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FFmpeg/libavformat/aacdec.c
James Almer 4de591e6fb Merge commit '83548fe894cdb455cc127f754d09905b6d23c173'
* commit '83548fe894cdb455cc127f754d09905b6d23c173':
  lavf: fix usage of AVIOContext.seekable

Merged-by: James Almer <jamrial@gmail.com>
2017-03-21 17:02:30 -03:00

116 lines
3.5 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) {
if (buf != buf0) {
// Found something that isn't an ADTS header, starting
// from a position other than the start of the buffer.
// Discard the count we've accumulated so far since it
// probably was a false positive.
frames = 0;
}
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 > 100)
return AVPROBE_SCORE_EXTENSION;
else if (max_frames >= 3)
return AVPROBE_SCORE_EXTENSION / 2;
else if (first_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->codecpar->codec_type = AVMEDIA_TYPE_AUDIO;
st->codecpar->codec_id = s->iformat->raw_codec_id;
st->need_parsing = AVSTREAM_PARSE_FULL_RAW;
ff_id3v1_read(s);
if ((s->pb->seekable & AVIO_SEEKABLE_NORMAL) &&
!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",
.mime_type = "audio/aac,audio/aacp,audio/x-aac",
.raw_codec_id = AV_CODEC_ID_AAC,
};