aboutsummaryrefslogtreecommitdiffstats
path: root/libavcodec/mlpdec.c
diff options
context:
space:
mode:
Diffstat (limited to 'libavcodec/mlpdec.c')
-rw-r--r--libavcodec/mlpdec.c119
1 files changed, 74 insertions, 45 deletions
diff --git a/libavcodec/mlpdec.c b/libavcodec/mlpdec.c
index 169437acef..b2d1e2f715 100644
--- a/libavcodec/mlpdec.c
+++ b/libavcodec/mlpdec.c
@@ -2,20 +2,20 @@
* MLP decoder
* Copyright (c) 2007-2008 Ian Caulfield
*
- * 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
*/
@@ -105,7 +105,7 @@ typedef struct SubStream {
/// Whether the LSBs of the matrix output are encoded in the bitstream.
uint8_t lsb_bypass[MAX_MATRICES];
/// Matrix coefficients, stored as 2.14 fixed point.
- int32_t matrix_coeff[MAX_MATRICES][MAX_CHANNELS];
+ DECLARE_ALIGNED(32, int32_t, matrix_coeff)[MAX_MATRICES][MAX_CHANNELS];
/// Left shift to apply to noise values in 0x31eb substreams.
uint8_t matrix_noise_shift[MAX_MATRICES];
//@}
@@ -144,6 +144,9 @@ typedef struct MLPDecodeContext {
/// Index of the last substream to decode - further substreams are skipped.
uint8_t max_decoded_substream;
+ /// Stream needs channel reordering to comply with FFmpeg's channel order
+ uint8_t needs_reordering;
+
/// number of PCM samples contained in each frame
int access_unit_size;
/// next power of two above the number of samples in each frame
@@ -156,7 +159,7 @@ typedef struct MLPDecodeContext {
int8_t noise_buffer[MAX_BLOCKSIZE_POW2];
int8_t bypassed_lsbs[MAX_BLOCKSIZE][MAX_CHANNELS];
- int32_t sample_buffer[MAX_BLOCKSIZE][MAX_CHANNELS];
+ DECLARE_ALIGNED(32, int32_t, sample_buffer)[MAX_BLOCKSIZE][MAX_CHANNELS];
MLPDSPContext dsp;
} MLPDecodeContext;
@@ -380,10 +383,22 @@ static int read_major_sync(MLPDecodeContext *m, GetBitContext *gb)
* substream is Stereo. Subsequent substreams' layouts are indicated in the
* major sync. */
if (m->avctx->codec_id == AV_CODEC_ID_MLP) {
+ if (mh.stream_type != 0xbb) {
+ avpriv_request_sample(m->avctx,
+ "unexpected stream_type %X in MLP",
+ mh.stream_type);
+ return AVERROR_PATCHWELCOME;
+ }
if ((substr = (mh.num_substreams > 1)))
m->substream[0].ch_layout = AV_CH_LAYOUT_STEREO;
m->substream[substr].ch_layout = mh.channel_layout_mlp;
} else {
+ if (mh.stream_type != 0xba) {
+ avpriv_request_sample(m->avctx,
+ "unexpected stream_type %X in !MLP",
+ mh.stream_type);
+ return AVERROR_PATCHWELCOME;
+ }
if ((substr = (mh.num_substreams > 1)))
m->substream[0].ch_layout = AV_CH_LAYOUT_STEREO;
if (mh.num_substreams > 2)
@@ -392,8 +407,17 @@ static int read_major_sync(MLPDecodeContext *m, GetBitContext *gb)
else
m->substream[2].ch_layout = mh.channel_layout_thd_stream1;
m->substream[substr].ch_layout = mh.channel_layout_thd_stream1;
+
+ if (m->avctx->channels<=2 && m->substream[substr].ch_layout == AV_CH_LAYOUT_MONO && m->max_decoded_substream == 1) {
+ av_log(m->avctx, AV_LOG_DEBUG, "Mono stream with 2 substreams, ignoring 2nd\n");
+ m->max_decoded_substream = 0;
+ if (m->avctx->channels==2)
+ m->avctx->channel_layout = AV_CH_LAYOUT_STEREO;
+ }
}
+ m->needs_reordering = mh.channel_arrangement >= 18 && mh.channel_arrangement <= 20;
+
/* Parse the TrueHD decoder channel modifiers and set each substream's
* AVMatrixEncoding accordingly.
