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Add MPEG-4 Audio Lossless Coding (ALS) decoder.

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Patch by Thilo Borgmann, thilo D borgmann A googlemail

Originally committed as revision 20517 to svn://svn.ffmpeg.org/ffmpeg/trunk
release/0.6
Thilo Borgmann 15 years ago committed by Carl Eugen Hoyos
parent 24026a2d49
commit 99971952b0
6 changed files with 1096 additions and 0 deletions
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  1. 1
      Changelog
  2. 1
      doc/general.texi
  3. 1
      libavcodec/Makefile
  4. 1
      libavcodec/allcodecs.c
  5. 95
      libavcodec/als_data.h
  6. 997
      libavcodec/alsdec.c

1
Changelog
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@ -41,6 +41,7 @@ version <next>:
- Atrac1 decoder
- MD STUDIO audio demuxer
- RF64 support in WAV demuxer
- MPEG-4 Audio Lossless Coding (ALS) decoder

1
doc/general.texi
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@ -569,6 +569,7 @@ following image formats are supported:
@item MP2 (MPEG audio layer 2) @tab IX @tab IX
@item MP3 (MPEG audio layer 3) @tab E @tab IX
@tab encoding supported through external library LAME, ADU MP3 and MP3onMP4 also supported
@item MPEG-4 Audio Lossless Coding (ALS) @tab @tab X
@item Musepack SV7 @tab @tab X
@item Musepack SV8 @tab @tab X
@item Nellymoser Asao @tab X @tab X

1
libavcodec/Makefile
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@ -51,6 +51,7 @@ OBJS-$(CONFIG_AC3_DECODER) += ac3dec.o ac3tab.o \
OBJS-$(CONFIG_AC3_ENCODER) += ac3enc.o ac3tab.o ac3.o
OBJS-$(CONFIG_ALAC_DECODER) += alac.o
OBJS-$(CONFIG_ALAC_ENCODER) += alacenc.o
OBJS-$(CONFIG_ALS_DECODER) += alsdec.o
OBJS-$(CONFIG_AMV_DECODER) += sp5xdec.o mjpegdec.o mjpeg.o
OBJS-$(CONFIG_APE_DECODER) += apedec.o
OBJS-$(CONFIG_ASV1_DECODER) += asv1.o mpeg12data.o

1
libavcodec/allcodecs.c
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@ -200,6 +200,7 @@ void avcodec_register_all(void)
REGISTER_ENCDEC (AAC, aac);
REGISTER_ENCDEC (AC3, ac3);
REGISTER_ENCDEC (ALAC, alac);
REGISTER_DECODER (ALS, als);
REGISTER_DECODER (APE, ape);
REGISTER_DECODER (ATRAC1, atrac1);
REGISTER_DECODER (ATRAC3, atrac3);

95
libavcodec/als_data.h
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@ -0,0 +1,95 @@
/*
* ALS header file for common data
* Copyright (c) 2009 Thilo Borgmann <thilo.borgmann _at_ googlemail.com>
*
* 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
*/
#ifndef AVCODEC_ALS_DATA_H
#define AVCODEC_ALS_DATA_H
/**
* @file libavcodec/als_data.h
* MPEG-4 ALS common data tables
* @author Thilo Borgmann <thilo.borgmann _at_ googlemail.com>
*/
#include <stdint.h>
/** Rice parameters and corresponding index offsets for decoding the
* indices of scaled PARCOR values. The table choosen is set globally
* by the encoder and stored in ALSSpecificConfig.
*/
static const int8_t parcor_rice_table[3][20][2] = {
{ {-52, 4}, {-29, 5}, {-31, 4}, { 19, 4}, {-16, 4},
{ 12, 3}, { -7, 3}, { 9, 3}, { -5, 3}, { 6, 3},
{ -4, 3}, { 3, 3}, { -3, 2}, { 3, 2}, { -2, 2},
{ 3, 2}, { -1, 2}, { 2, 2}, { -1, 2}, { 2, 2} },
{ {-58, 3}, {-42, 4}, {-46, 4}, { 37, 5}, {-36, 4},
{ 29, 4}, {-29, 4}, { 25, 4}, {-23, 4}, { 20, 4},
{-17, 4}, { 16, 4}, {-12, 4}, { 12, 3}, {-10, 4},
{ 7, 3}, { -4, 4}, { 3, 3}, { -1, 3}, { 1, 3} },
{ {-59, 3}, {-45, 5}, {-50, 4}, { 38, 4}, {-39, 4},
{ 32, 4}, {-30, 4}, { 25, 3}, {-23, 3}, { 20, 3},
{-20, 3}, { 16, 3}, {-13, 3}, { 10, 3}, { -7, 3},
{ 3, 3}, { 0, 3}, { -1, 3}, { 2, 3}, { -1, 2} }
};
/** Scaled PARCOR values used for the first two PARCOR coefficients.
* To be indexed by the Rice coded indices.
* Generated by: parcor_scaled_values[i] = 32 + ((i * (i+1)) << 7) - (1 << 20)
* Actual values are divided by 32 in order to be stored in 16 bits.
