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FFmpeg/libavcodec/flac.c
Michael Niedermayer 1e31d32c34 flac_flush() 
Originally committed as revision 2797 to svn://svn.ffmpeg.org/ffmpeg/trunk
2004-02-18 15:17:54 +00:00

774 lines
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/*
* FLAC (Free Lossless Audio Codec) decoder
* Copyright (c) 2003 Alex Beregszaszi
*
* This library 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 of the License, or (at your option) any later version.
*
* This library 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 this library; if not, write to the Free Software
* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
*/
/**
* @file flac.c
* FLAC (Free Lossless Audio Codec) decoder
* @author Alex Beregszaszi
*/
#include <limits.h>
#include "avcodec.h"
#include "golomb.h"
#undef NDEBUG
#include <assert.h>
#define MAX_CHANNELS 8
#define MAX_BLOCKSIZE 65535
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;
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 const uint8_t table_crc8[256] = {
0x00, 0x07, 0x0e, 0x09, 0x1c, 0x1b, 0x12, 0x15,
0x38, 0x3f, 0x36, 0x31, 0x24, 0x23, 0x2a, 0x2d,
0x70, 0x77, 0x7e, 0x79, 0x6c, 0x6b, 0x62, 0x65,
0x48, 0x4f, 0x46, 0x41, 0x54, 0x53, 0x5a, 0x5d,
0xe0, 0xe7, 0xee, 0xe9, 0xfc, 0xfb, 0xf2, 0xf5,
0xd8, 0xdf, 0xd6, 0xd1, 0xc4, 0xc3, 0xca, 0xcd,
0x90, 0x97, 0x9e, 0x99, 0x8c, 0x8b, 0x82, 0x85,
0xa8, 0xaf, 0xa6, 0xa1, 0xb4, 0xb3, 0xba, 0xbd,
0xc7, 0xc0, 0xc9, 0xce, 0xdb, 0xdc, 0xd5, 0xd2,
0xff, 0xf8, 0xf1, 0xf6, 0xe3, 0xe4, 0xed, 0xea,
0xb7, 0xb0, 0xb9, 0xbe, 0xab, 0xac, 0xa5, 0xa2,
0x8f, 0x88, 0x81, 0x86, 0x93, 0x94, 0x9d, 0x9a,
0x27, 0x20, 0x29, 0x2e, 0x3b, 0x3c, 0x35, 0x32,
0x1f, 0x18, 0x11, 0x16, 0x03, 0x04, 0x0d, 0x0a,
0x57, 0x50, 0x59, 0x5e, 0x4b, 0x4c, 0x45, 0x42,
0x6f, 0x68, 0x61, 0x66, 0x73, 0x74, 0x7d, 0x7a,
0x89, 0x8e, 0x87, 0x80, 0x95, 0x92, 0x9b, 0x9c,
0xb1, 0xb6, 0xbf, 0xb8, 0xad, 0xaa, 0xa3, 0xa4,
0xf9, 0xfe, 0xf7, 0xf0, 0xe5, 0xe2, 0xeb, 0xec,
0xc1, 0xc6, 0xcf, 0xc8, 0xdd, 0xda, 0xd3, 0xd4,
0x69, 0x6e, 0x67, 0x60, 0x75, 0x72, 0x7b, 0x7c,
0x51, 0x56, 0x5f, 0x58, 0x4d, 0x4a, 0x43, 0x44,
0x19, 0x1e, 0x17, 0x10, 0x05, 0x02, 0x0b, 0x0c,
0x21, 0x26, 0x2f, 0x28, 0x3d, 0x3a, 0x33, 0x34,
0x4e, 0x49, 0x40, 0x47, 0x52, 0x55, 0x5c, 0x5b,
0x76, 0x71, 0x78, 0x7f, 0x6a, 0x6d, 0x64, 0x63,
0x3e, 0x39, 0x30, 0x37, 0x22, 0x25, 0x2c, 0x2b,
0x06, 0x01, 0x08, 0x0f, 0x1a, 0x1d, 0x14, 0x13,
0xae, 0xa9, 0xa0, 0xa7, 0xb2, 0xb5, 0xbc, 0xbb,
0x96, 0x91, 0x98, 0x9f, 0x8a, 0x8d, 0x84, 0x83,
0xde, 0xd9, 0xd0, 0xd7, 0xc2, 0xc5, 0xcc, 0xcb,
0xe6, 0xe1, 0xe8, 0xef, 0xfa, 0xfd, 0xf4, 0xf3
};
static int64_t get_utf8(GetBitContext *gb)
{
uint64_t val;
int ones=0, bytes;
while(get_bits1(gb))
ones++;
if (ones==0) bytes=0;
else if(ones==1) return -1;
else bytes= ones - 1;
val= get_bits(gb, 7-ones);
while(bytes--){
const int tmp = get_bits(gb, 8);
if((tmp>>6) != 2)
return -1;
val<<=6;
val|= tmp&0x3F;
}
return val;
}
static int get_crc8(uint8_t *buf, int count){
int crc=0;
int i;
