mirror of
https://gitee.com/openharmony/third_party_ffmpeg
synced 2024-11-24 11:49:48 +00:00
faeb2bd41d
Originally committed as revision 16926 to svn://svn.ffmpeg.org/ffmpeg/trunk
712 lines
21 KiB
C
712 lines
21 KiB
C
/*
|
|
* 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 libavcodec/flacdec.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
|
|
|
|
enum decorrelation_type {
|
|
INDEPENDENT,
|
|
LEFT_SIDE,
|
|
RIGHT_SIDE,
|
|
MID_SIDE,
|
|
};
|
|
|
|
typedef struct FLACContext {
|
|
FLACSTREAMINFO
|
|
|
|
AVCodecContext *avctx; ///< parent AVCodecContext
|
|
GetBitContext gb; ///< GetBitContext initialized to start at the current frame
|
|
|
|
int blocksize; ///< number of samples in the current frame
|
|
int curr_bps; ///< bps for current subframe, adjusted for channel correlation and wasted bits
|
|
int sample_shift; ///< shift required to make output samples 16-bit or 32-bit
|
|
int is32; ///< flag to indicate if output should be 32-bit instead of 16-bit
|
|
enum decorrelation_type decorrelation; ///< channel decorrelation type in the current frame
|
|
|
|
int32_t *decoded[MAX_CHANNELS]; ///< decoded samples
|
|
uint8_t *bitstream;
|
|
unsigned int bitstream_size;
|
|
unsigned int bitstream_index;
|
|
unsigned int allocated_bitstream_size;
|
|
} FLACContext;
|
|
|
|
static const int sample_rate_table[] =
|
|
{ 0,
|
|
88200, 176400, 192000,
|
|
8000, 16000, 22050, 24000, 32000, 44100, 48000, 96000,
|
|
0, 0, 0, 0 };
|
|
|
|
static const int sample_size_table[] =
|
|
{ 0, 8, 12, 0, 16, 20, 24, 0 };
|
|
|
|
static const 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;
|
|
|
|
avctx->sample_fmt = SAMPLE_FMT_S16;
|
|
|
|
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) {
|
|
// FIXME header overhead
|
|
s->max_framesize= (s->channels * s->bps * s->max_blocksize + 7)/ 8;
|
|
}
|
|
|
|
for (i = 0; i < s->channels; i++) {
|
|
s->decoded[i] = av_realloc(s->decoded[i],
|
|
sizeof(int32_t)*s->max_blocksize);
|
|
}
|
|
|
|
if (s->allocated_bitstream_size < s->max_framesize)
|
|
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;
|
|
avctx->bits_per_raw_sample = s->bps;
|
|
if (s->bps > 16)
|
|
avctx->sample_fmt = SAMPLE_FMT_S32;
|
|
else
|
|
avctx->sample_fmt = SAMPLE_FMT_S16;
|
|
|
|
s->samples = get_bits_long(&gb, 32) << 4;
|
|
s->samples |= get_bits_long(&gb, 4);
|
|
|
|
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;
|
|
int initial_pos= get_bits_count(&s->gb);
|
|
|
|
if (show_bits_long(&s->gb, 32) == MKBETAG('f','L','a','C')) {
|
|
skip_bits(&s->gb, 32);
|
|
|
|
do {
|
|
metadata_last = get_bits1(&s->gb);
|
|
metadata_type = get_bits(&s->gb, 7);
|
|
metadata_size = get_bits_long(&s->gb, 24);
|
|
|
|
if (get_bits_count(&s->gb) + 8*metadata_size > s->gb.size_in_bits) {
|
|
skip_bits_long(&s->gb, initial_pos - get_bits_count(&s->gb));
|
|
break;
|
|
}
|
|
|
|
if (metadata_size) {
|
|
switch (metadata_type) {
|
|
case FLAC_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_ERROR, "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)) {
|
|
tmp = get_bits(&s->gb, 5);
|
|
for (; i < samples; i++, sample++)
|
|
s->decoded[channel][sample] = get_sbits(&s->gb, tmp);
|
|
} else {
|
|
for (; i < samples; i++, sample++) {
|
|
s->decoded[channel][sample] = get_sr_golomb_flac(&s->gb, tmp, INT_MAX, 0);
|
|
}
|
|
}
|
|
i= 0;
|
|
}
|
|
|
|
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;
|
|
|
|
/* warm up samples */
|
|
for (i = 0; i < pred_order; i++) {
|
|
decoded[i] = get_sbits(&s->gb, s->curr_bps);
|
|
}
|
|
|
|
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];
|
|
|
|
/* warm up samples */
|
|
for (i = 0; i < pred_order; i++) {
|
|
decoded[i] = get_sbits(&s->gb, s->curr_bps);
|
|
}
|