*/
static const int16_t parcor_scaled_values[] = {
-1048544 / 32, -1048288 / 32, -1047776 / 32, -1047008 / 32,
-1045984 / 32, -1044704 / 32, -1043168 / 32, -1041376 / 32,
-1039328 / 32, -1037024 / 32, -1034464 / 32, -1031648 / 32,
-1028576 / 32, -1025248 / 32, -1021664 / 32, -1017824 / 32,
-1013728 / 32, -1009376 / 32, -1004768 / 32, -999904 / 32,
-994784 / 32, -989408 / 32, -983776 / 32, -977888 / 32,
-971744 / 32, -965344 / 32, -958688 / 32, -951776 / 32,
-944608 / 32, -937184 / 32, -929504 / 32, -921568 / 32,
-913376 / 32, -904928 / 32, -896224 / 32, -887264 / 32,
-878048 / 32, -868576 / 32, -858848 / 32, -848864 / 32,
-838624 / 32, -828128 / 32, -817376 / 32, -806368 / 32,
-795104 / 32, -783584 / 32, -771808 / 32, -759776 / 32,
-747488 / 32, -734944 / 32, -722144 / 32, -709088 / 32,
-695776 / 32, -682208 / 32, -668384 / 32, -654304 / 32,
-639968 / 32, -625376 / 32, -610528 / 32, -595424 / 32,
-580064 / 32, -564448 / 32, -548576 / 32, -532448 / 32,
-516064 / 32, -499424 / 32, -482528 / 32, -465376 / 32,
-447968 / 32, -430304 / 32, -412384 / 32, -394208 / 32,
-375776 / 32, -357088 / 32, -338144 / 32, -318944 / 32,
-299488 / 32, -279776 / 32, -259808 / 32, -239584 / 32,
-219104 / 32, -198368 / 32, -177376 / 32, -156128 / 32,
-134624 / 32, -112864 / 32, -90848 / 32, -68576 / 32,
-46048 / 32, -23264 / 32, -224 / 32, 23072 / 32,
46624 / 32, 70432 / 32, 94496 / 32, 118816 / 32,
143392 / 32, 168224 / 32, 193312 / 32, 218656 / 32,
244256 / 32, 270112 / 32, 296224 / 32, 322592 / 32,
349216 / 32, 376096 / 32, 403232 / 32, 430624 / 32,
458272 / 32, 486176 / 32, 514336 / 32, 542752 / 32,
571424 / 32, 600352 / 32, 629536 / 32, 658976 / 32,
688672 / 32, 718624 / 32, 748832 / 32, 779296 / 32,
810016 / 32, 840992 / 32, 872224 / 32, 903712 / 32,
935456 / 32, 967456 / 32, 999712 / 32, 1032224 / 32
};
#endif /* AVCODEC_ALS_DATA_H */

997
libavcodec/alsdec.c
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@ -0,0 +1,997 @@
/*
* MPEG-4 ALS decoder
* Copyright (c) 2009 Thilo Borgmann <thilo.borgmann _at_ googlemail.com>
*
* 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/alsdec.c
* MPEG-4 ALS decoder
* @author Thilo Borgmann <thilo.borgmann _at_ googlemail.com>
*/
//#define DEBUG
#include "avcodec.h"
#include "get_bits.h"
#include "unary.h"
#include "mpeg4audio.h"
#include "bytestream.h"
#include "als_data.h"
enum RA_Flag {
RA_FLAG_NONE,
RA_FLAG_FRAMES,
RA_FLAG_HEADER
};
typedef struct {
uint32_t samples; ///< number of samples, 0xFFFFFFFF if unknown
int resolution; ///< 000 = 8-bit; 001 = 16-bit; 010 = 24-bit; 011 = 32-bit
int floating; ///< 1 = IEEE 32-bit floating-point, 0 = integer
int frame_length; ///< frame length for each frame (last frame may differ)
int ra_distance; ///< distance between RA frames (in frames, 0...255)
enum RA_Flag ra_flag; ///< indicates where the size of ra units is stored
int adapt_order; ///< adaptive order: 1 = on, 0 = off
int coef_table; ///< table index of Rice code parameters
int long_term_prediction; ///< long term prediction (LTP): 1 = on, 0 = off
int max_order; ///< maximum prediction order (0..1023)
int block_switching; ///< number of block switching levels
int bgmc; ///< "Block Gilbert-Moore Code": 1 = on, 0 = off (Rice coding only)
int sb_part; ///< sub-block partition
int joint_stereo; ///< joint stereo: 1 = on, 0 = off
int mc_coding; ///< extended inter-channel coding (multi channel coding): 1 = on, 0 = off
int chan_config; ///< indicates that a chan_config_info field is present
int chan_sort; ///< channel rearrangement: 1 = on, 0 = off
int rlslms; ///< use "Recursive Least Square-Least Mean Square" predictor: 1 = on, 0 = off
int chan_config_info; ///< mapping of channels to loudspeaker locations. Unused until setting channel configuration is implemented.
