Rename elements to avoid conflict with farsight elements with the same name. Fixes #430664.

Original commit message from CVS:
* docs/plugins/gst-plugins-bad-plugins-docs.sgml:
* docs/plugins/gst-plugins-bad-plugins-sections.txt:
* gst/rtpmanager/gstrtpbin.c: (create_session), (create_stream),
(gst_rtp_bin_class_init), (create_recv_rtp), (create_recv_rtcp),
(create_send_rtp), (create_rtcp), (gst_rtp_bin_request_new_pad):
* gst/rtpmanager/gstrtpclient.c: (create_stream),
(gst_rtp_client_request_new_pad):
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_clear_pt_map), (gst_rtp_jitter_buffer_loop):
* gst/rtpmanager/gstrtpmanager.c: (plugin_init):
* gst/rtpmanager/gstrtpptdemux.c:
* gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_class_init),
(gst_rtp_session_request_new_pad):
* gst/rtpmanager/gstrtpssrcdemux.c:
Rename elements to avoid conflict with farsight elements with the same
name. Fixes #430664.

7 files changed, 97 insertions, 93 deletions

diff --git a/gst/rtpmanager/gstrtpbin.c b/gst/rtpmanager/gstrtpbin.c
index 44fe3386d..cc0afd00b 100644
--- a/gst/rtpmanager/gstrtpbin.c
+++ b/gst/rtpmanager/gstrtpbin.c
@@ -18,34 +18,34 @@
  */
 
 /**
- * SECTION:element-rtpbin
+ * SECTION:element-gstrtpbin
  * @short_description: handle media from one RTP bin
- * @see_also: rtpjitterbuffer, rtpsession, rtpptdemux, rtpssrcdemux
+ * @see_also: gstrtpjitterbuffer, gstrtpsession, gstrtpptdemux, gstrtpssrcdemux
  *
  * <refsect2>
  * <para>
- * RTP bin combines the functions of rtpsession, rtpssrcdemux, rtpjitterbuffer
- * and rtpptdemux in one element. It allows for multiple rtpsessions that will
+ * RTP bin combines the functions of gstrtpsession, gstrtpssrcdemux, gstrtpjitterbuffer
+ * and gstrtpptdemux in one element. It allows for multiple RTP sessions that will
  * be synchronized together using RTCP SR packets.
  * </para>
  * <para>
- * rtpbin is configured with a number of request pads that define the
- * functionality that is activated, similar to the rtpsession element.
+ * gstrtpbin is configured with a number of request pads that define the
+ * functionality that is activated, similar to the gstrtpsession element.
  * </para>
  * <para>
- * To use rtpbin as an RTP receiver, request a recv_rtp_sink_%%d pad. The session
+ * To use gstrtpbin as an RTP receiver, request a recv_rtp_sink_%%d pad. The session
  * number must be specified in the pad name. 
- * Data received on the recv_rtp_sink_%%d pad will be processed in the rtpsession
- * manager and after being validated forwarded on rtpssrcdemuxer element. Each
- * RTP stream is demuxed based on the SSRC and send to a rtpjitterbuffer. After
- * the packets are released from the jitterbuffer, they will be forwarded to an
- * rtpptdemuxer element. The rtpptdemuxer element will demux the packets based
+ * Data received on the recv_rtp_sink_%%d pad will be processed in the gstrtpsession
+ * manager and after being validated forwarded on gstrtpssrcdemuxer element. Each
+ * RTP stream is demuxed based on the SSRC and send to a gstrtpjitterbuffer. After
+ * the packets are released from the jitterbuffer, they will be forwarded to a
+ * gstrtpptdemuxer element. The gstrtpptdemuxer element will demux the packets based
  * on the payload type and will create a unique pad recv_rtp_src_%%d_%%d_%%d on
- * rtpbin with the session number, SSRC and payload type respectively as the pad
+ * gstrtpbin with the session number, SSRC and payload type respectively as the pad
  * name.
  * </para>
  * <para>
- * To also use rtpbin as an RTCP receiver, request a recv_rtcp_sink_%%d pad. The
+ * To also use gstrtpbin as an RTCP receiver, request a recv_rtcp_sink_%%d pad. The
  * session number must be specified in the pad name.
  * </para>
  * <para>
@@ -55,7 +55,7 @@
  * in the session.
  * </para>
  * <para>
- * To use rtpbin as a sender, request a send_rtp_sink_%%d pad, which will
+ * To use gstrtpbin as a sender, request a send_rtp_sink_%%d pad, which will
  * automatically create a send_rtp_src_%%d pad. The session number must be specified when
  * requesting the sink pad. The session manager will modify the
  * SSRC in the RTP packets to its own SSRC and wil forward the packets on the
@@ -71,13 +71,13 @@
  * <para>
  * <programlisting>
  * gst-launch udpsrc port=5000 caps="application/x-rtp, ..." ! .recv_rtp_sink_0 \
- *     rtpbin ! rtptheoradepay ! theoradec ! xvimagesink
+ *     gstrtpbin ! rtptheoradepay ! theoradec ! xvimagesink
  * </programlisting>
- * Receive RTP data from port 5000 and send to the session 0 in rtpbin.
+ * Receive RTP data from port 5000 and send to the session 0 in gstrtpbin.
  * </para>
  * </refsect2>
  *
- * Last reviewed on 2007-05-23 (0.10.6)
+ * Last reviewed on 2007-05-28 (0.10.5)
  */
 
