gst/rtsp/: RTSP cleanups.

Original commit message from CVS:
* gst/rtsp/README:
* gst/rtsp/gstrtspsrc.c: (gst_rtsp_proto_get_type),
(gst_rtspsrc_class_init), (gst_rtspsrc_create_stream),
(gst_rtspsrc_add_element), (gst_rtspsrc_set_state),
(gst_rtspsrc_stream_setup_rtp),
(gst_rtspsrc_stream_configure_transport), (find_stream),
(gst_rtspsrc_loop), (gst_rtspsrc_open), (gst_rtspsrc_play),
(gst_rtspsrc_change_state):
* gst/rtsp/rtsptransport.c: (rtsp_transport_new),
(rtsp_transport_init), (parse_mode), (parse_range),
(rtsp_transport_parse), (rtsp_transport_free):
RTSP cleanups.

3 files changed, 18 insertions, 15 deletions

diff --git a/gst/rtsp/README b/gst/rtsp/README
index 285f4f43..2f7fde89 100644
--- a/gst/rtsp/README
+++ b/gst/rtsp/README
@@ -132,7 +132,7 @@ An RTSP session is created as follows:
   the data will be received on the same connection as the RTSP connection.
 
   At this point, we remove the UDP source elements as we don't need them anymore. We
-  setup the rtpdec session manager element though as follows:
+  set up the rtpsess session manager element though as follows:
 
     +---------------------------------------------+
     | +------------+                              |
diff --git a/gst/rtsp/gstrtspsrc.c b/gst/rtsp/gstrtspsrc.c
index ab2b36eb..5938608d 100644
--- a/gst/rtsp/gstrtspsrc.c
+++ b/gst/rtsp/gstrtspsrc.c
@@ -256,6 +256,8 @@ gst_rtspsrc_set_state (GstRTSPSrc * src, GstElementState state)
   GstElementStateReturn ret;
   GList *streams;
 
+  ret = GST_STATE_SUCCESS;
+
   /* for all streams */
   for (streams = src->streams; streams; streams = g_list_next (streams)) {
     GstRTSPStream *stream;
@@ -462,13 +464,14 @@ gst_rtspsrc_loop (GstRTSPSrc * src)
     outpad = stream->rtpdecrtcp;
   }
 
-  if ((chainfunc = GST_RPAD_CHAINFUNC (outpad))) {
+  /* and chain buffer to internal element */
+  {
     GstBuffer *buf;
 
     buf = gst_buffer_new_and_alloc (size);
     memcpy (GST_BUFFER_DATA (buf), data, size);
 
-    if (chainfunc (outpad, buf) != GST_FLOW_OK)
+    if (gst_pad_chain (outpad, buf) != GST_FLOW_OK)
       goto need_pause;
   }
 
@@ -689,9 +692,9 @@ gst_rtspsrc_open (GstRTSPSrc * src)
       }
     }
   }
-
   return TRUE;
 
+  /* ERRORS */
 invalid_url:
   {
     GST_ELEMENT_ERROR (src, RESOURCE, NOT_FOUND,
@@ -858,7 +861,8 @@ gst_rtspsrc_change_state (GstElement * element)
       break;
     case GST_STATE_READY_TO_PAUSED:
       rtspsrc->interleaved = FALSE;
-      gst_rtspsrc_open (rtspsrc);
+      if (!gst_rtspsrc_open (rtspsrc))
+        goto open_failed;
       /* need to play now for the preroll, might delay that to
        * next state when we have NO_PREROLL as a return value */
       gst_rtspsrc_play (rtspsrc);
@@ -872,7 +876,7 @@ gst_rtspsrc_change_state (GstElement * element)
 
   ret = gst_rtspsrc_set_state (rtspsrc, GST_STATE_PENDING (rtspsrc));
   if (ret != GST_STATE_SUCCESS)
-    goto error;
+    goto done;
   ret = GST_ELEMENT_CLASS (parent_class)->change_state (element);
 
   switch (transition) {
@@ -887,6 +891,11 @@ gst_rtspsrc_change_state (GstElement * element)
       break;
   }
 
-error:
+done:
   return ret;
+
+open_failed:
+  {
+    return GST_STATE_FAILURE;
+  }
 }
diff --git a/gst/rtsp/rtsptransport.c b/gst/rtsp/rtsptransport.c
index addf354f..592778bf 100644
--- a/gst/rtsp/rtsptransport.c
+++ b/gst/rtsp/rtsptransport.c
@@ -55,14 +55,8 @@ rtsp_transport_init (RTSPTransport * transport)
 static void
 parse_mode (RTSPTransport * transport, gchar * str)
 {
-  if (strstr (str, "\"PLAY\""))
-    transport->mode_play = TRUE;
-  else
-    transport->mode_play = FALSE;
-  if (strstr (str, "\"RECORD\""))
-    transport->mode_record = TRUE;
-  else
-    transport->mode_record = FALSE;
+  transport->mode_play = (strstr (str, "\"PLAY\"") != NULL);
+  transport->mode_record = (strstr (str, "\"RECORD\"") != NULL);
 }
 
 static void