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FFmpeg/libavformat/daud.c
Michael Niedermayer a2beabc231 Merge remote branch 'qatar/master'
* qatar/master:
  daud: Don't write packets that are too large to have their size expressed in the bytestream.
DUPLICATE  pthread: validate_thread_parameters() ignored slice-threading being intentionally off
DUPLICATE  Use consistent condition for whether to run slice-threading execute function.
  avformat.h: fix a misplaced #endif
DUPLCIATE  vaapi: cope with VA-API 0.32, i.e. fix VC-1 decoding on Sandy Bridge.
DUPLICATE  vaapi: filter out DELAYED_PIC_REF flag to determine field.

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2011-04-10 03:32:29 +02:00

101 lines
2.7 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 "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 = AVMEDIA_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) {
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);
avio_flush(s->pb);
return 0;
}
#if CONFIG_DAUD_DEMUXER
AVInputFormat ff_daud_demuxer = {
"daud",
NULL_IF_CONFIG_SMALL("D-Cinema audio format"),
0,
NULL,
daud_header,
daud_packet,
NULL,
NULL,
.extensions = "302",
};
#endif
#if CONFIG_DAUD_MUXER
AVOutputFormat ff_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