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-rw-r--r--Changelog1
-rwxr-xr-xconfigure1
-rw-r--r--doc/general.texi1
-rw-r--r--libavcodec/Makefile1
-rw-r--r--libavcodec/allcodecs.c1
-rw-r--r--libavcodec/avcodec.h4
-rw-r--r--libavcodec/mlpdec.c1180
7 files changed, 1187 insertions, 2 deletions
diff --git a/Changelog b/Changelog
index 0ff53d73c9..c86ed4dedc 100644
--- a/Changelog
+++ b/Changelog
@@ -122,6 +122,7 @@ version <next>
- MAXIS EA XA (.xa) demuxer / decoder
- BFI video decoder
- OMA demuxer
+- MLP/TrueHD decoder
version 0.4.9-pre1:
diff --git a/configure b/configure
index 4f92b8a8f8..b55c2fc353 100755
--- a/configure
+++ b/configure
@@ -832,6 +832,7 @@ ac3_decoder_deps="gpl"
dxa_decoder_deps="zlib"
flashsv_decoder_deps="zlib"
flashsv_encoder_deps="zlib"
+mlp_decoder_deps="mlp_parser"
mpeg_xvmc_decoder_deps="xvmc"
png_decoder_deps="zlib"
png_encoder_deps="zlib"
diff --git a/doc/general.texi b/doc/general.texi
index 9e96ea4f08..487354d802 100644
--- a/doc/general.texi
+++ b/doc/general.texi
@@ -259,6 +259,7 @@ following image formats are supported:
@item Renderware TXD @tab @tab X @tab Texture dictionaries used by the Renderware Engine.
@item AMV @tab @tab X @tab Used in Chinese MP3 players.
@item Mimic @tab @tab X @tab Used in MSN Messenger Webcam streams.
+@item MLP/TrueHD @tab @tab X @tab Used in DVD-Audio and Blu-Ray discs.
@end multitable
@code{X} means that encoding (resp. decoding) is supported.
diff --git a/libavcodec/Makefile b/libavcodec/Makefile
index da0122a785..bda6203d41 100644
--- a/libavcodec/Makefile
+++ b/libavcodec/Makefile
@@ -107,6 +107,7 @@ OBJS-$(CONFIG_MIMIC_DECODER) += mimic.o
OBJS-$(CONFIG_MJPEG_DECODER) += mjpegdec.o mjpeg.o
OBJS-$(CONFIG_MJPEG_ENCODER) += mjpegenc.o mjpeg.o mpegvideo_enc.o motion_est.o ratecontrol.o mpeg12data.o mpegvideo.o
OBJS-$(CONFIG_MJPEGB_DECODER) += mjpegbdec.o mjpegdec.o mjpeg.o
+OBJS-$(CONFIG_MLP_DECODER) += mlpdec.o
OBJS-$(CONFIG_MMVIDEO_DECODER) += mmvideo.o
OBJS-$(CONFIG_MP2_DECODER) += mpegaudiodec.o mpegaudiodecheader.o mpegaudio.o mpegaudiodata.o
OBJS-$(CONFIG_MP2_ENCODER) += mpegaudioenc.o mpegaudio.o mpegaudiodata.o
diff --git a/libavcodec/allcodecs.c b/libavcodec/allcodecs.c
index d15abe89ff..8051fae602 100644
--- a/libavcodec/allcodecs.c
+++ b/libavcodec/allcodecs.c
@@ -189,6 +189,7 @@ void avcodec_register_all(void)
REGISTER_DECODER (IMC, imc);
REGISTER_DECODER (MACE3, mace3);
REGISTER_DECODER (MACE6, mace6);
+ REGISTER_DECODER (MLP, mlp);
REGISTER_ENCDEC (MP2, mp2);
REGISTER_DECODER (MP3, mp3);
REGISTER_DECODER (MP3ADU, mp3adu);
diff --git a/libavcodec/avcodec.h b/libavcodec/avcodec.h
index 19b7e56dc6..35fe19e2d0 100644
--- a/libavcodec/avcodec.h
+++ b/libavcodec/avcodec.h
@@ -30,8 +30,8 @@
#include "libavutil/avutil.h"
#define LIBAVCODEC_VERSION_MAJOR 51
-#define LIBAVCODEC_VERSION_MINOR 57
-#define LIBAVCODEC_VERSION_MICRO 2
+#define LIBAVCODEC_VERSION_MINOR 58
+#define LIBAVCODEC_VERSION_MICRO 0
#define LIBAVCODEC_VERSION_INT AV_VERSION_INT(LIBAVCODEC_VERSION_MAJOR, \
LIBAVCODEC_VERSION_MINOR, \
diff --git a/libavcodec/mlpdec.c b/libavcodec/mlpdec.c
new file mode 100644
index 0000000000..07700a5c80
--- /dev/null
+++ b/libavcodec/mlpdec.c
@@ -0,0 +1,1180 @@
+/*
+ * MLP decoder
+ * Copyright (c) 2007-2008 Ian Caulfield
+ *
+ * This file is part of FFmpeg.
+ *
+ * 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.
+ *
+ * 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 FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+/**
+ * @file mlpdec.c
+ * MLP decoder
+ */
+
+#include "avcodec.h"
+#include "libavutil/intreadwrite.h"
+#include "bitstream.h"
+#include "libavutil/crc.h"
+#include "parser.h"
+#include "mlp_parser.h"
+
+/** Maximum number of channels that can be decoded. */
+#define MAX_CHANNELS 16
+
+/** Maximum number of matrices used in decoding. Most streams have one matrix
+ * per output channel, but some rematrix a channel (usually 0) more than once.
+ */
+
+#define MAX_MATRICES 15
+
+/** Maximum number of substreams that can be decoded. This could also be set
+ * higher, but again I haven't seen any examples with more than two. */
+#define MAX_SUBSTREAMS 2
+
+/** Maximum sample frequency seen in files. */
+#define MAX_SAMPLERATE 192000
+
+/** The maximum number of audio samples within one access unit. */
+#define MAX_BLOCKSIZE (40 * (MAX_SAMPLERATE / 48000))
+/** The next power of two greater than MAX_BLOCKSIZE. */
+#define MAX_BLOCKSIZE_POW2 (64 * (MAX_SAMPLERATE / 48000))
+
+/** Number of allowed filters. */
+#define NUM_FILTERS 2
+
+/** The maximum number of taps in either the IIR or FIR filter.
