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FFmpeg/libavformat/daud.c
Clément Bœsch 7308439158 lavf: Don't explicitly flush after each written packet in muxers
Since 596e5d4783, this is not necessary anymore. It also allows to
actually disable the flushing, improving write performance (but
possibly giving worse latency in real-time streaming).

Signed-off-by: Martin Storsjö <martin@martin.st>
2013-09-16 22:17:33 +03:00

96 lines
2.9 KiB
C

/*
* 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 "libavutil/channel_layout.h"
#include "avformat.h"
static int daud_header(AVFormatContext *s) {
AVStream *st = avformat_new_stream(s, NULL);
if (!st)
return AVERROR(ENOMEM);
st->codec->codec_type = AVMEDIA_TYPE_AUDIO;
st->codec->codec_id = AV_CODEC_ID_PCM_S24DAUD;
st->codec->codec_tag = MKTAG('d', 'a', 'u', 'd');
st->codec->channels = 6;
st->codec->channel_layout = AV_CH_LAYOUT_5POINT1;
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) {
AVIOContext *pb = s->pb;
int ret, size;
if (url_feof(pb))
return AVERROR(EIO);
size = avio_rb16(pb);
avio_rb16(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)
{
if (pkt->size > 65535) {
av_log(s, AV_LOG_ERROR,
"Packet size too large for s302m. (%d > 65535)\n", pkt->size);
return -1;
}
avio_wb16(s->pb, pkt->size);
avio_wb16(s->pb, 0x8010); // unknown
avio_write(s->pb, pkt->data, pkt->size);
return 0;
}
#if CONFIG_DAUD_DEMUXER
AVInputFormat ff_daud_demuxer = {
.name = "daud",
.long_name = NULL_IF_CONFIG_SMALL("D-Cinema audio"),
.read_header = daud_header,
.read_packet = daud_packet,
.extensions = "302,daud",
};
#endif
#if CONFIG_DAUD_MUXER
AVOutputFormat ff_daud_muxer = {
.name = "daud",
.long_name = NULL_IF_CONFIG_SMALL("D-Cinema audio"),
.extensions = "302",
.audio_codec = AV_CODEC_ID_PCM_S24DAUD,
.video_codec = AV_CODEC_ID_NONE,
.write_header = daud_write_header,
.write_packet = daud_write_packet,
.flags = AVFMT_NOTIMESTAMPS,
};
#endif