 #ifdef HAVE_CONFIG_H
@@ -265,10 +265,10 @@ create_session (GstRTPBin * rtpbin, gint id)
   GstRTPBinSession *sess;
   GstElement *session, *demux;
 
-  if (!(session = gst_element_factory_make ("rtpsession", NULL)))
+  if (!(session = gst_element_factory_make ("gstrtpsession", NULL)))
     goto no_session;
 
-  if (!(demux = gst_element_factory_make ("rtpssrcdemux", NULL)))
+  if (!(demux = gst_element_factory_make ("gstrtpssrcdemux", NULL)))
     goto no_demux;
 
   sess = g_new0 (GstRTPBinSession, 1);
@@ -294,13 +294,13 @@ create_session (GstRTPBin * rtpbin, gint id)
   /* ERRORS */
 no_session:
   {
-    g_warning ("rtpbin: could not create rtpsession element");
+    g_warning ("gstrtpbin: could not create gstrtpsession element");
     return NULL;
   }
 no_demux:
   {
     gst_object_unref (session);
-    g_warning ("rtpbin: could not create rtpssrcdemux element");
+    g_warning ("gstrtpbin: could not create gstrtpssrcdemux element");
     return NULL;
   }
 }
@@ -414,10 +414,10 @@ create_stream (GstRTPBinSession * session, guint32 ssrc)
   GstElement *buffer, *demux;
   GstRTPBinStream *stream;
 
-  if (!(buffer = gst_element_factory_make ("rtpjitterbuffer", NULL)))
+  if (!(buffer = gst_element_factory_make ("gstrtpjitterbuffer", NULL)))
     goto no_jitterbuffer;
 
-  if (!(demux = gst_element_factory_make ("rtpptdemux", NULL)))
+  if (!(demux = gst_element_factory_make ("gstrtpptdemux", NULL)))
     goto no_demux;
 
