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FFmpeg/libavcodec/mlpdec.c
Michael Niedermayer e4de71677f Merge remote-tracking branch 'qatar/master' 
* qatar/master:
  aac_latm: reconfigure decoder on audio specific config changes
  latmdec: fix audio specific config parsing
  Add avcodec_decode_audio4().
  avcodec: change number of plane pointers from 4 to 8 at next major bump.
  Update developers documentation with coding conventions.
  svq1dec: avoid undefined get_bits(0) call
  ARM: h264dsp_neon cosmetics
  ARM: make some NEON macros reusable
  Do not memcpy raw video frames when using null muxer
  fate: update asf seektest
  vp8: flush buffers on size changes.
  doc: improve general documentation for MacOSX
  asf: use packet dts as approximation of pts
  asf: do not call av_read_frame
  rtsp: Initialize the media_type_mask in the rtp guessing demuxer
  Cleaned up alacenc.c

Conflicts:
	doc/APIchanges
	doc/developer.texi
	libavcodec/8svx.c
	libavcodec/aacdec.c
	libavcodec/ac3dec.c
	libavcodec/avcodec.h
	libavcodec/nellymoserdec.c
	libavcodec/tta.c
	libavcodec/utils.c
	libavcodec/version.h
	libavcodec/wmadec.c
	libavformat/asfdec.c
	tests/ref/seek/lavf_asf

