/*
* 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 ( Free Lossless Audio Codec ) decoder
* @ author Alex Beregszaszi
* @ see 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>
# include "libavutil/avassert.h"
# include "libavutil/crc.h"
# include "libavutil/mem.h"
# include "libavutil/opt.h"
# include "avcodec.h"
# include "codec_internal.h"
# include "get_bits.h"
# include "golomb.h"
# include "flac.h"
# include "flacdsp.h"
# include "flac_parse.h"
# include "thread.h"
# include "unary.h"
typedef struct FLACContext {
AVClass * class ;
FLACStreaminfo stream_info ;
AVCodecContext * avctx ; ///< parent AVCodecContext
GetBitContext gb ; ///< GetBitContext initialized to start at the current frame
int blocksize ; ///< number of samples in the current frame
int sample_shift ; ///< shift required to make output samples 16-bit or 32-bit
int ch_mode ; ///< channel decorrelation type in the current frame
int got_streaminfo ; ///< indicates if the STREAMINFO has been read
int32_t * decoded [ FLAC_MAX_CHANNELS ] ; ///< decoded samples
uint8_t * decoded_buffer ;
unsigned int decoded_buffer_size ;
int64_t * decoded_33bps ; ///< decoded samples for a 33 bps subframe
uint8_t * decoded_buffer_33bps ;
unsigned int decoded_buffer_size_33bps ;
int buggy_lpc ; ///< use workaround for old lavc encoded files
FLACDSPContext dsp ;
} FLACContext ;
static int allocate_buffers ( FLACContext * s ) ;
static void flac_set_bps ( FLACContext * s )
{
enum AVSampleFormat req = s - > avctx - > request_sample_fmt ;
int need32 = s - > stream_info . bps > 16 ;
int want32 = av_get_bytes_per_sample ( req ) > 2 ;
int planar = av_sample_fmt_is_planar ( req ) ;
if ( need32 | | want32 ) {
if ( planar )
s - > avctx - > sample_fmt = AV_SAMPLE_FMT_S32P ;
else
s - > avctx - > sample_fmt = AV_SAMPLE_FMT_S32 ;
s - > sample_shift = 32 - s - > stream_info . bps ;
} else {
if ( planar )
s - > avctx - > sample_fmt = AV_SAMPLE_FMT_S16P ;
else
s - > avctx - > sample_fmt = AV_SAMPLE_FMT_S16 ;
s - > sample_shift = 16 - s - > stream_info . bps ;
}
}
static av_cold int flac_decode_init ( AVCodecContext * avctx )
{
uint8_t * streaminfo ;
int ret ;
FLACContext * s = avctx - > priv_data ;
s - > avctx = avctx ;
/* 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 , & streaminfo ) )
return AVERROR_INVALIDDATA ;
/* initialize based on the demuxer-supplied streamdata header */
ret = ff_flac_parse_streaminfo ( avctx , & s - > stream_info , streaminfo ) ;
if ( ret < 0 )
return ret ;
ret = allocate_buffers ( s ) ;
if ( ret < 0 )
return ret ;
flac_set_bps ( s ) ;