   stream = g_new0 (GstRTPBinStream, 1);
@@ -448,13 +448,13 @@ create_stream (GstRTPBinSession * session, guint32 ssrc)
   /* ERRORS */
 no_jitterbuffer:
   {
-    g_warning ("rtpbin: could not create rtpjitterbuffer element");
+    g_warning ("gstrtpbin: could not create gstrtpjitterbuffer element");
     return NULL;
   }
 no_demux:
   {
     gst_object_unref (buffer);
-    g_warning ("rtpbin: could not create rtpptdemux element");
+    g_warning ("gstrtpbin: could not create gstrtpptdemux element");
     return NULL;
   }
 }
@@ -834,7 +834,7 @@ create_recv_rtp (GstRTPBin * rtpbin, GstPadTemplate * templ, const gchar * name)
   /* ERRORS */
 no_name:
   {
-    g_warning ("rtpbin: invalid name given");
+    g_warning ("gstrtpbin: invalid name given");
     return NULL;
   }
 create_error:
@@ -844,17 +844,18 @@ create_error:
   }
 existed:
   {
-    g_warning ("rtpbin: recv_rtp pad already requested for session %d", sessid);
+    g_warning ("gstrtpbin: recv_rtp pad already requested for session %d",
+        sessid);
     return NULL;
   }
 pad_failed:
   {
-    g_warning ("rtpbin: failed to get session pad");
+    g_warning ("gstrtpbin: failed to get session pad");
     return NULL;
   }
 link_failed:
   {
-    g_warning ("rtpbin: failed to link pads");
+    g_warning ("gstrtpbin: failed to link pads");
     return NULL;
   }
 }
@@ -929,7 +930,7 @@ create_recv_rtcp (GstRTPBin * rtpbin, GstPadTemplate * templ,
   /* ERRORS */
 no_name:
   {
-    g_warning ("rtpbin: invalid name given");
+    g_warning ("gstrtpbin: invalid name given");
     return NULL;
   }
 create_error:
@@ -939,19 +940,19 @@ create_error:
   }
 existed:
   {
-    g_warning ("rtpbin: recv_rtcp pad already requested for session %d",
+    g_warning ("gstrtpbin: recv_rtcp pad already requested for session %d",
         sessid);
     return NULL;
   }
 pad_failed:
   {
-    g_warning ("rtpbin: failed to get session pad");
+    g_warning ("gstrtpbin: failed to get session pad");
     return NULL;
   }
 #if 0
 link_failed:
   {
-    g_warning ("rtpbin: failed to link pads");
+    g_warning ("gstrtpbin: failed to link pads");
     return NULL;
   }
 #endif
@@ -1018,7 +1019,7 @@ create_send_rtp (GstRTPBin * rtpbin, GstPadTemplate * templ, const gchar * name)
   /* ERRORS */
 no_name:
   {
-    g_warning ("rtpbin: invalid name given");
+    g_warning ("gstrtpbin: invalid name given");
     return NULL;
   }
 create_error:
@@ -1028,17 +1029,19 @@ create_error:
   }
 existed:
   {
-    g_warning ("rtpbin: send_rtp pad already requested for session %d", sessid);
+    g_warning ("gstrtpbin: send_rtp pad already requested for session %d",
+        sessid);
     return NULL;
   }
 pad_failed:
   {
-    g_warning ("rtpbin: failed to get session pad for session %d", sessid);
+    g_warning ("gstrtpbin: failed to get session pad for session %d", sessid);
     return NULL;
   }
 no_srcpad:
   {
-    g_warning ("rtpbin: failed to get rtp source pad for session %d", sessid);
+    g_warning ("gstrtpbin: failed to get rtp source pad for session %d",
+        sessid);
     return NULL;
   }
 }
@@ -1082,23 +1085,23 @@ create_rtcp (GstRTPBin * rtpbin, GstPadTemplate * templ, const gchar * name)
   /* ERRORS */
 no_name:
   {
-    g_warning ("rtpbin: invalid name given");
+    g_warning ("gstrtpbin: invalid name given");
     return NULL;
   }
 no_session:
   {
-    g_warning ("rtpbin: session with id %d does not exist", sessid);
+    g_warning ("gstrtpbin: session with id %d does not exist", sessid);
     return NULL;
   }
 existed:
   {
-    g_warning ("rtpbin: send_rtcp_src pad already requested for session %d",
+    g_warning ("gstrtpbin: send_rtcp_src pad already requested for session %d",
         sessid);
     return NULL;
   }
 pad_failed:
   {
-    g_warning ("rtpbin: failed to get rtcp pad for session %d", sessid);
+    g_warning ("gstrtpbin: failed to get rtcp pad for session %d", sessid);
     return NULL;
   }
 }
@@ -1144,7 +1147,7 @@ gst_rtp_bin_request_new_pad (GstElement * element,
 wrong_template:
   {
     GST_RTP_BIN_UNLOCK (rtpbin);
-    g_warning ("rtpbin: this is not our template");
+    g_warning ("gstrtpbin: this is not our template");
     return NULL;
   }
 }
diff --git a/gst/rtpmanager/gstrtpclient.c b/gst/rtpmanager/gstrtpclient.c
index 86c5f3ccb..ba18f9534 100644
--- a/gst/rtpmanager/gstrtpclient.c
+++ b/gst/rtpmanager/gstrtpclient.c
@@ -18,9 +18,9 @@
  */
 
