FFmpeg/libavformat/aacdec.c
Michael Niedermayer 2e0c360abd Merge remote-tracking branch 'qatar/master'
* qatar/master:
  cosmetics: Align muxer/demuxer declarations
  mpeg12: Do not change frame_pred_frame_dct flag and demote error into a warning
  avcodec: remove avcodec_guess_channel_layout()
  avutil: Add av_get_default_channel_layout()

Conflicts:
	doc/APIchanges
	libavcodec/mpeg12.c
	libavformat/cdg.c
	libavformat/matroskaenc.c
	libavformat/mpegts.c
	libavformat/nuv.c
	libavformat/wav.c
	libavutil/audioconvert.c
	libavutil/audioconvert.h
	libavutil/avutil.h

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2012-04-06 22:52:01 +02:00

95 lines
2.8 KiB
C

/*
* raw ADTS AAC demuxer
* Copyright (c) 2008 Michael Niedermayer <michaelni@gmx.at>
* Copyright (c) 2009 Robert Swain ( rob opendot cl )
*
* 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 "libavutil/intreadwrite.h"
#include "avformat.h"
#include "internal.h"
#include "rawdec.h"
#include "id3v1.h"
static int adts_aac_probe(AVProbeData *p)
{
int max_frames = 0, first_frames = 0;
int fsize, frames;
uint8_t *buf0 = p->buf;
uint8_t *buf2;
uint8_t *buf;
uint8_t *end = buf0 + p->buf_size - 7;
buf = buf0;
for(; buf < end; buf= buf2+1) {
buf2 = buf;
for(frames = 0; buf2 < end; frames++) {
uint32_t header = AV_RB16(buf2);
if((header&0xFFF6) != 0xFFF0)
break;
fsize = (AV_RB32(buf2 + 3) >> 13) & 0x1FFF;
if(fsize < 7)
break;
fsize = FFMIN(fsize, end - buf2);
buf2 += fsize;
}
max_frames = FFMAX(max_frames, frames);
if(buf == buf0)
first_frames= frames;
}
if (first_frames>=3) return AVPROBE_SCORE_MAX/2+1;
else if(max_frames>500)return AVPROBE_SCORE_MAX/2;
else if(max_frames>=3) return AVPROBE_SCORE_MAX/4;
else if(max_frames>=1) return 1;
else return 0;
}
static int adts_aac_read_header(AVFormatContext *s)
{
AVStream *st;
st = avformat_new_stream(s, NULL);
if (!st)
return AVERROR(ENOMEM);
st->codec->codec_type = AVMEDIA_TYPE_AUDIO;
st->codec->codec_id = s->iformat->raw_codec_id;
st->need_parsing = AVSTREAM_PARSE_FULL;
ff_id3v1_read(s);
//LCM of all possible ADTS sample rates
avpriv_set_pts_info(st, 64, 1, 28224000);
return 0;
}
AVInputFormat ff_aac_demuxer = {
.name = "aac",
.long_name = NULL_IF_CONFIG_SMALL("raw ADTS AAC"),
.read_probe = adts_aac_probe,
.read_header = adts_aac_read_header,
.read_packet = ff_raw_read_partial_packet,
.flags = AVFMT_GENERIC_INDEX,
.extensions = "aac",
.raw_codec_id = CODEC_ID_AAC,
};