mirror of
https://github.com/FFmpeg/FFmpeg.git
synced 2024-12-18 03:19:31 +02:00
d780511b94
message is pointless to ordinary users and is printed once per frame. Originally committed as revision 21486 to svn://svn.ffmpeg.org/ffmpeg/trunk
1162 lines
38 KiB
C
1162 lines
38 KiB
C
/*
|
|
* 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 libavcodec/mlpdec.c
|
|
* MLP decoder
|
|
*/
|
|
|
|
#include <stdint.h>
|
|
|
|
#include "avcodec.h"
|
|
#include "dsputil.h"
|
|
#include "libavutil/intreadwrite.h"
|
|
#include "get_bits.h"
|
|
#include "libavutil/crc.h"
|
|
#include "parser.h"
|
|
#include "mlp_parser.h"
|
|
#include "mlp.h"
|
|
|
|
/** 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.org/bugreports.html and include "
|
|
"a sample of this file.";
|
|
|
|
typedef struct SubStream {
|
|
//! Set if a valid restart header has been read. Otherwise the substream cannot 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;
|
|
//! For each channel output by the matrix, the output channel to map it to
|
|
uint8_t ch_assign[MAX_CHANNELS];
|
|
|
|
//! 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)
|
|
#define PARAM_PRESENCE (1 << 0)
|
|
//@}
|
|
|
|
//@{
|
|
/** 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];
|
|
//! 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;
|
|
|
|
//! Current access unit being read has a major sync.
|
|
int is_major_sync_unit;
|
|
|
|
//! 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];
|
|
|
|
ChannelParams channel_params[MAX_CHANNELS];
|
|
|
|
int matrix_changed;
|
|
int filter_changed[MAX_CHANNELS][NUM_FILTERS];
|
|
|
|
int8_t noise_buffer[MAX_BLOCKSIZE_POW2];
|
|
int8_t bypassed_lsbs[MAX_BLOCKSIZE][MAX_CHANNELS];
|
|
int32_t sample_buffer[MAX_BLOCKSIZE][MAX_CHANNELS];
|
|
|
|
DSPContext dsp;
|
|
} MLPDecodeContext;
|
|
|
|
static VLC huff_vlc[3];
|
|
|
|
/** Initialize static data, constant between all invocations of the codec. */
|
|
|
|
static av_cold void init_static(void)
|
|
{
|
|
if (!huff_vlc[0].bits) {
|
|
INIT_VLC_STATIC(&huff_vlc[0], VLC_BITS, 18,
|
|
&ff_mlp_huffman_tables[0][0][1], 2, 1,
|
|
&ff_mlp_huffman_tables[0][0][0], 2, 1, 512);
|
|
INIT_VLC_STATIC(&huff_vlc[1], VLC_BITS, 16,
|
|
&ff_mlp_huffman_tables[1][0][1], 2, 1,
|
|
&ff_mlp_huffman_tables[1][0][0], 2, 1, 512);
|
|
INIT_VLC_STATIC(&huff_vlc[2], VLC_BITS, 15,
|
|
&ff_mlp_huffman_tables[2][0][1], 2, 1,
|
|
&ff_mlp_huffman_tables[2][0][0], 2, 1, 512);
|
|
}
|
|
|
|
ff_mlp_init_crc();
|
|
}
|
|
|
|
static inline int32_t calculate_sign_huff(MLPDecodeContext *m,
|
|
unsigned int substr, unsigned int ch)
|
|
{
|
|
ChannelParams *cp = &m->channel_params[ch];
|
|
SubStream *s = &m->substream[substr];
|
|
int lsb_bits = cp->huff_lsbs - s->quant_step_size[ch];
|
|
int sign_shift = lsb_bits + (cp->codebook ? 2 - cp->codebook : -1);
|
|
int32_t sign_huff_offset = cp->huff_offset;
|
|
|
|
if (cp->codebook > 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++) {
|
|
ChannelParams *cp = &m->channel_params[channel];
|
|
int codebook = cp->codebook;
|
|
int quant_step_size = s->quant_step_size[channel];
|
|
int lsb_bits = cp->huff_lsbs - 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 += cp->sign_huff_offset;
|
|
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;
|
|
dsputil_init(&m->dsp, avctx);
|
|
|
|
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 are 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 the supported maximum (%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 the supported maximum (%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 the supported maximum (%d).\n",
