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FFmpeg/libavcodec/aac_ac3_parser.c
Vittorio Giovara dd3aa85b68 aac_ac3_parser: Drop in-parser downmix functionality
request_channel_layout is a decoder option and it makes no sense
to have it in a parser.

This feature was needed in the past when the decoder was allowed
to reuse the avctx from the demuxer. Nowadays the decoder receives
only the parameters from it, already containing the real channel
layout (and the correct request_channel_layout option).

After initialization the decoder overwrites the channel layout
with the downmixed one that is actually output, so there is no need
to preserve this functionality in the parser.

Signed-off-by: Vittorio Giovara <vittorio.giovara@gmail.com>
2017-04-27 14:19:50 -04:00

95 lines
3.1 KiB
C

/*
* Common AAC and AC-3 parser
* Copyright (c) 2003 Fabrice Bellard
* Copyright (c) 2003 Michael Niedermayer
*
* This file is part of Libav.
*
* Libav 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.
*
* Libav 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 Libav; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include "libavutil/channel_layout.h"
#include "libavutil/common.h"
#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 != AV_CODEC_ID_AAC) {
avctx->sample_rate = s->sample_rate;
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;
}