int *chan_pos; ///< original channel positions
uint32_t header_size; ///< header size of original audio file in bytes, provided for debugging
uint32_t trailer_size; ///< trailer size of original audio file in bytes, provided for debugging
} ALSSpecificConfig;
typedef struct {
AVCodecContext *avctx;
ALSSpecificConfig sconf;
GetBitContext gb;
unsigned int cur_frame_length; ///< length of the current frame to decode
unsigned int frame_id; ///< the frame ID / number of the current frame
unsigned int js_switch; ///< if true, joint-stereo decoding is enforced
unsigned int num_blocks; ///< number of blocks used in the current frame
int32_t *quant_cof; ///< quantized parcor coefficients
int32_t *lpc_cof; ///< coefficients of the direct form prediction filter
int32_t *prev_raw_samples; ///< contains unshifted raw samples from the previous block
int32_t **raw_samples; ///< decoded raw samples for each channel
int32_t *raw_buffer; ///< contains all decoded raw samples including carryover samples
} ALSDecContext;
static av_cold void dprint_specific_config(ALSDecContext *ctx)
{
#ifdef DEBUG
AVCodecContext *avctx = ctx->avctx;
ALSSpecificConfig *sconf = &ctx->sconf;
dprintf(avctx, "resolution = %i\n", sconf->resolution);
dprintf(avctx, "floating = %i\n", sconf->floating);
dprintf(avctx, "frame_length = %i\n", sconf->frame_length);
dprintf(avctx, "ra_distance = %i\n", sconf->ra_distance);
dprintf(avctx, "ra_flag = %i\n", sconf->ra_flag);
dprintf(avctx, "adapt_order = %i\n", sconf->adapt_order);
dprintf(avctx, "coef_table = %i\n", sconf->coef_table);
dprintf(avctx, "long_term_prediction = %i\n", sconf->long_term_prediction);
dprintf(avctx, "max_order = %i\n", sconf->max_order);
dprintf(avctx, "block_switching = %i\n", sconf->block_switching);
dprintf(avctx, "bgmc = %i\n", sconf->bgmc);
dprintf(avctx, "sb_part = %i\n", sconf->sb_part);
dprintf(avctx, "joint_stereo = %i\n", sconf->joint_stereo);
dprintf(avctx, "mc_coding = %i\n", sconf->mc_coding);
dprintf(avctx, "chan_config = %i\n", sconf->chan_config);
dprintf(avctx, "chan_sort = %i\n", sconf->chan_sort);
dprintf(avctx, "RLSLMS = %i\n", sconf->rlslms);
dprintf(avctx, "chan_config_info = %i\n", sconf->chan_config_info);
dprintf(avctx, "header_size = %i\n", sconf->header_size);
dprintf(avctx, "trailer_size = %i\n", sconf->trailer_size);
#endif
}
/** Reads an ALSSpecificConfig from a buffer into the output struct.
*/
static av_cold int read_specific_config(ALSDecContext *ctx)
{
GetBitContext gb;
uint64_t ht_size;
int i, config_offset, crc_enabled;
MPEG4AudioConfig m4ac;
ALSSpecificConfig *sconf = &ctx->sconf;
AVCodecContext *avctx = ctx->avctx;
uint32_t als_id;
init_get_bits(&gb, avctx->extradata, avctx->extradata_size * 8);
config_offset = ff_mpeg4audio_get_config(&m4ac, avctx->extradata,
avctx->extradata_size);
if (config_offset < 0)
return -1;
skip_bits_long(&gb, config_offset);
if (get_bits_left(&gb) < (30 << 3))
return -1;
// read the fixed items
als_id = get_bits_long(&gb, 32);
avctx->sample_rate = m4ac.sample_rate;
skip_bits_long(&gb, 32); // sample rate already known
sconf->samples = get_bits_long(&gb, 32);
avctx->channels = m4ac.channels;
skip_bits(&gb, 16); // number of channels already knwon
skip_bits(&gb, 3); // skip file_type
sconf->resolution = get_bits(&gb, 3);
sconf->floating = get_bits1(&gb);
skip_bits1(&gb); // skip msb_first
sconf->frame_length = get_bits(&gb, 16) + 1;
sconf->ra_distance = get_bits(&gb, 8);
sconf->ra_flag = get_bits(&gb, 2);
sconf->adapt_order = get_bits1(&gb);
sconf->coef_table = get_bits(&gb, 2);
sconf->long_term_prediction = get_bits1(&gb);
sconf->max_order = get_bits(&gb, 10);
sconf->block_switching = get_bits(&gb, 2);
sconf->bgmc = get_bits1(&gb);
sconf->sb_part = get_bits1(&gb);
sconf->joint_stereo = get_bits1(&gb);
sconf->mc_coding = get_bits1(&gb);
sconf->chan_config = get_bits1(&gb);
sconf->chan_sort = get_bits1(&gb);
crc_enabled = get_bits1(&gb);
sconf->rlslms = get_bits1(&gb);
skip_bits(&gb, 5); // skip 5 reserved bits
skip_bits1(&gb); // skip aux_data_enabled
// check for ALSSpecificConfig struct
if (als_id != MKBETAG('A','L','S','\0'))
return -1;
ctx->cur_frame_length = sconf->frame_length;
// allocate quantized parcor coefficient buffer
if (!(ctx->quant_cof = av_malloc(sizeof(*ctx->quant_cof) * sconf->max_order)) ||
!(ctx->lpc_cof = av_malloc(sizeof(*ctx->lpc_cof) * sconf->max_order))) {
av_log(avctx, AV_LOG_ERROR, "Allocating buffer memory failed.\n");
return AVERROR(ENOMEM);
}
// read channel config
if (sconf->chan_config)
sconf->chan_config_info = get_bits(&gb, 16);
// TODO: use this to set avctx->channel_layout
// read channel sorting
if (sconf->chan_sort && avctx->channels > 1) {
int chan_pos_bits = av_ceil_log2(avctx->channels);
int bits_needed = avctx->channels * chan_pos_bits + 7;
if (get_bits_left(&gb) < bits_needed)
return -1;
if (!(sconf->chan_pos = av_malloc(avctx->channels * sizeof(*sconf->chan_pos))))
return AVERROR(ENOMEM);
for (i = 0; i < avctx->channels; i++)
sconf->chan_pos[i] = get_bits(&gb, chan_pos_bits);
align_get_bits(&gb);
// TODO: use this to actually do channel sorting
} else {
sconf->chan_sort = 0;
}
// read fixed header and trailer sizes,
// if size = 0xFFFFFFFF then there is no data field!
