/*
* 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);
int ff_flac_is_extradata_valid(AVCodecContext *avctx,
enum FLACExtradataFormat *format,
uint8_t **streaminfo_start)
{
if (!avctx->extradata || avctx->extradata_size < FLAC_STREAMINFO_SIZE) {
av_log(avctx, AV_LOG_ERROR, "extradata NULL or too small.\n");
return 0;
}
if (AV_RL32(avctx->extradata) != MKTAG('f','L','a','C')) {
/* extradata contains STREAMINFO only */
if (avctx->extradata_size != FLAC_STREAMINFO_SIZE) {
av_log(avctx, AV_LOG_WARNING, "extradata contains %d bytes too many.\n",
FLAC_STREAMINFO_SIZE-avctx->extradata_size);
}
*format = FLAC_EXTRADATA_FORMAT_STREAMINFO;
*streaminfo_start = avctx->extradata;
} else {
if (avctx->extradata_size < 8+FLAC_STREAMINFO_SIZE) {
av_log(avctx, AV_LOG_ERROR, "extradata too small.\n");
return 0;
}
*format = FLAC_EXTRADATA_FORMAT_FULL_HEADER;
*streaminfo_start = &avctx->extradata[8];
}
return 1;
}
static av_cold int flac_decode_init(AVCodecContext *avctx)
{
enum FLACExtradataFormat format;
uint8_t *streaminfo;
FLACContext *s = avctx->priv_data;
s->avctx = avctx;
avctx->sample_fmt = SAMPLE_FMT_S16;
/* for now, the raw FLAC header is allowed to be passed to the decoder as
frame data instead of extradata. */
if (!avctx->extradata)
return 0;
if (!ff_flac_is_extradata_valid(avctx, &format, &streaminfo))
return -1;
/* initialize based on the demuxer-supplied streamdata header */
ff_flac_parse_streaminfo(avctx, (FLACStreaminfo *)s, streaminfo);
allocate_buffers(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 av_uninit(a), av_uninit(b), av_uninit(c), av_uninit(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)"),
};