*
@@ -479,7 +503,7 @@ static int read_restart_header(MLPDecodeContext *m, GetBitContext *gbp,
if (max_matrix_channel > std_max_matrix_channel) {
av_log(m->avctx, AV_LOG_ERROR,
"Max matrix channel cannot be greater than %d.\n",
- max_matrix_channel);
+ std_max_matrix_channel);
return AVERROR_INVALIDDATA;
}
@@ -491,11 +515,11 @@ static int read_restart_header(MLPDecodeContext *m, GetBitContext *gbp,
/* This should happen for TrueHD streams with >6 channels and MLP's noise
* type. It is not yet known if this is allowed. */
- if (s->max_channel > MAX_MATRIX_CHANNEL_MLP && !s->noise_type) {
+ if (max_channel > MAX_MATRIX_CHANNEL_MLP && !s->noise_type) {
avpriv_request_sample(m->avctx,
"%d channels (more than the "
"maximum supported by the decoder)",
- s->max_channel + 2);
+ max_channel + 2);
return AVERROR_PATCHWELCOME;
}
@@ -590,6 +614,20 @@ static int read_restart_header(MLPDecodeContext *m, GetBitContext *gbp,
s->output_shift,
s->max_matrix_channel,
m->avctx->sample_fmt == AV_SAMPLE_FMT_S32);
+
+ if (m->avctx->codec_id == AV_CODEC_ID_MLP && m->needs_reordering) {
+ if (m->avctx->channel_layout == (AV_CH_LAYOUT_QUAD|AV_CH_LOW_FREQUENCY) ||
+ m->avctx->channel_layout == AV_CH_LAYOUT_5POINT0_BACK) {
+ int i = s->ch_assign[4];
+ s->ch_assign[4] = s->ch_assign[3];
+ s->ch_assign[3] = s->ch_assign[2];
+ s->ch_assign[2] = i;
+ } else if (m->avctx->channel_layout == AV_CH_LAYOUT_5POINT1_BACK) {
+ FFSWAP(int, s->ch_assign[2], s->ch_assign[4]);
+ FFSWAP(int, s->ch_assign[3], s->ch_assign[5]);
+ }
+ }
+
}
return 0;
@@ -608,7 +646,7 @@ static int read_filter_params(MLPDecodeContext *m, GetBitContext *gbp,
int i, order;
// Filter is 0 for FIR, 1 for IIR.
- assert(filter < 2);
+ av_assert0(filter < 2);
if (m->filter_changed[channel][filter]++ > 1) {
av_log(m->avctx, AV_LOG_ERROR, "Filters may change only once per access unit.\n");
@@ -663,7 +701,7 @@ static int read_filter_params(MLPDecodeContext *m, GetBitContext *gbp,
/* TODO: Check validity of state data. */
for (i = 0; i < order; i++)
- fp->state[i] = get_sbits(gbp, state_bits) << state_shift;
+ fp->state[i] = state_bits ? get_sbits(gbp, state_bits) << state_shift : 0;
}
}
@@ -782,6 +820,7 @@ static int read_channel_params(MLPDecodeContext *m, unsigned int substr,
if (cp->huff_lsbs > 24) {
av_log(m->avctx, AV_LOG_ERROR, "Invalid huff_lsbs.\n");
+ cp->huff_lsbs = 0;
return AVERROR_INVALIDDATA;
}
@@ -808,7 +847,7 @@ static int read_decoding_params(MLPDecodeContext *m, GetBitContext *gbp,
if (get_bits1(gbp)) {
s->blocksize = get_bits(gbp, 9);
if (s->blocksize < 8 || s->blocksize > m->access_unit_size) {
- av_log(m->avctx, AV_LOG_ERROR, "Invalid blocksize.");
+ av_log(m->avctx, AV_LOG_ERROR, "Invalid blocksize.\n");
s->blocksize = 0;
return AVERROR_INVALIDDATA;
}
@@ -848,7 +887,7 @@ static int read_decoding_params(MLPDecodeContext *m, GetBitContext *gbp,
return 0;
}
-#define MSB_MASK(bits) (-1u << bits)
+#define MSB_MASK(bits) (-1u << (bits))
/** Generate PCM samples using the prediction filters and residual values
* read from the data stream, and update the filter state. */
@@ -986,15 +1025,27 @@ static void fill_noise_buffer(MLPDecodeContext *m, unsigned int substr)
s->noisegen_seed = seed;
