docs: update docs with 1.0 element names

gst/rtpmanager/gstrtpbin.c

@@ -633,13 +633,13 @@ create_session (GstRtpBin * rtpbin, gint id)
   /* ERRORS */
 no_session:
   {
-    g_warning ("rtpbin: could not create gstrtpsession element");
+    g_warning ("rtpbin: could not create rtpsession element");
     return NULL;
   }
 no_demux:
   {
     gst_object_unref (session);
-    g_warning ("rtpbin: could not create gstrtpssrcdemux element");
+    g_warning ("rtpbin: could not create rtpssrcdemux element");
     return NULL;
   }
 }
@@ -1465,13 +1465,13 @@ create_stream (GstRtpBinSession * session, guint32 ssrc)
   /* ERRORS */
 no_jitterbuffer:
   {
-    g_warning ("rtpbin: could not create gstrtpjitterbuffer element");
+    g_warning ("rtpbin: could not create rtpjitterbuffer element");
     return NULL;
   }
 no_demux:
   {
     gst_object_unref (buffer);
-    g_warning ("rtpbin: could not create gstrtpptdemux element");
+    g_warning ("rtpbin: could not create rtpptdemux element");
     return NULL;
   }
 }
@@ -2598,7 +2598,7 @@ new_ssrc_pad_found (GstElement * element, guint ssrc, GstPad * pad,
     stream->demux_ptchange_sig = g_signal_connect (stream->demux,
         "payload-type-change", (GCallback) payload_type_change, session);
   } else {
-    /* add gstrtpjitterbuffer src pad to pads */
+    /* add rtpjitterbuffer src pad to pads */
     GstElementClass *klass;
     GstPadTemplate *templ;
     gchar *padname;

gst/rtpmanager/gstrtpjitterbuffer.c

@@ -24,7 +24,7 @@
  */
 
 /**
- * SECTION:element-gstrtpjitterbuffer
+ * SECTION:element-rtpjitterbuffer
  *
  * 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
@@ -39,12 +39,12 @@
  * when no caps are present, from the #GstRtpJitterBuffer::request-pt-map signal.
  * To clear the previous pt-map use the #GstRtpJitterBuffer::clear-pt-map signal.
  *
- * This element will automatically be used inside gstrtpbin.
+ * This element will automatically be used inside rtpbin.
  *
  * <refsect2>
  * <title>Example pipelines</title>
  * |[
- * gst-launch-1.0 rtspsrc location=rtsp://192.168.1.133:8554/mpeg1or2AudioVideoTest ! gstrtpjitterbuffer ! rtpmpvdepay ! mpeg2dec ! xvimagesink
+ * gst-launch-1.0 rtspsrc location=rtsp://192.168.1.133:8554/mpeg1or2AudioVideoTest ! rtpjitterbuffer ! rtpmpvdepay ! mpeg2dec ! xvimagesink
  * ]| 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
  * out-of-order RTP packets.
@@ -593,7 +593,7 @@ gst_rtp_jitter_buffer_class_init (GstRtpJitterBufferClass * klass)
   klass->set_active = GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_set_active);
 
   GST_DEBUG_CATEGORY_INIT
-      (rtpjitterbuffer_debug, "gstrtpjitterbuffer", 0, "RTP Jitter Buffer");
+      (rtpjitterbuffer_debug, "rtpjitterbuffer", 0, "RTP Jitter Buffer");
 }
 
 static void
@@ -801,12 +801,12 @@ gst_rtp_jitter_buffer_request_new_pad (GstElement * element,
   /* ERRORS */
 wrong_template:
   {
-    g_warning ("gstrtpjitterbuffer: this is not our template");
+    g_warning ("rtpjitterbuffer: this is not our template");
     return NULL;
   }
 exists:
   {
-    g_warning ("gstrtpjitterbuffer: pad already requested");
+    g_warning ("rtpjitterbuffer: pad already requested");
     return NULL;
   }
 }