 /**
- * SECTION:element-rtpclient
+ * SECTION:element-gstrtpclient
  * @short_description: handle media from one RTP client
- * @see_also: rtpjitterbuffer, rtpbin, rtpsession
+ * @see_also: gstrtpjitterbuffer, gstrtpbin, gstrtpsession
  *
  * <refsect2>
  * <para>
@@ -38,7 +38,7 @@
  * </para>
  * </refsect2>
  *
- * Last reviewed on 2007-04-02 (0.10.6)
+ * Last reviewed on 2007-04-02 (0.10.5)
  */
 
 #ifdef HAVE_CONFIG_H
@@ -136,11 +136,11 @@ create_stream (GstRTPClient * rtpclient, guint32 ssrc)
   stream->ssrc = ssrc;
   stream->client = rtpclient;
 
-  stream->jitterbuffer = gst_element_factory_make ("rtpjitterbuffer", NULL);
+  stream->jitterbuffer = gst_element_factory_make ("gstrtpjitterbuffer", NULL);
   if (!stream->jitterbuffer)
     goto no_jitterbuffer;
 
-  stream->ptdemux = gst_element_factory_make ("rtpptdemux", NULL);
+  stream->ptdemux = gst_element_factory_make ("gstrtpptdemux", NULL);
   if (!stream->ptdemux)
     goto no_ptdemux;
 
@@ -180,14 +180,14 @@ create_stream (GstRTPClient * rtpclient, guint32 ssrc)
 no_jitterbuffer:
   {
     g_free (stream);
-    g_warning ("could not create rtpjitterbuffer element");
+    g_warning ("gstrtpclient: could not create gstrtpjitterbuffer element");
     return NULL;
   }
 no_ptdemux:
   {
     gst_object_unref (stream->jitterbuffer);
     g_free (stream);
-    g_warning ("could not create rtpptdemux element");
+    g_warning ("gstrtpclient: could not create gstrtpptdemux element");
     return NULL;
   }
 could_not_link:
@@ -195,7 +195,7 @@ could_not_link:
     gst_bin_remove (GST_BIN_CAST (rtpclient), stream->jitterbuffer);
     gst_bin_remove (GST_BIN_CAST (rtpclient), stream->ptdemux);
     g_free (stream);
-    g_warning ("could not link jitterbuffer and rtpptdemux element");
+    g_warning ("gstrtpclient: could not link jitterbuffer and ptdemux element");
     return NULL;
   }
 }
@@ -455,27 +455,27 @@ gst_rtp_client_request_new_pad (GstElement * element,
   /* ERRORS */
 wrong_direction:
   {
-    g_warning ("rtpclient: request pad that is not a SINK pad");
+    g_warning ("gstrtpclient: request pad that is not a SINK pad");
     return NULL;
   }
 wrong_template:
   {
-    g_warning ("rtpclient: this is not our template");
+    g_warning ("gstrtpclient: this is not our template");
     return NULL;
   }
 no_name:
   {
-    g_warning ("rtpclient: no padname was specified");
+    g_warning ("gstrtpclient: no padname was specified");
     return NULL;
   }
 stream_exists:
   {
-    g_warning ("rtpclient: stream with SSRC %d already registered", ssrc);
+    g_warning ("gstrtpclient: stream with SSRC %d already registered", ssrc);
     return NULL;
   }
 stream_not_found:
   {
-    g_warning ("rtpclient: stream with SSRC %d not yet registered", ssrc);
+    g_warning ("gstrtpclient: stream with SSRC %d not yet registered", ssrc);
     return NULL;
   }
 }
diff --git a/gst/rtpmanager/gstrtpjitterbuffer.c b/gst/rtpmanager/gstrtpjitterbuffer.c
index 1838ba0b3..33f78741c 100644
--- a/gst/rtpmanager/gstrtpjitterbuffer.c
+++ b/gst/rtpmanager/gstrtpjitterbuffer.c
@@ -24,7 +24,7 @@
  */
 