|
|
mh.access_unit_size_pow2, MAX_BLOCKSIZE_POW2);
|
|
return -1;
|
|
}
|
|
|
|
if (mh.num_substreams == 0)
|
|
return -1;
|
|
if (m->avctx->codec_id == CODEC_ID_MLP && mh.num_substreams > 2) {
|
|
av_log(m->avctx, AV_LOG_ERROR, "MLP only supports up to 2 substreams.\n");
|
|
return -1;
|
|
}
|
|
if (mh.num_substreams > MAX_SUBSTREAMS) {
|
|
av_log(m->avctx, AV_LOG_ERROR,
|
|
"Number of substreams %d is larger than the maximum supported "
|
|
"by the 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;
|
|
|
|
m->avctx->bits_per_raw_sample = mh.group1_bits;
|
|
if (mh.group1_bits > 16)
|
|
m->avctx->sample_fmt = SAMPLE_FMT_S32;
|
|
else
|
|
m->avctx->sample_fmt = SAMPLE_FMT_S16;
|
|
|
|
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);
|
|
const int max_matrix_channel = m->avctx->codec_id == CODEC_ID_MLP
|
|
? MAX_MATRIX_CHANNEL_MLP
|
|
: MAX_MATRIX_CHANNEL_TRUEHD;
|
|
|
|
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);
|
|
|
|
if (m->avctx->codec_id == CODEC_ID_MLP && s->noise_type) {
|
|
av_log(m->avctx, AV_LOG_ERROR, "MLP must have 0x31ea sync word.\n");
|
|
return -1;
|
|
}
|
|
|
|
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->max_matrix_channel > max_matrix_channel) {
|
|
av_log(m->avctx, AV_LOG_ERROR,
|
|
"Max matrix channel cannot be greater than %d.\n",
|
|
max_matrix_channel);
|
|
return -1;
|
|
}
|
|
|
|
if (s->max_channel != s->max_matrix_channel) {
|
|
av_log(m->avctx, AV_LOG_ERROR,
|
|
"Max channel must be equal max matrix channel.\n");
|
|
return -1;
|
|
}
|
|
|
|
/* 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) {
|
|
av_log(m->avctx, AV_LOG_ERROR,
|
|
"Number of channels %d is larger than the maximum supported "
|
|
"by the decoder. %s\n", s->max_channel+2, sample_message);
|
|
return -1;
|
|
}
|
|
|
|
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_DEBUG,
|
|
"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 = xor_32_to_8(s->lossless_check_data);
|
|
if (tmp != lossless_check)
|
|
av_log(m->avctx, AV_LOG_WARNING,
|
|
"Lossless check failed - expected %02x, calculated %02x.\n",
|
|
lossless_check, tmp);
|
|
}
|
|
|
|
skip_bits(gbp, 16);
|
|
|
|
memset(s->ch_assign, 0, sizeof(s->ch_assign));
|
|
|
|
for (ch = 0; ch <= s->max_matrix_channel; ch++) {
|
|
int ch_assign = get_bits(gbp, 6);
|
|
if (ch_assign > s->max_matrix_channel) {
|
|
av_log(m->avctx, AV_LOG_ERROR,
|
|
"Assignment of matrix channel %d to invalid output channel %d. %s\n",
|
|
ch, ch_assign, sample_message);
|
|
return -1;
|
|
}
|
|
s->ch_assign[ch_assign] = ch;
|
|
}
|
|
|
|
checksum = ff_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++) {
|
|
ChannelParams *cp = &m->channel_params[ch];
|
|
cp->filter_params[FIR].order = 0;
|
|
cp->filter_params[IIR].order = 0;
|
|
cp->filter_params[FIR].shift = 0;
|
|
cp->filter_params[IIR].shift = 0;
|
|
|
|
/* Default audio coding is 24-bit raw PCM. */
|
|
cp->huff_offset = 0;
|
|
cp->sign_huff_offset = (-1) << 23;
|
|
cp->codebook = 0;
|
|
cp->huff_lsbs = 24;
|
|
}
|
|
|
|
if (substr == m->max_decoded_substream)
|
|
m->avctx->channels = s->max_matrix_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)
|
|
{
|
|
FilterParams *fp = &m->channel_params[channel].filter_params[filter];
|
|
const int max_order = filter ? MAX_IIR_ORDER : MAX_FIR_ORDER;
|
|
const char fchar = filter ? 'I' : 'F';
|
|
int i, order;
|
|
|
|
// Filter is 0 for FIR, 1 for IIR.
