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https://github.com/xenia-project/FFmpeg.git
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8df8a92993
If a frame starts very close to a packet boundary, the start code may already have been added to the parsing buffer, indicated by a small negative value of "i", while the header is still being tracked in the "state" variable. Reduce the remaining size accordingly, otherwise trying to find the next frame could skip over the frame header and lump two frames together as one.
106 lines
3.4 KiB
C
106 lines
3.4 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 "libavutil/channel_layout.h"
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#include "libavutil/common.h"
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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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int got_frame = 0;
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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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got_frame = 1;
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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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else if (i < 0) {
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s->remaining_size += i;
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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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if (got_frame) {
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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
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duration in seconds is still correct (as is the number of bits in
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the frame). */
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if (avctx->codec_id != AV_CODEC_ID_AAC) {
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avctx->sample_rate = s->sample_rate;
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if (avctx->codec_id != AV_CODEC_ID_EAC3) {
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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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s1->duration = s->samples;
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avctx->audio_service_type = s->service_type;
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}
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if (avctx->codec_id != AV_CODEC_ID_EAC3)
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avctx->bit_rate = s->bit_rate;
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}
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return i;
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}
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