if (get_bits_left(&gb) < 64)
return -1;
sconf->header_size = get_bits_long(&gb, 32);
sconf->trailer_size = get_bits_long(&gb, 32);
if (sconf->header_size == 0xFFFFFFFF)
sconf->header_size = 0;
if (sconf->trailer_size == 0xFFFFFFFF)
sconf->trailer_size = 0;
ht_size = ((int64_t)(sconf->header_size) + (int64_t)(sconf->trailer_size)) << 3;
// skip the header and trailer data
if (get_bits_left(&gb) < ht_size)
return -1;
if (ht_size > INT32_MAX)
return -1;
skip_bits_long(&gb, ht_size);
// skip the crc data
if (crc_enabled) {
if (get_bits_left(&gb) < 32)
return -1;
skip_bits_long(&gb, 32);
}
// no need to read the rest of ALSSpecificConfig (ra_unit_size & aux data)
dprint_specific_config(ctx);
return 0;
}
/** Checks the ALSSpecificConfig for unsupported features.
*/
static int check_specific_config(ALSDecContext *ctx)
{
ALSSpecificConfig *sconf = &ctx->sconf;
int error = 0;
// report unsupported feature and set error value
#define MISSING_ERR(cond, str, errval) \
{ \
if (cond) { \
av_log_missing_feature(ctx->avctx, str, 0); \
error = errval; \
} \
}
MISSING_ERR(sconf->floating, "Floating point decoding", -1);
MISSING_ERR(sconf->long_term_prediction, "Long-term prediction", -1);
MISSING_ERR(sconf->bgmc, "BGMC entropy decoding", -1);
MISSING_ERR(sconf->mc_coding, "Multi-channel correlation", -1);
MISSING_ERR(sconf->rlslms, "Adaptive RLS-LMS prediction", -1);
MISSING_ERR(sconf->chan_sort, "Channel sorting", 0);
return error;
}
/** Parses the bs_info field to extract the block partitioning used in
* block switching mode, refer to ISO/IEC 14496-3, section 11.6.2.
*/
static void parse_bs_info(const uint32_t bs_info, unsigned int n,
unsigned int div, unsigned int **div_blocks,
unsigned int *num_blocks)
{
if (n < 31 && ((bs_info << n) & 0x40000000)) {
// if the level is valid and the investigated bit n is set
// then recursively check both children at bits (2n+1) and (2n+2)
n *= 2;
div += 1;
parse_bs_info(bs_info, n + 1, div, div_blocks, num_blocks);
parse_bs_info(bs_info, n + 2, div, div_blocks, num_blocks);
} else {
// else the bit is not set or the last level has been reached
// (bit implicitly not set)
**div_blocks = div;
(*div_blocks)++;
(*num_blocks)++;
}
}
/** Reads and decodes a Rice codeword.
*/
static int32_t decode_rice(GetBitContext *gb, unsigned int k)
{
int max = gb->size_in_bits - get_bits_count(gb) - k;
int q = get_unary(gb, 0, max);
int r = k ? get_bits1(gb) : !(q & 1);
if (k > 1) {
q <<= (k - 1);
q += get_bits_long(gb, k - 1);
} else if (!k) {
q >>= 1;
}
return r ? q : ~q;
}
/** Converts PARCOR coefficient k to direct filter coefficient.
*/
static void parcor_to_lpc(unsigned int k, const int32_t *par, int32_t *cof)
{
int i, j;
for (i = 0, j = k - 1; i < j; i++, j--) {
int tmp1 = ((MUL64(par[k], cof[j]) + (1 << 19)) >> 20);
cof[j] += ((MUL64(par[k], cof[i]) + (1 << 19)) >> 20);
cof[i] += tmp1;
}
if (i == j)
cof[i] += ((MUL64(par[k], cof[j]) + (1 << 19)) >> 20);
cof[k] = par[k];
}
/** Reads block switching field if necessary and sets actual block sizes.
* Also assures that the block sizes of the last frame correspond to the
* actual number of samples.
