Mercurial > libavformat.hg
view oggparsespeex.c @ 4332:0d776969b708 libavformat
Fix the Transport: line in the SETUP request so that it works with WMS
servers when trying to set up a session over TCP:
- add the interleave property
- add unicast, only for WMS (since it is normally only UDP, but WMS expects it
for UDP and TCP)
- add mode=play
See discussion in "[PATCH] RTSP-MS 9/15: add interleave property to the TCP
transport line of the SETUP request" thread on mailinglist.
author | rbultje |
---|---|
date | Sun, 01 Feb 2009 13:37:45 +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 };