+ * I believe MLP actually specifies the maximum order for IIR filters as four,
+ * and that the sum of the orders of both filters must be <= 8. */
+#define MAX_FILTER_ORDER 8
+
+/** Number of bits used for VLC lookup - longest huffman code is 9. */
+#define VLC_BITS 9
+
+
+static const char* sample_message =
+ "Please file a bug report following the instructions at "
+ "http://ffmpeg.mplayerhq.hu/bugreports.html and include "
+ "a sample of this file.";
+
+typedef struct SubStream {
+ //! Set if a valid restart header has been read. Otherwise the substream can not be decoded.
+ uint8_t restart_seen;
+
+ //@{
+ /** restart header data */
+ //! The type of noise to be used in the rematrix stage.
+ uint16_t noise_type;
+
+ //! The index of the first channel coded in this substream.
+ uint8_t min_channel;
+ //! The index of the last channel coded in this substream.
+ uint8_t max_channel;
+ //! The number of channels input into the rematrix stage.
+ uint8_t max_matrix_channel;
+
+ //! The left shift applied to random noise in 0x31ea substreams.
+ uint8_t noise_shift;
+ //! The current seed value for the pseudorandom noise generator(s).
+ uint32_t noisegen_seed;
+
+ //! Set if the substream contains extra info to check the size of VLC blocks.
+ uint8_t data_check_present;
+
+ //! Bitmask of which parameter sets are conveyed in a decoding parameter block.
+ uint8_t param_presence_flags;
+#define PARAM_BLOCKSIZE (1 << 7)
+#define PARAM_MATRIX (1 << 6)
+#define PARAM_OUTSHIFT (1 << 5)
+#define PARAM_QUANTSTEP (1 << 4)
+#define PARAM_FIR (1 << 3)
+#define PARAM_IIR (1 << 2)
+#define PARAM_HUFFOFFSET (1 << 1)
+ //@}
+
+ //@{
+ /** matrix data */
+
+ //! Number of matrices to be applied.
+ uint8_t num_primitive_matrices;
+
+ //! matrix output channel
+ uint8_t matrix_out_ch[MAX_MATRICES];
+
+ //! 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+2];
+ //! Left shift to apply to noise values in 0x31eb substreams.
+ uint8_t matrix_noise_shift[MAX_MATRICES];
+ //@}
+
+ //! Left shift to apply to huffman-decoded residuals.
+ uint8_t quant_step_size[MAX_CHANNELS];
+
+ //! Number of PCM samples in current audio block.
+ uint16_t blocksize;
+ //! Number of PCM samples decoded so far in this frame.
+ uint16_t blockpos;
+
+ //! Left shift to apply to decoded PCM values to get final 24-bit output.
+ int8_t output_shift[MAX_CHANNELS];
+
+ //! Running XOR of all output samples.
+ int32_t lossless_check_data;
+
+} SubStream;
+
+typedef struct MLPDecodeContext {
+ AVCodecContext *avctx;
+
+ //! Set if a valid major sync block has been read. Otherwise no decoding is possible.
+ uint8_t params_valid;
+
+ //! Number of substreams contained within this stream.
+ uint8_t num_substreams;
+
+ //! Index of the last substream to decode - further substreams are skipped.
+ uint8_t max_decoded_substream;
+
+ //! Number of PCM samples contained in each frame.
+ int access_unit_size;
+ //! Next power of two above the number of samples in each frame.
+ int access_unit_size_pow2;
+
+ SubStream substream[MAX_SUBSTREAMS];
+
+ //@{
+ /** filter data */
+#define FIR 0
+#define IIR 1
+ //! Number of taps in filter.
+ uint8_t filter_order[MAX_CHANNELS][NUM_FILTERS];
+ //! Right shift to apply to output of filter.
+ uint8_t filter_shift[MAX_CHANNELS][NUM_FILTERS];
+
+ int32_t filter_coeff[MAX_CHANNELS][NUM_FILTERS][MAX_FILTER_ORDER];
+ int32_t filter_state[MAX_CHANNELS][NUM_FILTERS][MAX_FILTER_ORDER];
+ //@}
+
+ //@{
+ /** sample data coding infomation */
+ //! Offset to apply to residual values.
+ int16_t huff_offset[MAX_CHANNELS];
+ //! Sign/rounding corrected version of huff_offset.
+ int32_t sign_huff_offset[MAX_CHANNELS];
+ //! Which VLC codebook to use to read residuals.
+ uint8_t codebook[MAX_CHANNELS];
+ //! Size of residual suffix not encoded using VLC.
+ uint8_t huff_lsbs[MAX_CHANNELS];
+ //@}
+
+ int8_t noise_buffer[MAX_BLOCKSIZE_POW2];
+ int8_t bypassed_lsbs[MAX_BLOCKSIZE][MAX_CHANNELS];
+ int32_t sample_buffer[MAX_BLOCKSIZE][MAX_CHANNELS+2];
+} MLPDecodeContext;
+
+/** Tables defining the huffman codes.
