rtspsrc: Fix handling of interleaved (TCP) streams

We need to set up the transport in any case, not just if we have a container
stream or a non-interleaved stream. Only if we have an interleaved stream and
are retrying, we should not set up the stream again.

https://bugzilla.gnome.org/show_bug.cgi?id=745599

diff --git a/gst/rtsp/gstrtspsrc.c b/gst/rtsp/gstrtspsrc.c
index 552bfde3cd864ff234e4d74549c434d63238ff27..3d9852c8551a7b8cc692b31ebaa76fd2a734ab8f 100644 (file)
--- a/gst/rtsp/gstrtspsrc.c
+++ b/gst/rtsp/gstrtspsrc.c
@@ -6523,7 +6523,7 @@ gst_rtspsrc_setup_streams (GstRTSPSrc * src, gboolean async)
           break;
       }
 
-      if (stream->container || (!src->interleaved && !retry)) {
+      if (!src->interleaved || !retry) {
         /* now configure the stream with the selected transport */
         if (!gst_rtspsrc_stream_configure_transport (stream, &transport)) {
           GST_DEBUG_OBJECT (src,