view daud.c @ 3957:9f943bb755f9 libavformat

Rename RTSPProtocol to RTSPLowerTransport, so that its name properly tells us that it only describes the lower-level transport (TCP vs. UDP) and not the actual data layout (e.g. RDT vs. RTP). See discussion in "Realmedia patch" thread on ML.
author rbultje
date Tue, 30 Sep 2008 13:18:41 +0000
parents 1d3d17de20ba
children c3102b189cb6
line wrap: on
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/*
 * 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_coded_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