ff_flacdsp_init ( & s - > dsp , avctx - > sample_fmt ,
s - > stream_info . channels ) ;
s - > got_streaminfo = 1 ;
return 0 ;
}
static void dump_headers ( AVCodecContext * avctx , FLACStreaminfo * s )
{
av_log ( avctx , AV_LOG_DEBUG , " Max Blocksize: %d \n " , 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 int allocate_buffers ( FLACContext * s )
{
int buf_size ;
int ret ;
av_assert0 ( s - > stream_info . max_blocksize ) ;
buf_size = av_samples_get_buffer_size ( NULL , s - > stream_info . channels ,
s - > stream_info . max_blocksize ,
AV_SAMPLE_FMT_S32P , 0 ) ;
if ( buf_size < 0 )
return buf_size ;
av_fast_malloc ( & s - > decoded_buffer , & s - > decoded_buffer_size , buf_size ) ;
if ( ! s - > decoded_buffer )
return AVERROR ( ENOMEM ) ;
ret = av_samples_fill_arrays ( ( uint8_t * * ) s - > decoded , NULL ,
s - > decoded_buffer ,
s - > stream_info . channels ,
s - > stream_info . max_blocksize ,
AV_SAMPLE_FMT_S32P , 0 ) ;
if ( ret > = 0 & & s - > stream_info . bps = = 32 & & s - > stream_info . channels = = 2 ) {
buf_size = av_samples_get_buffer_size ( NULL , 1 ,
s - > stream_info . max_blocksize ,
AV_SAMPLE_FMT_S64P , 0 ) ;
if ( buf_size < 0 )
return buf_size ;
av_fast_malloc ( & s - > decoded_buffer_33bps , & s - > decoded_buffer_size_33bps , buf_size ) ;
if ( ! s - > decoded_buffer_33bps )
return AVERROR ( ENOMEM ) ;
ret = av_samples_fill_arrays ( ( uint8_t * * ) & s - > decoded_33bps , NULL ,
s - > decoded_buffer_33bps ,
1 ,
s - > stream_info . max_blocksize ,
AV_SAMPLE_FMT_S64P , 0 ) ;
}
return ret < 0 ? ret : 0 ;
}
/**
* Parse the STREAMINFO from an inline header .
* @ param s the flac decoding context
* @ param buf input buffer , starting with the " fLaC " marker
* @ param buf_size buffer size
* @ return non - zero if metadata is invalid
*/
static int parse_streaminfo ( FLACContext * s , const uint8_t * buf , int buf_size )
{
int metadata_type , metadata_size , ret ;
if ( buf_size < FLAC_STREAMINFO_SIZE + 8 ) {
/* need more data */
return 0 ;
}
flac_parse_block_header ( & buf [ 4 ] , NULL , & metadata_type , & metadata_size ) ;
if ( metadata_type ! = FLAC_METADATA_TYPE_STREAMINFO | |
metadata_size ! = FLAC_STREAMINFO_SIZE ) {
return AVERROR_INVALIDDATA ;
}
ret = ff_flac_parse_streaminfo ( s - > avctx , & s - > stream_info , & buf [ 8 ] ) ;
if ( ret < 0 )
return ret ;
ret = allocate_buffers ( s ) ;
if ( ret < 0 )
return ret ;
flac_set_bps ( s ) ;
ff_flacdsp_init ( & s - > dsp , s - > avctx - > sample_fmt ,
s - > stream_info . channels ) ;
s - > got_streaminfo = 1 ;
return 0 ;
}
/**
* Determine the size of an inline header .
