[Bug 758179] GstRTSPStream : Create pipeline based on enabled transport type

[GStreamer (GNOME Bugzilla)]

https://bugzilla.gnome.org/show_bug.cgi?id=758179

Olivier Crête <olivier.crete at ocrete.ca> changed:

           What    |Removed                     |Added
----------------------------------------------------------------------------
 Attachment #315688|none                        |needs-work
             status|                            |

--- Comment #3 from Olivier Crête <olivier.crete at ocrete.ca> ---
Review of attachment 315688:
 --> (https://bugzilla.gnome.org/review?bug=758179&attachment=315688)

::: gst/rtsp-server/rtsp-client.c
@@ +1826,3 @@
+    /* need to suspend the media, if the protocol has changed */
+    if (media != NULL)
+      gst_rtsp_media_suspend (media);

You can't do that in the general case, because the media may be shared between
multiple sessions. For example if you have multiple viewers for the same
camera.

You can probably just always leave the tee there and then add or remove
branches. If you gst_element_release_request_pad() on a tee src pad, then you
can unlink the downstream bits while it's playing.

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