*/
static void get_block_sizes(ALSDecContext *ctx, unsigned int *div_blocks,
uint32_t *bs_info)
{
ALSSpecificConfig *sconf = &ctx->sconf;
GetBitContext *gb = &ctx->gb;
unsigned int *ptr_div_blocks = div_blocks;
unsigned int b;
if (sconf->block_switching) {
unsigned int bs_info_len = 1 << (sconf->block_switching + 2);
*bs_info = get_bits_long(gb, bs_info_len);
*bs_info <<= (32 - bs_info_len);
}
ctx->num_blocks = 0;
parse_bs_info(*bs_info, 0, 0, &ptr_div_blocks, &ctx->num_blocks);
// The last frame may have an overdetermined block structure given in
// the bitstream. In that case the defined block structure would need
// more samples than available to be consistent.
// The block structure is actually used but the block sizes are adapted
// to fit the actual number of available samples.
// Example: 5 samples, 2nd level block sizes: 2 2 2 2.
// This results in the actual block sizes: 2 2 1 0.
// This is not specified in 14496-3 but actually done by the reference
// codec RM22 revision 2.
// This appears to happen in case of an odd number of samples in the last
// frame which is actually not allowed by the block length switching part
// of 14496-3.
// The ALS conformance files feature an odd number of samples in the last
// frame.
for (b = 0; b < ctx->num_blocks; b++)
div_blocks[b] = ctx->sconf.frame_length >> div_blocks[b];
if (ctx->cur_frame_length != ctx->sconf.frame_length) {
unsigned int remaining = ctx->cur_frame_length;
for (b = 0; b < ctx->num_blocks; b++) {
if (remaining < div_blocks[b]) {
div_blocks[b] = remaining;
ctx->num_blocks = b + 1;
break;
}
remaining -= div_blocks[b];
}
}
}
/** Reads the block data for a constant block
*/
static void read_const_block(ALSDecContext *ctx, int32_t *raw_samples,
unsigned int block_length, unsigned int *js_blocks)
{
ALSSpecificConfig *sconf = &ctx->sconf;
AVCodecContext *avctx = ctx->avctx;
GetBitContext *gb = &ctx->gb;
int32_t const_val = 0;
unsigned int const_block, k;
const_block = get_bits1(gb); // 1 = constant value, 0 = zero block (silence)
*js_blocks = get_bits1(gb);
// skip 5 reserved bits
skip_bits(gb, 5);
if (const_block) {
unsigned int const_val_bits = sconf->floating ? 24 : avctx->bits_per_raw_sample;
const_val = get_sbits_long(gb, const_val_bits);
}
// write raw samples into buffer
for (k = 0; k < block_length; k++)
raw_samples[k] = const_val;
}
/** Reads the block data for a non-constant block
*/
static int read_var_block(ALSDecContext *ctx, unsigned int ra_block,
int32_t *raw_samples, unsigned int block_length,
unsigned int *js_blocks, int32_t *raw_other,
unsigned int *shift_lsbs)
{
ALSSpecificConfig *sconf = &ctx->sconf;
AVCodecContext *avctx = ctx->avctx;
GetBitContext *gb = &ctx->gb;
unsigned int k;
unsigned int s[8];
unsigned int sub_blocks, log2_sub_blocks, sb_length;
unsigned int opt_order = 1;
int32_t *quant_cof = ctx->quant_cof;
int32_t *lpc_cof = ctx->lpc_cof;
unsigned int start = 0;
int smp = 0;
int sb, store_prev_samples;
int64_t y;
*js_blocks = get_bits1(gb);
// determine the number of subblocks for entropy decoding
if (!sconf->bgmc && !sconf->sb_part) {
log2_sub_blocks = 0;
} else {
if (sconf->bgmc && sconf->sb_part)
log2_sub_blocks = get_bits(gb, 2);
else
log2_sub_blocks = 2 * get_bits1(gb);
}
sub_blocks = 1 << log2_sub_blocks;
// do not continue in case of a damaged stream since
// block_length must be evenly divisible by sub_blocks
if (block_length & (sub_blocks - 1)) {
av_log(avctx, AV_LOG_WARNING,
"Block length is not evenly divisible by the number of subblocks.\n");
return -1;
}
sb_length = block_length >> log2_sub_blocks;
if (sconf->bgmc) {
// TODO: BGMC mode
} else {
s[0] = get_bits(gb, 4 + (sconf->resolution > 1));
for (k = 1; k < sub_blocks; k++)
s[k] = s[k - 1] + decode_rice(gb, 0);
}
if (get_bits1(gb))
*shift_lsbs = get_bits(gb, 4) + 1;
store_prev_samples = (*js_blocks && raw_other) || *shift_lsbs;
if (!sconf->rlslms) {
if (sconf->adapt_order) {
int opt_order_length = av_ceil_log2(av_clip((block_length >> 3) - 1,
2, sconf->max_order + 1));
opt_order = get_bits(gb, opt_order_length);
} else {
opt_order = sconf->max_order;
}
if (opt_order) {
int add_base;
if (sconf->coef_table == 3) {
add_base = 0x7F;
