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

[GStreamer (GNOME Bugzilla)]

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

--- Comment #5 from Srimanta Panda <panda_srimanta at yahoo.co.in> ---
(In reply to Olivier Crête from comment #3)
> Review of attachment 315688 [details] [review]:
> 
> ::: 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.

I should have submitted the above particular change as a separate patch
instead. handle_play_request() and handle_setup_request() expect the media to
be suspended (for example block-size from setup request will not affect preroll
buffer unless the media is suspended).

Shared medias are no problems since rtsp_media_suspend() takes care about that
(priv->n_active).

> 
> 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.

Adding and removing branches dynamically and reconfiguring the pipeline is the
right solution and we agree on that. But this patch is still an improvement.

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