|
|
assert(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");
|
|
return -1;
|
|
}
|
|
|
|
order = get_bits(gbp, 4);
|
|
if (order > max_order) {
|
|
av_log(m->avctx, AV_LOG_ERROR,
|
|
"%cIR filter order %d is greater than maximum %d.\n",
|
|
fchar, order, max_order);
|
|
return -1;
|
|
}
|
|
fp->order = order;
|
|
|
|
if (order > 0) {
|
|
int32_t *fcoeff = m->channel_params[channel].coeff[filter];
|
|
int coeff_bits, coeff_shift;
|
|
|
|
fp->shift = 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++)
|
|
fcoeff[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++)
|
|
fp->state[i] = get_sbits(gbp, state_bits) << state_shift;
|
|
}
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
/** Read parameters for primitive matrices. */
|
|
|
|
static int read_matrix_params(MLPDecodeContext *m, unsigned int substr, GetBitContext *gbp)
|
|
{
|
|
SubStream *s = &m->substream[substr];
|
|
unsigned int mat, ch;
|
|
const int max_primitive_matrices = m->avctx->codec_id == CODEC_ID_MLP
|
|
? MAX_MATRICES_MLP
|
|
: MAX_MATRICES_TRUEHD;
|
|
|
|
if (m->matrix_changed++ > 1) {
|
|
av_log(m->avctx, AV_LOG_ERROR, "Matrices may change only once per access unit.\n");
|
|
return -1;
|
|
}
|
|
|
|
s->num_primitive_matrices = get_bits(gbp, 4);
|
|
|
|
if (s->num_primitive_matrices > max_primitive_matrices) {
|
|
av_log(m->avctx, AV_LOG_ERROR,
|
|
"Number of primitive matrices cannot be greater than %d.\n",
|
|
max_primitive_matrices);
|
|
return -1;
|
|
}
|
|
|
|
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_matrix_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;
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
/** Read channel parameters. */
|
|
|
|
static int read_channel_params(MLPDecodeContext *m, unsigned int substr,
|
|
GetBitContext *gbp, unsigned int ch)
|
|
{
|
|
ChannelParams *cp = &m->channel_params[ch];
|
|
FilterParams *fir = &cp->filter_params[FIR];
|
|
FilterParams *iir = &cp->filter_params[IIR];
|
|
SubStream *s = &m->substream[substr];
|
|
|
|
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 (fir->order + iir->order > 8) {
|
|
av_log(m->avctx, AV_LOG_ERROR, "Total filter orders too high.\n");
|
|
return -1;
|
|
}
|
|
|
|
if (fir->order && iir->order &&
|
|
fir->shift != iir->shift) {
|
|
av_log(m->avctx, AV_LOG_ERROR,
|
|
"FIR and IIR filters must use the 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 (!fir->order && iir->order)
|
|
fir->shift = iir->shift;
|
|
|
|
if (s->param_presence_flags & PARAM_HUFFOFFSET)
|
|
if (get_bits1(gbp))
|
|
cp->huff_offset = get_sbits(gbp, 15);
|
|
|
|
cp->codebook = get_bits(gbp, 2);
|
|
cp->huff_lsbs = get_bits(gbp, 5);
|
|
|
|
if (cp->huff_lsbs > 24) {
|
|
av_log(m->avctx, AV_LOG_ERROR, "Invalid huff_lsbs.\n");
|
|
return -1;
|
|
}
|
|
|
|
cp->sign_huff_offset = calculate_sign_huff(m, substr, ch);
|
|
|
|
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 ch;
|
|
|
|
if (s->param_presence_flags & PARAM_PRESENCE)
|
|
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 < 8 || s->blocksize > m->access_unit_size) {
|
|
av_log(m->avctx, AV_LOG_ERROR, "Invalid blocksize.");
|
|
s->blocksize = 0;
|
|
return -1;
|
|
}
|
|
}
|
|
|
|
if (s->param_presence_flags & PARAM_MATRIX)
|
|
if (get_bits1(gbp))
|
|
if (read_matrix_params(m, substr, gbp) < 0)
|
|
return -1;
|
|
|
|
