aboutsummaryrefslogtreecommitdiffstats
path: root/libavformat/aacdec.c
blob: 8737e56409871ce17c8a0952f7d1988f412b55ae (plain) (blame)
1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
73
74
75
76
77
78
79
80
81
82
83
84
85
86
87
88
89
90
91
92
93
94
95
96
/*
 * 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 "raw.h"
#include "id3v2.h"
#include "id3v1.h"


static int adts_aac_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 - 7;

    if (ff_id3v2_match(buf0, ID3v2_DEFAULT_MAGIC)) {
        buf0 += ff_id3v2_tag_len(buf0);
    }
    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)
                break;
            fsize = (AV_RB32(buf2+3)>>13) & 0x8FFF;
            if(fsize < 7)
                break;
            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>500)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 adts_aac_read_header(AVFormatContext *s,
                                AVFormatParameters *ap)
{
    AVStream *st;

    st = av_new_stream(s, 0);
    if (!st)
        return AVERROR(ENOMEM);

    st->codec->codec_type = AVMEDIA_TYPE_AUDIO;
    st->codec->codec_id = s->iformat->value;
    st->need_parsing = AVSTREAM_PARSE_FULL;

    ff_id3v1_read(s);
    ff_id3v2_read(s, ID3v2_DEFAULT_MAGIC);

    return 0;
}

AVInputFormat aac_demuxer = {
    "aac",
    NULL_IF_CONFIG_SMALL("raw ADTS AAC"),
    0,
    adts_aac_probe,
    adts_aac_read_header,
    ff_raw_read_partial_packet,
    .flags= AVFMT_GENERIC_INDEX,
    .extensions = "aac",
    .value = CODEC_ID_AAC,
};