+ * There are three entropy coding methods used in MLP (four if you count
+ * "none" as a method). These use the same sequences for codes starting with
+ * 00 or 01, but have different codes starting with 1. */
+
+static const uint8_t huffman_tables[3][18][2] = {
+ { /* huffman table 0, -7 - +10 */
+ {0x01, 9}, {0x01, 8}, {0x01, 7}, {0x01, 6}, {0x01, 5}, {0x01, 4}, {0x01, 3},
+ {0x04, 3}, {0x05, 3}, {0x06, 3}, {0x07, 3},
+ {0x03, 3}, {0x05, 4}, {0x09, 5}, {0x11, 6}, {0x21, 7}, {0x41, 8}, {0x81, 9},
+ }, { /* huffman table 1, -7 - +8 */
+ {0x01, 9}, {0x01, 8}, {0x01, 7}, {0x01, 6}, {0x01, 5}, {0x01, 4}, {0x01, 3},
+ {0x02, 2}, {0x03, 2},
+ {0x03, 3}, {0x05, 4}, {0x09, 5}, {0x11, 6}, {0x21, 7}, {0x41, 8}, {0x81, 9},
+ }, { /* huffman table 2, -7 - +7 */
+ {0x01, 9}, {0x01, 8}, {0x01, 7}, {0x01, 6}, {0x01, 5}, {0x01, 4}, {0x01, 3},
+ {0x01, 1},
+ {0x03, 3}, {0x05, 4}, {0x09, 5}, {0x11, 6}, {0x21, 7}, {0x41, 8}, {0x81, 9},
+ }
+};
+
+static VLC huff_vlc[3];
+
+static int crc_init = 0;
+static AVCRC crc_63[1024];
+static AVCRC crc_1D[1024];
+
+
+/** Initialize static data, constant between all invocations of the codec. */
+
+static av_cold void init_static()
+{
+ INIT_VLC_STATIC(&huff_vlc[0], VLC_BITS, 18,
+ &huffman_tables[0][0][1], 2, 1,
+ &huffman_tables[0][0][0], 2, 1, 512);
+ INIT_VLC_STATIC(&huff_vlc[1], VLC_BITS, 16,
+ &huffman_tables[1][0][1], 2, 1,
+ &huffman_tables[1][0][0], 2, 1, 512);
+ INIT_VLC_STATIC(&huff_vlc[2], VLC_BITS, 15,
+ &huffman_tables[2][0][1], 2, 1,
+ &huffman_tables[2][0][0], 2, 1, 512);
+
+ if (!crc_init) {
+ av_crc_init(crc_63, 0, 8, 0x63, sizeof(crc_63));
+ av_crc_init(crc_1D, 0, 8, 0x1D, sizeof(crc_1D));
+ crc_init = 1;
+ }
+}
+
+
+/** MLP uses checksums that seem to be based on the standard CRC algorithm,
+ * but not (in implementation terms, the table lookup and XOR are reversed).
+ * We can implement this behavior using a standard av_crc on all but the
+ * last element, then XOR that with the last element. */
+
+static uint8_t mlp_checksum8(const uint8_t *buf, unsigned int buf_size)
+{
+ uint8_t checksum = av_crc(crc_63, 0x3c, buf, buf_size - 1); // crc_63[0xa2] == 0x3c
+ checksum ^= buf[buf_size-1];
+ return checksum;
+}
+
+/** Calculate an 8-bit checksum over a restart header -- a non-multiple-of-8
+ * number of bits, starting two bits into the first byte of buf. */
+
+static uint8_t mlp_restart_checksum(const uint8_t *buf, unsigned int bit_size)
+{
+ int i;
+ int num_bytes = (bit_size + 2) / 8;
+
+ int crc = crc_1D[buf[0] & 0x3f];
+ crc = av_crc(crc_1D, crc, buf + 1, num_bytes - 2);
+ crc ^= buf[num_bytes - 1];
+
+ for (i = 0; i < ((bit_size + 2) & 7); i++) {
+ crc <<= 1;
+ if (crc & 0x100)
+ crc ^= 0x11D;
+ crc ^= (buf[num_bytes] >> (7 - i)) & 1;
+ }
+
+ return crc;
+}
+
+static inline int32_t calculate_sign_huff(MLPDecodeContext *m,
+ unsigned int substr, unsigned int ch)
+{
+ SubStream *s = &m->substream[substr];
+ int lsb_bits = m->huff_lsbs[ch] - s->quant_step_size[ch];
+ int sign_shift = lsb_bits + (m->codebook[ch] ? 2 - m->codebook[ch] : -1);
+ int32_t sign_huff_offset = m->huff_offset[ch];
+
+ if (m->codebook[ch] > 0)
+ sign_huff_offset -= 7 << lsb_bits;
+
+ if (sign_shift >= 0)
+ sign_huff_offset -= 1 << sign_shift;
+
+ return sign_huff_offset;
+}
+
+/** Read a sample, consisting of either, both or neither of entropy-coded MSBs
+ * and plain LSBs. */
+
+static inline int read_huff_channels(MLPDecodeContext *m, GetBitContext *gbp,
+ unsigned int substr, unsigned int pos)
+{
+ SubStream *s = &m->substream[substr];
+ unsigned int mat, channel;
+
+ for (mat = 0; mat < s->num_primitive_matrices; mat++)
+ if (s->lsb_bypass[mat])
+ m->bypassed_lsbs[pos + s->blockpos][mat] = get_bits1(gbp);
+
+ for (channel = s->min_channel; channel <= s->max_channel; channel++) {
+ int codebook = m->codebook[channel];
+ int quant_step_size = s->quant_step_size[channel];
+ int lsb_bits = m->huff_lsbs[channel] - quant_step_size;
+ int result = 0;
+
+ if (codebook > 0)
+ result = get_vlc2(gbp, huff_vlc[codebook-1].table,
+ VLC_BITS, (9 + VLC_BITS - 1) / VLC_BITS);
+
+ if (result < 0)
+ return -1;
+
+ if (lsb_bits > 0)
+ result = (result << lsb_bits) + get_bits(gbp, lsb_bits);
+
+ result += m->sign_huff_offset[channel];
+ result <<= quant_step_size;
+
+ m->sample_buffer[pos + s->blockpos][channel] = result;
+ }
+
+ return 0;
+}
+
+static av_cold int mlp_decode_init(AVCodecContext *avctx)
+{
+ MLPDecodeContext *m = avctx->priv_data;
+ int substr;
+
+ init_static();
+ m->avctx = avctx;
+ for (substr = 0; substr < MAX_SUBSTREAMS; substr++)
+ m->substream[substr].lossless_check_data = 0xffffffff;
+ return 0;
+}
+
+/** Read a major sync info header - contains high level information about
+ * the stream - sample rate, channel arrangement etc. Most of this
+ * information is not actually necessary for decoding, only for playback.
