More cleanups

Don't keep a reference to the GstRTSPMedia in the stream.
Free more things when freeing the GstRTSPMedia.

gst/rtsp-server/rtsp-client.c

@@ -435,19 +435,16 @@ handle_setup_request (GstRTSPClient *client, GstRTSPUrl *uri, GstRTSPMessage *re
   }
   g_strfreev (transports);
 
-  g_free (ct->destination);
-  ct->destination = g_strdup (inet_ntoa (client->address.sin_addr));
-
   /* we have not found anything usable, error out */
-  if (!have_transport) {
+  if (!have_transport) 
     goto unsupported_transports;
-  }
 
   /* we have a valid transport, check if we can handle it */
   if (ct->trans != GST_RTSP_TRANS_RTP)
     goto unsupported_transports;
   if (ct->profile != GST_RTSP_PROFILE_AVP)
     goto unsupported_transports;
+
   supported = GST_RTSP_LOWER_TRANS_UDP |
 	GST_RTSP_LOWER_TRANS_UDP_MCAST | GST_RTSP_LOWER_TRANS_TCP;
   if (!(ct->lower_transport & supported))
@@ -456,6 +453,10 @@ handle_setup_request (GstRTSPClient *client, GstRTSPUrl *uri, GstRTSPMessage *re
   if (client->session_pool == NULL)
     goto no_pool;
 
+  /* we have a valid transport now, set the destination of the client. */
+  g_free (ct->destination);
+  ct->destination = g_strdup (inet_ntoa (client->address.sin_addr));
+
   /* a setup request creates a session for a client, check if the client already
    * sent a session id to us */
   res = gst_rtsp_message_get_header (request, GST_RTSP_HDR_SESSION, &sessid, 0);
@@ -478,6 +479,7 @@ handle_setup_request (GstRTSPClient *client, GstRTSPUrl *uri, GstRTSPMessage *re
 
     /* get a handle to the configuration of the media in the session */
     if ((m = find_media (client, uri, request))) {
+      /* manage the media in our session now */
       media = gst_rtsp_session_manage_media (session, uri, m);
     }
   }

gst/rtsp-server/rtsp-media-factory.c

@@ -416,7 +416,6 @@ default_construct (GstRTSPMediaFactory *factory, const GstRTSPUrl *url)
     
     /* create the stream */
     stream = g_new0 (GstRTSPMediaStream, 1);
-    stream->media = media;
     stream->payloader = pay;
 
     pad = gst_element_get_static_pad (pay, "src");

gst/rtsp-server/rtsp-media.c

@@ -56,11 +56,16 @@ static void
 gst_rtsp_media_init (GstRTSPMedia * media)
 {
   media->streams = g_array_new (FALSE, TRUE, sizeof (GstRTSPMediaStream *));
+  media->complete = FALSE;
 }
 
 static void
 gst_rtsp_media_stream_free (GstRTSPMediaStream *stream)
 {
+  if (stream->caps)
+    gst_caps_unref (stream->caps);
+
+  g_free (stream);
 }
 
 static void
@@ -80,6 +85,8 @@ gst_rtsp_media_finalize (GObject * obj)
   }
   g_array_free (media->streams, TRUE);
 
+  gst_object_unref (media->pipeline);
+
   G_OBJECT_CLASS (gst_rtsp_media_parent_class)->finalize (obj);
 }
 
@@ -188,7 +195,8 @@ gst_rtsp_media_n_streams (GstRTSPMedia *media)
  *
  * Retrieve the stream with index @idx from @media.
  *
- * Returns: the #GstRTSPMediaStream at index @idx.
+ * Returns: the #GstRTSPMediaStream at index @idx or %NULL when a stream with
+ * that index did not exist.
  */
 GstRTSPMediaStream *
 gst_rtsp_media_get_stream (GstRTSPMedia *media, guint idx)
@@ -196,9 +204,11 @@ gst_rtsp_media_get_stream (GstRTSPMedia *media, guint idx)
   GstRTSPMediaStream *res;
   
   g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
-  g_return_val_if_fail (idx < media->streams->len, NULL);
 
-  res = g_array_index (media->streams, GstRTSPMediaStream *, idx);
+  if (idx < media->streams->len)
+    res = g_array_index (media->streams, GstRTSPMediaStream *, idx);
+  else
+    res = NULL;
 
   return res;
 }
@@ -315,19 +325,13 @@ again:
 
   /* we keep these elements, we configure all in configure_transport when the
    * server told us to really use the UDP ports. */
-  stream->udpsrc[0] = gst_object_ref (udpsrc0);
-  stream->udpsrc[1] = gst_object_ref (udpsrc1);
-  stream->udpsink[0] = gst_object_ref (udpsink0);
-  stream->udpsink[1] = gst_object_ref (udpsink1);
+  stream->udpsrc[0] = udpsrc0;
+  stream->udpsrc[1] = udpsrc1;
+  stream->udpsink[0] = udpsink0;
+  stream->udpsink[1] = udpsink1;
   stream->server_port.min = rtpport;
   stream->server_port.max = rtcpport;
 
-  /* they are ours now */
-  gst_object_sink (udpsrc0);
-  gst_object_sink (udpsrc1);
-  gst_object_sink (udpsink0);
-  gst_object_sink (udpsink1);
-
   return TRUE;
 
   /* ERRORS */
@@ -504,6 +508,7 @@ gst_rtsp_media_prepare (GstRTSPMedia *media)
   /* and back to PAUSED for live pipelines */
   ret = gst_element_set_state (media->pipeline, GST_STATE_PAUSED);
 
+  /* unlock the udp src elements */
   n_streams = gst_rtsp_media_n_streams (media);
   for (i = 0; i < n_streams; i++) {
     GstRTSPMediaStream *stream;

gst/rtsp-server/rtsp-media.h

@@ -59,14 +59,10 @@ typedef struct _GstRTSPMediaClass GstRTSPMediaClass;
  * The definition of a media stream. The streams are identified by @id.
  */
 struct _GstRTSPMediaStream {
-  GstRTSPMedia *media;
-
   GstPad       *srcpad;
   GstElement   *payloader;
   gboolean      prepared;
 
-  GstRTSPRange  server_port;
-
   /* pads on the rtpbin */
   GstPad       *recv_rtcp_sink;
   GstPad       *send_rtp_sink;
@@ -78,6 +74,9 @@ struct _GstRTSPMediaStream {
   GstElement   *udpsrc[2];
   GstElement   *udpsink[2];
 
+  /* server ports for sending/receiving */
+  GstRTSPRange  server_port;
+
   /* the caps of the stream */
   gulong        caps_sig;
   GstCaps      *caps;
@@ -102,6 +101,7 @@ struct _GstRTSPMedia {
   GObject       parent;
 
   gboolean      shared;
+  gboolean      complete;
 
   GstElement   *element;
   GArray       *streams;