ffmpeg/libavformat/loasdec.c

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/*
* 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"
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#include "rawdec.h"
static int loas_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 - 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_MAX/2+1;
else if(max_frames>100)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;
}
Merge remote-tracking branch 'qatar/master' * qatar/master: (71 commits) movenc: Allow writing to a non-seekable output if using empty moov movenc: Support adding isml (smooth streaming live) metadata libavcodec: Don't crash in avcodec_encode_audio if time_base isn't set sunrast: Document the different Sun Raster file format types. sunrast: Add a check for experimental type. libspeexenc: use AVSampleFormat instead of deprecated/removed SampleFormat lavf: remove disabled FF_API_SET_PTS_INFO cruft lavf: remove disabled FF_API_OLD_INTERRUPT_CB cruft lavf: remove disabled FF_API_REORDER_PRIVATE cruft lavf: remove disabled FF_API_SEEK_PUBLIC cruft lavf: remove disabled FF_API_STREAM_COPY cruft lavf: remove disabled FF_API_PRELOAD cruft lavf: remove disabled FF_API_NEW_STREAM cruft lavf: remove disabled FF_API_RTSP_URL_OPTIONS cruft lavf: remove disabled FF_API_MUXRATE cruft lavf: remove disabled FF_API_FILESIZE cruft lavf: remove disabled FF_API_TIMESTAMP cruft lavf: remove disabled FF_API_LOOP_OUTPUT cruft lavf: remove disabled FF_API_LOOP_INPUT cruft lavf: remove disabled FF_API_AVSTREAM_QUALITY cruft ... Conflicts: doc/APIchanges libavcodec/8bps.c libavcodec/avcodec.h libavcodec/libx264.c libavcodec/mjpegbdec.c libavcodec/options.c libavcodec/sunrast.c libavcodec/utils.c libavcodec/version.h libavcodec/x86/h264_deblock.asm libavdevice/libdc1394.c libavdevice/v4l2.c libavformat/avformat.h libavformat/avio.c libavformat/avio.h libavformat/aviobuf.c libavformat/dv.c libavformat/mov.c libavformat/utils.c libavformat/version.h libavformat/wtv.c libavutil/Makefile libavutil/file.c libswscale/x86/input.asm libswscale/x86/swscale_mmx.c libswscale/x86/swscale_template.c tests/ref/lavf/ffm Merged-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-28 03:23:26 +00:00
static int loas_read_header(AVFormatContext *s)
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{
AVStream *st;
st = avformat_new_stream(s, NULL);
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if (!st)
return AVERROR(ENOMEM);
st->codec->codec_type = AVMEDIA_TYPE_AUDIO;
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st->codec->codec_id = s->iformat->raw_codec_id;
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st->need_parsing = AVSTREAM_PARSE_FULL;
//LCM of all possible AAC sample rates
avpriv_set_pts_info(st, 64, 1, 28224000);
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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,
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.raw_codec_id = CODEC_ID_AAC_LATM,
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};