+ */
+
+static int read_major_sync(MLPDecodeContext *m, GetBitContext *gb)
+{
+ MLPHeaderInfo mh;
+ int substr;
+
+ if (ff_mlp_read_major_sync(m->avctx, &mh, gb) != 0)
+ return -1;
+
+ if (mh.group1_bits == 0) {
+ av_log(m->avctx, AV_LOG_ERROR, "Invalid/unknown bits per sample\n");
+ return -1;
+ }
+ if (mh.group2_bits > mh.group1_bits) {
+ av_log(m->avctx, AV_LOG_ERROR,
+ "Channel group 2 cannot have more bits per sample than group 1\n");
+ return -1;
+ }
+
+ if (mh.group2_samplerate && mh.group2_samplerate != mh.group1_samplerate) {
+ av_log(m->avctx, AV_LOG_ERROR,
+ "Channel groups with differing sample rates not currently supported\n");
+ return -1;
+ }
+
+ if (mh.group1_samplerate == 0) {
+ av_log(m->avctx, AV_LOG_ERROR, "Invalid/unknown sampling rate\n");
+ return -1;
+ }
+ if (mh.group1_samplerate > MAX_SAMPLERATE) {
+ av_log(m->avctx, AV_LOG_ERROR,
+ "Sampling rate %d is greater than maximum supported (%d)\n",
+ mh.group1_samplerate, MAX_SAMPLERATE);
+ return -1;
+ }
+ if (mh.access_unit_size > MAX_BLOCKSIZE) {
+ av_log(m->avctx, AV_LOG_ERROR,
+ "Block size %d is greater than maximum supported (%d)\n",
+ mh.access_unit_size, MAX_BLOCKSIZE);
+ return -1;
+ }
+ if (mh.access_unit_size_pow2 > MAX_BLOCKSIZE_POW2) {
+ av_log(m->avctx, AV_LOG_ERROR,
+ "Block size pow2 %d is greater than maximum supported (%d)\n",
+ mh.access_unit_size_pow2, MAX_BLOCKSIZE_POW2);
+ return -1;
+ }
+
+ if (mh.num_substreams == 0)
+ return -1;
+ if (mh.num_substreams > MAX_SUBSTREAMS) {
+ av_log(m->avctx, AV_LOG_ERROR,
+ "Number of substreams %d is more than maximum supported by "
+ "decoder. %s\n", mh.num_substreams, sample_message);
+ return -1;
+ }
+
+ m->access_unit_size = mh.access_unit_size;
+ m->access_unit_size_pow2 = mh.access_unit_size_pow2;
+
+ m->num_substreams = mh.num_substreams;
+ m->max_decoded_substream = m->num_substreams - 1;
+
+ m->avctx->sample_rate = mh.group1_samplerate;
+ m->avctx->frame_size = mh.access_unit_size;
+
+#ifdef CONFIG_AUDIO_NONSHORT
+ m->avctx->bits_per_sample = mh.group1_bits;
+ if (mh.group1_bits > 16) {
+ m->avctx->sample_fmt = SAMPLE_FMT_S32;
+ }
+#endif
+
+ m->params_valid = 1;
+ for (substr = 0; substr < MAX_SUBSTREAMS; substr++)
+ m->substream[substr].restart_seen = 0;
+
+ return 0;
+}
+
+/** Read a restart header from a block in a substream. This contains parameters
+ * required to decode the audio that do not change very often. Generally
+ * (always) present only in blocks following a major sync. */
+
+static int read_restart_header(MLPDecodeContext *m, GetBitContext *gbp,
+ const uint8_t *buf, unsigned int substr)
+{
+ SubStream *s = &m->substream[substr];
+ unsigned int ch;
+ int sync_word, tmp;
+ uint8_t checksum;
+ uint8_t lossless_check;
+ int start_count = get_bits_count(gbp);
+
+ sync_word = get_bits(gbp, 13);
+
+ if (sync_word != 0x31ea >> 1) {
+ av_log(m->avctx, AV_LOG_ERROR,
+ "Restart header sync incorrect (got 0x%04x)\n", sync_word);
+ return -1;
+ }
+ s->noise_type = get_bits1(gbp);
+
+ skip_bits(gbp, 16); /* Output timestamp */
+
+ s->min_channel = get_bits(gbp, 4);
+ s->max_channel = get_bits(gbp, 4);
+ s->max_matrix_channel = get_bits(gbp, 4);
+
+ if (s->min_channel > s->max_channel) {
+ av_log(m->avctx, AV_LOG_ERROR,
+ "Substream min channel cannot be greater than max channel.\n");
+ return -1;
+ }
+
+ if (m->avctx->request_channels > 0
+ && s->max_channel + 1 >= m->avctx->request_channels
+ && substr < m->max_decoded_substream) {
+ av_log(m->avctx, AV_LOG_INFO,
+ "Extracting %d channel downmix from substream %d. "
+ "Further substreams will be skipped.\n",
+ s->max_channel + 1, substr);
+ m->max_decoded_substream = substr;
+ }
+
+ s->noise_shift = get_bits(gbp, 4);
+ s->noisegen_seed = get_bits(gbp, 23);
+
+ skip_bits(gbp, 19);
+
+ s->data_check_present = get_bits1(gbp);
+ lossless_check = get_bits(gbp, 8);
+ if (substr == m->max_decoded_substream
+ && s->lossless_check_data != 0xffffffff) {
+ tmp = s->lossless_check_data;
+ tmp ^= tmp >> 16;
+ tmp ^= tmp >> 8;
+ tmp &= 0xff;
+ if (tmp != lossless_check)
+ av_log(m->avctx, AV_LOG_WARNING,
+ "Lossless check failed - expected %02x, calculated %02x\n",
+ lossless_check, tmp);
+ else
+ dprintf(m->avctx, "Lossless check passed for substream %d (%x)\n",
+ substr, tmp);
+ }
+
+ skip_bits(gbp, 16);
+
+ for (ch = 0; ch <= s->max_matrix_channel; ch++) {
+ int ch_assign = get_bits(gbp, 6);
+ dprintf(m->avctx, "ch_assign[%d][%d] = %d\n", substr, ch,
+ ch_assign);
+ if (ch_assign != ch) {
+ av_log(m->avctx, AV_LOG_ERROR,
+ "Non 1:1 channel assignments are used in this stream. %s\n",
+ sample_message);
+ return -1;
+ }
+ }
+
+ checksum = mlp_restart_checksum(buf, get_bits_count(gbp) - start_count);
+
+ if (checksum != get_bits(gbp, 8))
+ av_log(m->avctx, AV_LOG_ERROR, "Restart header checksum error\n");
+
+ /* Set default decoding parameters */
+ s->param_presence_flags = 0xff;
+ s->num_primitive_matrices = 0;
+ s->blocksize = 8;
+ s->lossless_check_data = 0;