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_sbits(gbp, 4);
|
|
|
|
if (s->param_presence_flags & PARAM_QUANTSTEP)
|
|
if (get_bits1(gbp))
|
|
for (ch = 0; ch <= s->max_channel; ch++) {
|
|
ChannelParams *cp = &m->channel_params[ch];
|
|
|
|
s->quant_step_size[ch] = get_bits(gbp, 4);
|
|
|
|
cp->sign_huff_offset = calculate_sign_huff(m, substr, ch);
|
|
}
|
|
|
|
for (ch = s->min_channel; ch <= s->max_channel; ch++)
|
|
if (get_bits1(gbp))
|
|
if (read_channel_params(m, substr, gbp, ch) < 0)
|
|
return -1;
|
|
|
|
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];
|
|
const int32_t *fircoeff = m->channel_params[channel].coeff[FIR];
|
|
int32_t state_buffer[NUM_FILTERS][MAX_BLOCKSIZE + MAX_FIR_ORDER];
|
|
int32_t *firbuf = state_buffer[FIR] + MAX_BLOCKSIZE;
|
|
int32_t *iirbuf = state_buffer[IIR] + MAX_BLOCKSIZE;
|
|
FilterParams *fir = &m->channel_params[channel].filter_params[FIR];
|
|
FilterParams *iir = &m->channel_params[channel].filter_params[IIR];
|
|
unsigned int filter_shift = fir->shift;
|
|
int32_t mask = MSB_MASK(s->quant_step_size[channel]);
|
|
|
|
memcpy(firbuf, fir->state, MAX_FIR_ORDER * sizeof(int32_t));
|
|
memcpy(iirbuf, iir->state, MAX_IIR_ORDER * sizeof(int32_t));
|
|
|
|
m->dsp.mlp_filter_channel(firbuf, fircoeff,
|
|
fir->order, iir->order,
|
|
filter_shift, mask, s->blocksize,
|
|
&m->sample_buffer[s->blockpos][channel]);
|
|
|
|
memcpy(fir->state, firbuf - s->blocksize, MAX_FIR_ORDER * sizeof(int32_t));
|
|
memcpy(iir->state, iirbuf - s->blocksize, MAX_IIR_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]);
|
|
int32_t *coeffs = s->matrix_coeff[mat];
|
|
int index = s->num_primitive_matrices - mat;
|
|
int index2 = 2 * index + 1;
|
|
|
|
/* TODO: DSPContext? */
|
|
|
|
for (i = 0; i < s->blockpos; i++) {
|
|
int32_t bypassed_lsb = m->bypassed_lsbs[i][mat];
|
|
int32_t *samples = m->sample_buffer[i];
|
|
int64_t accum = 0;
|
|
|
|
for (src_ch = 0; src_ch <= maxchan; src_ch++)
|
|
accum += (int64_t) samples[src_ch] * coeffs[src_ch];
|
|
|
|
if (matrix_noise_shift) {
|
|
index &= m->access_unit_size_pow2 - 1;
|
|
accum += m->noise_buffer[index] << (matrix_noise_shift + 7);
|
|
index += index2;
|
|
}
|
|
|
|
samples[dest_ch] = ((accum >> 14) & mask) + bypassed_lsb;
|
|
}
|
|
}
|
|
}
|
|
|
|
/** 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, out_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 (out_ch = 0; out_ch <= s->max_matrix_channel; out_ch++) {
|
|
int mat_ch = s->ch_assign[out_ch];
|
|
int32_t sample = m->sample_buffer[i][mat_ch]
|
|
<< s->output_shift[mat_ch];
|
|
s->lossless_check_data ^= (sample & 0xffffff) << mat_ch;
|
|
if (is32) *data_32++ = sample << 8;
|
|
else *data_16++ = sample >> 8;
|
|
}
|
|
}
|
|
|
|
*data_size = i * out_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);
|
|
}
|
|
|
|
|
|
/** 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,
|
|
AVPacket *avpkt)
|
|
{
|
|
const uint8_t *buf = avpkt->data;
|
|
int buf_size = avpkt->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 < 4 || length > buf_size)
|
|
return -1;
|
|
|
|
init_get_bits(&gb, (buf + 4), (length - 4) * 8);
|
|
|
|
m->is_major_sync_unit = 0;
|
|
if (show_bits_long(&gb, 31) == (0xf8726fba >> 1)) {
|
|
if (read_major_sync(m, &gb) < 0)
|
|
goto error;
|
|
m->is_major_sync_unit = 1;
|
|
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, nonrestart_substr;
