move mp3_header_decompress bitstream filter in its own file

Originally committed as revision 9062 to svn://svn.ffmpeg.org/ffmpeg/trunk
pull/126/head
Aurelien Jacobs 18 years ago
parent 677fe2e226
commit b57eed12da
  1. 2
      libavcodec/Makefile
  2. 80
      libavcodec/bitstream_filter.c
  3. 99
      libavcodec/mp3_header_decompress_bsf.c
  4. 2
      libavcodec/mpegaudio.h

@ -308,7 +308,7 @@ OBJS-$(CONFIG_DUMP_EXTRADATA_BSF) += bitstream_filter.o
OBJS-$(CONFIG_REMOVE_EXTRADATA_BSF) += bitstream_filter.o
OBJS-$(CONFIG_NOISE_BSF) += bitstream_filter.o
OBJS-$(CONFIG_MP3_HEADER_COMPRESS_BSF) += bitstream_filter.o
OBJS-$(CONFIG_MP3_HEADER_DECOMPRESS_BSF) += bitstream_filter.o mpegaudiodata.o
OBJS-$(CONFIG_MP3_HEADER_DECOMPRESS_BSF) += mp3_header_decompress_bsf.o mpegaudiodata.o
OBJS-$(CONFIG_MJPEGA_DUMP_HEADER_BSF) += mjpegdec.o mjpeg.o jpeglsdec.o jpegls.o
OBJS-$(CONFIG_IMX_DUMP_HEADER_BSF) += mpeg12.o

@ -20,7 +20,6 @@
#include "avcodec.h"
#include "mpegaudio.h"
#include "mpegaudiodata.h"
AVBitStreamFilter *first_bitstream_filter= NULL;
@ -126,8 +125,6 @@ static int noise(AVBitStreamFilterContext *bsfc, AVCodecContext *avctx, const ch
return 1;
}
#define MP3_MASK 0xFFFE0CCF
static int mp3_header_compress(AVBitStreamFilterContext *bsfc, AVCodecContext *avctx, const char *args,
uint8_t **poutbuf, int *poutbuf_size,
const uint8_t *buf, int buf_size, int keyframe){
@ -185,75 +182,6 @@ output_unchanged:
return 1;
}
static int mp3_header_decompress(AVBitStreamFilterContext *bsfc, AVCodecContext *avctx, const char *args,
uint8_t **poutbuf, int *poutbuf_size,
const uint8_t *buf, int buf_size, int keyframe){
uint32_t header;
int sample_rate= avctx->sample_rate;
int sample_rate_index=0;
int lsf, mpeg25, bitrate_index, frame_size;
header = AV_RB32(buf);
if(ff_mpa_check_header(header) >= 0){
*poutbuf= (uint8_t *) buf;
*poutbuf_size= buf_size;
return 0;
}
if(avctx->extradata_size != 15 || strcmp(avctx->extradata, "FFCMP3 0.0")){
av_log(avctx, AV_LOG_ERROR, "Extradata invalid %d\n", avctx->extradata_size);
return -1;
}
header= AV_RB32(avctx->extradata+11) & MP3_MASK;
lsf = sample_rate < (24000+32000)/2;
mpeg25 = sample_rate < (12000+16000)/2;
sample_rate_index= (header>>10)&3;
sample_rate= ff_mpa_freq_tab[sample_rate_index] >> (lsf + mpeg25); //in case sample rate is a little off
for(bitrate_index=2; bitrate_index<30; bitrate_index++){
frame_size = ff_mpa_bitrate_tab[lsf][2][bitrate_index>>1];
frame_size = (frame_size * 144000) / (sample_rate << lsf) + (bitrate_index&1);
if(frame_size == buf_size + 4)
break;
if(frame_size == buf_size + 6)
break;
}
if(bitrate_index == 30){
av_log(avctx, AV_LOG_ERROR, "couldnt find bitrate_index\n");
return -1;
}
header |= (bitrate_index&1)<<9;
header |= (bitrate_index>>1)<<12;
header |= (frame_size == buf_size + 4)<<16; //FIXME actually set a correct crc instead of 0
*poutbuf_size= frame_size;
*poutbuf= av_malloc(frame_size + FF_INPUT_BUFFER_PADDING_SIZE);
memcpy(*poutbuf + frame_size - buf_size, buf, buf_size + FF_INPUT_BUFFER_PADDING_SIZE);
if(avctx->channels==2){
uint8_t *p= *poutbuf + frame_size - buf_size;
if(lsf){
FFSWAP(int, p[1], p[2]);
header |= (p[1] & 0xC0)>>2;
p[1] &= 0x3F;
}else{
header |= p[1] & 0x30;
p[1] &= 0xCF;
}
}
(*poutbuf)[0]= header>>24;
(*poutbuf)[1]= header>>16;
(*poutbuf)[2]= header>> 8;
(*poutbuf)[3]= header ;
return 1;
}
#ifdef CONFIG_DUMP_EXTRADATA_BSF
AVBitStreamFilter dump_extradata_bsf={
"dump_extra",
@ -285,11 +213,3 @@ AVBitStreamFilter mp3_header_compress_bsf={
mp3_header_compress,
};
#endif
#ifdef CONFIG_MP3_HEADER_DECOMPRESS_BSF
AVBitStreamFilter mp3_header_decompress_bsf={
"mp3decomp",
0,
mp3_header_decompress,
};
#endif

