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mirror of https://git.ffmpeg.org/ffmpeg.git synced 2024-12-23 15:53:08 +00:00
ffmpeg/libavformat/loasdec.c
Michael Niedermayer a369a6b858 Merge remote-tracking branch 'qatar/master'
* qatar/master: (29 commits)
  fate: add golomb-test
  golomb-test: K&R formatting cosmetics
  h264: Split h264-test off into a separate file - golomb-test.c.
  h264-test: cleanup: drop timer invocations, commented out code and other cruft
  h264-test: Remove unused DSP and AVCodec contexts and related init calls.
  adpcm: Add missing stdint.h #include to fix standalone header compilation.
  lavf: add functions for accessing the fourcc<->CodecID mapping tables.
  lavc: set AVCodecContext.codec in avcodec_get_context_defaults3().
  lavc: make avcodec_close() work properly on unopened codecs.
  lavc: add avcodec_is_open().
  lavf: rename AVInputFormat.value to raw_codec_id.
  lavf: remove the pointless value field from flv and iv8
  lavc/lavf: remove unnecessary symbols from the symbol version script.
  lavc: reorder AVCodec fields.
  lavf: reorder AVInput/OutputFormat fields.
  mp3dec: Fix a heap-buffer-overflow
  adpcmenc: remove some unneeded casts
  adpcmenc: use int16_t and uint8_t instead of short and unsigned char.
  adpcmenc: fix adpcm_ms extradata allocation
  adpcmenc: return proper AVERROR codes instead of -1
  ...

Conflicts:
	doc/APIchanges
	libavcodec/Makefile
	libavcodec/adpcmenc.c
	libavcodec/avcodec.h
	libavcodec/h264.c
	libavcodec/libavcodec.v
	libavcodec/mpc7.c
	libavcodec/mpegaudiodec.c
	libavcodec/options.c
	libavformat/Makefile
	libavformat/avformat.h
	libavformat/flvdec.c
	libavformat/libavformat.v

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2012-02-01 02:36:09 +01:00

89 lines
2.6 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;
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;
}
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;
//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 = CODEC_ID_AAC_LATM,
};