|
|
|
|
extraword_present = get_bits1(&gb);
|
|
nonrestart_substr = get_bits1(&gb);
|
|
checkdata_present = get_bits1(&gb);
|
|
skip_bits1(&gb);
|
|
|
|
end = get_bits(&gb, 12) * 2;
|
|
|
|
substr_header_size += 2;
|
|
|
|
if (extraword_present) {
|
|
if (m->avctx->codec_id == CODEC_ID_MLP) {
|
|
av_log(m->avctx, AV_LOG_ERROR, "There must be no extraword for MLP.\n");
|
|
goto error;
|
|
}
|
|
skip_bits(&gb, 16);
|
|
substr_header_size += 2;
|
|
}
|
|
|
|
if (!(nonrestart_substr ^ m->is_major_sync_unit)) {
|
|
av_log(m->avctx, AV_LOG_ERROR, "Invalid nonrestart_substr.\n");
|
|
goto error;
|
|
}
|
|
|
|
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 = ff_mlp_calculate_parity(buf, 4);
|
|
parity_bits ^= ff_mlp_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);
|
|
|
|
m->matrix_changed = 0;
|
|
memset(m->filter_changed, 0, sizeof(m->filter_changed));
|
|
|
|
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)
|
|
goto next_substr;
|
|
if (read_decoding_params(m, &gb, substr) < 0)
|
|
goto next_substr;
|
|
}
|
|
|
|
if (!s->restart_seen)
|
|
goto next_substr;
|
|
|
|
if (read_block_data(m, &gb, substr) < 0)
|
|
return -1;
|
|
|
|
if (get_bits_count(&gb) >= substream_data_len[substr] * 8)
|
|
goto substream_length_mismatch;
|
|
|
|
} while (!get_bits1(&gb));
|
|
|
|
skip_bits(&gb, (-get_bits_count(&gb)) & 15);
|
|
|
|
if (substream_data_len[substr] * 8 - get_bits_count(&gb) >= 32) {
|
|
int shorten_by;
|
|
|
|
if (get_bits(&gb, 16) != 0xD234)
|
|
return -1;
|
|
|
|
shorten_by = get_bits(&gb, 16);
|
|
if (m->avctx->codec_id == CODEC_ID_TRUEHD && shorten_by & 0x2000)
|
|
s->blockpos -= FFMIN(shorten_by & 0x1FFF, s->blockpos);
|
|
else if (m->avctx->codec_id == CODEC_ID_MLP && shorten_by != 0xD234)
|
|
return -1;
|
|
|
|
if (substr == m->max_decoded_substream)
|
|
av_log(m->avctx, AV_LOG_INFO, "End of stream indicated.\n");
|
|
}
|
|
|
|
if (substream_parity_present[substr]) {
|
|
uint8_t parity, checksum;
|
|
|
|
if (substream_data_len[substr] * 8 - get_bits_count(&gb) != 16)
|
|
goto substream_length_mismatch;
|
|
|
|
parity = ff_mlp_calculate_parity(buf, substream_data_len[substr] - 2);
|
|
checksum = ff_mlp_checksum8 (buf, substream_data_len[substr] - 2);
|
|
|
|
if ((get_bits(&gb, 8) ^ parity) != 0xa9 )
|
|
av_log(m->avctx, AV_LOG_ERROR, "Substream %d parity check failed.\n", substr);
|
|
if ( get_bits(&gb, 8) != checksum)
|
|
av_log(m->avctx, AV_LOG_ERROR, "Substream %d checksum failed.\n" , substr);
|
|
}
|
|
|
|
if (substream_data_len[substr] * 8 != get_bits_count(&gb))
|
|
goto substream_length_mismatch;
|
|
|
|
next_substr:
|
|
if (!s->restart_seen)
|
|
av_log(m->avctx, AV_LOG_ERROR,
|
|
"No restart header present in substream %d.\n", 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;
|
|
|
|
substream_length_mismatch:
|
|
av_log(m->avctx, AV_LOG_ERROR, "substream %d length mismatch\n", substr);
|
|
return -1;
|
|
|
|
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("MLP (Meridian Lossless Packing)"),
|
|
};
|
|
|
|
#if CONFIG_TRUEHD_DECODER
|
|
AVCodec truehd_decoder = {
|
|
"truehd",
|
|
CODEC_TYPE_AUDIO,
|
|
CODEC_ID_TRUEHD,
|
|
sizeof(MLPDecodeContext),
|
|
mlp_decode_init,
|
|
NULL,
|
|
NULL,
|
|
read_access_unit,
|
|
.long_name = NULL_IF_CONFIG_SMALL("TrueHD"),
|
|
};
|
|
#endif /* CONFIG_TRUEHD_DECODER */
|