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  • Wim Taymans's avatar
    gst/rtsp/gstrtspsrc.*: Reorganize stream parsing and creation. · a7d7309e
    Wim Taymans authored
    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.
    a7d7309e