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

[GStreamer (GNOME Bugzilla)]

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

Sebastian Dröge (slomo) <slomo at coaxion.net> changed:

           What    |Removed                     |Added
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 Attachment #315837|none                        |reviewed
             status|                            |

--- Comment #11 from Sebastian Dröge (slomo) <slomo at coaxion.net> ---
Review of attachment 315837:
 --> (https://bugzilla.gnome.org/review?bug=758179&attachment=315837)

Olivier, looks good to you?

Srimanta, is there a unit test already that tests all 3 variants? UDP-only,
TCP-only, UDP & TCP.

::: gst/rtsp-server/rtsp-stream.c
@@ +2218,3 @@
+
+    if (is_udp && is_tcp) {
+      g_object_set (priv->appsink[i], "async", FALSE, "sync", FALSE, NULL);

Why and why only on the appsink and not both? As we have queues after the tee
now on both branches, this should not be required anymore AFAIU and might even
cause problems.

@@ +2274,3 @@
+       * sink used for RTP data, not the RTCP data. */
+      if (i == 1)
+        g_object_set (priv->appsink[i], "async", FALSE, "sync", FALSE, NULL);

The comment is wrong I think? This i==1 is the RTCP sink, i==0 is the RTP sink.
For RTCP it makes sense to disable both (we should do that for RTCP! But also
for the udpsink)

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