+
+ memset(s->output_shift , 0, sizeof(s->output_shift ));
+ memset(s->quant_step_size, 0, sizeof(s->quant_step_size));
+
+ for (ch = s->min_channel; ch <= s->max_channel; ch++) {
+ m->filter_order[ch][FIR] = 0;
+ m->filter_order[ch][IIR] = 0;
+ m->filter_shift[ch][FIR] = 0;
+ m->filter_shift[ch][IIR] = 0;
+
+ /* Default audio coding is 24-bit raw PCM */
+ m->huff_offset [ch] = 0;
+ m->sign_huff_offset[ch] = (-1) << 23;
+ m->codebook [ch] = 0;
+ m->huff_lsbs [ch] = 24;
+ }
+
+ if (substr == m->max_decoded_substream) {
+ m->avctx->channels = s->max_channel + 1;
+ }
+
+ return 0;
+}
+
+/** Read parameters for one of the prediction filters. */
+
+static int read_filter_params(MLPDecodeContext *m, GetBitContext *gbp,
+ unsigned int channel, unsigned int filter)
+{
+ const char fchar = filter ? 'I' : 'F';
+ int i, order;
+
+ // filter is 0 for FIR, 1 for IIR
+ assert(filter < 2);
+
+ order = get_bits(gbp, 4);
+ if (order > MAX_FILTER_ORDER) {
+ av_log(m->avctx, AV_LOG_ERROR,
+ "%cIR filter order %d is greater than maximum %d\n",
+ fchar, order, MAX_FILTER_ORDER);
+ return -1;
+ }
+ m->filter_order[channel][filter] = order;
+
+ if (order > 0) {
+ int coeff_bits, coeff_shift;
+
+ m->filter_shift[channel][filter] = get_bits(gbp, 4);
+
+ coeff_bits = get_bits(gbp, 5);
+ coeff_shift = get_bits(gbp, 3);
+ if (coeff_bits < 1 || coeff_bits > 16) {
+ av_log(m->avctx, AV_LOG_ERROR,
+ "%cIR filter coeff_bits must be between 1 and 16\n",
+ fchar);
+ return -1;
+ }
+ if (coeff_bits + coeff_shift > 16) {
+ av_log(m->avctx, AV_LOG_ERROR,
+ "Sum of coeff_bits and coeff_shift for %cIR filter must be 16 or less\n",
+ fchar);
+ return -1;
+ }
+
+ for (i = 0; i < order; i++)
+ m->filter_coeff[channel][filter][i] =
+ get_sbits(gbp, coeff_bits) << coeff_shift;
+
+ if (get_bits1(gbp)) {
+ int state_bits, state_shift;
+
+ if (filter == FIR) {
+ av_log(m->avctx, AV_LOG_ERROR,
+ "FIR filter has state data specified\n");
+ return -1;
+ }
+
+ state_bits = get_bits(gbp, 4);
+ state_shift = get_bits(gbp, 4);
+
+ /* TODO: check validity of state data */
+
+ for (i = 0; i < order; i++)
+ m->filter_state[channel][filter][i] =
+ get_sbits(gbp, state_bits) << state_shift;
+ }
+ }
+
+ return 0;
+}
+
+/** Read decoding parameters that change more often than those in the restart
+ * header. */
+
+static int read_decoding_params(MLPDecodeContext *m, GetBitContext *gbp,
+ unsigned int substr)
+{
+ SubStream *s = &m->substream[substr];
+ unsigned int mat, ch;
+
+ if (get_bits1(gbp))
+ s->param_presence_flags = get_bits(gbp, 8);
+
+ if (s->param_presence_flags & PARAM_BLOCKSIZE)
+ if (get_bits1(gbp)) {
+ s->blocksize = get_bits(gbp, 9);
+ if (s->blocksize > MAX_BLOCKSIZE) {
+ av_log(m->avctx, AV_LOG_ERROR, "Block size too large\n");
+ s->blocksize = 0;
+ return -1;
+ }
+ }
+
+ if (s->param_presence_flags & PARAM_MATRIX)
+ if (get_bits1(gbp)) {
+ s->num_primitive_matrices = get_bits(gbp, 4);
+
+ for (mat = 0; mat < s->num_primitive_matrices; mat++) {
+ int frac_bits, max_chan;
+ s->matrix_out_ch[mat] = get_bits(gbp, 4);
+ frac_bits = get_bits(gbp, 4);
+ s->lsb_bypass [mat] = get_bits1(gbp);
+
+ if (s->matrix_out_ch[mat] > s->max_channel) {
+ av_log(m->avctx, AV_LOG_ERROR,
+ "Invalid channel %d specified as output from matrix\n",
+ s->matrix_out_ch[mat]);
+ return -1;
+ }
+ if (frac_bits > 14) {
+ av_log(m->avctx, AV_LOG_ERROR,
+ "Too many fractional bits specified\n");
+ return -1;
+ }
+
+ max_chan = s->max_matrix_channel;
+ if (!s->noise_type)
+ max_chan+=2;
+
+ for (ch = 0; ch <= max_chan; ch++) {
+ int coeff_val = 0;
+ if (get_bits1(gbp))
+ coeff_val = get_sbits(gbp, frac_bits + 2);
+
+ s->matrix_coeff[mat][ch] = coeff_val << (14 - frac_bits);
+ }
+
+ if (s->noise_type)
+ s->matrix_noise_shift[mat] = get_bits(gbp, 4);
+ else
+ s->matrix_noise_shift[mat] = 0;
+ }
+ }
+
+ if (s->param_presence_flags & PARAM_OUTSHIFT)
+ if (get_bits1(gbp))
+ for (ch = 0; ch <= s->max_matrix_channel; ch++) {
+ s->output_shift[ch] = get_bits(gbp, 4);
+ dprintf(m->avctx, "output shift[%d] = %d\n",
+ ch, s->output_shift[ch]);
+ /* TODO: validate */
+ }
+
+ if (s->param_presence_flags & PARAM_QUANTSTEP)
+ if (get_bits1(gbp))
+ for (ch = 0; ch <= s->max_channel; ch++) {
+ s->quant_step_size[ch] = get_bits(gbp, 4);
+ /* TODO: validate */
+
+ m->sign_huff_offset[ch] = calculate_sign_huff(m, substr, ch);
+ }
+
+ for (ch = s->min_channel; ch <= s->max_channel; ch++)
+ if (get_bits1(gbp)) {
+ if (s->param_presence_flags & PARAM_FIR)
+ if (get_bits1(gbp))
+ if (read_filter_params(m, gbp, ch, FIR) < 0)
+ return -1;
+
+ if (s->param_presence_flags & PARAM_IIR)
+ if (get_bits1(gbp))
+ if (read_filter_params(m, gbp, ch, IIR) < 0)
+ return -1;
+
+ if (m->filter_order[ch][FIR] && m->filter_order[ch][IIR] &&
+ m->filter_shift[ch][FIR] != m->filter_shift[ch][IIR]) {
+ av_log(m->avctx, AV_LOG_ERROR,
+ "FIR and IIR filters must use same precision\n");
+ return -1;
+ }
+ /* The FIR and IIR filters must have the same precision.