for(i=0; i<count; i++){
crc = table_crc8[crc ^ buf[i]];
}
return crc;
}
static int flac_decode_init(AVCodecContext * avctx)
{
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 */
allocate_buffers(s);
}
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 sum, 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;
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_DEBUG, "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_DEBUG, "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 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_DEBUG, "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_DEBUG, "invalid sample size code (%d)\n", sample_size_code);
return -1;
}
if (get_bits1(&s->gb))
{
av_log(s->avctx, AV_LOG_DEBUG, "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 (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= get_crc8(s->gb.buffer, get_bits_count(&s->gb)/8);
if(crc8){
av_log(s->avctx, AV_LOG_ERROR, "header crc missmatch 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 int flac_decode_frame(AVCodecContext *avctx,
void *data, int *data_size,
uint8_t *buf, int buf_size)
{
FLACContext *s = avctx->priv_data;
int metadata_last, metadata_type, metadata_size;
int tmp = 0, i, j = 0, input_buf_size;
int16_t *samples = data, *left, *right;
*data_size = 0;
s->avctx = avctx;
if(s->max_framesize == 0){
s->max_framesize= 8192; // 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= FFMIN(buf_size, s->max_framesize - s->bitstream_size);
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);
/* fLaC signature (be) */
if (show_bits_long(&s->gb, 32) == bswap_32(ff_get_fourcc("fLaC")))
{
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);
dump_headers(s);
break;
default:
for(i=0; i<metadata_size; i++)
skip_bits(&s->gb, 8);
}
}
} while(!metadata_last);
}
else
{
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) < 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
switch(s->decorrelation)
{
case INDEPENDENT:
for (j = 0; j < s->blocksize; j++)
{
for (i = 0; i < s->channels; i++)
*(samples++) = s->decoded[i][j];
}
break;
case LEFT_SIDE:
assert(s->channels == 2);
for (i = 0; i < s->blocksize; i++)
{
*(samples++) = s->decoded[0][i];
*(samples++) = s->decoded[0][i] - s->decoded[1][i];
}
break;
case RIGHT_SIDE:
assert(s->channels == 2);
for (i = 0; i < s->blocksize; i++)
{
*(samples++) = s->decoded[0][i] + s->decoded[1][i];
*(samples++) = s->decoded[1][i];
}
break;
case MID_SIDE:
assert(s->channels == 2);
for (i = 0; i < s->blocksize; i++)
{
int mid, side;
mid = s->decoded[0][i];
side = s->decoded[1][i];
#if 1 //needs to be checked but IMHO it should be binary identical
mid -= side>>1;
*(samples++) = mid + side;
*(samples++) = mid;
#else
mid <<= 1;
if (side & 1)
mid++;
*(samples++) = (mid + side) >> 1;
*(samples++) = (mid - side) >> 1;
#endif
}
break;
}
#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){
int i;
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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