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FFmpeg/libavcodec/aac_ac3_parser.c
Michael Niedermayer eadd4264ee Merge remote-tracking branch 'qatar/master'
* qatar/master: (36 commits)
  adpcmenc: Use correct frame_size for Yamaha ADPCM.
  avcodec: add ff_samples_to_time_base() convenience function to internal.h
  adx parser: set duration
  mlp parser: set duration instead of frame_size
  gsm parser: set duration
  mpegaudio parser: set duration instead of frame_size
  (e)ac3 parser: set duration instead of frame_size
  flac parser: set duration instead of frame_size
  avcodec: add duration field to AVCodecParserContext
  avutil: add av_rescale_q_rnd() to allow different rounding
  pnmdec: remove useless .pix_fmts
  libmp3lame: support float and s32 sample formats
  libmp3lame: renaming, rearrangement, alignment, and comments
  libmp3lame: use the LAME default bit rate
  libmp3lame: use avpriv_mpegaudio_decode_header() for output frame parsing
  libmp3lame: cosmetics: remove some pointless comments
  libmp3lame: convert some debugging code to av_dlog()
  libmp3lame: remove outdated comment.
  libmp3lame: do not set coded_frame->key_frame.
  libmp3lame: improve error handling in MP3lame_encode_init()
  ...

Conflicts:
	doc/APIchanges
	libavcodec/libmp3lame.c
	libavcodec/pcxenc.c
	libavcodec/pnmdec.c
	libavcodec/pnmenc.c
	libavcodec/sgienc.c
	libavcodec/utils.c
	libavformat/hls.c
	libavutil/avutil.h
	libswscale/x86/swscale_mmx.c

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2012-02-21 05:10:12 +01:00

104 lines
3.4 KiB
C

/*
* Common AAC and AC-3 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 "aac_ac3_parser.h"
int ff_aac_ac3_parse(AVCodecParserContext *s1,
AVCodecContext *avctx,
const uint8_t **poutbuf, int *poutbuf_size,
const uint8_t *buf, int buf_size)
{
AACAC3ParseContext *s = s1->priv_data;
ParseContext *pc = &s->pc;
int len, i;
int new_frame_start;
get_next:
i=END_NOT_FOUND;
if(s->remaining_size <= buf_size){
if(s->remaining_size && !s->need_next_header){
i= s->remaining_size;
s->remaining_size = 0;
}else{ //we need a header first
len=0;
for(i=s->remaining_size; i<buf_size; i++){
s->state = (s->state<<8) + buf[i];
if((len=s->sync(s->state, s, &s->need_next_header, &new_frame_start)))
break;
}
if(len<=0){
i=END_NOT_FOUND;
}else{
s->state=0;
i-= s->header_size -1;
s->remaining_size = len;
if(!new_frame_start || pc->index+i<=0){
s->remaining_size += i;
goto get_next;
}
}
}
}
if(ff_combine_frame(pc, i, &buf, &buf_size)<0){
s->remaining_size -= FFMIN(s->remaining_size, buf_size);
*poutbuf = NULL;
*poutbuf_size = 0;
return buf_size;
}
*poutbuf = buf;
*poutbuf_size = buf_size;
/* update codec info */
if(s->codec_id)
avctx->codec_id = s->codec_id;
/* Due to backwards compatible HE-AAC the sample rate, channel count,
and total number of samples found in an AAC ADTS header are not
reliable. Bit rate is still accurate because the total frame duration in
seconds is still correct (as is the number of bits in the frame). */
if (avctx->codec_id != CODEC_ID_AAC) {
avctx->sample_rate = s->sample_rate;
/* allow downmixing to stereo (or mono for AC-3) */
if(avctx->request_channels > 0 &&
avctx->request_channels < s->channels &&
(avctx->request_channels <= 2 ||
(avctx->request_channels == 1 &&
(avctx->codec_id == CODEC_ID_AC3 ||
avctx->codec_id == CODEC_ID_EAC3)))) {
avctx->channels = avctx->request_channels;
} else {
avctx->channels = s->channels;
avctx->channel_layout = s->channel_layout;
}
s1->duration = s->samples;
avctx->audio_service_type = s->service_type;
}
avctx->bit_rate = s->bit_rate;
return i;
}