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author | Wim Taymans <wim.taymans@gmail.com> | 2006-09-19 17:25:15 +0000 |
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committer | Wim Taymans <wim.taymans@gmail.com> | 2006-09-19 17:25:15 +0000 |
commit | a7d7309e18dca433661058e54651c2d1b78edc6a (patch) | |
tree | dac7af9ddafd02e800262888aebbc4feb9542abc /gst/rtp/gstrtp.c | |
parent | 34fd3a416d9249ea78df1a3fcbc8584afc8887dd (diff) |
gst/rtsp/gstrtspsrc.*: Reorganize stream parsing and creation.
Original commit message from CVS:
* 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.
Diffstat (limited to 'gst/rtp/gstrtp.c')
0 files changed, 0 insertions, 0 deletions