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FFmpeg/libavcodec/mlpdec.c

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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 mlpdec.c
* MLP decoder
*/
#include <stdint.h>
#include "avcodec.h"
#include "libavutil/intreadwrite.h"
#include "bitstream.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;
//! 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];
ChannelParams channel_params[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;
static VLC huff_vlc[3];
/** Initialize static data, constant between all invocations of the codec. */
static av_cold void init_static(void)
{
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;
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 (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);
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 = 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);
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 = 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_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 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;
}
fp->order = order;
if (order > 0) {
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++)
fp->coeff[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 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++) {
ChannelParams *cp = &m->channel_params[ch];
s->quant_step_size[ch] = get_bits(gbp, 4);
/* TODO: validate */
cp->sign_huff_offset = calculate_sign_huff(m, substr, ch);
}
for (ch = s->min_channel; ch <= s->max_channel; ch++)
if (get_bits1(gbp)) {
ChannelParams *cp = &m->channel_params[ch];
FilterParams *fir = &cp->filter_params[FIR];
FilterParams *iir = &cp->filter_params[IIR];
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 &&
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);
cp->sign_huff_offset = 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];
FilterParams *fp[NUM_FILTERS] = { &m->channel_params[channel].filter_params[FIR],
&m->channel_params[channel].filter_params[IIR], };
unsigned int filter_shift = fp[FIR]->shift;
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], &fp[j]->state[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 < fp[j]->order; order++)
accum += (int64_t)filter_state_buffer[j][index + order] *
fp[j]->coeff[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(&fp[j]->state[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);
}
/** 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 = 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);
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) >= 32 &&
(show_bits_long(&gb, 32) == END_OF_STREAM ||
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_data_len[substr] * 8 - get_bits_count(&gb) >= 16 &&
substream_parity_present[substr]) {
uint8_t parity, checksum;
parity = ff_mlp_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 = ff_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"),
};