}
+/** Write the audio data into the output buffer. */
-/** Apply the channel matrices in turn to reconstruct the original audio
- * samples. */
-
-static void rematrix_channels(MLPDecodeContext *m, unsigned int substr)
+static int output_data(MLPDecodeContext *m, unsigned int substr,
+ AVFrame *frame, int *got_frame_ptr)
{
+ AVCodecContext *avctx = m->avctx;
SubStream *s = &m->substream[substr];
unsigned int mat;
unsigned int maxchan;
+ int ret;
+ int is32 = (m->avctx->sample_fmt == AV_SAMPLE_FMT_S32);
+
+ if (m->avctx->channels != s->max_matrix_channel + 1) {
+ av_log(m->avctx, AV_LOG_ERROR, "channel count mismatch\n");
+ return AVERROR_INVALIDDATA;
+ }
+
+ if (!s->blockpos) {
+ av_log(avctx, AV_LOG_ERROR, "No samples to output.\n");
+ return AVERROR_INVALIDDATA;
+ }
maxchan = s->max_matrix_channel;
if (!s->noise_type) {
@@ -1004,6 +1055,8 @@ static void rematrix_channels(MLPDecodeContext *m, unsigned int substr)
fill_noise_buffer(m, substr);
}
+ /* Apply the channel matrices in turn to reconstruct the original audio
+ * samples. */
for (mat = 0; mat < s->num_primitive_matrices; mat++) {
unsigned int dest_ch = s->matrix_out_ch[mat];
m->dsp.mlp_rematrix_channel(&m->sample_buffer[0][0],
@@ -1018,34 +1071,11 @@ static void rematrix_channels(MLPDecodeContext *m, unsigned int substr)
m->access_unit_size_pow2,
MSB_MASK(s->quant_step_size[dest_ch]));
}
-}
-
-/** Write the audio data into the output buffer. */
-
-static int output_data(MLPDecodeContext *m, unsigned int substr,
- AVFrame *frame, int *got_frame_ptr)
-{
- AVCodecContext *avctx = m->avctx;
- SubStream *s = &m->substream[substr];
- int ret;
- int is32 = (m->avctx->sample_fmt == AV_SAMPLE_FMT_S32);
-
- if (m->avctx->channels != s->max_matrix_channel + 1) {
- av_log(m->avctx, AV_LOG_ERROR, "channel count mismatch\n");
- return AVERROR_INVALIDDATA;
- }
-
- if (!s->blockpos) {
- av_log(avctx, AV_LOG_ERROR, "No samples to output.\n");
- return AVERROR_INVALIDDATA;
- }
/* get output buffer */
frame->nb_samples = s->blockpos;
- if ((ret = ff_get_buffer(avctx, frame, 0)) < 0) {
- av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
+ if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
return ret;
- }
s->lossless_check_data = m->dsp.mlp_pack_output(s->lossless_check_data,
s->blockpos,
m->sample_buffer,
@@ -1085,7 +1115,7 @@ static int read_access_unit(AVCodecContext *avctx, void* data,
int ret;
if (buf_size < 4)
- return 0;
+ return AVERROR_INVALIDDATA;
length = (AV_RB16(buf) & 0xfff) * 2;
@@ -1248,8 +1278,6 @@ next_substr:
buf += substream_data_len[substr];
}
- rematrix_channels(m, m->max_decoded_substream);
-
if ((ret = output_data(m, m->max_decoded_substream, data, got_frame_ptr)) < 0)
return ret;
@@ -1264,6 +1292,7 @@ error:
return AVERROR_INVALIDDATA;
}
+#if CONFIG_MLP_DECODER
AVCodec ff_mlp_decoder = {
.name = "mlp",
.long_name = NULL_IF_CONFIG_SMALL("MLP (Meridian Lossless Packing)"),
@@ -1274,7 +1303,7 @@ AVCodec ff_mlp_decoder = {
.decode = read_access_unit,
.capabilities = AV_CODEC_CAP_DR1,
};
-
+#endif
#if CONFIG_TRUEHD_DECODER
AVCodec ff_truehd_decoder = {
.name = "truehd",
generated by cgit v1.2.3 (git 2.25.1) at 2024-06-08 18:39:21 +0000