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FFmpeg/libavcodec/mpegaudio_parser.c
Michael Niedermayer 15c6be8c7d Merge remote-tracking branch 'qatar/master'
* qatar/master:
  tiertexseq: set correct block_align for audio
  tiertexseq: set audio stream start time to 0
  voc/avs: Do not change the sample rate mid-stream.
  segafilm: use the sample rate as the time base for audio streams
  ea: fix audio pts
  psx-str: fix audio pts
  vqf: set packet duration
  tta demuxer: set packet duration
  mpegaudio_parser: do not ignore information from the first parsed frame
  mpegaudio_parser: be less picky about the start position
  thp: set audio packet durations
  avcodec: add a Vorbis parser to get packet duration
  vorbisdec: read the previous window flag for long windows
  lavc: free the output packet when encoding failed or produced no output.
  lavc: preserve avpkt->destruct in ff_alloc_packet().
  lavc: clarify the meaning of AVCodecContext.frame_number.
  mpegts: Pad the packet buffer in handle_packet().
  mpegts: Do not call read_sl_header() when no bytes remain in the buffer.

Conflicts:
	libavcodec/mpegaudio_parser.c
	libavcodec/version.h
	libavformat/mpegts.c
	tests/ref/fate/pva-demux

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2012-03-04 04:26:04 +01:00

109 lines
3.3 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 "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))
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 = avpriv_mpa_decode_header(avctx, state, &sr, &channels, &frame_size, &bit_rate);
if (ret < 4) {
if (i > 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;
s1->duration = 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_ids = { CODEC_ID_MP1, CODEC_ID_MP2, CODEC_ID_MP3 },
.priv_data_size = sizeof(MpegAudioParseContext),
.parser_parse = mpegaudio_parse,
.parser_close = ff_parse_close,
};