+ * To simplify the filtering code, only the precision of the
+ * FIR filter is considered. If only the IIR filter is employed,
+ * the FIR filter precision is set to that of the IIR filter, so
+ * that the filtering code can use it. */
+ if (!m->filter_order[ch][FIR] && m->filter_order[ch][IIR])
+ m->filter_shift[ch][FIR] = m->filter_shift[ch][IIR];
+
+ if (s->param_presence_flags & PARAM_HUFFOFFSET)
+ if (get_bits1(gbp))
+ m->huff_offset[ch] = get_sbits(gbp, 15);
+
+ m->codebook [ch] = get_bits(gbp, 2);
+ m->huff_lsbs[ch] = get_bits(gbp, 5);
+
+ m->sign_huff_offset[ch] = calculate_sign_huff(m, substr, ch);
+
+ /* TODO: validate */
+ }
+
+ return 0;
+}
+
+#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. */
+
+static void filter_channel(MLPDecodeContext *m, unsigned int substr,
+ unsigned int channel)
+{
+ SubStream *s = &m->substream[substr];
+ int32_t filter_state_buffer[NUM_FILTERS][MAX_BLOCKSIZE + MAX_FILTER_ORDER];
+ unsigned int filter_shift = m->filter_shift[channel][FIR];
+ int32_t mask = MSB_MASK(s->quant_step_size[channel]);
+ int index = MAX_BLOCKSIZE;
+ int j, i;
+
+ for (j = 0; j < NUM_FILTERS; j++) {
+ memcpy(& filter_state_buffer [j][MAX_BLOCKSIZE],
+ &m->filter_state[channel][j][0],
+ MAX_FILTER_ORDER * sizeof(int32_t));
+ }
+
+ for (i = 0; i < s->blocksize; i++) {
+ int32_t residual = m->sample_buffer[i + s->blockpos][channel];
+ unsigned int order;
+ int64_t accum = 0;
+ int32_t result;
+
+ /* TODO: Move this code to DSPContext? */
+
+ for (j = 0; j < NUM_FILTERS; j++)
+ for (order = 0; order < m->filter_order[channel][j]; order++)
+ accum += (int64_t)filter_state_buffer[j][index + order] *
+ m->filter_coeff[channel][j][order];
+
+ accum = accum >> filter_shift;
+ result = (accum + residual) & mask;
+
+ --index;
+
+ filter_state_buffer[FIR][index] = result;
+ filter_state_buffer[IIR][index] = result - accum;
+
+ m->sample_buffer[i + s->blockpos][channel] = result;
+ }
+
+ for (j = 0; j < NUM_FILTERS; j++) {
+ memcpy(&m->filter_state[channel][j][0],
+ & filter_state_buffer [j][index],
+ MAX_FILTER_ORDER * sizeof(int32_t));
+ }
+}
+
+/** Read a block of PCM residual data (or actual if no filtering active). */
+
+static int read_block_data(MLPDecodeContext *m, GetBitContext *gbp,
+ unsigned int substr)
+{
+ SubStream *s = &m->substream[substr];
+ unsigned int i, ch, expected_stream_pos = 0;
+
+ if (s->data_check_present) {
+ expected_stream_pos = get_bits_count(gbp);
+ expected_stream_pos += get_bits(gbp, 16);
+ av_log(m->avctx, AV_LOG_WARNING, "This file contains some features "
+ "we have not tested yet. %s\n", sample_message);
+ }
+
+ if (s->blockpos + s->blocksize > m->access_unit_size) {
+ av_log(m->avctx, AV_LOG_ERROR, "Too many audio samples in frame\n");
+ return -1;
+ }
+
+ memset(&m->bypassed_lsbs[s->blockpos][0], 0,
+ s->blocksize * sizeof(m->bypassed_lsbs[0]));
+
+ for (i = 0; i < s->blocksize; i++) {
+ if (read_huff_channels(m, gbp, substr, i) < 0)
+ return -1;
+ }
+
+ for (ch = s->min_channel; ch <= s->max_channel; ch++) {
+ filter_channel(m, substr, ch);
+ }
+
+ s->blockpos += s->blocksize;
+
+ if (s->data_check_present) {
+ if (get_bits_count(gbp) != expected_stream_pos)
+ av_log(m->avctx, AV_LOG_ERROR, "Block data length mismatch\n");
+ skip_bits(gbp, 8);
+ }
+
+ return 0;
+}
+
+/** Data table used for TrueHD noise generation function */
+
+static const int8_t noise_table[256] = {
+ 30, 51, 22, 54, 3, 7, -4, 38, 14, 55, 46, 81, 22, 58, -3, 2,
+ 52, 31, -7, 51, 15, 44, 74, 30, 85, -17, 10, 33, 18, 80, 28, 62,
+ 10, 32, 23, 69, 72, 26, 35, 17, 73, 60, 8, 56, 2, 6, -2, -5,
+ 51, 4, 11, 50, 66, 76, 21, 44, 33, 47, 1, 26, 64, 48, 57, 40,
+ 38, 16, -10, -28, 92, 22, -18, 29, -10, 5, -13, 49, 19, 24, 70, 34,
+ 61, 48, 30, 14, -6, 25, 58, 33, 42, 60, 67, 17, 54, 17, 22, 30,
+ 67, 44, -9, 50, -11, 43, 40, 32, 59, 82, 13, 49, -14, 55, 60, 36,
+ 48, 49, 31, 47, 15, 12, 4, 65, 1, 23, 29, 39, 45, -2, 84, 69,
+ 0, 72, 37, 57, 27, 41, -15, -16, 35, 31, 14, 61, 24, 0, 27, 24,
+ 16, 41, 55, 34, 53, 9, 56, 12, 25, 29, 53, 5, 20, -20, -8, 20,
+ 13, 28, -3, 78, 38, 16, 11, 62, 46, 29, 21, 24, 46, 65, 43, -23,
+ 89, 18, 74, 21, 38, -12, 19, 12, -19, 8, 15, 33, 4, 57, 9, -8,
+ 36, 35, 26, 28, 7, 83, 63, 79, 75, 11, 3, 87, 37, 47, 34, 40,
+ 39, 19, 20, 42, 27, 34, 39, 77, 13, 42, 59, 64, 45, -1, 32, 37,
+ 45, -5, 53, -6, 7, 36, 50, 23, 6, 32, 9, -21, 18, 71, 27, 52,
+ -25, 31, 35, 42, -1, 68, 63, 52, 26, 43, 66, 37, 41, 25, 40, 70,
+};
+
+/** Noise generation functions.