* @ param buf input buffer , starting with the " fLaC " marker
* @ param buf_size buffer size
* @ return number of bytes in the header , or 0 if more data is needed
*/
static int get_metadata_size ( const uint8_t * buf , int buf_size )
{
int metadata_last , metadata_size ;
const uint8_t * buf_end = buf + buf_size ;
buf + = 4 ;
do {
if ( buf_end - buf < 4 )
return AVERROR_INVALIDDATA ;
flac_parse_block_header ( buf , & metadata_last , NULL , & metadata_size ) ;
buf + = 4 ;
if ( buf_end - buf < metadata_size ) {
/* need more data in order to read the complete header */
return AVERROR_INVALIDDATA ;
}
buf + = metadata_size ;
} while ( ! metadata_last ) ;
return buf_size - ( buf_end - buf ) ;
}
static int decode_residuals ( FLACContext * s , int32_t * decoded , int pred_order )
{
GetBitContext gb = s - > gb ;
int i , tmp , partition , method_type , rice_order ;
int rice_bits , rice_esc ;
int samples ;
method_type = get_bits ( & gb , 2 ) ;
rice_order = get_bits ( & gb , 4 ) ;
samples = s - > blocksize > > rice_order ;
rice_bits = 4 + method_type ;
rice_esc = ( 1 < < rice_bits ) - 1 ;
decoded + = pred_order ;
i = pred_order ;
if ( method_type > 1 ) {
av_log ( s - > avctx , AV_LOG_ERROR , " illegal residual coding method %d \n " ,
method_type ) ;
return AVERROR_INVALIDDATA ;
}
if ( samples < < rice_order ! = s - > blocksize ) {
av_log ( s - > avctx , AV_LOG_ERROR , " invalid rice order: %i blocksize %i \n " ,
rice_order , s - > blocksize ) ;
return AVERROR_INVALIDDATA ;
}
if ( pred_order > samples ) {
av_log ( s - > avctx , AV_LOG_ERROR , " invalid predictor order: %i > %i \n " ,
pred_order , samples ) ;
return AVERROR_INVALIDDATA ;
}
for ( partition = 0 ; partition < ( 1 < < rice_order ) ; partition + + ) {
tmp = get_bits ( & gb , rice_bits ) ;
if ( tmp = = rice_esc ) {
tmp = get_bits ( & gb , 5 ) ;
for ( ; i < samples ; i + + )
* decoded + + = get_sbits_long ( & gb , tmp ) ;
} else {
int real_limit = ( tmp > 1 ) ? ( INT_MAX > > ( tmp - 1 ) ) + 2 : INT_MAX ;
for ( ; i < samples ; i + + ) {
int v = get_sr_golomb_flac ( & gb , tmp , real_limit , 1 ) ;
if ( v = = 0x80000000 ) {
av_log ( s - > avctx , AV_LOG_ERROR , " invalid residual \n " ) ;
return AVERROR_INVALIDDATA ;
}
* decoded + + = v ;
}
}
i = 0 ;
}
s - > gb = gb ;
return 0 ;
}
static int decode_subframe_fixed ( FLACContext * s , int32_t * decoded ,
int pred_order , int bps )
{
const int blocksize = s - > blocksize ;
unsigned av_uninit ( a ) , av_uninit ( b ) , av_uninit ( c ) , av_uninit ( d ) ;
int i ;
int ret ;
/* warm up samples */
for ( i = 0 ; i < pred_order ; i + + ) {
decoded [ i ] = get_sbits_long ( & s - > gb , bps ) ;
}
if ( ( ret = decode_residuals ( s , decoded , pred_order ) ) < 0 )
return ret ;
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 ] + 2U * 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 AVERROR_INVALIDDATA ;
