Mercurial > libavformat.hg
view rsodec.c @ 6349:93c7a56fa912 libavformat
Add RTP packetization of Theora and Vorbis
Patch by Josh Allmann, joshua dot allmann at gmail
author | mstorsjo |
---|---|
date | Sat, 07 Aug 2010 11:16:07 +0000 |
parents | ab5b82b7b8e6 |
children | 4aaed59641ff |
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/* * RSO demuxer * Copyright (c) 2001 Fabrice Bellard (original AU code) * Copyright (c) 2010 Rafael Carre * * 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/intreadwrite.h" #include "avformat.h" #include "internal.h" #include "raw.h" #include "riff.h" #include "rso.h" static int rso_read_header(AVFormatContext *s, AVFormatParameters *ap) { ByteIOContext *pb = s->pb; int id, rate, bps; unsigned int size; enum CodecID codec; AVStream *st; id = get_be16(pb); size = get_be16(pb); rate = get_be16(pb); get_be16(pb); /* play mode ? (0x0000 = don't loop) */ codec = ff_codec_get_id(ff_codec_rso_tags, id); if (codec == CODEC_ID_ADPCM_IMA_WAV) { av_log(s, AV_LOG_ERROR, "ADPCM in RSO not implemented\n"); return AVERROR_PATCHWELCOME; } bps = av_get_bits_per_sample(codec); if (!bps) { av_log_ask_for_sample(s, "could not determine bits per sample\n"); return AVERROR_INVALIDDATA; } /* now we are ready: build format streams */ st = av_new_stream(s, 0); if (!st) return AVERROR(ENOMEM); st->duration = (size * 8) / bps; st->codec->codec_type = AVMEDIA_TYPE_AUDIO; st->codec->codec_tag = id; st->codec->codec_id = codec; st->codec->channels = 1; st->codec->sample_rate = rate; av_set_pts_info(st, 64, 1, rate); return 0; } #define BLOCK_SIZE 1024 /* in samples */ static int rso_read_packet(AVFormatContext *s, AVPacket *pkt) { int bps = av_get_bits_per_sample(s->streams[0]->codec->codec_id); int ret = av_get_packet(s->pb, pkt, BLOCK_SIZE * bps >> 3); if (ret < 0) return ret; pkt->stream_index = 0; /* note: we need to modify the packet size here to handle the last packet */ pkt->size = ret; return 0; } AVInputFormat rso_demuxer = { .name = "rso", .long_name = NULL_IF_CONFIG_SMALL("Lego Mindstorms RSO format"), .extensions = "rso", .priv_data_size = 0, .read_probe = NULL, /* no magic value in this format */ .read_header = rso_read_header, .read_packet = rso_read_packet, .read_close = NULL, .read_seek = pcm_read_seek, .codec_tag = (const AVCodecTag* const []){ff_codec_rso_tags, 0}, };