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2011-12-03 03:00:30 +01:00

1217 lines
41 KiB
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/*
* 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
* 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];
//! Channel coding parameters for channels in the substream
ChannelParams channel_params[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;
AVFrame frame;
//! 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;
//! 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
int access_unit_size_pow2;
SubStream substream[MAX_SUBSTREAMS];
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)
{
SubStream *s = &m->substream[substr];
ChannelParams *cp = &s->channel_params[ch];
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 = &s->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 AVERROR_INVALIDDATA;
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);
avcodec_get_frame_defaults(&m->frame);
avctx->coded_frame = &m->frame;
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, ret;
if ((ret = ff_mlp_read_major_sync(m->avctx, &mh, gb)) != 0)
return ret;
if (mh.group1_bits == 0) {
av_log(m->avctx, AV_LOG_ERROR, "invalid/unknown bits per sample\n");
return AVERROR_INVALIDDATA;
}
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 AVERROR_INVALIDDATA;
}
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 AVERROR_INVALIDDATA;
}
if (mh.group1_samplerate == 0) {
av_log(m->avctx, AV_LOG_ERROR, "invalid/unknown sampling rate\n");
return AVERROR_INVALIDDATA;
}
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 AVERROR_INVALIDDATA;
}
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 AVERROR_INVALIDDATA;
}
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 AVERROR_INVALIDDATA;
}
if (mh.num_substreams == 0)
return AVERROR_INVALIDDATA;
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 AVERROR_INVALIDDATA;
}
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 AVERROR_INVALIDDATA;
}
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 = AV_SAMPLE_FMT_S32;
else
m->avctx->sample_fmt = AV_SAMPLE_FMT_S16;
m->params_valid = 1;
for (substr = 0; substr < MAX_SUBSTREAMS; substr++)
m->substream[substr].restart_seen = 0;
if (mh.stream_type == 0xbb) {
/* MLP stream */
m->avctx->channel_layout = ff_mlp_layout[mh.channels_mlp];
} else { /* mh.stream_type == 0xba */
/* TrueHD stream */
if (mh.channels_thd_stream2) {
m->avctx->channel_layout = ff_truehd_layout(mh.channels_thd_stream2);
} else {
m->avctx->channel_layout = ff_truehd_layout(mh.channels_thd_stream1);
}
if (m->avctx->channels &&
!m->avctx->request_channels && !m->avctx->request_channel_layout &&
av_get_channel_layout_nb_channels(m->avctx->channel_layout) != m->avctx->channels) {
m->avctx->channel_layout = 0;
av_log_ask_for_sample(m->avctx, "Unknown channel layout.");
}
}
m->needs_reordering = mh.channels_mlp >= 18 && mh.channels_mlp <= 20;
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 AVERROR_INVALIDDATA;
}
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 AVERROR_INVALIDDATA;
}
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 AVERROR_INVALIDDATA;
}
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 AVERROR_INVALIDDATA;
}
/* 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 AVERROR_INVALIDDATA;
}
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 AVERROR_INVALIDDATA;
}
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 AVERROR_INVALIDDATA;
}
s->ch_assign[ch_assign] = ch;
}
if (m->avctx->codec_id == 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]);
}
}
if (m->avctx->codec_id == CODEC_ID_TRUEHD &&
m->avctx->channel_layout == AV_CH_LAYOUT_7POINT1) {
FFSWAP(int, s->ch_assign[4], s->ch_assign[6]);
FFSWAP(int, s->ch_assign[5], s->ch_assign[7]);
}
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 = &s->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 substr, unsigned int channel,
unsigned int filter)
{
SubStream *s = &m->substream[substr];
FilterParams *fp = &s->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 AVERROR_INVALIDDATA;
}
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 AVERROR_INVALIDDATA;
}
fp->order = order;
if (order > 0) {
int32_t *fcoeff = s->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 AVERROR_INVALIDDATA;
}
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 AVERROR_INVALIDDATA;
}
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 AVERROR_INVALIDDATA;
}
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 AVERROR_INVALIDDATA;
}
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 AVERROR_INVALIDDATA;
}
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 AVERROR_INVALIDDATA;
}
if (frac_bits > 14) {
av_log(m->avctx, AV_LOG_ERROR,
"Too many fractional bits specified.\n");
return AVERROR_INVALIDDATA;
}
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)
{
SubStream *s = &m->substream[substr];
ChannelParams *cp = &s->channel_params[ch];
FilterParams *fir = &cp->filter_params[FIR];
FilterParams *iir = &cp->filter_params[IIR];
int ret;
if (s->param_presence_flags & PARAM_FIR)
if (get_bits1(gbp))
if ((ret = read_filter_params(m, gbp, substr, ch, FIR)) < 0)
return ret;
if (s->param_presence_flags & PARAM_IIR)
if (get_bits1(gbp))
if ((ret = read_filter_params(m, gbp, substr, ch, IIR)) < 0)
return ret;
if (fir->order + iir->order > 8) {
av_log(m->avctx, AV_LOG_ERROR, "Total filter orders too high.\n");
return AVERROR_INVALIDDATA;
}
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 AVERROR_INVALIDDATA;
}
/* 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 AVERROR_INVALIDDATA;
}
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;
int ret;
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 AVERROR_INVALIDDATA;
}
}
if (s->param_presence_flags & PARAM_MATRIX)
if (get_bits1(gbp))
if ((ret = read_matrix_params(m, substr, gbp)) < 0)
return ret;
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 = &s->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 ((ret = read_channel_params(m, substr, gbp, ch)) < 0)
return ret;
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 = s->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 = &s->channel_params[channel].filter_params[FIR];
FilterParams *iir = &s->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;
int ret;
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 AVERROR_INVALIDDATA;
}
memset(&m->bypassed_lsbs[s->blockpos][0], 0,
s->blocksize * sizeof(m->bypassed_lsbs[0]));
for (i = 0; i < s->blocksize; i++)
if ((ret = read_huff_channels(m, gbp, substr, i)) < 0)
return ret;
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(MLPDecodeContext *m, unsigned int substr,
void *data, int *got_frame_ptr)
{
AVCodecContext *avctx = m->avctx;
SubStream *s = &m->substream[substr];
unsigned int i, out_ch = 0;
int32_t *data_32;
int16_t *data_16;
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;
}
/* get output buffer */
m->frame.nb_samples = s->blockpos;
if ((ret = avctx->get_buffer(avctx, &m->frame)) < 0) {
av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
return ret;
}
data_32 = (int32_t *)m->frame.data[0];
data_16 = (int16_t *)m->frame.data[0];
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;
}
}
*got_frame_ptr = 1;
*(AVFrame *)data = m->frame;
return 0;
}
/** Read an access unit from the stream.
* @return negative on error, 0 if not enough data is present in the input stream,
* otherwise the number of bytes consumed. */
static int read_access_unit(AVCodecContext *avctx, void* data,
int *got_frame_ptr, 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;
int ret;
if (buf_size < 4)
return 0;
length = (AV_RB16(buf) & 0xfff) * 2;
if (length < 4 || length > buf_size)
return AVERROR_INVALIDDATA;
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");
*got_frame_ptr = 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 ((ret = read_block_data(m, &gb, substr)) < 0)
return ret;
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 AVERROR_INVALIDDATA;
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 AVERROR_INVALIDDATA;
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 ((ret = output_data(m, m->max_decoded_substream, data, got_frame_ptr)) < 0)
return ret;
return length;
substream_length_mismatch:
av_log(m->avctx, AV_LOG_ERROR, "substream %d length mismatch\n", substr);
return AVERROR_INVALIDDATA;
error:
m->params_valid = 0;
return AVERROR_INVALIDDATA;
}
AVCodec ff_mlp_decoder = {
.name = "mlp",
.type = AVMEDIA_TYPE_AUDIO,
.id = CODEC_ID_MLP,
.priv_data_size = sizeof(MLPDecodeContext),
.init = mlp_decode_init,
.decode = read_access_unit,
.capabilities = CODEC_CAP_DR1,
.long_name = NULL_IF_CONFIG_SMALL("MLP (Meridian Lossless Packing)"),
};
#if CONFIG_TRUEHD_DECODER
AVCodec ff_truehd_decoder = {
.name = "truehd",
.type = AVMEDIA_TYPE_AUDIO,
.id = CODEC_ID_TRUEHD,
.priv_data_size = sizeof(MLPDecodeContext),
.init = mlp_decode_init,
.decode = read_access_unit,
.capabilities = CODEC_CAP_DR1,
.long_name = NULL_IF_CONFIG_SMALL("TrueHD"),
};
#endif /* CONFIG_TRUEHD_DECODER */
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