}
return 0 ;
}
# define DECODER_SUBFRAME_FIXED_WIDE(residual) { \
const int blocksize = s - > blocksize ; \
int ret ; \
\
if ( ( ret = decode_residuals ( s , residual , pred_order ) ) < 0 ) \
return ret ; \
\
switch ( pred_order ) { \
case 0 : \
for ( int i = pred_order ; i < blocksize ; i + + ) \
decoded [ i ] = residual [ i ] ; \
break ; \
case 1 : \
for ( int i = pred_order ; i < blocksize ; i + + ) \
decoded [ i ] = ( uint64_t ) residual [ i ] + ( uint64_t ) decoded [ i - 1 ] ; \
break ; \
case 2 : \
for ( int i = pred_order ; i < blocksize ; i + + ) \
decoded [ i ] = ( uint64_t ) residual [ i ] + 2 * ( uint64_t ) decoded [ i - 1 ] - ( uint64_t ) decoded [ i - 2 ] ; \
break ; \
case 3 : \
for ( int i = pred_order ; i < blocksize ; i + + ) \
decoded [ i ] = ( uint64_t ) residual [ i ] + 3 * ( uint64_t ) decoded [ i - 1 ] - 3 * ( uint64_t ) decoded [ i - 2 ] + ( uint64_t ) decoded [ i - 3 ] ; \
break ; \
case 4 : \
for ( int i = pred_order ; i < blocksize ; i + + ) \
decoded [ i ] = ( uint64_t ) residual [ i ] + 4 * ( uint64_t ) decoded [ i - 1 ] - 6 * ( uint64_t ) decoded [ i - 2 ] + 4 * ( uint64_t ) decoded [ i - 3 ] - ( uint64_t ) decoded [ i - 4 ] ; \
break ; \
default : \
av_log ( s - > avctx , AV_LOG_ERROR , " illegal pred order %d \n " , pred_order ) ; \
return AVERROR_INVALIDDATA ; \
} \
return 0 ; \
}
static int decode_subframe_fixed_wide ( FLACContext * s , int32_t * decoded ,
int pred_order , int bps )
{
/* warm up samples */
for ( int i = 0 ; i < pred_order ; i + + ) {
decoded [ i ] = get_sbits_long ( & s - > gb , bps ) ;
}
DECODER_SUBFRAME_FIXED_WIDE ( decoded ) ;
}
static int decode_subframe_fixed_33bps ( FLACContext * s , int64_t * decoded ,
int32_t * residual , int pred_order )
{
/* warm up samples */ \
for ( int i = 0 ; i < pred_order ; i + + ) { \
decoded [ i ] = get_sbits64 ( & s - > gb , 33 ) ; \
} \
DECODER_SUBFRAME_FIXED_WIDE ( residual ) ;
}
static void lpc_analyze_remodulate ( SUINT32 * decoded , const int coeffs [ 32 ] ,
int order , int qlevel , int len , int bps )
{
int i , j ;
int ebps = 1 < < ( bps - 1 ) ;
unsigned sigma = 0 ;
for ( i = order ; i < len ; i + + )
sigma | = decoded [ i ] + ebps ;
if ( sigma < 2 * ebps )
return ;
for ( i = len - 1 ; i > = order ; i - - ) {
int64_t p = 0 ;
for ( j = 0 ; j < order ; j + + )
p + = coeffs [ j ] * ( int64_t ) ( int32_t ) decoded [ i - order + j ] ;
decoded [ i ] - = p > > qlevel ;
}
for ( i = order ; i < len ; i + + , decoded + + ) {
int32_t p = 0 ;
for ( j = 0 ; j < order ; j + + )
p + = coeffs [ j ] * ( uint32_t ) decoded [ j ] ;
decoded [ j ] + = p > > qlevel ;
}
}
static int decode_subframe_lpc ( FLACContext * s , int32_t * decoded , int pred_order ,
int bps )
{
int i , ret ;
int coeff_prec , qlevel ;
int coeffs [ 32 ] ;
