view daud.c @ 4095:27f4b31bc790 libavformat

Separate the packet fetching from the data reading, so that the data reading function is assured to parse at most one packet. This makes this function useful for ASF data packet parsing in a "push-mode" in addition to the current "pull-mode", and therefore allows for use of these functions in, for example, the RTSP demuxer (for MS-RTSP support). Tested to give identical output before and after for regular ASF playback, also see discussion in the ML thread "[PATCH] asf.c: move packet_time_start=0 statement". Testsuite also works after the patch, tested by Benoit Fouet.

/*
 * 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