 /**
- * SECTION:element-rtpjitterbuffer
+ * SECTION:element-gstrtpjitterbuffer
  * @short_description: buffer, reorder and remove duplicate RTP packets to
  * compensate for network oddities.
  *
@@ -33,7 +33,7 @@
  * This element reorders and removes duplicate RTP packets as they are received
  * from a network source. It will also wait for missing packets up to a
  * configurable time limit using the ::latency property. Packets arriving too
- * late are considered as lost packets.
+ * late are considered to be lost packets.
  * </para>
  * <para>
  * This element acts as a live element and so adds ::latency to the pipeline.
@@ -45,12 +45,12 @@
  * previous pt-map use the ::clear-pt-map signal.
  * </para>
  * <para>
- * This element will automatically be used inside rtpbin.
+ * This element will automatically be used inside gstrtpbin.
  * </para>
  * <title>Example pipelines</title>
  * <para>
  * <programlisting>
- * gst-launch rtspsrc location=rtsp://192.168.1.133:8554/mpeg1or2AudioVideoTest ! rtpjitterbuffer ! rtpmpvdepay ! mpeg2dec ! xvimagesink
+ * gst-launch rtspsrc location=rtsp://192.168.1.133:8554/mpeg1or2AudioVideoTest ! gstrtpjitterbuffer ! rtpmpvdepay ! mpeg2dec ! xvimagesink
  * </programlisting>
  * Connect to a streaming server and decode the MPEG video. The jitterbuffer is
  * inserted into the pipeline to smooth out network jitter and to reorder the
@@ -58,7 +58,7 @@
  * </para>
  * </refsect2>
  *
- * Last reviewed on 2007-05-22 (0.10.6)
+ * Last reviewed on 2007-05-28 (0.10.5)
  */
 
 #ifdef HAVE_CONFIG_H
diff --git a/gst/rtpmanager/gstrtpmanager.c b/gst/rtpmanager/gstrtpmanager.c
index 1490c4cbe..eed5c25f7 100644
--- a/gst/rtpmanager/gstrtpmanager.c
+++ b/gst/rtpmanager/gstrtpmanager.c
@@ -31,26 +31,27 @@
 static gboolean
 plugin_init (GstPlugin * plugin)
 {
-  if (!gst_element_register (plugin, "rtpbin", GST_RANK_NONE, GST_TYPE_RTP_BIN))
+  if (!gst_element_register (plugin, "gstrtpbin", GST_RANK_NONE,
+          GST_TYPE_RTP_BIN))
     return FALSE;
 
-  if (!gst_element_register (plugin, "rtpclient", GST_RANK_NONE,
+  if (!gst_element_register (plugin, "gstrtpclient", GST_RANK_NONE,
           GST_TYPE_RTP_CLIENT))
     return FALSE;
 
-  if (!gst_element_register (plugin, "rtpjitterbuffer", GST_RANK_NONE,
+  if (!gst_element_register (plugin, "gstrtpjitterbuffer", GST_RANK_NONE,
           GST_TYPE_RTP_JITTER_BUFFER))
     return FALSE;
 
-  if (!gst_element_register (plugin, "rtpptdemux", GST_RANK_NONE,
+  if (!gst_element_register (plugin, "gstrtpptdemux", GST_RANK_NONE,
           GST_TYPE_RTP_PT_DEMUX))
     return FALSE;
 
-  if (!gst_element_register (plugin, "rtpsession", GST_RANK_NONE,
+  if (!gst_element_register (plugin, "gstrtpsession", GST_RANK_NONE,
           GST_TYPE_RTP_SESSION))
     return FALSE;
 
-  if (!gst_element_register (plugin, "rtpssrcdemux", GST_RANK_NONE,
+  if (!gst_element_register (plugin, "gstrtpssrcdemux", GST_RANK_NONE,
           GST_TYPE_RTP_SSRC_DEMUX))
     return FALSE;
 
@@ -59,6 +60,6 @@ plugin_init (GstPlugin * plugin)
 
 GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
     GST_VERSION_MINOR,
-    "rtpmanager",
+    "gstrtpmanager",
     "RTP session management plugin library",
     plugin_init, VERSION, "LGPL", GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN)
diff --git a/gst/rtpmanager/gstrtpptdemux.c b/gst/rtpmanager/gstrtpptdemux.c
index 578f03532..235bbdd4a 100644
--- a/gst/rtpmanager/gstrtpptdemux.c
+++ b/gst/rtpmanager/gstrtpptdemux.c
@@ -24,12 +24,12 @@
  */
 