/* warm up samples */
for ( i = 0 ; i < pred_order ; i + + ) {
decoded [ i ] = get_sbits_long ( & s - > gb , 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 AVERROR_INVALIDDATA ;
}
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 AVERROR_INVALIDDATA ;
}
for ( i = 0 ; i < pred_order ; i + + ) {
coeffs [ pred_order - i - 1 ] = get_sbits ( & s - > gb , coeff_prec ) ;
}
if ( ( ret = decode_residuals ( s , decoded , pred_order ) ) < 0 )
return ret ;
if ( ( s - > buggy_lpc & & s - > stream_info . bps < = 16 )
| | ( ! s - > buggy_lpc & & bps < = 16
& & bps + coeff_prec + av_log2 ( pred_order ) < = 32 ) ) {
s - > dsp . lpc16 ( decoded , coeffs , pred_order , qlevel , s - > blocksize ) ;
} else {
s - > dsp . lpc32 ( decoded , coeffs , pred_order , qlevel , s - > blocksize ) ;
if ( s - > stream_info . bps < = 16 )
lpc_analyze_remodulate ( decoded , coeffs , pred_order , qlevel , s - > blocksize , bps ) ;
}
return 0 ;
}
static int decode_subframe_lpc_33bps ( FLACContext * s , int64_t * decoded ,
int32_t * residual , int pred_order )
{
int i , ret ;
int coeff_prec , qlevel ;
int coeffs [ 32 ] ;
/* warm up samples */
for ( i = 0 ; i < pred_order ; i + + ) {
decoded [ i ] = get_sbits64 ( & s - > gb , 33 ) ;
}
coeff_prec = get_bits ( & s - > gb , 4 ) + 1 ;
if ( coeff_prec = = 16 ) {
av_log ( s - > avctx , AV_LOG_ERROR , " invalid coeff precision \n " ) ;
return AVERROR_INVALIDDATA ;
}
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 AVERROR_INVALIDDATA ;
}
for ( i = 0 ; i < pred_order ; i + + ) {
coeffs [ pred_order - i - 1 ] = get_sbits ( & s - > gb , coeff_prec ) ;
}
if ( ( ret = decode_residuals ( s , residual , pred_order ) ) < 0 )
return ret ;
s - > dsp . lpc33 ( decoded , residual , coeffs , pred_order , qlevel , s - > blocksize ) ;
return 0 ;
}
static inline int decode_subframe ( FLACContext * s , int channel )
{
int32_t * decoded = s - > decoded [ channel ] ;
int type , wasted = 0 ;
int bps = s - > stream_info . bps ;
int i , ret ;
if ( channel = = 0 ) {
if ( s - > ch_mode = = FLAC_CHMODE_RIGHT_SIDE )
bps + + ;
} else {
if ( s - > ch_mode = = FLAC_CHMODE_LEFT_SIDE | | s - > ch_mode = = FLAC_CHMODE_MID_SIDE )
bps + + ;
}
if ( get_bits1 ( & s - > gb ) ) {
av_log ( s - > avctx , AV_LOG_ERROR , " invalid subframe padding \n " ) ;
return AVERROR_INVALIDDATA ;
}
type = get_bits ( & s - > gb , 6 ) ;
if ( get_bits1 ( & s - > gb ) ) {
int left = get_bits_left ( & s - > gb ) ;
if ( left < = 0 | |
( left < bps & & ! show_bits_long ( & s - > gb , left ) ) | |
! show_bits_long ( & s - > gb , bps - 1 ) ) {
av_log ( s - > avctx , AV_LOG_ERROR ,
" Invalid number of wasted bits > available bits (%d) - left=%d \n " ,
bps , left ) ;
return AVERROR_INVALIDDATA ;
}
wasted = 1 + get_unary ( & s - > gb , 1 , get_bits_left ( & s - > gb ) ) ;
bps - = wasted ;