+ * I'm not sure what these are for - they seem to be some kind of pseudorandom
+ * sequence generators, used to generate noise data which is used when the
+ * channels are rematrixed. I'm not sure if they provide a practical benefit
+ * to compression, or just obfuscate the decoder. Are they for some kind of
+ * dithering? */
+
+/** Generate two channels of noise, used in the matrix when
+ * restart sync word == 0x31ea. */
+
+static void generate_2_noise_channels(MLPDecodeContext *m, unsigned int substr)
+{
+ SubStream *s = &m->substream[substr];
+ unsigned int i;
+ uint32_t seed = s->noisegen_seed;
+ unsigned int maxchan = s->max_matrix_channel;
+
+ for (i = 0; i < s->blockpos; i++) {
+ uint16_t seed_shr7 = seed >> 7;
+ m->sample_buffer[i][maxchan+1] = ((int8_t)(seed >> 15)) << s->noise_shift;
+ m->sample_buffer[i][maxchan+2] = ((int8_t) seed_shr7) << s->noise_shift;
+
+ seed = (seed << 16) ^ seed_shr7 ^ (seed_shr7 << 5);
+ }
+
+ s->noisegen_seed = seed;
+}
+
+/** Generate a block of noise, used when restart sync word == 0x31eb. */
+
+static void fill_noise_buffer(MLPDecodeContext *m, unsigned int substr)
+{
+ SubStream *s = &m->substream[substr];
+ unsigned int i;
+ uint32_t seed = s->noisegen_seed;
+
+ for (i = 0; i < m->access_unit_size_pow2; i++) {
+ uint8_t seed_shr15 = seed >> 15;
+ m->noise_buffer[i] = noise_table[seed_shr15];
+ seed = (seed << 8) ^ seed_shr15 ^ (seed_shr15 << 5);
+ }
+
+ s->noisegen_seed = seed;
+}
+
+
+/** Apply the channel matrices in turn to reconstruct the original audio
+ * samples. */
+
+static void rematrix_channels(MLPDecodeContext *m, unsigned int substr)
+{
+ SubStream *s = &m->substream[substr];
+ unsigned int mat, src_ch, i;
+ unsigned int maxchan;
+
+ maxchan = s->max_matrix_channel;
+ if (!s->noise_type) {
+ generate_2_noise_channels(m, substr);
+ maxchan += 2;
+ } else {
+ fill_noise_buffer(m, substr);
+ }
+
+ for (mat = 0; mat < s->num_primitive_matrices; mat++) {
+ int matrix_noise_shift = s->matrix_noise_shift[mat];
+ unsigned int dest_ch = s->matrix_out_ch[mat];
+ int32_t mask = MSB_MASK(s->quant_step_size[dest_ch]);
+
+ /* TODO: DSPContext? */
+
+ for (i = 0; i < s->blockpos; i++) {
+ int64_t accum = 0;
+ for (src_ch = 0; src_ch <= maxchan; src_ch++) {
+ accum += (int64_t)m->sample_buffer[i][src_ch]
+ * s->matrix_coeff[mat][src_ch];
+ }
+ if (matrix_noise_shift) {
+ uint32_t index = s->num_primitive_matrices - mat;
+ index = (i * (index * 2 + 1) + index) & (m->access_unit_size_pow2 - 1);
+ accum += m->noise_buffer[index] << (matrix_noise_shift + 7);
+ }
+ m->sample_buffer[i][dest_ch] = ((accum >> 14) & mask)
+ + m->bypassed_lsbs[i][mat];
+ }
+ }
+}
+
+/** Write the audio data into the output buffer. */
+
+static int output_data_internal(MLPDecodeContext *m, unsigned int substr,
+ uint8_t *data, unsigned int *data_size, int is32)
+{
+ SubStream *s = &m->substream[substr];
+ unsigned int i, ch = 0;
+ int32_t *data_32 = (int32_t*) data;
+ int16_t *data_16 = (int16_t*) data;
+
+ if (*data_size < (s->max_channel + 1) * s->blockpos * (is32 ? 4 : 2))
+ return -1;
+
+ for (i = 0; i < s->blockpos; i++) {
+ for (ch = 0; ch <= s->max_channel; ch++) {
+ int32_t sample = m->sample_buffer[i][ch] << s->output_shift[ch];
+ s->lossless_check_data ^= (sample & 0xffffff) << ch;
+ if (is32) *data_32++ = sample << 8;
+ else *data_16++ = sample >> 8;
+ }
+ }
+
+ *data_size = i * ch * (is32 ? 4 : 2);
+
+ return 0;
+}
+
+static int output_data(MLPDecodeContext *m, unsigned int substr,
+ uint8_t *data, unsigned int *data_size)
+{
+ if (m->avctx->sample_fmt == SAMPLE_FMT_S32)
+ return output_data_internal(m, substr, data, data_size, 1);
+ else
+ return output_data_internal(m, substr, data, data_size, 0);
+}
+
+
+/** XOR together all the bytes of a buffer.
+ * Does this belong in dspcontext? */
+
+static uint8_t calculate_parity(const uint8_t *buf, unsigned int buf_size)
+{
+ uint32_t scratch = 0;
+ const uint8_t *buf_end = buf + buf_size;
+
+ for (; buf < buf_end - 3; buf += 4)
+ scratch ^= *((const uint32_t*)buf);
+
+ scratch ^= scratch >> 16;
+ scratch ^= scratch >> 8;
+
+ for (; buf < buf_end; buf++)
+ scratch ^= *buf;
+
+ return scratch;
+}
+
+/** Read an access unit from the stream.
