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https://github.com/FFmpeg/FFmpeg.git
synced 2024-12-23 12:43:46 +02:00
move mp3_header_decompress bitstream filter in its own file
Originally committed as revision 9062 to svn://svn.ffmpeg.org/ffmpeg/trunk
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677fe2e226
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@ -308,7 +308,7 @@ OBJS-$(CONFIG_DUMP_EXTRADATA_BSF) += bitstream_filter.o
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OBJS-$(CONFIG_REMOVE_EXTRADATA_BSF) += bitstream_filter.o
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OBJS-$(CONFIG_NOISE_BSF) += bitstream_filter.o
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OBJS-$(CONFIG_MP3_HEADER_COMPRESS_BSF) += bitstream_filter.o
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OBJS-$(CONFIG_MP3_HEADER_DECOMPRESS_BSF) += bitstream_filter.o mpegaudiodata.o
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OBJS-$(CONFIG_MP3_HEADER_DECOMPRESS_BSF) += mp3_header_decompress_bsf.o mpegaudiodata.o
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OBJS-$(CONFIG_MJPEGA_DUMP_HEADER_BSF) += mjpegdec.o mjpeg.o jpeglsdec.o jpegls.o
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OBJS-$(CONFIG_IMX_DUMP_HEADER_BSF) += mpeg12.o
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@ -20,7 +20,6 @@
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#include "avcodec.h"
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#include "mpegaudio.h"
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#include "mpegaudiodata.h"
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AVBitStreamFilter *first_bitstream_filter= NULL;
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@ -126,8 +125,6 @@ static int noise(AVBitStreamFilterContext *bsfc, AVCodecContext *avctx, const ch
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return 1;
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}
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#define MP3_MASK 0xFFFE0CCF
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static int mp3_header_compress(AVBitStreamFilterContext *bsfc, AVCodecContext *avctx, const char *args,
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uint8_t **poutbuf, int *poutbuf_size,
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const uint8_t *buf, int buf_size, int keyframe){
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@ -185,75 +182,6 @@ output_unchanged:
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return 1;
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}
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static int mp3_header_decompress(AVBitStreamFilterContext *bsfc, AVCodecContext *avctx, const char *args,
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uint8_t **poutbuf, int *poutbuf_size,
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const uint8_t *buf, int buf_size, int keyframe){
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uint32_t header;
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int sample_rate= avctx->sample_rate;
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int sample_rate_index=0;
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int lsf, mpeg25, bitrate_index, frame_size;
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header = AV_RB32(buf);
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if(ff_mpa_check_header(header) >= 0){
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*poutbuf= (uint8_t *) buf;
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*poutbuf_size= buf_size;
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return 0;
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}
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if(avctx->extradata_size != 15 || strcmp(avctx->extradata, "FFCMP3 0.0")){
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av_log(avctx, AV_LOG_ERROR, "Extradata invalid %d\n", avctx->extradata_size);
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return -1;
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}
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header= AV_RB32(avctx->extradata+11) & MP3_MASK;
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lsf = sample_rate < (24000+32000)/2;
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mpeg25 = sample_rate < (12000+16000)/2;
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sample_rate_index= (header>>10)&3;
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sample_rate= ff_mpa_freq_tab[sample_rate_index] >> (lsf + mpeg25); //in case sample rate is a little off
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for(bitrate_index=2; bitrate_index<30; bitrate_index++){
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frame_size = ff_mpa_bitrate_tab[lsf][2][bitrate_index>>1];
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frame_size = (frame_size * 144000) / (sample_rate << lsf) + (bitrate_index&1);
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if(frame_size == buf_size + 4)
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break;
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if(frame_size == buf_size + 6)
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break;
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}
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if(bitrate_index == 30){
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av_log(avctx, AV_LOG_ERROR, "couldnt find bitrate_index\n");
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return -1;
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}
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header |= (bitrate_index&1)<<9;
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header |= (bitrate_index>>1)<<12;
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header |= (frame_size == buf_size + 4)<<16; //FIXME actually set a correct crc instead of 0
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*poutbuf_size= frame_size;
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*poutbuf= av_malloc(frame_size + FF_INPUT_BUFFER_PADDING_SIZE);
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memcpy(*poutbuf + frame_size - buf_size, buf, buf_size + FF_INPUT_BUFFER_PADDING_SIZE);
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if(avctx->channels==2){
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uint8_t *p= *poutbuf + frame_size - buf_size;
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if(lsf){
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FFSWAP(int, p[1], p[2]);
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header |= (p[1] & 0xC0)>>2;
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p[1] &= 0x3F;
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}else{
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header |= p[1] & 0x30;
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p[1] &= 0xCF;
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}
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}
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(*poutbuf)[0]= header>>24;
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(*poutbuf)[1]= header>>16;
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(*poutbuf)[2]= header>> 8;
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(*poutbuf)[3]= header ;
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return 1;
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}