}
//FIXME use av_log2 for types
if ( type = = 0 ) {
if ( bps < 33 ) {
int32_t tmp = get_sbits_long ( & s - > gb , bps ) ;
for ( i = 0 ; i < s - > blocksize ; i + + )
decoded [ i ] = tmp ;
} else {
int64_t tmp = get_sbits64 ( & s - > gb , 33 ) ;
for ( i = 0 ; i < s - > blocksize ; i + + )
s - > decoded_33bps [ i ] = tmp ;
}
} else if ( type = = 1 ) {
if ( bps < 33 ) {
for ( i = 0 ; i < s - > blocksize ; i + + )
decoded [ i ] = get_sbits_long ( & s - > gb , bps ) ;
} else {
for ( i = 0 ; i < s - > blocksize ; i + + )
s - > decoded_33bps [ i ] = get_sbits64 ( & s - > gb , 33 ) ;
}
} else if ( ( type > = 8 ) & & ( type < = 12 ) ) {
int order = type & ~ 0x8 ;
if ( bps < 33 ) {
if ( bps + order < = 32 ) {
if ( ( ret = decode_subframe_fixed ( s , decoded , order , bps ) ) < 0 )
return ret ;
} else {
if ( ( ret = decode_subframe_fixed_wide ( s , decoded , order , bps ) ) < 0 )
return ret ;
}
} else {
if ( ( ret = decode_subframe_fixed_33bps ( s , s - > decoded_33bps , decoded , order ) ) < 0 )
return ret ;
}
} else if ( type > = 32 ) {
if ( bps < 33 ) {
if ( ( ret = decode_subframe_lpc ( s , decoded , ( type & ~ 0x20 ) + 1 , bps ) ) < 0 )
return ret ;
} else {
if ( ( ret = decode_subframe_lpc_33bps ( s , s - > decoded_33bps , decoded , ( type & ~ 0x20 ) + 1 ) ) < 0 )
return ret ;
}
} else {
av_log ( s - > avctx , AV_LOG_ERROR , " invalid coding type \n " ) ;
return AVERROR_INVALIDDATA ;
}
if ( wasted ) {
if ( wasted + bps = = 33 ) {
s - > dsp . wasted33 ( s - > decoded_33bps , decoded , wasted , s - > blocksize ) ;
} else if ( wasted < 32 ) {
s - > dsp . wasted32 ( decoded , wasted , s - > blocksize ) ;
}
}
return 0 ;
}
static int decode_frame ( FLACContext * s )
{
int i , ret ;
GetBitContext * gb = & s - > gb ;
FLACFrameInfo fi ;
if ( ( ret = ff_flac_decode_frame_header ( s - > avctx , gb , & fi , 0 ) ) < 0 ) {
av_log ( s - > avctx , AV_LOG_ERROR , " invalid frame header \n " ) ;
return ret ;
}
if ( s - > stream_info . channels
& & fi . channels ! = s - > stream_info . channels
& & s - > got_streaminfo ) {
s - > stream_info . channels = fi . channels ;
ff_flac_set_channel_layout ( s - > avctx , fi . channels ) ;
ret = allocate_buffers ( s ) ;
if ( ret < 0 )
return ret ;
}
s - > stream_info . channels = fi . channels ;
ff_flac_set_channel_layout ( s - > avctx , fi . channels ) ;
s - > ch_mode = fi . ch_mode ;
if ( ! s - > stream_info . bps & & ! fi . bps ) {
av_log ( s - > avctx , AV_LOG_ERROR , " bps not found in STREAMINFO or frame header \n " ) ;
return AVERROR_INVALIDDATA ;
}
if ( ! fi . bps ) {
fi . bps = s - > stream_info . bps ;
} else if ( s - > stream_info . bps & & fi . bps ! = s - > stream_info . bps ) {
av_log ( s - > avctx , AV_LOG_ERROR , " switching bps mid-stream is not "
" supported \n " ) ;
return AVERROR_INVALIDDATA ;
}