@ -0,0 +1,99 @@
/*
* copyright (c) 2006 Michael Niedermayer <michaelni@gmx.at>
*
* 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
*/
#include "avcodec.h"
#include "mpegaudio.h"
#include "mpegaudiodata.h"
static int mp3_header_decompress(AVBitStreamFilterContext *bsfc, AVCodecContext *avctx, const char *args,
uint8_t **poutbuf, int *poutbuf_size,
const uint8_t *buf, int buf_size, int keyframe){
uint32_t header;
int sample_rate= avctx->sample_rate;
int sample_rate_index=0;
int lsf, mpeg25, bitrate_index, frame_size;
header = AV_RB32(buf);
if(ff_mpa_check_header(header) >= 0){
*poutbuf= (uint8_t *) buf;
*poutbuf_size= buf_size;
return 0;
}
if(avctx->extradata_size != 15 || strcmp(avctx->extradata, "FFCMP3 0.0")){
av_log(avctx, AV_LOG_ERROR, "Extradata invalid %d\n", avctx->extradata_size);
return -1;
}
header= AV_RB32(avctx->extradata+11) & MP3_MASK;
lsf = sample_rate < (24000+32000)/2;
mpeg25 = sample_rate < (12000+16000)/2;
sample_rate_index= (header>>10)&3;
sample_rate= ff_mpa_freq_tab[sample_rate_index] >> (lsf + mpeg25); //in case sample rate is a little off
for(bitrate_index=2; bitrate_index<30; bitrate_index++){
frame_size = ff_mpa_bitrate_tab[lsf][2][bitrate_index>>1];
frame_size = (frame_size * 144000) / (sample_rate << lsf) + (bitrate_index&1);
if(frame_size == buf_size + 4)
break;
if(frame_size == buf_size + 6)
break;
}
if(bitrate_index == 30){
av_log(avctx, AV_LOG_ERROR, "couldnt find bitrate_index\n");
return -1;
}
header |= (bitrate_index&1)<<9;
header |= (bitrate_index>>1)<<12;
header |= (frame_size == buf_size + 4)<<16; //FIXME actually set a correct crc instead of 0
*poutbuf_size= frame_size;
*poutbuf= av_malloc(frame_size + FF_INPUT_BUFFER_PADDING_SIZE);
memcpy(*poutbuf + frame_size - buf_size, buf, buf_size + FF_INPUT_BUFFER_PADDING_SIZE);
if(avctx->channels==2){
uint8_t *p= *poutbuf + frame_size - buf_size;
if(lsf){
FFSWAP(int, p[1], p[2]);
header |= (p[1] & 0xC0)>>2;
p[1] &= 0x3F;
}else{
header |= p[1] & 0x30;
p[1] &= 0xCF;
}
}
(*poutbuf)[0]= header>>24;
(*poutbuf)[1]= header>>16;
(*poutbuf)[2]= header>> 8;
(*poutbuf)[3]= header ;
return 1;
}
AVBitStreamFilter mp3_header_decompress_bsf={
"mp3decomp",
0,
mp3_header_decompress,
};

@ -42,6 +42,8 @@
#define SAME_HEADER_MASK \
(0xffe00000 | (3 << 17) | (0xf << 12) | (3 << 10) | (3 << 19))
#define MP3_MASK 0xFFFE0CCF
/* define USE_HIGHPRECISION to have a bit exact (but slower) mpeg
audio decoder */

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