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#ifdef CONFIG_DUMP_EXTRADATA_BSF
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AVBitStreamFilter dump_extradata_bsf={
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"dump_extra",
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@ -285,11 +213,3 @@ AVBitStreamFilter mp3_header_compress_bsf={
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mp3_header_compress,
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};
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#endif
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#ifdef CONFIG_MP3_HEADER_DECOMPRESS_BSF
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AVBitStreamFilter mp3_header_decompress_bsf={
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"mp3decomp",
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0,
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mp3_header_decompress,
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};
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#endif
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99
libavcodec/mp3_header_decompress_bsf.c
Normal file
99
libavcodec/mp3_header_decompress_bsf.c
Normal file
@ -0,0 +1,99 @@
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/*
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* copyright (c) 2006 Michael Niedermayer <michaelni@gmx.at>
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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#include "avcodec.h"
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#include "mpegaudio.h"
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#include "mpegaudiodata.h"
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static int mp3_header_decompress(AVBitStreamFilterContext *bsfc, AVCodecContext *avctx, const char *args,
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uint8_t **poutbuf, int *poutbuf_size,
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const uint8_t *buf, int buf_size, int keyframe){
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uint32_t header;
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int sample_rate= avctx->sample_rate;
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int sample_rate_index=0;
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int lsf, mpeg25, bitrate_index, frame_size;
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header = AV_RB32(buf);
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if(ff_mpa_check_header(header) >= 0){
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*poutbuf= (uint8_t *) buf;
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*poutbuf_size= buf_size;
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return 0;
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}
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if(avctx->extradata_size != 15 || strcmp(avctx->extradata, "FFCMP3 0.0")){
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av_log(avctx, AV_LOG_ERROR, "Extradata invalid %d\n", avctx->extradata_size);
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return -1;
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}
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header= AV_RB32(avctx->extradata+11) & MP3_MASK;
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lsf = sample_rate < (24000+32000)/2;
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mpeg25 = sample_rate < (12000+16000)/2;
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sample_rate_index= (header>>10)&3;
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sample_rate= ff_mpa_freq_tab[sample_rate_index] >> (lsf + mpeg25); //in case sample rate is a little off
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for(bitrate_index=2; bitrate_index<30; bitrate_index++){
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frame_size = ff_mpa_bitrate_tab[lsf][2][bitrate_index>>1];
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frame_size = (frame_size * 144000) / (sample_rate << lsf) + (bitrate_index&1);
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if(frame_size == buf_size + 4)
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break;
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if(frame_size == buf_size + 6)
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break;
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}
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if(bitrate_index == 30){
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av_log(avctx, AV_LOG_ERROR, "couldnt find bitrate_index\n");
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return -1;
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}
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header |= (bitrate_index&1)<<9;
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header |= (bitrate_index>>1)<<12;
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header |= (frame_size == buf_size + 4)<<16; //FIXME actually set a correct crc instead of 0
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*poutbuf_size= frame_size;
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*poutbuf= av_malloc(frame_size + FF_INPUT_BUFFER_PADDING_SIZE);
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memcpy(*poutbuf + frame_size - buf_size, buf, buf_size + FF_INPUT_BUFFER_PADDING_SIZE);
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if(avctx->channels==2){
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uint8_t *p= *poutbuf + frame_size - buf_size;
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if(lsf){
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FFSWAP(int, p[1], p[2]);
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header |= (p[1] & 0xC0)>>2;
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p[1] &= 0x3F;
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}else{
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header |= p[1] & 0x30;
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p[1] &= 0xCF;
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}
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}
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(*poutbuf)[0]= header>>24;
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(*poutbuf)[1]= header>>16;
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(*poutbuf)[2]= header>> 8;
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(*poutbuf)[3]= header ;
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return 1;
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}
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AVBitStreamFilter mp3_header_decompress_bsf={
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"mp3decomp",
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0,
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mp3_header_decompress,
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};
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@ -42,6 +42,8 @@
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#define SAME_HEADER_MASK \
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(0xffe00000 | (3 << 17) | (0xf << 12) | (3 << 10) | (3 << 19))
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#define MP3_MASK 0xFFFE0CCF
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/* define USE_HIGHPRECISION to have a bit exact (but slower) mpeg
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audio decoder */
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