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ffmpeg/libavcodec/flac.c
Michael Niedermayer ac66834c75 avcodec_decode_audio2() 
difference to avcodec_decode_audio() is that the user can pass the allocated size of the output buffer to the decoder and the decoder can check if theres enough space

Originally committed as revision 7518 to svn://svn.ffmpeg.org/ffmpeg/trunk
2007-01-14 23:50:06 +00:00

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/*
* FLAC (Free Lossless Audio Codec) decoder
* Copyright (c) 2003 Alex Beregszaszi
*
* 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 flac.c
* FLAC (Free Lossless Audio Codec) decoder
* @author Alex Beregszaszi
*
* For more information on the FLAC format, visit:
* http://flac.sourceforge.net/
*
* This decoder can be used in 1 of 2 ways: Either raw FLAC data can be fed
* through, starting from the initial 'fLaC' signature; or by passing the
* 34-byte streaminfo structure through avctx->extradata[_size] followed
* by data starting with the 0xFFF8 marker.
*/
#include <limits.h>
#define ALT_BITSTREAM_READER
#include "avcodec.h"
#include "bitstream.h"
#include "golomb.h"
#include "crc.h"
#undef NDEBUG
#include <assert.h>
#define MAX_CHANNELS 8
#define MAX_BLOCKSIZE 65535
#define FLAC_STREAMINFO_SIZE 34
enum decorrelation_type {
INDEPENDENT,
LEFT_SIDE,
RIGHT_SIDE,
MID_SIDE,
};
typedef struct FLACContext {
AVCodecContext *avctx;
GetBitContext gb;
int min_blocksize, max_blocksize;
int min_framesize, max_framesize;
int samplerate, channels;
int blocksize/*, last_blocksize*/;
int bps, curr_bps;
enum decorrelation_type decorrelation;
int32_t *decoded[MAX_CHANNELS];
uint8_t *bitstream;
int bitstream_size;
int bitstream_index;
unsigned int allocated_bitstream_size;
} FLACContext;
#define METADATA_TYPE_STREAMINFO 0
static int sample_rate_table[] =
{ 0, 0, 0, 0,
8000, 16000, 22050, 24000, 32000, 44100, 48000, 96000,
0, 0, 0, 0 };
static int sample_size_table[] =
{ 0, 8, 12, 0, 16, 20, 24, 0 };
static int blocksize_table[] = {
0, 192, 576<<0, 576<<1, 576<<2, 576<<3, 0, 0,
256<<0, 256<<1, 256<<2, 256<<3, 256<<4, 256<<5, 256<<6, 256<<7
};
static int64_t get_utf8(GetBitContext *gb){
int64_t val;
GET_UTF8(val, get_bits(gb, 8), return -1;)
return val;
}
static void metadata_streaminfo(FLACContext *s);
static void allocate_buffers(FLACContext *s);
static int metadata_parse(FLACContext *s);
static int flac_decode_init(AVCodecContext * avctx)
{
FLACContext *s = avctx->priv_data;
s->avctx = avctx;
if (avctx->extradata_size > 4) {
/* initialize based on the demuxer-supplied streamdata header */
init_get_bits(&s->gb, avctx->extradata, avctx->extradata_size*8);
if (avctx->extradata_size == FLAC_STREAMINFO_SIZE) {
metadata_streaminfo(s);
allocate_buffers(s);
} else {
metadata_parse(s);
}
}
return 0;
}
static void dump_headers(FLACContext *s)
{
av_log(s->avctx, AV_LOG_DEBUG, " Blocksize: %d .. %d (%d)\n", s->min_blocksize, s->max_blocksize, s->blocksize);
