mirror of
https://git.ffmpeg.org/ffmpeg.git
synced 2024-12-18 05:24:42 +00:00
b800327f4c
This commit does for AVInputFormat what commit59c9dc82f4did for AVOutputFormat: It adds a new type FFInputFormat, moves all the internals of AVInputFormat to it and adds a now reduced AVInputFormat as first member. This does not affect/improve extensibility of both public or private fields for demuxers (it is still a mess due to lavd). This is possible since50f34172e0(which removed the last usage of an internal field of AVInputFormat in fftools). (Hint: tools/probetest.c accesses the internals of FFInputFormat as well, but given that it is a testing tool this is not considered a problem.) Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
98 lines
2.9 KiB
C
98 lines
2.9 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 "demux.h"
|
|
#include "internal.h"
|
|
#include "rawdec.h"
|
|
|
|
#define LOAS_SYNC_WORD 0x2b7
|
|
|
|
static int loas_probe(const 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) != LOAS_SYNC_WORD)
|
|
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->codecpar->codec_type = AVMEDIA_TYPE_AUDIO;
|
|
st->codecpar->codec_id = AV_CODEC_ID_AAC_LATM;
|
|
ffstream(st)->need_parsing = AVSTREAM_PARSE_FULL_RAW;
|
|
|
|
//LCM of all possible AAC sample rates
|
|
avpriv_set_pts_info(st, 64, 1, 28224000);
|
|
|
|
return 0;
|
|
}
|
|
|
|
const FFInputFormat ff_loas_demuxer = {
|
|
.p.name = "loas",
|
|
.p.long_name = NULL_IF_CONFIG_SMALL("LOAS AudioSyncStream"),
|
|
.p.flags = AVFMT_GENERIC_INDEX,
|
|
.p.priv_class = &ff_raw_demuxer_class,
|
|
.read_probe = loas_probe,
|
|
.read_header = loas_read_header,
|
|
.read_packet = ff_raw_read_partial_packet,
|
|
.raw_codec_id = AV_CODEC_ID_AAC_LATM,
|
|
.priv_data_size = sizeof(FFRawDemuxerContext),
|
|
};
|