if ( ! s - > stream_info . bps ) {
s - > stream_info . bps = s - > avctx - > bits_per_raw_sample = fi . bps ;
flac_set_bps ( s ) ;
}
if ( ! s - > stream_info . max_blocksize )
s - > stream_info . max_blocksize = FLAC_MAX_BLOCKSIZE ;
if ( fi . blocksize > s - > stream_info . max_blocksize ) {
av_log ( s - > avctx , AV_LOG_ERROR , " blocksize %d > %d \n " , fi . blocksize ,
s - > stream_info . max_blocksize ) ;
return AVERROR_INVALIDDATA ;
}
s - > blocksize = fi . blocksize ;
if ( ! s - > stream_info . samplerate & & ! fi . samplerate ) {
av_log ( s - > avctx , AV_LOG_ERROR , " sample rate not found in STREAMINFO "
" or frame header \n " ) ;
return AVERROR_INVALIDDATA ;
}
if ( fi . samplerate = = 0 )
fi . samplerate = s - > stream_info . samplerate ;
s - > stream_info . samplerate = s - > avctx - > sample_rate = fi . samplerate ;
if ( ! s - > got_streaminfo ) {
ret = allocate_buffers ( s ) ;
if ( ret < 0 )
return ret ;
s - > got_streaminfo = 1 ;
dump_headers ( s - > avctx , & s - > stream_info ) ;
}
ff_flacdsp_init ( & s - > dsp , s - > avctx - > sample_fmt ,
s - > stream_info . channels ) ;
// dump_headers(s->avctx, &s->stream_info);
/* subframes */
for ( i = 0 ; i < s - > stream_info . channels ; i + + ) {
if ( ( ret = decode_subframe ( s , i ) ) < 0 )
return ret ;
}
align_get_bits ( gb ) ;
/* frame footer */
skip_bits ( gb , 16 ) ; /* data crc */
return 0 ;
}
static void decorrelate_33bps ( int ch_mode , int32_t * * decoded , int64_t * decoded_33bps , int len )
{
int i ;
if ( ch_mode = = FLAC_CHMODE_LEFT_SIDE ) {
for ( i = 0 ; i < len ; i + + )
decoded [ 1 ] [ i ] = decoded [ 0 ] [ i ] - ( uint64_t ) decoded_33bps [ i ] ;
} else if ( ch_mode = = FLAC_CHMODE_RIGHT_SIDE ) {
for ( i = 0 ; i < len ; i + + )
decoded [ 0 ] [ i ] = decoded [ 1 ] [ i ] + ( uint64_t ) decoded_33bps [ i ] ;
} else if ( ch_mode = = FLAC_CHMODE_MID_SIDE ) {
for ( i = 0 ; i < len ; i + + ) {
uint64_t a = decoded [ 0 ] [ i ] ;
int64_t b = decoded_33bps [ i ] ;
a - = b > > 1 ;
decoded [ 0 ] [ i ] = ( a + b ) ;
decoded [ 1 ] [ i ] = a ;
}
}
}
static int flac_decode_frame ( AVCodecContext * avctx , AVFrame * frame ,
int * got_frame_ptr , AVPacket * avpkt )
{
const uint8_t * buf = avpkt - > data ;
int buf_size = avpkt - > size ;
FLACContext * s = avctx - > priv_data ;
int bytes_read = 0 ;
int ret ;
* got_frame_ptr = 0 ;
if ( buf_size > 5 & & ! memcmp ( buf , " \177 FLAC " , 5 ) ) {
av_log ( s - > avctx , AV_LOG_DEBUG , " skipping flac header packet 1 \n " ) ;
return buf_size ;
}
if ( buf_size > 0 & & ( * buf & 0x7F ) = = FLAC_METADATA_TYPE_VORBIS_COMMENT ) {
av_log ( s - > avctx , AV_LOG_DEBUG , " skipping vorbis comment \n " ) ;
return buf_size ;
}
/* check that there is at least the smallest decodable amount of data.
this amount corresponds to the smallest valid FLAC frame possible .