av_log(s->avctx, AV_LOG_DEBUG, " Framesize: %d .. %d\n", s->min_framesize, s->max_framesize);
av_log(s->avctx, AV_LOG_DEBUG, " Samplerate: %d\n", s->samplerate);
av_log(s->avctx, AV_LOG_DEBUG, " Channels: %d\n", s->channels);
av_log(s->avctx, AV_LOG_DEBUG, " Bits: %d\n", s->bps);
}
static void allocate_buffers(FLACContext *s){
int i;
assert(s->max_blocksize);
if(s->max_framesize == 0 && s->max_blocksize){
s->max_framesize= (s->channels * s->bps * s->max_blocksize + 7)/ 8; //FIXME header overhead
}
for (i = 0; i < s->channels; i++)
{
s->decoded[i] = av_realloc(s->decoded[i], sizeof(int32_t)*s->max_blocksize);
}
s->bitstream= av_fast_realloc(s->bitstream, &s->allocated_bitstream_size, s->max_framesize);
}
static void metadata_streaminfo(FLACContext *s)
{
/* mandatory streaminfo */
s->min_blocksize = get_bits(&s->gb, 16);
s->max_blocksize = get_bits(&s->gb, 16);
s->min_framesize = get_bits_long(&s->gb, 24);
s->max_framesize = get_bits_long(&s->gb, 24);
s->samplerate = get_bits_long(&s->gb, 20);
s->channels = get_bits(&s->gb, 3) + 1;
s->bps = get_bits(&s->gb, 5) + 1;
s->avctx->channels = s->channels;
s->avctx->sample_rate = s->samplerate;
skip_bits(&s->gb, 36); /* total num of samples */
skip_bits(&s->gb, 64); /* md5 sum */
skip_bits(&s->gb, 64); /* md5 sum */
dump_headers(s);
}
/**
* Parse a list of metadata blocks. This list of blocks must begin with
* the fLaC marker.
* @param s the flac decoding context containing the gb bit reader used to
* parse metadata
* @return 1 if some metadata was read, 0 if no fLaC marker was found
*/
static int metadata_parse(FLACContext *s)
{
int i, metadata_last, metadata_type, metadata_size, streaminfo_updated=0;
if (show_bits_long(&s->gb, 32) == MKBETAG('f','L','a','C')) {
skip_bits(&s->gb, 32);
av_log(s->avctx, AV_LOG_DEBUG, "STREAM HEADER\n");
do {
metadata_last = get_bits(&s->gb, 1);
metadata_type = get_bits(&s->gb, 7);
metadata_size = get_bits_long(&s->gb, 24);
av_log(s->avctx, AV_LOG_DEBUG,
" metadata block: flag = %d, type = %d, size = %d\n",
metadata_last, metadata_type, metadata_size);
if (metadata_size) {
switch (metadata_type) {
case METADATA_TYPE_STREAMINFO:
metadata_streaminfo(s);
streaminfo_updated = 1;
break;
default:
for (i=0; i<metadata_size; i++)
skip_bits(&s->gb, 8);
}
}
} while (!metadata_last);
if (streaminfo_updated)
allocate_buffers(s);
return 1;
}
return 0;
}
static int decode_residuals(FLACContext *s, int channel, int pred_order)
{
int i, tmp, partition, method_type, rice_order;
int sample = 0, samples;
method_type = get_bits(&s->gb, 2);
if (method_type != 0){
av_log(s->avctx, AV_LOG_DEBUG, "illegal residual coding method %d\n", method_type);
return -1;
}
rice_order = get_bits(&s->gb, 4);
samples= s->blocksize >> rice_order;
sample=
i= pred_order;
for (partition = 0; partition < (1 << rice_order); partition++)
{
tmp = get_bits(&s->gb, 4);
if (tmp == 15)
{
av_log(s->avctx, AV_LOG_DEBUG, "fixed len partition\n");
tmp = get_bits(&s->gb, 5);
