aacdec.c 5.84 KB
Newer Older
1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16 17 18 19 20 21 22 23 24
/*
 * 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"
25
#include "avio_internal.h"
26
#include "internal.h"
27
#include "id3v1.h"
28
#include "id3v2.h"
29
#include "apetag.h"
30

31 32
#define ADTS_HEADER_SIZE 7

33 34 35 36
static int adts_aac_probe(AVProbeData *p)
{
    int max_frames = 0, first_frames = 0;
    int fsize, frames;
37 38 39 40
    const uint8_t *buf0 = p->buf;
    const uint8_t *buf2;
    const uint8_t *buf;
    const uint8_t *end = buf0 + p->buf_size - 7;
41 42 43

    buf = buf0;

44
    for (; buf < end; buf = buf2 + 1) {
45 46
        buf2 = buf;

47
        for (frames = 0; buf2 < end; frames++) {
48
            uint32_t header = AV_RB16(buf2);
49 50 51 52 53 54 55 56
            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;
                }
57
                break;
58
            }
59
            fsize = (AV_RB32(buf2 + 3) >> 13) & 0x1FFF;
60
            if (fsize < 7)
61
                break;
62
            fsize = FFMIN(fsize, end - buf2);
63 64 65
            buf2 += fsize;
        }
        max_frames = FFMAX(max_frames, frames);
66 67
        if (buf == buf0)
            first_frames = frames;
68
    }
69 70 71

    if (first_frames >= 3)
        return AVPROBE_SCORE_EXTENSION + 1;
72
    else if (max_frames > 100)
73 74 75
        return AVPROBE_SCORE_EXTENSION;
    else if (max_frames >= 3)
        return AVPROBE_SCORE_EXTENSION / 2;
76
    else if (first_frames >= 1)
77 78 79
        return 1;
    else
        return 0;
80 81
}

82
static int adts_aac_read_header(AVFormatContext *s)
83 84
{
    AVStream *st;
85
    uint16_t state;
86

87
    st = avformat_new_stream(s, NULL);
88 89 90
    if (!st)
        return AVERROR(ENOMEM);

91 92
    st->codecpar->codec_type = AVMEDIA_TYPE_AUDIO;
    st->codecpar->codec_id   = s->iformat->raw_codec_id;
93
    st->need_parsing         = AVSTREAM_PARSE_FULL_RAW;
94 95

    ff_id3v1_read(s);
James Almer's avatar
James Almer committed
96
    if ((s->pb->seekable & AVIO_SEEKABLE_NORMAL) &&
97 98 99 100 101
        !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);
    }
102

103 104 105 106 107 108 109 110 111 112 113 114
    // skip data until the first ADTS frame is found
    state = avio_r8(s->pb);
    while (!avio_feof(s->pb) && avio_tell(s->pb) < s->probesize) {
        state = (state << 8) | avio_r8(s->pb);
        if ((state >> 4) != 0xFFF)
            continue;
        avio_seek(s->pb, -2, SEEK_CUR);
        break;
    }
    if ((state >> 4) != 0xFFF)
        return AVERROR_INVALIDDATA;

115
    // LCM of all possible ADTS sample rates
116
    avpriv_set_pts_info(st, 64, 1, 28224000);
117

118 119 120
    return 0;
}

121 122 123 124 125 126 127 128 129 130 131 132 133 134 135 136 137 138 139 140 141 142 143 144 145 146 147 148 149 150 151 152
static int handle_id3(AVFormatContext *s, AVPacket *pkt)
{
    AVDictionary *metadata = NULL;
    AVIOContext ioctx;
    ID3v2ExtraMeta *id3v2_extra_meta = NULL;
    int ret;

    ret = av_append_packet(s->pb, pkt, ff_id3v2_tag_len(pkt->data) - pkt->size);
    if (ret < 0) {
        av_packet_unref(pkt);
        return ret;
    }

    ffio_init_context(&ioctx, pkt->data, pkt->size, 0, NULL, NULL, NULL, NULL);
    ff_id3v2_read_dict(&ioctx, &metadata, ID3v2_DEFAULT_MAGIC, &id3v2_extra_meta);
    if ((ret = ff_id3v2_parse_priv_dict(&metadata, &id3v2_extra_meta)) < 0)
        goto error;

    if (metadata) {
        if ((ret = av_dict_copy(&s->metadata, metadata, 0)) < 0)
            goto error;
        s->event_flags |= AVFMT_EVENT_FLAG_METADATA_UPDATED;
    }

error:
    av_packet_unref(pkt);
    ff_id3v2_free_extra_meta(&id3v2_extra_meta);
    av_dict_free(&metadata);

    return ret;
}

153 154 155 156
static int adts_aac_read_packet(AVFormatContext *s, AVPacket *pkt)
{
    int ret, fsize;

157 158 159 160 161 162 163 164 165
    // Parse all the ID3 headers between frames
    while (1) {
        ret = av_get_packet(s->pb, pkt, FFMAX(ID3v2_HEADER_SIZE, ADTS_HEADER_SIZE));
        if (ret >= ID3v2_HEADER_SIZE && ff_id3v2_match(pkt->data, ID3v2_DEFAULT_MAGIC)) {
            if ((ret = handle_id3(s, pkt)) >= 0) {
                continue;
            }
        }
        break;
166 167
    }

168 169
    if (ret < 0)
        return ret;
170

171 172 173 174 175 176 177 178 179 180 181 182 183 184 185 186
    if (ret < ADTS_HEADER_SIZE) {
        av_packet_unref(pkt);
        return AVERROR(EIO);
    }

    if ((AV_RB16(pkt->data) >> 4) != 0xfff) {
        av_packet_unref(pkt);
        return AVERROR_INVALIDDATA;
    }

    fsize = (AV_RB32(pkt->data + 3) >> 13) & 0x1FFF;
    if (fsize < ADTS_HEADER_SIZE) {
        av_packet_unref(pkt);
        return AVERROR_INVALIDDATA;
    }

187
    ret = av_append_packet(s->pb, pkt, fsize - pkt->size);
188 189 190 191
    if (ret < 0)
        av_packet_unref(pkt);

    return ret;
192 193
}

194
AVInputFormat ff_aac_demuxer = {
195 196 197 198
    .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,
199
    .read_packet  = adts_aac_read_packet,
200 201
    .flags        = AVFMT_GENERIC_INDEX,
    .extensions   = "aac",
202
    .mime_type    = "audio/aac,audio/aacp,audio/x-aac",
203
    .raw_codec_id = AV_CODEC_ID_AAC,
204
};