]; /* not used */ +struct hdspm_peak_rms { + uint32_t input_peaks[64]; + uint32_t playback_peaks[64]; + uint32_t output_peaks[64]; - unsigned int input_rms_h[64]; - unsigned int playback_rms_h[64]; - unsigned int output_rms_h[64]; - unsigned int xxx_rms_h[64]; /* not used */ -}; + uint64_t input_rms[64]; + uint64_t playback_rms[64]; + uint64_t output_rms[64]; -struct hdspm_peak_rms_ioctl { - struct hdspm_peak_rms *peak; + uint8_t speed; /* enum {ss, ds, qs} */ + int status2; }; -/* use indirect access due to the limit of ioctl bit size */ #define SNDRV_HDSPM_IOCTL_GET_PEAK_RMS \ - _IOR('H', 0x40, struct hdspm_peak_rms_ioctl) + _IOR('H', 0x42, struct hdspm_peak_rms) /* ------------ CONFIG block IOCTL ---------------------- */ -struct hdspm_config_info { +struct hdspm_config { unsigned char pref_sync_ref; unsigned char wordclock_sync_check; unsigned char madi_sync_check; @@ -80,18 +71,121 @@ struct hdspm_config_info { unsigned int analog_out; }; -#define SNDRV_HDSPM_IOCTL_GET_CONFIG_INFO \ - _IOR('H', 0x41, struct hdspm_config_info) +#define SNDRV_HDSPM_IOCTL_GET_CONFIG \ + _IOR('H', 0x41, struct hdspm_config) + +/** + * If there's a TCO (TimeCode Option) board installed, + * there are further options and status data available. + * The hdspm_ltc structure contains the current SMPTE + * timecode and some status information and can be + * obtained via SNDRV_HDSPM_IOCTL_GET_LTC or in the + * hdspm_status struct. + **/ + +enum hdspm_ltc_format { + format_invalid, + fps_24, + fps_25, + fps_2997, + fps_30 +}; + +enum hdspm_ltc_frame { + frame_invalid, + drop_frame, + full_frame +}; + +enum hdspm_ltc_input_format { + ntsc, + pal, + no_video +}; + +struct hdspm_ltc { + unsigned int ltc; + + enum hdspm_ltc_format format; + enum hdspm_ltc_frame frame; + enum hdspm_ltc_input_format input_format; +}; + +#define SNDRV_HDSPM_IOCTL_GET_LTC _IOR('H', 0x46, struct hdspm_mixer_ioctl) + +/** + * The status data reflects the device's current state + * as determined by the card's configuration and + * connection status. + **/ + +enum hdspm_sync { + hdspm_sync_no_lock = 0, + hdspm_sync_lock = 1, + hdspm_sync_sync = 2 +}; + +enum hdspm_madi_input { + hdspm_input_optical = 0, + hdspm_input_coax = 1 +}; + +enum hdspm_madi_channel_format { + hdspm_format_ch_64 = 0, + hdspm_format_ch_56 = 1 +}; + +enum hdspm_madi_frame_format { + hdspm_frame_48 = 0, + hdspm_frame_96 = 1 +}; + +enum hdspm_syncsource { + syncsource_wc = 0, + syncsource_madi = 1, + syncsource_tco = 2, + syncsource_sync = 3, + syncsource_none = 4 +}; + +struct hdspm_status { + uint8_t card_type; /* enum hdspm_io_type */ + enum hdspm_syncsource autosync_source; + uint64_t card_clock; + uint32_t master_period; + + union { + struct { + uint8_t sync_wc; /* enum hdspm_sync */ + uint8_t sync_madi; /* enum hdspm_sync */ + uint8_t sync_tco; /* enum hdspm_sync */ + uint8_t sync_in; /* enum hdspm_sync */ + uint8_t madi_input; /* enum hdspm_madi_input */ + uint8_t channel_format; /* enum hdspm_madi_channel_format */ + uint8_t frame_format; /* enum hdspm_madi_frame_format */ + } madi; + } card_specific; +}; -/* get Soundcard Version */ +#define SNDRV_HDSPM_IOCTL_GET_STATUS \ + _IOR('H', 0x47, struct hdspm_status) + +/** + * Get information about the card and its add-ons. + **/ + +#define HDSPM_ADDON_TCO 1 struct hdspm_version { + uint8_t card_type; /* enum hdspm_io_type */ + char cardname[20]; + unsigned int serial; unsigned short firmware_rev; + int addons; }; -#define SNDRV_HDSPM_IOCTL_GET_VERSION _IOR('H', 0x43, struct hdspm_version) - +#define SNDRV_HDSPM_IOCTL_GET_VERSION _IOR('H', 0x48, struct hdspm_version) /* ------------- get Matrix Mixer IOCTL --------------- */ @@ -103,7 +197,7 @@ struct hdspm_version { /* equivalent to hardware definition, maybe for future feature of mmap of * them */ -/* each of 64 outputs has 64 infader and 64 outfader: +/* each of 64 outputs has 64 infader and 64 outfader: Ins to Outs mixer[out].in[in], Outstreams to Outs mixer[out].pb[pb] */ #define HDSPM_MIXER_CHANNELS HDSPM_MAX_CHANNELS @@ -131,4 +225,175 @@ typedef struct hdspm_version