// read coefficient 0
quant_cof[0] = 32 * parcor_scaled_values[get_bits(gb, 7)];
// read coefficient 1
if (opt_order > 1)
quant_cof[1] = -32 * parcor_scaled_values[get_bits(gb, 7)];
// read coefficients 2 to opt_order
for (k = 2; k < opt_order; k++)
quant_cof[k] = get_bits(gb, 7);
} else {
int k_max;
add_base = 1;
// read coefficient 0 to 19
k_max = FFMIN(opt_order, 20);
for (k = 0; k < k_max; k++) {
int rice_param = parcor_rice_table[sconf->coef_table][k][1];
int offset = parcor_rice_table[sconf->coef_table][k][0];
quant_cof[k] = decode_rice(gb, rice_param) + offset;
}
// read coefficients 20 to 126
k_max = FFMIN(opt_order, 127);
for (; k < k_max; k++)
quant_cof[k] = decode_rice(gb, 2) + (k & 1);
// read coefficients 127 to opt_order
for (; k < opt_order; k++)
quant_cof[k] = decode_rice(gb, 1);
quant_cof[0] = 32 * parcor_scaled_values[quant_cof[0] + 64];
if (opt_order > 1)
quant_cof[1] = -32 * parcor_scaled_values[quant_cof[1] + 64];
}
for (k = 2; k < opt_order; k++)
quant_cof[k] = (quant_cof[k] << 14) + (add_base << 13);
}
}
// TODO: LTP mode
// read first value and residuals in case of a random access block
if (ra_block) {
if (opt_order)
raw_samples[0] = decode_rice(gb, avctx->bits_per_raw_sample - 4);
if (opt_order > 1)
raw_samples[1] = decode_rice(gb, s[0] + 3);
if (opt_order > 2)
raw_samples[2] = decode_rice(gb, s[0] + 1);
start = FFMIN(opt_order, 3);
}
// read all residuals
if (sconf->bgmc) {
// TODO: BGMC mode
} else {
int32_t *current_res = raw_samples + start;
for (sb = 0; sb < sub_blocks; sb++, start = 0)
for (; start < sb_length; start++)
*current_res++ = decode_rice(gb, s[sb]);
}
// reconstruct all samples from residuals
if (ra_block) {
for (smp = 0; smp < opt_order; smp++) {
y = 1 << 19;
for (sb = 0; sb < smp; sb++)
y += MUL64(lpc_cof[sb],raw_samples[smp - (sb + 1)]);
raw_samples[smp] -= y >> 20;
parcor_to_lpc(smp, quant_cof, lpc_cof);
}
} else {
for (k = 0; k < opt_order; k++)
parcor_to_lpc(k, quant_cof, lpc_cof);
// store previous samples in case that they have to be altered
if (store_prev_samples)
memcpy(ctx->prev_raw_samples, raw_samples - sconf->max_order,
sizeof(*ctx->prev_raw_samples) * sconf->max_order);
// reconstruct difference signal for prediction (joint-stereo)
if (*js_blocks && raw_other) {
int32_t *left, *right;
if (raw_other > raw_samples) { // D = R - L
left = raw_samples;
right = raw_other;
} else { // D = R - L
left = raw_other;
right = raw_samples;
}
for (sb = -1; sb >= -sconf->max_order; sb--)
raw_samples[sb] = right[sb] - left[sb];
}
// reconstruct shifted signal
if (*shift_lsbs)
for (sb = -1; sb >= -sconf->max_order; sb--)
raw_samples[sb] >>= *shift_lsbs;
}
// reconstruct raw samples
for (; smp < block_length; smp++) {
y = 1 << 19;
for (sb = 0; sb < opt_order; sb++)
y += MUL64(lpc_cof[sb],raw_samples[smp - (sb + 1)]);
raw_samples[smp] -= y >> 20;
}
// restore previous samples in case that they have been altered
if (store_prev_samples)
memcpy(raw_samples - sconf->max_order, ctx->prev_raw_samples,
sizeof(*raw_samples) * sconf->max_order);
return 0;
}
/** Reads the block data.
*/
static int read_block_data(ALSDecContext *ctx, unsigned int ra_block,
int32_t *raw_samples, unsigned int block_length,
unsigned int *js_blocks, int32_t *raw_other)
{
ALSSpecificConfig *sconf = &ctx->sconf;
GetBitContext *gb = &ctx->gb;
unsigned int shift_lsbs = 0;
unsigned int k;
// read block type flag and read the samples accordingly
if (get_bits1(gb)) {
if (read_var_block(ctx, ra_block, raw_samples, block_length, js_blocks,
raw_other, &shift_lsbs))
return -1;
} else {
read_const_block(ctx, raw_samples, block_length, js_blocks);
}
// TODO: read RLSLMS extension data
if (!sconf->mc_coding || ctx->js_switch)
align_get_bits(gb);
if (shift_lsbs)
for (k = 0; k < block_length; k++)
raw_samples[k] <<= shift_lsbs;
return 0;
}
/** Computes the number of samples left to decode for the current frame and
* sets these samples to zero.
*/
static void zero_remaining(unsigned int b, unsigned int b_max,
const unsigned int *div_blocks, int32_t *buf)
{
unsigned int count = 0;
while (b < b_max)
count += div_blocks[b];
memset(buf, 0, sizeof(*buf) * count);
}
/** Decodes blocks independently.