 /**
- * SECTION:element-rtpptdemux
+ * SECTION:element-gstrtpptdemux
  * @short_description: separate RTP payloads based on the payload type
  *
  * <refsect2>
  * <para>
- * rtpptdemux acts as a demuxer for RTP packets based on the payload type of the
+ * gstrtpptdemux acts as a demuxer for RTP packets based on the payload type of the
  * packets. Its main purpose is to allow an application to easily receive and
  * decode an RTP stream with multiple payload types.
  * </para>
@@ -45,14 +45,14 @@
  * <title>Example pipelines</title>
  * <para>
  * <programlisting>
- * gst-launch udpsrc caps="application/x-rtp" ! rtpptdemux ! fakesink
+ * gst-launch udpsrc caps="application/x-rtp" ! gstrtpptdemux ! fakesink
  * </programlisting>
  * Takes an RTP stream and send the RTP packets with the first detected payload
  * type to fakesink, discarding the other payload types.
  * </para>
  * </refsect2>
  *
- * Last reviewed on 2007-05-22 (0.10.6)
+ * Last reviewed on 2007-05-28 (0.10.5)
  */
 
 /*
diff --git a/gst/rtpmanager/gstrtpsession.c b/gst/rtpmanager/gstrtpsession.c
index 431098d92..3e33cf6af 100644
--- a/gst/rtpmanager/gstrtpsession.c
+++ b/gst/rtpmanager/gstrtpsession.c
@@ -18,9 +18,9 @@
  */
 
 /**
- * SECTION:element-rtpsession
+ * SECTION:element-gstrtpsession
  * @short_description: an RTP session manager
- * @see_also: rtpjitterbuffer, rtpbin, rtpptdemux, rtpssrcdemux
+ * @see_also: gstrtpjitterbuffer, gstrtpbin, gstrtpptdemux, gstrtpssrcdemux
  *
  * <refsect2>
  * <para>
@@ -47,20 +47,20 @@
  * </itemizedlist>
  * </para>
  * <para>
- * The rtpsession will not demux packets based on SSRC or payload type, nor will
- * it correct for packet reordering and jitter. Use rtpssrcdemux, rtpptdemux and
- * rtpjitterbuffer in addition to rtpsession to perform these tasks. It is
- * usually a good idea to use rtpbin, which combines all these features in one
+ * The gstrtpsession will not demux packets based on SSRC or payload type, nor will
+ * it correct for packet reordering and jitter. Use gstrtpssrcdemux, gstrtpptdemux and
+ * gstrtpjitterbuffer in addition to gstrtpsession to perform these tasks. It is
+ * usually a good idea to use gstrtpbin, which combines all these features in one
  * element.
  * </para>
  * <para>
- * To use rtpsession as an RTP receiver, request a recv_rtp_sink pad, which will
+ * To use gstrtpsession as an RTP receiver, request a recv_rtp_sink pad, which will
  * automatically create recv_rtp_src pad. Data received on the recv_rtp_sink pad
  * will be processed in the session and after being validated forwarded on the
  * recv_rtp_src pad.
  * </para>
  * <para>
- * To also use rtpsession as an RTCP receiver, request a recv_rtcp_sink pad,
+ * To also use gstrtpsession as an RTCP receiver, request a recv_rtcp_sink pad,
  * which will automatically create a sync_src pad. Packets received on the RTCP
  * pad will be used by the session manager to update the stats and database of
  * the other participants. SR packets will be forwarded on the sync_src pad
@@ -72,7 +72,7 @@
  * that should be sent to all participants in the session.
  * </para>
  * <para>
- * To use rtpsession as a sender, request a send_rtp_sink pad, which will
+ * To use gstrtpsession as a sender, request a send_rtp_sink pad, which will
  * automatically create a send_rtp_src pad. The session manager will modify the
  * SSRC in the RTP packets to its own SSRC and wil forward the packets on the
  * send_rtp_src pad after updating its internal state.
@@ -86,7 +86,7 @@
  * <title>Example pipelines</title>
  * <para>
  * <programlisting>
- * gst-launch udpsrc port=5000 caps="application/x-rtp, ..." ! .recv_rtp_sink rtpsession .recv_rtp_src ! rtptheoradepay ! theoradec ! xvimagesink
+ * gst-launch udpsrc port=5000 caps="application/x-rtp, ..." ! .recv_rtp_sink gstrtpsession .recv_rtp_src ! rtptheoradepay ! theoradec ! xvimagesink
  * </programlisting>
  * Receive theora RTP packets from port 5000 and send them to the depayloader,
  * decoder and display. Note that the application/x-rtp caps on udpsrc should be
@@ -95,7 +95,7 @@
  * </para>
  * <para>
  * <programlisting>
- * gst-launch udpsrc port=5000 caps="application/x-rtp, ..." ! .recv_rtp_sink rtpsession name=session \
+ * gst-launch udpsrc port=5000 caps="application/x-rtp, ..." ! .recv_rtp_sink gstrtpsession name=session \
  *        .recv_rtp_src ! rtptheoradepay ! theoradec ! xvimagesink \
  *     udpsrc port=5001 caps="application/x-rtcp" ! session.recv_rtcp_sink
  * </programlisting>
@@ -108,24 +108,24 @@
  * </para>
  * <para>
  * <programlisting>
- * gst-launch videotestsrc ! theoraenc ! rtptheorapay ! .send_rtp_sink rtpsession .send_rtp_src ! udpsink port=5000
+ * gst-launch videotestsrc ! theoraenc ! rtptheorapay ! .send_rtp_sink gstrtpsession .send_rtp_src ! udpsink port=5000
  * </programlisting>
  * Send theora RTP packets through the session manager and out on UDP port 5000.
  * </para>
  * <para>
  * <programlisting>
- * gst-launch videotestsrc ! theoraenc ! rtptheorapay ! .send_rtp_sink rtpsession name=session .send_rtp_src \
+ * gst-launch videotestsrc ! theoraenc ! rtptheorapay ! .send_rtp_sink gstrtpsession name=session .send_rtp_src \
  *     ! udpsink port=5000  session.send_rtcp_src ! udpsink port=5001
  * </programlisting>
  * Send theora RTP packets through the session manager and out on UDP port 5000.
- * Send RTCP packets on port 5001. Not that this pipeline will not preroll
+ * Send RTCP packets on port 5001. Note that this pipeline will not preroll
  * correctly because the second udpsink will not preroll correctly (no RTCP
  * packets are sent in the PAUSED state). Applications should manually set and
  * keep (see #gst_element_set_locked_state()) the RTCP udpsink to the PLAYING state.
  * </para>
  * </refsect2>
  *
- * Last reviewed on 2007-05-23 (0.10.6)
+ * Last reviewed on 2007-05-28 (0.10.5)
  */
 
