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/*
* 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 "rawdec.h"
#include "id3v2.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;
if (ff_id3v2_match(buf0, ID3v2_DEFAULT_MAGIC)) {
buf0 += ff_id3v2_tag_len(buf0);
}
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) & 0x8FFF;
if(fsize < 7)
break;
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,
AVFormatParameters *ap)
{
AVStream *st;
st = av_new_stream(s, 0);
if (!st)
return AVERROR(ENOMEM);
st->codec->codec_type = AVMEDIA_TYPE_AUDIO;
st->codec->codec_id = s->iformat->value;
st->need_parsing = AVSTREAM_PARSE_FULL;
ff_id3v1_read(s);
ff_id3v2_read(s, ID3v2_DEFAULT_MAGIC);
//LCM of all possible ADTS sample rates
av_set_pts_info(st, 64, 1, 28224000);
return 0;
}
AVInputFormat aac_demuxer = {
"aac",
NULL_IF_CONFIG_SMALL("raw ADTS AAC"),
0,
adts_aac_probe,
adts_aac_read_header,
ff_raw_read_partial_packet,
.flags= AVFMT_GENERIC_INDEX,
.extensions = "aac",
.value = CODEC_ID_AAC,
};
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