+ * Returns < 0 on error, 0 if not enough data is present in the input stream
+ * otherwise returns the number of bytes consumed. */
+
+static int read_access_unit(AVCodecContext *avctx, void* data, int *data_size,
+ const uint8_t *buf, int buf_size)
+{
+ MLPDecodeContext *m = avctx->priv_data;
+ GetBitContext gb;
+ unsigned int length, substr;
+ unsigned int substream_start;
+ unsigned int header_size = 4;
+ unsigned int substr_header_size = 0;
+ uint8_t substream_parity_present[MAX_SUBSTREAMS];
+ uint16_t substream_data_len[MAX_SUBSTREAMS];
+ uint8_t parity_bits;
+
+ if (buf_size < 4)
+ return 0;
+
+ length = (AV_RB16(buf) & 0xfff) * 2;
+
+ if (length > buf_size)
+ return -1;
+
+ init_get_bits(&gb, (buf + 4), (length - 4) * 8);
+
+ if (show_bits_long(&gb, 31) == (0xf8726fba >> 1)) {
+ dprintf(m->avctx, "Found major sync\n");
+ if (read_major_sync(m, &gb) < 0)
+ goto error;
+ header_size += 28;
+ }
+
+ if (!m->params_valid) {
+ av_log(m->avctx, AV_LOG_WARNING,
+ "Stream parameters not seen; skipping frame\n");
+ *data_size = 0;
+ return length;
+ }
+
+ substream_start = 0;
+
+ for (substr = 0; substr < m->num_substreams; substr++) {
+ int extraword_present, checkdata_present, end;
+
+ extraword_present = get_bits1(&gb);
+ skip_bits1(&gb);
+ checkdata_present = get_bits1(&gb);
+ skip_bits1(&gb);
+
+ end = get_bits(&gb, 12) * 2;
+
+ substr_header_size += 2;
+
+ if (extraword_present) {
+ skip_bits(&gb, 16);
+ substr_header_size += 2;
+ }
+
+ if (end + header_size + substr_header_size > length) {
+ av_log(m->avctx, AV_LOG_ERROR,
+ "Indicated length of substream %d data goes off end of "
+ "packet.\n", substr);
+ end = length - header_size - substr_header_size;
+ }
+
+ if (end < substream_start) {
+ av_log(avctx, AV_LOG_ERROR,
+ "Indicated end offset of substream %d data "
+ "is smaller than calculated start offset.\n",
+ substr);
+ goto error;
+ }
+
+ if (substr > m->max_decoded_substream)
+ continue;
+
+ substream_parity_present[substr] = checkdata_present;
+ substream_data_len[substr] = end - substream_start;
+ substream_start = end;
+ }
+
+ parity_bits = calculate_parity(buf, 4);
+ parity_bits ^= calculate_parity(buf + header_size, substr_header_size);
+
+ if ((((parity_bits >> 4) ^ parity_bits) & 0xF) != 0xF) {
+ av_log(avctx, AV_LOG_ERROR, "Parity check failed.\n");
+ goto error;
+ }
+
+ buf += header_size + substr_header_size;
+
+ for (substr = 0; substr <= m->max_decoded_substream; substr++) {
+ SubStream *s = &m->substream[substr];
+ init_get_bits(&gb, buf, substream_data_len[substr] * 8);
+
+ s->blockpos = 0;
+ do {
+ if (get_bits1(&gb)) {
+ if (get_bits1(&gb)) {
+ /* A restart header should be present */
+ if (read_restart_header(m, &gb, buf, substr) < 0)
+ goto next_substr;
+ s->restart_seen = 1;
+ }
+
+ if (!s->restart_seen) {
+ av_log(m->avctx, AV_LOG_ERROR,
+ "No restart header present in substream %d.\n",
+ substr);
+ goto next_substr;
+ }
+
+ if (read_decoding_params(m, &gb, substr) < 0)
+ goto next_substr;
+ }
+
+ if (!s->restart_seen) {
+ av_log(m->avctx, AV_LOG_ERROR,
+ "No restart header present in substream %d.\n",
+ substr);
+ goto next_substr;
+ }
+
+ if (read_block_data(m, &gb, substr) < 0)
+ return -1;
+
+ } while ((get_bits_count(&gb) < substream_data_len[substr] * 8)
+ && get_bits1(&gb) == 0);
+
+ skip_bits(&gb, (-get_bits_count(&gb)) & 15);
+ if (substream_data_len[substr] * 8 - get_bits_count(&gb) >= 48 &&
+ (show_bits_long(&gb, 32) == 0xd234d234 ||
+ show_bits_long(&gb, 20) == 0xd234e)) {
+ skip_bits(&gb, 18);
+ if (substr == m->max_decoded_substream)
+ av_log(m->avctx, AV_LOG_INFO, "End of stream indicated\n");
+
+ if (get_bits1(&gb)) {
+ int shorten_by = get_bits(&gb, 13);
+ shorten_by = FFMIN(shorten_by, s->blockpos);
+ s->blockpos -= shorten_by;
+ } else
+ skip_bits(&gb, 13);
+ }
+ if (substream_parity_present[substr]) {
+ uint8_t parity, checksum;
+
+ parity = calculate_parity(buf, substream_data_len[substr] - 2);
+ if ((parity ^ get_bits(&gb, 8)) != 0xa9)
+ av_log(m->avctx, AV_LOG_ERROR,
+ "Substream %d parity check failed\n", substr);
+
+ checksum = mlp_checksum8(buf, substream_data_len[substr] - 2);
+ if (checksum != get_bits(&gb, 8))
+ av_log(m->avctx, AV_LOG_ERROR, "Substream %d checksum failed\n",
+ substr);
+ }
+ if (substream_data_len[substr] * 8 != get_bits_count(&gb)) {
+ av_log(m->avctx, AV_LOG_ERROR, "Substream %d length mismatch.\n",
+ substr);
+ return -1;
+ }
+
+next_substr:
+ buf += substream_data_len[substr];
+ }
+
+ rematrix_channels(m, m->max_decoded_substream);
+
+ if (output_data(m, m->max_decoded_substream, data, data_size) < 0)
+ return -1;
+
+ return length;
+
+error:
+ m->params_valid = 0;
+ return -1;
+}
+
+AVCodec mlp_decoder = {
+ "mlp",
+ CODEC_TYPE_AUDIO,
+ CODEC_ID_MLP,
+ sizeof(MLPDecodeContext),
+ mlp_decode_init,
+ NULL,
+ NULL,
+ read_access_unit,
+ .long_name = NULL_IF_CONFIG_SMALL("Meridian Lossless Packing"),
+};
+
generated by cgit v1.2.3 (git 2.39.1) at 2024-06-12 16:36:23 +0000