for (; i < samples; i++, sample++)
s->decoded[channel][sample] = get_sbits(&s->gb, tmp);
}
else
{
// av_log(s->avctx, AV_LOG_DEBUG, "rice coded partition k=%d\n", tmp);
for (; i < samples; i++, sample++){
s->decoded[channel][sample] = get_sr_golomb_flac(&s->gb, tmp, INT_MAX, 0);
}
}
i= 0;
}
// av_log(s->avctx, AV_LOG_DEBUG, "partitions: %d, samples: %d\n", 1 << rice_order, sample);
return 0;
}
static int decode_subframe_fixed(FLACContext *s, int channel, int pred_order)
{
int i;
// av_log(s->avctx, AV_LOG_DEBUG, " SUBFRAME FIXED\n");
/* warm up samples */
// av_log(s->avctx, AV_LOG_DEBUG, " warm up samples: %d\n", pred_order);
for (i = 0; i < pred_order; i++)
{
s->decoded[channel][i] = get_sbits(&s->gb, s->curr_bps);
// av_log(s->avctx, AV_LOG_DEBUG, " %d: %d\n", i, s->decoded[channel][i]);
}
if (decode_residuals(s, channel, pred_order) < 0)
return -1;
switch(pred_order)
{
case 0:
break;
case 1:
for (i = pred_order; i < s->blocksize; i++)
s->decoded[channel][i] += s->decoded[channel][i-1];
break;
case 2:
for (i = pred_order; i < s->blocksize; i++)
s->decoded[channel][i] += 2*s->decoded[channel][i-1]
- s->decoded[channel][i-2];
break;
case 3:
for (i = pred_order; i < s->blocksize; i++)
s->decoded[channel][i] += 3*s->decoded[channel][i-1]
- 3*s->decoded[channel][i-2]
+ s->decoded[channel][i-3];
break;
case 4:
for (i = pred_order; i < s->blocksize; i++)
s->decoded[channel][i] += 4*s->decoded[channel][i-1]
- 6*s->decoded[channel][i-2]
+ 4*s->decoded[channel][i-3]
- s->decoded[channel][i-4];
break;
default:
av_log(s->avctx, AV_LOG_ERROR, "illegal pred order %d\n", pred_order);
return -1;
}
return 0;
}
static int decode_subframe_lpc(FLACContext *s, int channel, int pred_order)
{
int i, j;
int coeff_prec, qlevel;
int coeffs[pred_order];
// av_log(s->avctx, AV_LOG_DEBUG, " SUBFRAME LPC\n");
/* warm up samples */
// av_log(s->avctx, AV_LOG_DEBUG, " warm up samples: %d\n", pred_order);
for (i = 0; i < pred_order; i++)
{
s->decoded[channel][i] = get_sbits(&s->gb, s->curr_bps);
// av_log(s->avctx, AV_LOG_DEBUG, " %d: %d\n", i, s->decoded[channel][i]);
}
coeff_prec = get_bits(&s->gb, 4) + 1;
if (coeff_prec == 16)
{
av_log(s->avctx, AV_LOG_DEBUG, "invalid coeff precision\n");
return -1;
}
// av_log(s->avctx, AV_LOG_DEBUG, " qlp coeff prec: %d\n", coeff_prec);
qlevel = get_sbits(&s->gb, 5);
// av_log(s->avctx, AV_LOG_DEBUG, " quant level: %d\n", qlevel);
if(qlevel < 0){
av_log(s->avctx, AV_LOG_DEBUG, "qlevel %d not supported, maybe buggy stream\n", qlevel);
return -1;
}
for (i = 0; i < pred_order; i++)
{
coeffs[i] = get_sbits(&s->gb, coeff_prec);
// av_log(s->avctx, AV_LOG_DEBUG, " %d: %d\n", i, coeffs[i]);
}
if (decode_residuals(s, channel, pred_order) < 0)
return -1;
if (s->bps > 16) {
int64_t sum;
for (i = pred_order; i < s->blocksize; i++)
{
sum = 0;
for (j = 0; j < pred_order; j++)
sum += (int64_t)coeffs[j] * s->decoded[channel][i-j-1];
s->decoded[channel][i] += sum >> qlevel;
}
} else {
int sum;
for (i = pred_order; i < s->blocksize; i++)