*/
static int decode_blocks_ind(ALSDecContext *ctx, unsigned int ra_frame,
unsigned int c, const unsigned int *div_blocks,
unsigned int *js_blocks)
{
int32_t *raw_sample;
unsigned int b;
raw_sample = ctx->raw_samples[c];
for (b = 0; b < ctx->num_blocks; b++) {
if (read_block_data(ctx, ra_frame, raw_sample,
div_blocks[b], &js_blocks[0], NULL)) {
// damaged block, write zero for the rest of the frame
zero_remaining(b, ctx->num_blocks, div_blocks, raw_sample);
return -1;
}
raw_sample += div_blocks[b];
ra_frame = 0;
}
return 0;
}
/** Decodes blocks dependently.
*/
static int decode_blocks(ALSDecContext *ctx, unsigned int ra_frame,
unsigned int c, const unsigned int *div_blocks,
unsigned int *js_blocks)
{
ALSSpecificConfig *sconf = &ctx->sconf;
unsigned int offset = 0;
int32_t *raw_samples_R;
int32_t *raw_samples_L;
unsigned int b;
// decode all blocks
for (b = 0; b < ctx->num_blocks; b++) {
unsigned int s;
raw_samples_L = ctx->raw_samples[c ] + offset;
raw_samples_R = ctx->raw_samples[c + 1] + offset;
if (read_block_data(ctx, ra_frame, raw_samples_L, div_blocks[b],
&js_blocks[0], raw_samples_R) ||
read_block_data(ctx, ra_frame, raw_samples_R, div_blocks[b],
&js_blocks[1], raw_samples_L)) {
// damaged block, write zero for the rest of the frame
zero_remaining(b, ctx->num_blocks, div_blocks, raw_samples_L);
zero_remaining(b, ctx->num_blocks, div_blocks, raw_samples_R);
return -1;
}
// reconstruct joint-stereo blocks
if (js_blocks[0]) {
if (js_blocks[1])
av_log(ctx->avctx, AV_LOG_WARNING, "Invalid channel pair!\n");
for (s = 0; s < div_blocks[b]; s++)
raw_samples_L[s] = raw_samples_R[s] - raw_samples_L[s];
} else if (js_blocks[1]) {
for (s = 0; s < div_blocks[b]; s++)
raw_samples_R[s] = raw_samples_R[s] + raw_samples_L[s];
}
offset += div_blocks[b];
ra_frame = 0;
}
// store carryover raw samples,
// the others channel raw samples are stored by the calling function.
memmove(ctx->raw_samples[c] - sconf->max_order,
ctx->raw_samples[c] - sconf->max_order + sconf->frame_length,
sizeof(*ctx->raw_samples[c]) * sconf->max_order);
return 0;
}
/** Reads the frame data.
*/
static int read_frame_data(ALSDecContext *ctx, unsigned int ra_frame)
{
ALSSpecificConfig *sconf = &ctx->sconf;
AVCodecContext *avctx = ctx->avctx;
GetBitContext *gb = &ctx->gb;
unsigned int div_blocks[32]; ///< block sizes.
unsigned int c;
unsigned int js_blocks[2];
uint32_t bs_info = 0;
// skip the size of the ra unit if present in the frame
if (sconf->ra_flag == RA_FLAG_FRAMES && ra_frame)
skip_bits_long(gb, 32);
if (sconf->mc_coding && sconf->joint_stereo) {
ctx->js_switch = get_bits1(gb);
align_get_bits(gb);
}
if (!sconf->mc_coding || ctx->js_switch) {
int independent_bs = !sconf->joint_stereo;
for (c = 0; c < avctx->channels; c++) {
js_blocks[0] = 0;
js_blocks[1] = 0;
get_block_sizes(ctx, div_blocks, &bs_info);
// if joint_stereo and block_switching is set, independent decoding
// is signaled via the first bit of bs_info
if (sconf->joint_stereo && sconf->block_switching)
if (bs_info >> 31)
independent_bs = 2;
// if this is the last channel, it has to be decoded independently
if (c == avctx->channels - 1)
independent_bs = 1;
if (independent_bs) {
if (decode_blocks_ind(ctx, ra_frame, c, div_blocks, js_blocks))
return -1;
independent_bs--;
} else {
if (decode_blocks(ctx, ra_frame, c, div_blocks, js_blocks))
return -1;
c++;
}
// store carryover raw samples
memmove(ctx->raw_samples[c] - sconf->max_order,
ctx->raw_samples[c] - sconf->max_order + sconf->frame_length,
sizeof(*ctx->raw_samples[c]) * sconf->max_order);
}
} else { // multi-channel coding
get_block_sizes(ctx, div_blocks, &bs_info);
// TODO: multi channel coding might use a temporary buffer instead as
// the actual channel is not known when read_block-data is called
if (decode_blocks_ind(ctx, ra_frame, 0, div_blocks, js_blocks))
return -1;
// TODO: read_channel_data
}
// TODO: read_diff_float_data
return 0;
}
/** Decodes an ALS frame.
*/
static int decode_frame(AVCodecContext *avctx,
void *data, int *data_size,
AVPacket *avpkt)
{
ALSDecContext *ctx = avctx->priv_data;
ALSSpecificConfig *sconf = &ctx->sconf;
const uint8_t *buffer = avpkt->data;
int buffer_size = avpkt->size;
int invalid_frame, size;
unsigned int c, sample, ra_frame, bytes_read, shift;
init_get_bits(&ctx->gb, buffer, buffer_size * 8);
// In the case that the distance between random access frames is set to zero
// (sconf->ra_distance == 0) no frame is treated as a random access frame.
