From: <wt...@ke...> - 2006-09-19 17:25:31
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CVS Root: /cvs/gstreamer Module: gst-plugins-good Changes by: wtay Date: Tue Sep 19 2006 17:25:27 UTC Log message: * gst/rtsp/gstrtspsrc.c: (find_stream_by_pt), (gst_rtspsrc_create_stream), (gst_rtspsrc_free_stream), (gst_rtspsrc_media_to_caps), (gst_rtspsrc_stream_setup_rtp), (gst_rtspsrc_stream_configure_transport), (find_stream_by_channel), (gst_rtspsrc_push_event), (gst_rtspsrc_loop), (gst_rtspsrc_send), (gst_rtspsrc_parse_methods), (gst_rtspsrc_open), (gst_rtspsrc_parse_rtpinfo), (gst_rtspsrc_play): * gst/rtsp/gstrtspsrc.h: Reorganize stream parsing and creation. Detect container formats in interleaved mode. Keep more state about the streams. Assume a server also supports PLAY if it does not say. Add unicast and interleaved properties to TCP transport requests to make some servers happy (WMServer). * gst/rtsp/sdpmessage.h: Add some defines for the standard Bandwidth types. Modified files: . : ChangeLog gst/rtsp : gstrtspsrc.c gstrtspsrc.h sdpmessage.h Links: http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gst-plugins-good/ChangeLog.diff?r1=1.2548&r2=1.2549 http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gst-plugins-good/gst/rtsp/gstrtspsrc.c.diff?r1=1.37&r2=1.38 http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gst-plugins-good/gst/rtsp/gstrtspsrc.h.diff?r1=1.14&r2=1.15 http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gst-plugins-good/gst/rtsp/sdpmessage.h.diff?r1=1.5&r2=1.6 |