{
sum = 0;
for (j = 0; j < pred_order; j++)
sum += coeffs[j] * s->decoded[channel][i-j-1];
s->decoded[channel][i] += sum >> qlevel;
}
}
return 0;
}
static inline int decode_subframe(FLACContext *s, int channel)
{
int type, wasted = 0;
int i, tmp;
s->curr_bps = s->bps;
if(channel == 0){
if(s->decorrelation == RIGHT_SIDE)
s->curr_bps++;
}else{
if(s->decorrelation == LEFT_SIDE || s->decorrelation == MID_SIDE)
s->curr_bps++;
}
if (get_bits1(&s->gb))
{
av_log(s->avctx, AV_LOG_ERROR, "invalid subframe padding\n");
return -1;
}
type = get_bits(&s->gb, 6);
// wasted = get_bits1(&s->gb);
// if (wasted)
// {
// while (!get_bits1(&s->gb))
// wasted++;
// if (wasted)
// wasted++;
// s->curr_bps -= wasted;
// }
#if 0
wasted= 16 - av_log2(show_bits(&s->gb, 17));
skip_bits(&s->gb, wasted+1);
s->curr_bps -= wasted;
#else
if (get_bits1(&s->gb))
{
wasted = 1;
while (!get_bits1(&s->gb))
wasted++;
s->curr_bps -= wasted;
av_log(s->avctx, AV_LOG_DEBUG, "%d wasted bits\n", wasted);
}
#endif
//FIXME use av_log2 for types
if (type == 0)
{
av_log(s->avctx, AV_LOG_DEBUG, "coding type: constant\n");
tmp = get_sbits(&s->gb, s->curr_bps);
for (i = 0; i < s->blocksize; i++)
s->decoded[channel][i] = tmp;
}
else if (type == 1)
{
av_log(s->avctx, AV_LOG_DEBUG, "coding type: verbatim\n");
for (i = 0; i < s->blocksize; i++)
s->decoded[channel][i] = get_sbits(&s->gb, s->curr_bps);
}
else if ((type >= 8) && (type <= 12))
{
// av_log(s->avctx, AV_LOG_DEBUG, "coding type: fixed\n");
if (decode_subframe_fixed(s, channel, type & ~0x8) < 0)
return -1;
}
else if (type >= 32)
{
// av_log(s->avctx, AV_LOG_DEBUG, "coding type: lpc\n");
if (decode_subframe_lpc(s, channel, (type & ~0x20)+1) < 0)
return -1;
}
else
{
av_log(s->avctx, AV_LOG_ERROR, "invalid coding type\n");
return -1;
}
if (wasted)
{
int i;
for (i = 0; i < s->blocksize; i++)
s->decoded[channel][i] <<= wasted;
}
return 0;
}
static int decode_frame(FLACContext *s, int alloc_data_size)
{
int blocksize_code, sample_rate_code, sample_size_code, assignment, i, crc8;
int decorrelation, bps, blocksize, samplerate;
blocksize_code = get_bits(&s->gb, 4);
sample_rate_code = get_bits(&s->gb, 4);
assignment = get_bits(&s->gb, 4); /* channel assignment */
if (assignment < 8 && s->channels == assignment+1)
decorrelation = INDEPENDENT;
else if (assignment >=8 && assignment < 11 && s->channels == 2)
decorrelation = LEFT_SIDE + assignment - 8;
else
{
av_log(s->avctx, AV_LOG_ERROR, "unsupported channel assignment %d (channels=%d)\n", assignment, s->channels);
return -1;
}
sample_size_code = get_bits(&s->gb, 3);
if(sample_size_code == 0)
bps= s->bps;
else if((sample_size_code != 3) && (sample_size_code != 7))
bps = sample_size_table[sample_size_code];
else
{
av_log(s->avctx, AV_LOG_ERROR, "invalid sample size code (%d)\n", sample_size_code);
return -1;
}
if (get_bits1(&s->gb))
{
av_log(s->avctx, AV_LOG_ERROR, "broken stream, invalid padding\n");
return -1;
}
if(get_utf8(&s->gb) < 0){
