view daud.c @ 3894:30c8c9f53b9d libavformat

matroskadec: fix ASS subtitle track packets before emitting them Matroska does some butchering when storing the ASS lines. The start and end time are removed (because they are duplicated in the container). The matroska_fix_ass_packet() function simply restore those start and end time in ASS lines to ensure our ASS packets comply with the ASS spec.

/*
 * D-Cinema audio demuxer
 * Copyright (c) 2005 Reimar Döffinger
 *
 * 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 "avformat.h"

static int daud_header(AVFormatContext *s, AVFormatParameters *ap) {
    AVStream *st = av_new_stream(s, 0);
    if (!st)
        return AVERROR(ENOMEM);
    st->codec->codec_type = CODEC_TYPE_AUDIO;
    st->codec->codec_id = CODEC_ID_PCM_S24DAUD;
    st->codec->codec_tag = MKTAG('d', 'a', 'u', 'd');
    st->codec->channels = 6;
    st->codec->sample_rate = 96000;
    st->codec->bit_rate = 3 * 6 * 96000 * 8;
    st->codec->block_align = 3 * 6;
    st->codec->bits_per_sample = 24;
    return 0;
}

static int daud_packet(AVFormatContext *s, AVPacket *pkt) {
    ByteIOContext *pb = s->pb;
    int ret, size;
    if (url_feof(pb))
        return AVERROR(EIO);
    size = get_be16(pb);
    get_be16(pb); // unknown
    ret = av_get_packet(pb, pkt, size);
    pkt->stream_index = 0;
    return ret;
}

static int daud_write_header(struct AVFormatContext *s)
{
    AVCodecContext *codec = s->streams[0]->codec;
    if (codec->channels!=6 || codec->sample_rate!=96000)
        return -1;
    return 0;
}

static int daud_write_packet(struct AVFormatContext *s, AVPacket *pkt)
{
    put_be16(s->pb, pkt->size);
    put_be16(s->pb, 0x8010); // unknown
    put_buffer(s->pb, pkt->data, pkt->size);
    put_flush_packet(s->pb);
    return 0;
}

#if CONFIG_DAUD_DEMUXER
AVInputFormat daud_demuxer = {
    "daud",
    NULL_IF_CONFIG_SMALL("D-Cinema audio format"),
    0,
    NULL,
    daud_header,
    daud_packet,
    NULL,
    NULL,
    .extensions = "302",
};
#endif

#ifdef CONFIG_DAUD_MUXER
AVOutputFormat daud_muxer =
{
    "daud",
    NULL_IF_CONFIG_SMALL("D-Cinema audio format"),
    NULL,
    "302",
    0,
    CODEC_ID_PCM_S24DAUD,
    CODEC_ID_NONE,
    daud_write_header,
    daud_write_packet,
    .flags= AVFMT_NOTIMESTAMPS,
};
#endif