ffmpeg/libavcodec/mp3_header_decompress_bsf.c
Michael Niedermayer f8ae3a2108 Merge remote branch 'qatar/master'
12 files changed, 36 insertions(+), 81 deletions(-)
yes thats 36 new lines in 14 commits

* qatar/master:
  ffmpeg: fix -aspect cli option
  Restructure video filter implementation in ffmpeg.c.
  ffplay: remove audio_write_get_buf_size() forward declaration
  lavfi: print key-frame and picture type information in ff_dlog_ref()
  mathops: remove ancient confusing comment
  cws2fws: Improve error message wording.
  tools: Check the return value of write().
  mpegaudio: move OUT_FMT macro to mpegaudiodec.c
  mpegaudio: remove OUT_MIN/MAX macros
  Add missing #includes to mp3_header_(de)compress bsf
  dct: fix indentation
  dct: bypass table allocation for DCT_II of size 32
  h264dsp_mmx: Add #ifdefs around some mmxext functions on x86_64.
  Remove unused header mpegaudio3.h.

Conflicts:
	ffmpeg.c
	libavcodec/mpegaudio.h
	libavcodec/mpegaudio3.h
	libavfilter/avfilter.c

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2011-05-17 04:51:33 +02:00

98 lines
3.2 KiB
C

/*
* 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 "libavutil/intreadwrite.h"
#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, "Could not 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;
}
}
AV_WB32(*poutbuf, header);
return 1;
}
AVBitStreamFilter ff_mp3_header_decompress_bsf={
"mp3decomp",
0,
mp3_header_decompress,
};