av_log(s->avctx, AV_LOG_ERROR, "utf8 fscked\n");
return -1;
}
#if 0
if (/*((blocksize_code == 6) || (blocksize_code == 7)) &&*/
(s->min_blocksize != s->max_blocksize)){
}else{
}
#endif
if (blocksize_code == 0)
blocksize = s->min_blocksize;
else if (blocksize_code == 6)
blocksize = get_bits(&s->gb, 8)+1;
else if (blocksize_code == 7)
blocksize = get_bits(&s->gb, 16)+1;
else
blocksize = blocksize_table[blocksize_code];
if(blocksize > s->max_blocksize){
av_log(s->avctx, AV_LOG_ERROR, "blocksize %d > %d\n", blocksize, s->max_blocksize);
return -1;
}
if(blocksize * s->channels * sizeof(int16_t) > alloc_data_size)
return -1;
if (sample_rate_code == 0){
samplerate= s->samplerate;
}else if ((sample_rate_code > 3) && (sample_rate_code < 12))
samplerate = sample_rate_table[sample_rate_code];
else if (sample_rate_code == 12)
samplerate = get_bits(&s->gb, 8) * 1000;
else if (sample_rate_code == 13)
samplerate = get_bits(&s->gb, 16);
else if (sample_rate_code == 14)
samplerate = get_bits(&s->gb, 16) * 10;
else{
av_log(s->avctx, AV_LOG_ERROR, "illegal sample rate code %d\n", sample_rate_code);
return -1;
}
skip_bits(&s->gb, 8);
crc8= av_crc(av_crc07, 0, s->gb.buffer, get_bits_count(&s->gb)/8);
if(crc8){
av_log(s->avctx, AV_LOG_ERROR, "header crc mismatch crc=%2X\n", crc8);
return -1;
}
s->blocksize = blocksize;
s->samplerate = samplerate;
s->bps = bps;
s->decorrelation= decorrelation;
// dump_headers(s);
/* subframes */
for (i = 0; i < s->channels; i++)
{
// av_log(s->avctx, AV_LOG_DEBUG, "decoded: %x residual: %x\n", s->decoded[i], s->residual[i]);
if (decode_subframe(s, i) < 0)
return -1;
}
align_get_bits(&s->gb);
/* frame footer */
skip_bits(&s->gb, 16); /* data crc */
return 0;
}
static inline int16_t shift_to_16_bits(int32_t data, int bps)
{
if (bps == 24) {
return (data >> 8);
} else if (bps == 20) {
return (data >> 4);
} else {
return data;
}
}
static int flac_decode_frame(AVCodecContext *avctx,
void *data, int *data_size,
uint8_t *buf, int buf_size)
{
FLACContext *s = avctx->priv_data;
int tmp = 0, i, j = 0, input_buf_size = 0;
int16_t *samples = data;
int alloc_data_size= *data_size;
*data_size=0;
if(s->max_framesize == 0){
s->max_framesize= 65536; // should hopefully be enough for the first header
s->bitstream= av_fast_realloc(s->bitstream, &s->allocated_bitstream_size, s->max_framesize);
}
if(1 && s->max_framesize){//FIXME truncated
buf_size= FFMAX(FFMIN(buf_size, s->max_framesize - s->bitstream_size), 0);
input_buf_size= buf_size;
if(s->bitstream_index + s->bitstream_size + buf_size > s->allocated_bitstream_size){
// printf("memmove\n");
memmove(s->bitstream, &s->bitstream[s->bitstream_index], s->bitstream_size);
s->bitstream_index=0;
}
memcpy(&s->bitstream[s->bitstream_index + s->bitstream_size], buf, buf_size);
buf= &s->bitstream[s->bitstream_index];
buf_size += s->bitstream_size;
s->bitstream_size= buf_size;
if(buf_size < s->max_framesize){
// printf("wanna more data ...\n");
return input_buf_size;