FF F8 69 02 00 00 9 A 00 00 34 */
if ( buf_size < FLAC_MIN_FRAME_SIZE )
return buf_size ;
/* check for inline header */
if ( AV_RB32 ( buf ) = = MKBETAG ( ' f ' , ' L ' , ' a ' , ' C ' ) ) {
if ( ! s - > got_streaminfo & & ( ret = parse_streaminfo ( s , buf , buf_size ) ) ) {
av_log ( s - > avctx , AV_LOG_ERROR , " invalid header \n " ) ;
return ret ;
}
return get_metadata_size ( buf , buf_size ) ;
}
/* decode frame */
if ( ( ret = init_get_bits8 ( & s - > gb , buf , buf_size ) ) < 0 )
return ret ;
if ( ( ret = decode_frame ( s ) ) < 0 ) {
av_log ( s - > avctx , AV_LOG_ERROR , " decode_frame() failed \n " ) ;
return ret ;
}
bytes_read = get_bits_count ( & s - > gb ) / 8 ;
if ( ( s - > avctx - > err_recognition & ( AV_EF_CRCCHECK | AV_EF_COMPLIANT ) ) & &
av_crc ( av_crc_get_table ( AV_CRC_16_ANSI ) ,
0 , buf , bytes_read ) ) {
av_log ( s - > avctx , AV_LOG_ERROR , " CRC error at PTS % " PRId64 " \n " , avpkt - > pts ) ;
if ( s - > avctx - > err_recognition & AV_EF_EXPLODE )
return AVERROR_INVALIDDATA ;
}
/* get output buffer */
frame - > nb_samples = s - > blocksize ;
if ( ( ret = ff_thread_get_buffer ( avctx , frame , 0 ) ) < 0 )
return ret ;
if ( s - > stream_info . bps = = 32 & & s - > ch_mode > 0 ) {
decorrelate_33bps ( s - > ch_mode , s - > decoded , s - > decoded_33bps , s - > blocksize ) ;
s - > dsp . decorrelate [ 0 ] ( frame - > data , s - > decoded , s - > stream_info . channels ,
s - > blocksize , s - > sample_shift ) ;
} else {
s - > dsp . decorrelate [ s - > ch_mode ] ( frame - > data , s - > decoded ,
s - > stream_info . channels ,
s - > blocksize , s - > sample_shift ) ;
}
if ( bytes_read > buf_size ) {
av_log ( s - > avctx , AV_LOG_ERROR , " overread: %d \n " , bytes_read - buf_size ) ;
return AVERROR_INVALIDDATA ;
}
if ( bytes_read < buf_size ) {
av_log ( s - > avctx , AV_LOG_DEBUG , " underread: %d orig size: %d \n " ,
buf_size - bytes_read , buf_size ) ;
}
* got_frame_ptr = 1 ;
return bytes_read ;
}
static av_cold int flac_decode_close ( AVCodecContext * avctx )
{
FLACContext * s = avctx - > priv_data ;
av_freep ( & s - > decoded_buffer ) ;
av_freep ( & s - > decoded_buffer_33bps ) ;
return 0 ;
}
static const AVOption options [ ] = {
{ " use_buggy_lpc " , " emulate old buggy lavc behavior " , offsetof ( FLACContext , buggy_lpc ) , AV_OPT_TYPE_BOOL , { . i64 = 0 } , 0 , 1 , AV_OPT_FLAG_DECODING_PARAM | AV_OPT_FLAG_AUDIO_PARAM } ,
{ NULL } ,
} ;
static const AVClass flac_decoder_class = {
. class_name = " FLAC decoder " ,
. item_name = av_default_item_name ,
. option = options ,
. version = LIBAVUTIL_VERSION_INT ,
} ;
const FFCodec ff_flac_decoder = {
. p . name = " flac " ,
CODEC_LONG_NAME ( " FLAC (Free Lossless Audio Codec) " ) ,
. p . type = AVMEDIA_TYPE_AUDIO ,
. p . id = AV_CODEC_ID_FLAC ,
. priv_data_size = sizeof ( FLACContext ) ,
. init = flac_decode_init ,
. close = flac_decode_close ,
FF_CODEC_DECODE_CB ( flac_decode_frame ) ,
. p . capabilities = AV_CODEC_CAP_CHANNEL_CONF |
AV_CODEC_CAP_DR1 |
AV_CODEC_CAP_FRAME_THREADS ,
. p . sample_fmts = ( const enum AVSampleFormat [ ] ) { AV_SAMPLE_FMT_S16 ,
AV_SAMPLE_FMT_S16P ,
AV_SAMPLE_FMT_S32 ,
AV_SAMPLE_FMT_S32P ,
AV_SAMPLE_FMT_NONE } ,
. p . priv_class = & flac_decoder_class ,
} ;