kolibrios-gitea/contrib/sdk/sources/ffmpeg/libavformat/aacdec.c
Sergey Semyonov (Serge) 754f9336f0 upload sdk
git-svn-id: svn://kolibrios.org@4349 a494cfbc-eb01-0410-851d-a64ba20cac60
2013-12-15 08:09:20 +00:00

101 lines
3.1 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"
#include "apetag.h"
static int adts_aac_probe(AVProbeData *p)
{
int max_frames = 0, first_frames = 0;
int fsize, frames;
const uint8_t *buf0 = p->buf;
const uint8_t *buf2;
const uint8_t *buf;
const 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_EXTENSION + 1;
else if(max_frames>500)return AVPROBE_SCORE_EXTENSION;
else if(max_frames>=3) return AVPROBE_SCORE_EXTENSION / 2;
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_RAW;
ff_id3v1_read(s);
if (s->pb->seekable &&
!av_dict_get(s->metadata, "", NULL, AV_DICT_IGNORE_SUFFIX)) {
int64_t cur = avio_tell(s->pb);
ff_ape_parse_tag(s);
avio_seek(s->pb, cur, SEEK_SET);
}
//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 (Advanced Audio Coding)"),
.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 = AV_CODEC_ID_AAC,
};