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ffmpeg/libavcodec/flac.c

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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 "libavutil/crc.h"
#include "avcodec.h"
#include "bitstream.h"
#include "golomb.h"
#include "flac.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 {
FLACSTREAMINFO
AVCodecContext *avctx;
GetBitContext gb;
int blocksize/*, last_blocksize*/;
int 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 allocate_buffers(FLACContext *s);
static int metadata_parse(FLACContext *s);
static av_cold 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 */
if (avctx->extradata_size == FLAC_STREAMINFO_SIZE) {
ff_flac_parse_streaminfo(avctx, (FLACStreaminfo *)s, avctx->extradata);
allocate_buffers(s);
} else {
init_get_bits(&s->gb, avctx->extradata, avctx->extradata_size*8);
metadata_parse(s);
}
}
return 0;
}
static void dump_headers(AVCodecContext *avctx, FLACStreaminfo *s)
{
av_log(avctx, AV_LOG_DEBUG, " Blocksize: %d .. %d\n", s->min_blocksize, s->max_blocksize);
av_log(avctx, AV_LOG_DEBUG, " Max Framesize: %d\n", s->max_framesize);
av_log(avctx, AV_LOG_DEBUG, " Samplerate: %d\n", s->samplerate);
av_log(avctx, AV_LOG_DEBUG, " Channels: %d\n", s->channels);
av_log(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);
}
void ff_flac_parse_streaminfo(AVCodecContext *avctx, struct FLACStreaminfo *s,
const uint8_t *buffer)
{
GetBitContext gb;
init_get_bits(&gb, buffer, FLAC_STREAMINFO_SIZE*8);
/* mandatory streaminfo */
s->min_blocksize = get_bits(&gb, 16);
s->max_blocksize = get_bits(&gb, 16);
skip_bits(&gb, 24); /* skip min frame size */
s->max_framesize = get_bits_long(&gb, 24);
s->samplerate = get_bits_long(&gb, 20);
s->channels = get_bits(&gb, 3) + 1;
s->bps = get_bits(&gb, 5) + 1;
avctx->channels = s->channels;
avctx->sample_rate = s->samplerate;
skip_bits(&gb, 36); /* total num of samples */
skip_bits(&gb, 64); /* md5 sum */
skip_bits(&gb, 64); /* md5 sum */
dump_headers(avctx, 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_bits1(&s->gb);
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:
ff_flac_parse_streaminfo(s->avctx, (FLACStreaminfo *)s, s->gb.buffer+get_bits_count(&s->gb)/8);
streaminfo_updated = 1;
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 > 1){
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;
if (pred_order > samples) {
av_log(s->avctx, AV_LOG_ERROR, "invalid predictor order: %i > %i\n", pred_order, samples);
return -1;
}
sample=
i= pred_order;
for (partition = 0; partition < (1 << rice_order); partition++)
{
tmp = get_bits(&s->gb, method_type == 0 ? 4 : 5);
if (tmp == (method_type == 0 ? 15 : 31))
{
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)
{
const int blocksize = s->blocksize;
int32_t *decoded = s->decoded[channel];
int a, b, c, d, 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++)
{
decoded[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;
if(pred_order > 0)
a = decoded[pred_order-1];
if(pred_order > 1)
b = a - decoded[pred_order-2];
if(pred_order > 2)
c = b - decoded[pred_order-2] + decoded[pred_order-3];
if(pred_order > 3)
d = c - decoded[pred_order-2] + 2*decoded[pred_order-3] - decoded[pred_order-4];
switch(pred_order)
{
case 0:
break;
case 1:
for (i = pred_order; i < blocksize; i++)
decoded[i] = a += decoded[i];
break;
case 2:
for (i = pred_order; i < blocksize; i++)
decoded[i] = a += b += decoded[i];
break;
case 3:
for (i = pred_order; i < blocksize; i++)
decoded[i] = a += b += c += decoded[i];
break;
case 4:
for (i = pred_order; i < blocksize; i++)
decoded[i] = a += b += c += d += decoded[i];
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];
int32_t *decoded = s->decoded[channel];
// 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++)
{
decoded[i] = get_sbits(&s->gb, s->curr_bps);
// av_log(s->avctx, AV_LOG_DEBUG, " %d: %d\n", i, decoded[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] * decoded[i-j-1];
decoded[i] += sum >> qlevel;
}
} else {
for (i = pred_order; i < s->blocksize-1; i += 2)
{
int c;
int d = decoded[i-pred_order];
int s0 = 0, s1 = 0;
for (j = pred_order-1; j > 0; j--)
{
c = coeffs[j];
s0 += c*d;
d = decoded[i-j];
s1 += c*d;
}
c = coeffs[0];
s0 += c*d;
d = decoded[i] += s0 >> qlevel;
s1 += c*d;
decoded[i+1] += s1 >> qlevel;
}
if (i < s->blocksize)
{
int sum = 0;
for (j = 0; j < pred_order; j++)
sum += coeffs[j] * decoded[i-j-1];
decoded[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_crc_get_table(AV_CRC_8_ATM), 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->avctx, (FLACStreaminfo *)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,
const 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 & 0xFFFE) != 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) & 0xFFFE) != 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) << (24 - s->bps)) >> 8;\
*samples++ = ((right) << (24 - s->bps)) >> 8;\
}\
break;
switch(s->decorrelation)
{
case INDEPENDENT:
for (j = 0; j < s->blocksize; j++)
{
for (i = 0; i < s->channels; i++)
*samples++ = (s->decoded[i][j] << (24 - s->bps)) >> 8;
}
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 av_cold 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,
.long_name= "FLAC (Free Lossless Audio Codec)"
};
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