// For the first frame, if prediction is used, all samples used from the
// previous frame are assumed to be zero.
ra_frame = sconf->ra_distance && !(ctx->frame_id % sconf->ra_distance);
// the last frame to decode might have a different length
if (sconf->samples != 0xFFFFFFFF)
ctx->cur_frame_length = FFMIN(sconf->samples - ctx->frame_id * (uint64_t) sconf->frame_length,
sconf->frame_length);
else
ctx->cur_frame_length = sconf->frame_length;
// decode the frame data
if ((invalid_frame = read_frame_data(ctx, ra_frame) < 0))
av_log(ctx->avctx, AV_LOG_WARNING,
"Reading frame data failed. Skipping RA unit.\n");
ctx->frame_id++;
// check for size of decoded data
size = ctx->cur_frame_length * avctx->channels *
(av_get_bits_per_sample_format(avctx->sample_fmt) >> 3);
if (size > *data_size) {
av_log(avctx, AV_LOG_ERROR, "Decoded data exceeds buffer size.\n");
return -1;
}
*data_size = size;
// transform decoded frame into output format
#define INTERLEAVE_OUTPUT(bps) \
{ \
int##bps##_t *dest = (int##bps##_t*) data; \
shift = bps - ctx->avctx->bits_per_raw_sample; \
for (sample = 0; sample < ctx->cur_frame_length; sample++) \
for (c = 0; c < avctx->channels; c++) \
*dest++ = ctx->raw_samples[c][sample] << shift; \
}
if (ctx->avctx->bits_per_raw_sample <= 16) {
INTERLEAVE_OUTPUT(16)
} else {
INTERLEAVE_OUTPUT(32)
}
bytes_read = invalid_frame ? buffer_size :
(get_bits_count(&ctx->gb) + 7) >> 3;
return bytes_read;
}
/** Uninitializes the ALS decoder.
*/
static av_cold int decode_end(AVCodecContext *avctx)
{
ALSDecContext *ctx = avctx->priv_data;
av_freep(&ctx->sconf.chan_pos);
av_freep(&ctx->quant_cof);
av_freep(&ctx->lpc_cof);
av_freep(&ctx->prev_raw_samples);
av_freep(&ctx->raw_samples);
av_freep(&ctx->raw_buffer);
return 0;
}
/** Initializes the ALS decoder.
*/
static av_cold int decode_init(AVCodecContext *avctx)
{
unsigned int c;
unsigned int channel_size;
ALSDecContext *ctx = avctx->priv_data;
ALSSpecificConfig *sconf = &ctx->sconf;
ctx->avctx = avctx;
if (!avctx->extradata) {
av_log(avctx, AV_LOG_ERROR, "Missing required ALS extradata.\n");
return -1;
}
if (read_specific_config(ctx)) {
av_log(avctx, AV_LOG_ERROR, "Reading ALSSpecificConfig failed.\n");
decode_end(avctx);
return -1;
}
if (check_specific_config(ctx)) {
decode_end(avctx);
return -1;
}
if (sconf->floating) {
avctx->sample_fmt = SAMPLE_FMT_FLT;
avctx->bits_per_raw_sample = 32;
} else {
avctx->sample_fmt = sconf->resolution > 1
? SAMPLE_FMT_S32 : SAMPLE_FMT_S16;
avctx->bits_per_raw_sample = (sconf->resolution + 1) * 8;
}
avctx->frame_size = sconf->frame_length;
channel_size = sconf->frame_length + sconf->max_order;
ctx->prev_raw_samples = av_malloc (sizeof(*ctx->prev_raw_samples) * sconf->max_order);
ctx->raw_buffer = av_mallocz(sizeof(*ctx-> raw_buffer) * avctx->channels * channel_size);
ctx->raw_samples = av_malloc (sizeof(*ctx-> raw_samples) * avctx->channels);
// allocate previous raw sample buffer
if (!ctx->prev_raw_samples || !ctx->raw_buffer|| !ctx->raw_samples) {
av_log(avctx, AV_LOG_ERROR, "Allocating buffer memory failed.\n");
decode_end(avctx);
return AVERROR(ENOMEM);
}
// assign raw samples buffers
ctx->raw_samples[0] = ctx->raw_buffer + sconf->max_order;
for (c = 1; c < avctx->channels; c++)
ctx->raw_samples[c] = ctx->raw_samples[c - 1] + channel_size;
return 0;
}
/** Flushes (resets) the frame ID after seeking.
*/
static av_cold void flush(AVCodecContext *avctx)
{
ALSDecContext *ctx = avctx->priv_data;
ctx->frame_id = 0;
}
AVCodec als_decoder = {
"als",
CODEC_TYPE_AUDIO,
CODEC_ID_MP4ALS,
sizeof(ALSDecContext),
decode_init,
NULL,
decode_end,
decode_frame,
.flush = flush,
.capabilities = CODEC_CAP_SUBFRAMES,
.long_name = NULL_IF_CONFIG_SMALL("MPEG-4 Audio Lossless Coding (ALS)"),
};
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