}
}
init_get_bits(&s->gb, buf, buf_size*8);
if (!metadata_parse(s))
{
tmp = show_bits(&s->gb, 16);
if(tmp != 0xFFF8){
av_log(s->avctx, AV_LOG_ERROR, "FRAME HEADER not here\n");
while(get_bits_count(&s->gb)/8+2 < buf_size && show_bits(&s->gb, 16) != 0xFFF8)
skip_bits(&s->gb, 8);
goto end; // we may not have enough bits left to decode a frame, so try next time
}
skip_bits(&s->gb, 16);
if (decode_frame(s, alloc_data_size) < 0){
av_log(s->avctx, AV_LOG_ERROR, "decode_frame() failed\n");
s->bitstream_size=0;
s->bitstream_index=0;
return -1;
}
}
#if 0
/* fix the channel order here */
if (s->order == MID_SIDE)
{
short *left = samples;
short *right = samples + s->blocksize;
for (i = 0; i < s->blocksize; i += 2)
{
uint32_t x = s->decoded[0][i];
uint32_t y = s->decoded[0][i+1];
right[i] = x - (y / 2);
left[i] = right[i] + y;
}
*data_size = 2 * s->blocksize;
}
else
{
for (i = 0; i < s->channels; i++)
{
switch(s->order)
{
case INDEPENDENT:
for (j = 0; j < s->blocksize; j++)
samples[(s->blocksize*i)+j] = s->decoded[i][j];
break;
case LEFT_SIDE:
case RIGHT_SIDE:
if (i == 0)
for (j = 0; j < s->blocksize; j++)
samples[(s->blocksize*i)+j] = s->decoded[0][j];
else
for (j = 0; j < s->blocksize; j++)
samples[(s->blocksize*i)+j] = s->decoded[0][j] - s->decoded[i][j];
break;
// case MID_SIDE:
// av_log(s->avctx, AV_LOG_DEBUG, "mid-side unsupported\n");
}
*data_size += s->blocksize;
}
}
#else
#define DECORRELATE(left, right)\
assert(s->channels == 2);\
for (i = 0; i < s->blocksize; i++)\
{\
int a= s->decoded[0][i];\
int b= s->decoded[1][i];\
*(samples++) = (left ) >> (16 - s->bps);\
*(samples++) = (right) >> (16 - s->bps);\
}\
break;
switch(s->decorrelation)
{
case INDEPENDENT:
for (j = 0; j < s->blocksize; j++)
{
for (i = 0; i < s->channels; i++)
*(samples++) = shift_to_16_bits(s->decoded[i][j], s->bps);
}
break;
case LEFT_SIDE:
DECORRELATE(a,a-b)
case RIGHT_SIDE:
DECORRELATE(a+b,b)
case MID_SIDE:
DECORRELATE( (a-=b>>1) + b, a)
}
#endif
*data_size = (int8_t *)samples - (int8_t *)data;
// av_log(s->avctx, AV_LOG_DEBUG, "data size: %d\n", *data_size);
// s->last_blocksize = s->blocksize;
end:
i= (get_bits_count(&s->gb)+7)/8;;
if(i > buf_size){
av_log(s->avctx, AV_LOG_ERROR, "overread: %d\n", i - buf_size);
s->bitstream_size=0;
s->bitstream_index=0;
return -1;
}
if(s->bitstream_size){
s->bitstream_index += i;
s->bitstream_size -= i;
return input_buf_size;
}else
return i;
}
static int flac_decode_close(AVCodecContext *avctx)
{
FLACContext *s = avctx->priv_data;
int i;
for (i = 0; i < s->channels; i++)
{
av_freep(&s->decoded[i]);
}
av_freep(&s->bitstream);
return 0;
}
static void flac_flush(AVCodecContext *avctx){
FLACContext *s = avctx->priv_data;
s->bitstream_size=
s->bitstream_index= 0;
}
AVCodec flac_decoder = {
"flac",
CODEC_TYPE_AUDIO,
CODEC_ID_FLAC,
sizeof(FLACContext),
flac_decode_init,
NULL,
flac_decode_close,
flac_decode_frame,
.flush= flac_flush,
};
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