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https://github.com/FFmpeg/FFmpeg.git
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27ef7b1bcd
* newdev/master: mov: set audio service type for AC-3 from bitstream mode in the 'dac3' atom. Get audio_service_type for AC-3 based on bitstream mode in the AC-3 parser and decoder, and vice-versa for the AC-3 encoder. Use audio_service_type to set stream disposition. Add APIchanges entry for audio_service_type. Add audio_service_type field to AVCodecContext for encoding and reporting of the service type in the audio bitstream. configure: in check_ld, place new -l flags before existing ones support @heading, @subheading, @subsubheading, and @subsubsection in texi2pod.pl doc: update build system documentation aacenc: indentation aacenc: fix the side calculation in search_for_ms vp8.c: rename EDGE_* to VP8_EDGE_*. Conflicts: doc/APIchanges libavcodec/avcodec.h libavcodec/version.h libavcodec/vp8.c Merged-by: Michael Niedermayer <michaelni@gmx.at>
104 lines
3.5 KiB
C
104 lines
3.5 KiB
C
/*
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* Common AAC and AC-3 parser
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* Copyright (c) 2003 Fabrice Bellard
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* Copyright (c) 2003 Michael Niedermayer
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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#include "parser.h"
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#include "aac_ac3_parser.h"
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int ff_aac_ac3_parse(AVCodecParserContext *s1,
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AVCodecContext *avctx,
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const uint8_t **poutbuf, int *poutbuf_size,
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const uint8_t *buf, int buf_size)
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{
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AACAC3ParseContext *s = s1->priv_data;
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ParseContext *pc = &s->pc;
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int len, i;
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int new_frame_start;
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get_next:
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i=END_NOT_FOUND;
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if(s->remaining_size <= buf_size){
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if(s->remaining_size && !s->need_next_header){
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i= s->remaining_size;
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s->remaining_size = 0;
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}else{ //we need a header first
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len=0;
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for(i=s->remaining_size; i<buf_size; i++){
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s->state = (s->state<<8) + buf[i];
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if((len=s->sync(s->state, s, &s->need_next_header, &new_frame_start)))
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break;
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}
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if(len<=0){
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i=END_NOT_FOUND;
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}else{
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s->state=0;
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i-= s->header_size -1;
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s->remaining_size = len;
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if(!new_frame_start || pc->index+i<=0){
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s->remaining_size += i;
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goto get_next;
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}
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}
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}
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}
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if(ff_combine_frame(pc, i, &buf, &buf_size)<0){
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s->remaining_size -= FFMIN(s->remaining_size, buf_size);
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*poutbuf = NULL;
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*poutbuf_size = 0;
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return buf_size;
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}
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*poutbuf = buf;
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*poutbuf_size = buf_size;
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/* update codec info */
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if(s->codec_id)
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avctx->codec_id = s->codec_id;
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/* Due to backwards compatible HE-AAC the sample rate, channel count,
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and total number of samples found in an AAC ADTS header are not
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reliable. Bit rate is still accurate because the total frame duration in
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seconds is still correct (as is the number of bits in the frame). */
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if (avctx->codec_id != CODEC_ID_AAC) {
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avctx->sample_rate = s->sample_rate;
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/* allow downmixing to stereo (or mono for AC-3) */
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if(avctx->request_channels > 0 &&
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avctx->request_channels < s->channels &&
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(avctx->request_channels <= 2 ||
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(avctx->request_channels == 1 &&
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(avctx->codec_id == CODEC_ID_AC3 ||
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avctx->codec_id == CODEC_ID_EAC3)))) {
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avctx->channels = avctx->request_channels;
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} else {
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avctx->channels = s->channels;
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avctx->channel_layout = s->channel_layout;
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}
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avctx->frame_size = s->samples;
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avctx->audio_service_type = s->service_type;
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}
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avctx->bit_rate = s->bit_rate;
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return i;
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}
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