FFmpeg/libavformat/loasdec.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"

#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 = s->iformat->raw_codec_id;
    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 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 = AV_CODEC_ID_AAC_LATM,
    .priv_data_size = sizeof(FFRawDemuxerContext),
    .priv_class     = &ff_raw_demuxer_class,
};