|
|
|
coeff_prec = get_bits(&s->gb, 4) + 1;
|
|
if (coeff_prec == 16) {
|
|
av_log(s->avctx, AV_LOG_ERROR, "invalid coeff precision\n");
|
|
return -1;
|
|
}
|
|
qlevel = get_sbits(&s->gb, 5);
|
|
if (qlevel < 0) {
|
|
av_log(s->avctx, AV_LOG_ERROR, "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);
|
|
}
|
|
|
|
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);
|
|
|
|
if (get_bits1(&s->gb)) {
|
|
wasted = 1;
|
|
while (!get_bits1(&s->gb))
|
|
wasted++;
|
|
s->curr_bps -= wasted;
|
|
}
|
|
|
|
//FIXME use av_log2 for types
|
|
if (type == 0) {
|
|
tmp = get_sbits(&s->gb, s->curr_bps);
|
|
for (i = 0; i < s->blocksize; i++)
|
|
s->decoded[channel][i] = tmp;
|
|
} else if (type == 1) {
|
|
for (i = 0; i < s->blocksize; i++)
|
|
s->decoded[channel][i] = get_sbits(&s->gb, s->curr_bps);
|
|
} else if ((type >= 8) && (type <= 12)) {
|
|
if (decode_subframe_fixed(s, channel, type & ~0x8) < 0)
|
|
return -1;
|
|
} else if (type >= 32) {
|
|
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 (bps > 16) {
|
|
s->avctx->sample_fmt = SAMPLE_FMT_S32;
|
|
s->sample_shift = 32 - bps;
|
|
s->is32 = 1;
|
|
} else {
|
|
s->avctx->sample_fmt = SAMPLE_FMT_S16;
|
|
s->sample_shift = 16 - bps;
|
|
s->is32 = 0;
|
|
}
|
|
s->bps = s->avctx->bits_per_raw_sample = bps;
|
|
|
|
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 (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 < 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++) {
|
|
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_16 = data;
|
|
int32_t *samples_32 = data;
|
|
int alloc_data_size= *data_size;
|
|
|
|
*data_size=0;
|
|
|
|
if (s->max_framesize == 0) {
|
|
s->max_framesize= FFMAX(4, buf_size); // 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
|
|
if (s->bitstream_size < 4 || AV_RL32(s->bitstream) != MKTAG('f','L','a','C'))
|
|
buf_size= FFMIN(buf_size, s->max_framesize - FFMIN(s->bitstream_size, s->max_framesize));
|
|
input_buf_size= buf_size;
|
|
|
|
if (s->bitstream_size + buf_size < buf_size || s->bitstream_index + s->bitstream_size + buf_size < s->bitstream_index)
|
|
return -1;
|
|
|
|
if (s->allocated_bitstream_size < s->bitstream_size + buf_size)
|
|
s->bitstream= av_fast_realloc(s->bitstream, &s->allocated_bitstream_size, s->bitstream_size + buf_size);
|
|
|
|
if (s->bitstream_index + s->bitstream_size + buf_size > s->allocated_bitstream_size) {
|
|
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 && input_buf_size) {
|
|
return input_buf_size;
|
|
}
|
|
}
|
|
|
|
init_get_bits(&s->gb, buf, buf_size*8);
|
|
|
|
if (metadata_parse(s))
|
|
goto end;
|
|
|
|
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;
|
|
}
|
|
|
|
#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];\
|
|
if (s->is32) {\
|
|
*samples_32++ = (left) << s->sample_shift;\
|
|
*samples_32++ = (right) << s->sample_shift;\
|
|
} else {\
|
|
*samples_16++ = (left) << s->sample_shift;\
|
|
*samples_16++ = (right) << s->sample_shift;\
|
|
}\
|
|
}\
|
|
break;
|
|
|
|
switch (s->decorrelation) {
|
|
case INDEPENDENT:
|
|
for (j = 0; j < s->blocksize; j++) {
|
|
for (i = 0; i < s->channels; i++) {
|
|
if (s->is32)
|
|
*samples_32++ = s->decoded[i][j] << s->sample_shift;
|
|
else
|
|
*samples_16++ = s->decoded[i][j] << s->sample_shift;
|
|
}
|
|
}
|
|
break;
|
|
case LEFT_SIDE:
|
|
DECORRELATE(a,a-b)
|
|
case RIGHT_SIDE:
|
|
DECORRELATE(a+b,b)
|
|
case MID_SIDE:
|
|
DECORRELATE( (a-=b>>1) + b, a)
|
|
}
|
|
|
|
*data_size = s->blocksize * s->channels * (s->is32 ? 4 : 2);
|
|
|
|
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,
|
|
CODEC_CAP_DELAY,
|
|
.flush= flac_flush,
|
|
.long_name= NULL_IF_CONFIG_SMALL("FLAC (Free Lossless Audio Codec)"),
|
|
};
|