hdspm_version_t; typedef struct hdspm_channelfader snd_hdspm_channelfader_t; typedef struct hdspm_mixer hdspm_mixer_t; -#endif /* __SOUND_HDSPM_H */ +/* These tables map the ALSA channels 1..N to the channels that we + need to use in order to find the relevant channel buffer. RME + refers to this kind of mapping as between "the ADAT channel and + the DMA channel." We index it using the logical audio channel, + and the value is the DMA channel (i.e. channel buffer number) + where the data for that channel can be read/written from/to. +*/ + +char channel_map_unity_ss[HDSPM_MAX_CHANNELS] = { + 0, 1, 2, 3, 4, 5, 6, 7, + 8, 9, 10, 11, 12, 13, 14, 15, + 16, 17, 18, 19, 20, 21, 22, 23, + 24, 25, 26, 27, 28, 29, 30, 31, + 32, 33, 34, 35, 36, 37, 38, 39, + 40, 41, 42, 43, 44, 45, 46, 47, + 48, 49, 50, 51, 52, 53, 54, 55, + 56, 57, 58, 59, 60, 61, 62, 63 +}; + +char channel_map_unity_ds[HDSPM_MAX_CHANNELS] = { + 0, 2, 4, 6, 8, 10, 12, 14, + 16, 18, 20, 22, 24, 26, 28, 30, + 32, 34, 36, 38, 40, 42, 44, 46, + 48, 50, 52, 54, 56, 58, 60, 62, + -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, +}; + +char channel_map_unity_qs[HDSPM_MAX_CHANNELS] = { + 0, 4, 8, 12, 16, 20, 24, 28, + 32, 36, 40, 44, 48, 52, 56, 60, + -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, -1, -1, -1, + -1, -1, -1, -1, -1, -1, -1, -1, +}; + +char channel_map_raydat_ss[HDSPM_MAX_CHANNELS] = { + 4, 5, 6, 7, 8, 9, 10, 11, /* ADAT 1 */ + 12, 13, 14, 15, 16, 17, 18, 19, /* ADAT 2 */ + 20, 21, 22, 23, 24, 25, 26, 27, /* ADAT 3 */ + 28, 29, 30, 31, 32, 33, 34, 35, /* ADAT 4 */ + 0, 1, /* AES */ + 2, 3, /* SPDIF */ + -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, +}; + +char channel_map_raydat_ds[HDSPM_MAX_CHANNELS] = { + 4, 5, 6, 7, /* ADAT 1 */ + 8, 9, 10, 11, /* ADAT 2 */ + 12, 13, 14, 15, /* ADAT 3 */ + 16, 17, 18, 19, /* ADAT 4 */ + 0, 1, /* AES */ + 2, 3, /* SPDIF */ + -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, -1, -1, -1, -1, -1, -1, -1, +}; + +char channel_map_raydat_qs[HDSPM_MAX_CHANNELS] = { + 4, 5, /* ADAT 1 */ + 6, 7, /* ADAT 2 */ + 8, 9, /* ADAT 3 */ + 10, 11, /* ADAT 4 */ + 0, 1, /* AES */ + 2, 3, /* SPDIF */ + -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, -1, -1, -1, -1, -1, -1, -1, + -1, -1, -1, -1, -1, -1, -1, -1, +}; + +char channel_map_aio_in_ss[HDSPM_MAX_CHANNELS] = { + 0, 1, /* line in */ + 8, 9, /* aes in, */ + 10, 11, /* spdif in */ + 12, 13, 14, 15, 16, 17, 18, 19, /* ADAT in */ + -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, -1, -1, -1, -1, -1, + -1, -1, -1, -1, -1, -1, -1, -1, +}; + +char channel_map_aio_out_ss[HDSPM_MAX_CHANNELS] = { + 0, 1, /* line out */ + 8, 9, /* aes out */ + 10, 11, /* spdif out */ + 12, 13, 14, 15, 16, 17, 18, 19, /* ADAT out */ + 6, 7, /* phone out */ + -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, -1, -1, -1, + -1, -1, -1, -1, -1, -1, -1, -1, +}; + +char channel_map_aio_in_ds[HDSPM_MAX_CHANNELS] = { + 0, 1, /* line in */ + 8, 9, /* aes in */ + 10, 11, /* spdif in */ + 12, 14, 16, 18, /* adat in */ + -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, -1, + -1, -1, -1, -1, -1, -1, -1, -1, + -1, -1, -1, -1, -1, -1, -1, -1 +}; + +char channel_map_aio_out_ds[HDSPM_MAX_CHANNELS] = { + 0, 1, /* line out */ + 8, 9, /* aes out */ + 10, 11, /* spdif out */ + 12, 14, 16, 18, /* adat out */ + 6, 7, /* phone out */ + -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, -1, -1, -1, -1, -1, -1, -1, + -1, -1, -1, -1, -1, -1, -1, -1 +}; + +char channel_map_aio_in_qs[HDSPM_MAX_CHANNELS] = { + 0, 1, /* line in */ + 8, 9, /* aes in */ + 10, 11, /* spdif in */ + 12, 16, /* adat in */ + -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, -1, -1, -1, + -1, -1, -1, -1, -1, -1, -1, -1, + -1, -1, -1, -1, -1, -1, -1, -1 +}; + +char channel_map_aio_out_qs[HDSPM_MAX_CHANNELS] = { + 0, 1, /* line out */ + 8, 9, /* aes out */ + 10, 11, /* spdif out */ + 12, 16, /* adat out */ + 6, 7, /* phone out */ + -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, -1, + -1, -1, -1, -1, -1, -1, -1, -1, + -1, -1, -1, -1, -1, -1, -1, -1 +}; + +#endif |