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eadd4264ee
* 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cosmetics: remove some pointless comments libmp3lame: convert some debugging code to av_dlog() libmp3lame: remove outdated comment.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>
104 lines
3.4 KiB
C
104 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 "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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s1->duration = 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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