 #ifdef HAVE_CONFIG_H
@@ -1001,13 +1001,13 @@ gst_rtp_session_request_new_pad (GstElement * element,
 wrong_template:
   {
     GST_RTP_SESSION_UNLOCK (rtpsession);
-    g_warning ("rtpsession: this is not our template");
+    g_warning ("gstrtpsession: this is not our template");
     return NULL;
   }
 exists:
   {
     GST_RTP_SESSION_UNLOCK (rtpsession);
-    g_warning ("rtpsession: pad already requested");
+    g_warning ("gstrtpsession: pad already requested");
     return NULL;
   }
 }
diff --git a/gst/rtpmanager/gstrtpssrcdemux.c b/gst/rtpmanager/gstrtpssrcdemux.c
index bd0c6b05c..e1100be4f 100644
--- a/gst/rtpmanager/gstrtpssrcdemux.c
+++ b/gst/rtpmanager/gstrtpssrcdemux.c
@@ -20,12 +20,12 @@
  */
 
 /**
- * SECTION:element-rtpssrcdemux
+ * SECTION:element-gstrtpssrcdemux
  * @short_description: separate RTP payloads based on the SSRC
  *
  * <refsect2>
  * <para>
- * rtpssrcdemux acts as a demuxer for RTP packets based on the SSRC of the
+ * gstrtpssrcdemux acts as a demuxer for RTP packets based on the SSRC of the
  * packets. Its main purpose is to allow an application to easily receive and
  * decode an RTP stream with multiple SSRCs.
  * </para>
@@ -36,14 +36,14 @@
  * <title>Example pipelines</title>
  * <para>
  * <programlisting>
- * gst-launch udpsrc caps="application/x-rtp" ! rtpssrcdemux ! fakesink
+ * gst-launch udpsrc caps="application/x-rtp" ! gstrtpssrcdemux ! fakesink
  * </programlisting>
  * Takes an RTP stream and send the RTP packets with the first detected SSRC
  * to fakesink, discarding the other SSRCs.
  * </para>
  * </refsect2>
  *
- * Last reviewed on 2007-05-23 (0.10.6)
+ * Last reviewed on 2007-05-28 (0.10.5)
  */
 
 #ifdef HAVE_CONFIG_H