3194ab78a6
* qatar/master: avcodec: add a cook parser to get subpacket duration FATE: allow lavf tests to alter input parameters FATE: replace the acodec-pcm_s24daud test with an enc_dec_pcm checksum test FATE: replace the acodec-g726 test with 4 new encode/decode tests FATE: replace current g722 encoding tests with an encode/decode test FATE: add a pattern rule for generating asynth wav files FATE: optionally write a WAVE header in audiogen avutil: add audio fifo buffer Conflicts: doc/APIchanges libavcodec/version.h libavutil/avutil.h tests/Makefile tests/codec-regression.sh tests/fate/voice.mak tests/lavf-regression.sh tests/ref/acodec/g722 tests/ref/acodec/g726 tests/ref/acodec/pcm_s24daud tests/ref/lavf/dv_fmt tests/ref/lavf/gxf tests/ref/lavf/mxf tests/ref/lavf/mxf_d10 tests/ref/seek/lavf_dv Merged-by: Michael Niedermayer <michaelni@gmx.at>
94 lines
2.8 KiB
C
94 lines
2.8 KiB
C
/*
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* D-Cinema audio demuxer
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* Copyright (c) 2005 Reimar Döffinger
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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#include "avformat.h"
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static int daud_header(AVFormatContext *s) {
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AVStream *st = avformat_new_stream(s, NULL);
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if (!st)
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return AVERROR(ENOMEM);
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st->codec->codec_type = AVMEDIA_TYPE_AUDIO;
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st->codec->codec_id = CODEC_ID_PCM_S24DAUD;
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st->codec->codec_tag = MKTAG('d', 'a', 'u', 'd');
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st->codec->channels = 6;
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st->codec->sample_rate = 96000;
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st->codec->bit_rate = 3 * 6 * 96000 * 8;
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st->codec->block_align = 3 * 6;
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st->codec->bits_per_coded_sample = 24;
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return 0;
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}
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static int daud_packet(AVFormatContext *s, AVPacket *pkt) {
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AVIOContext *pb = s->pb;
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int ret, size;
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if (url_feof(pb))
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return AVERROR(EIO);
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size = avio_rb16(pb);
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avio_rb16(pb); // unknown
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ret = av_get_packet(pb, pkt, size);
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pkt->stream_index = 0;
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return ret;
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}
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static int daud_write_header(struct AVFormatContext *s)
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{
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AVCodecContext *codec = s->streams[0]->codec;
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if (codec->channels!=6 || codec->sample_rate!=96000)
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return -1;
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return 0;
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}
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static int daud_write_packet(struct AVFormatContext *s, AVPacket *pkt)
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{
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if (pkt->size > 65535) {
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av_log(s, AV_LOG_ERROR,
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"Packet size too large for s302m. (%d > 65535)\n", pkt->size);
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return -1;
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}
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avio_wb16(s->pb, pkt->size);
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avio_wb16(s->pb, 0x8010); // unknown
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avio_write(s->pb, pkt->data, pkt->size);
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avio_flush(s->pb);
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return 0;
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}
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#if CONFIG_DAUD_DEMUXER
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AVInputFormat ff_daud_demuxer = {
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.name = "daud",
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.long_name = NULL_IF_CONFIG_SMALL("D-Cinema audio format"),
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.read_header = daud_header,
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.read_packet = daud_packet,
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.extensions = "302,daud",
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};
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#endif
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#if CONFIG_DAUD_MUXER
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AVOutputFormat ff_daud_muxer = {
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.name = "daud",
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.long_name = NULL_IF_CONFIG_SMALL("D-Cinema audio format"),
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.extensions = "302",
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.audio_codec = CODEC_ID_PCM_S24DAUD,
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.video_codec = CODEC_ID_NONE,
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.write_header = daud_write_header,
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.write_packet = daud_write_packet,
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.flags = AVFMT_NOTIMESTAMPS,
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};
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#endif
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