ffmpeg/libavcodec/mpegaudio_parser.c
Michael Niedermayer 80d156d7fd Merge remote-tracking branch 'qatar/master'
* qatar/master:
  qdm2: Use floating point synthesis filter.
  h264: correct border check.
  h264: fix loopfilter with threading at slice boundaries.
  Fix ff_mpa_synth_filter_fixed() prototype
  Rename costablegen.c ---> cos_tablegen.c.
  Collapse tableprint.c into tableprint.h.
  Simplify trig table rules
  Remove potentially unstable filenames from comments in generated files.
  Ignore generated tables and generated table generator programs.
  Simplify CLEANFILES make variable by using wildcards.
  Remove silly insults from avformat_version() Doxygen documentation.
  mpegaudiodsp: fix x86 and ppc makefiles
  configure: Adjust AVX assembler check.
  mpegaudio: remove unused version of SAME_HEADER_MASK
  mpegaudio: remove useless #undef at end of file
  asfdec: add missing #include for av_bswap32()
  mpegaudio: merge two #if CONFIG_FLOAT blocks
  mpegaudio: move some struct definitions from mpegaudio.h
  Move some mpegaudio functions to new mpegaudiodsp subsystem

Conflicts:
	libavcodec/h264.c
	libavcodec/x86/Makefile

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2011-05-20 05:48:22 +02:00

149 lines
4.2 KiB
C

/*
* MPEG Audio parser
* Copyright (c) 2003 Fabrice Bellard
* Copyright (c) 2003 Michael Niedermayer
*
* 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 "parser.h"
#include "mpegaudio.h"
#include "mpegaudiodecheader.h"
typedef struct MpegAudioParseContext {
ParseContext pc;
int frame_size;
uint32_t header;
int header_count;
} MpegAudioParseContext;
#define MPA_HEADER_SIZE 4
/* header + layer + bitrate + freq + lsf/mpeg25 */
#define SAME_HEADER_MASK \
(0xffe00000 | (3 << 17) | (3 << 10) | (3 << 19))
/* useful helper to get mpeg audio stream infos. Return -1 if error in
header, otherwise the coded frame size in bytes */
int ff_mpa_decode_header(AVCodecContext *avctx, uint32_t head, int *sample_rate, int *channels, int *frame_size, int *bit_rate)
{
MPADecodeHeader s1, *s = &s1;
if (ff_mpa_check_header(head) != 0)
return -1;
if (ff_mpegaudio_decode_header(s, head) != 0) {
return -1;
}
switch(s->layer) {
case 1:
avctx->codec_id = CODEC_ID_MP1;
*frame_size = 384;
break;
case 2:
avctx->codec_id = CODEC_ID_MP2;
*frame_size = 1152;
break;
default:
case 3:
avctx->codec_id = CODEC_ID_MP3;
if (s->lsf)
*frame_size = 576;
else
*frame_size = 1152;
break;
}
*sample_rate = s->sample_rate;
*channels = s->nb_channels;
*bit_rate = s->bit_rate;
avctx->sub_id = s->layer;
return s->frame_size;
}
static int mpegaudio_parse(AVCodecParserContext *s1,
AVCodecContext *avctx,
const uint8_t **poutbuf, int *poutbuf_size,
const uint8_t *buf, int buf_size)
{
MpegAudioParseContext *s = s1->priv_data;
ParseContext *pc = &s->pc;
uint32_t state= pc->state;
int i;
int next= END_NOT_FOUND;
for(i=0; i<buf_size; ){
if(s->frame_size){
int inc= FFMIN(buf_size - i, s->frame_size);
i += inc;
s->frame_size -= inc;
if(!s->frame_size){
next= i;
break;
}
}else{
while(i<buf_size){
int ret, sr, channels, bit_rate, frame_size;
state= (state<<8) + buf[i++];
ret = ff_mpa_decode_header(avctx, state, &sr, &channels, &frame_size, &bit_rate);
if (ret < 4) {
s->header_count= -2;
} else {
if((state&SAME_HEADER_MASK) != (s->header&SAME_HEADER_MASK) && s->header)
s->header_count= -3;
s->header= state;
s->header_count++;
s->frame_size = ret-4;
if(s->header_count > 1){
avctx->sample_rate= sr;
avctx->channels = channels;
avctx->frame_size = frame_size;
avctx->bit_rate = bit_rate;
}
break;
}
}
}
}
pc->state= state;
if (ff_combine_frame(pc, next, &buf, &buf_size) < 0) {
*poutbuf = NULL;
*poutbuf_size = 0;
return buf_size;
}
*poutbuf = buf;
*poutbuf_size = buf_size;
return next;
}
AVCodecParser ff_mpegaudio_parser = {
{ CODEC_ID_MP1, CODEC_ID_MP2, CODEC_ID_MP3 },
sizeof(MpegAudioParseContext),
NULL,
mpegaudio_parse,
ff_parse_close,
};