kolibrios/contrib/sdk/sources/ffmpeg/libavformat/loasdec.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

88 lines
2.7 KiB
C

/*
* LOAS AudioSyncStream demuxer
* Copyright (c) 2008 Michael Niedermayer <michaelni@gmx.at>
*
* 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 "libavutil/internal.h"
#include "avformat.h"
#include "internal.h"
#include "rawdec.h"
static int loas_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 - 3;
buf = buf0;
for(; buf < end; buf= buf2+1) {
buf2 = buf;
for(frames = 0; buf2 < end; frames++) {
uint32_t header = AV_RB24(buf2);
if((header >> 13) != 0x2B7)
break;
fsize = (header & 0x1FFF) + 3;
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>100)return AVPROBE_SCORE_EXTENSION;
else if(max_frames>=3) return AVPROBE_SCORE_EXTENSION / 2;
else return 0;
}
static int loas_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;
//LCM of all possible AAC sample rates
avpriv_set_pts_info(st, 64, 1, 28224000);
return 0;
}
AVInputFormat ff_loas_demuxer = {
.name = "loas",
.long_name = NULL_IF_CONFIG_SMALL("LOAS AudioSyncStream"),
.read_probe = loas_probe,
.read_header = loas_read_header,
.read_packet = ff_raw_read_partial_packet,
.flags= AVFMT_GENERIC_INDEX,
.raw_codec_id = AV_CODEC_ID_AAC_LATM,
};