Mercurial > libavformat.hg
view daud.c @ 3894:30c8c9f53b9d libavformat
matroskadec: fix ASS subtitle track packets before emitting them
Matroska does some butchering when storing the ASS lines. The start and end
time are removed (because they are duplicated in the container).
The matroska_fix_ass_packet() function simply restore those start and end
time in ASS lines to ensure our ASS packets comply with the ASS spec.
author | aurel |
---|---|
date | Thu, 04 Sep 2008 23:26:12 +0000 |
parents | 0dd9806a26ea |
children | 1d3d17de20ba |
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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_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