Mercurial > libavformat.hg
view oggparsespeex.c @ 4164:56b7ebdf9ef4 libavformat
Parse the bitrate field in the ASMRuleBook ("AverageBandwidth") to fill in
the AVStream->AVCodecContext->bit_rate field, which is not in the MDPR block
(the "OpaqueData" SDP field). This allows clients to choose streams based
on their bitrate, which is what most network-players base stream selection
on. (Of course, it is also possible to select based on anything else, that
is entirely up to the client.) See "[PATCH] rdt.c: ASM rulebook bitrate
reading" thread on mailinglist.
author | rbultje |
---|---|
date | Wed, 07 Jan 2009 14:41:40 +0000 |
parents | 6cd006bc2de9 |
children | 6ab95f681099 |
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/* Copyright (C) 2008 Reimar Döffinger Permission is hereby granted, free of charge, to any person obtaining a copy of this software and associated documentation files (the "Software"), to deal in the Software without restriction, including without limitation the rights to use, copy, modify, merge, publish, distribute, sublicense, and/or sell copies of the Software, and to permit persons to whom the Software is furnished to do so, subject to the following conditions: The above copyright notice and this permission notice shall be included in all copies or substantial portions of the Software. THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE. **/ #include <stdlib.h> #include "libavutil/bswap.h" #include "libavutil/avstring.h" #include "libavcodec/bitstream.h" #include "libavcodec/bytestream.h" #include "avformat.h" #include "oggdec.h" static int speex_header(AVFormatContext *s, int idx) { struct ogg *ogg = s->priv_data; struct ogg_stream *os = ogg->streams + idx; AVStream *st = s->streams[idx]; uint8_t *p = os->buf + os->pstart; if (os->psize < 80) return 1; st->codec->codec_type = CODEC_TYPE_AUDIO; st->codec->codec_id = CODEC_ID_SPEEX; st->codec->sample_rate = AV_RL32(p + 36); st->codec->channels = AV_RL32(p + 48); st->codec->extradata_size = os->psize; st->codec->extradata = av_malloc(st->codec->extradata_size); memcpy(st->codec->extradata, p, st->codec->extradata_size); st->time_base.num = 1; st->time_base.den = st->codec->sample_rate; return 0; } const struct ogg_codec ff_speex_codec = { .magic = "Speex ", .magicsize = 8, .header = speex_header };