gst/rtpmanager/gstrtpptdemux.c

@@ -24,9 +24,9 @@
  */
 
 /**
- * SECTION:element-gstrtpptdemux
+ * SECTION:element-rtpptdemux
  *
- * gstrtpptdemux acts as a demuxer for RTP packets based on the payload type of
+ * rtpptdemux 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.
  * 
@@ -42,7 +42,7 @@
  * <refsect2>
  * <title>Example pipelines</title>
  * |[
- * gst-launch-1.0 udpsrc caps="application/x-rtp" ! gstrtpptdemux ! fakesink
+ * gst-launch-1.0 udpsrc caps="application/x-rtp" ! rtpptdemux ! fakesink
  * ]| Takes an RTP stream and send the RTP packets with the first detected
  * payload type to fakesink, discarding the other payload types.
  * </refsect2>

gst/rtpmanager/gstrtpsession.c

@@ -18,8 +18,8 @@
  */
 
 /**
- * SECTION:element-gstrtpsession
- * @see_also: gstrtpjitterbuffer, gstrtpbin, gstrtpptdemux, gstrtpssrcdemux
+ * SECTION:element-rtpsession
+ * @see_also: rtpjitterbuffer, rtpbin, rtpptdemux, rtpssrcdemux
  *
  * The RTP session manager models one participant with a unique SSRC in an RTP
  * session. This session can be used to send and receive RTP and RTCP packets.
@@ -42,7 +42,7 @@
  *   </listitem>
  * </itemizedlist>
  *
- * The gstrtpsession will not demux packets based on SSRC or payload type, nor will
+ * The rtpsession 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
@@ -76,13 +76,13 @@
  * <refsect2>
  * <title>Example pipelines</title>
  * |[
- * gst-launch-1.0 udpsrc port=5000 caps="application/x-rtp, ..." ! .recv_rtp_sink gstrtpsession .recv_rtp_src ! rtptheoradepay ! theoradec ! xvimagesink
+ * gst-launch-1.0 udpsrc port=5000 caps="application/x-rtp, ..." ! .recv_rtp_sink rtpsession .recv_rtp_src ! rtptheoradepay ! theoradec ! xvimagesink
  * ]| 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
  * configured based on some negotiation process such as RTSP for this pipeline
  * to work correctly.
  * |[
- * gst-launch-1.0 udpsrc port=5000 caps="application/x-rtp, ..." ! .recv_rtp_sink gstrtpsession name=session \
+ * gst-launch-1.0 udpsrc port=5000 caps="application/x-rtp, ..." ! .recv_rtp_sink rtpsession name=session \
  *        .recv_rtp_src ! rtptheoradepay ! theoradec ! xvimagesink \
  *     udpsrc port=5001 caps="application/x-rtcp" ! session.recv_rtcp_sink
  * ]| Receive theora RTP packets from port 5000 and send them to the depayloader,
@@ -92,11 +92,11 @@
  * configured based on some negotiation process such as RTSP for this pipeline
  * to work correctly.
  * |[
- * gst-launch-1.0 videotestsrc ! theoraenc ! rtptheorapay ! .send_rtp_sink gstrtpsession .send_rtp_src ! udpsink port=5000
+ * gst-launch-1.0 videotestsrc ! theoraenc ! rtptheorapay ! .send_rtp_sink rtpsession .send_rtp_src ! udpsink port=5000
  * ]| Send theora RTP packets through the session manager and out on UDP port
  * 5000.
  * |[
- * gst-launch-1.0 videotestsrc ! theoraenc ! rtptheorapay ! .send_rtp_sink gstrtpsession name=session .send_rtp_src \
+ * gst-launch-1.0 videotestsrc ! theoraenc ! rtptheorapay ! .send_rtp_sink rtpsession name=session .send_rtp_src \
  *     ! udpsink port=5000  session.send_rtcp_src ! udpsink port=5001
  * ]| Send theora RTP packets through the session manager and out on UDP port
  * 5000. Send RTCP packets on port 5001. Note that this pipeline will not preroll
@@ -2269,13 +2269,13 @@ gst_rtp_session_request_new_pad (GstElement * element,
 wrong_template:
   {
     GST_RTP_SESSION_UNLOCK (rtpsession);
-    g_warning ("gstrtpsession: this is not our template");
+    g_warning ("rtpsession: this is not our template");
     return NULL;
   }
 exists:
   {
     GST_RTP_SESSION_UNLOCK (rtpsession);
-    g_warning ("gstrtpsession: pad already requested");
+    g_warning ("rtpsession: pad already requested");
     return NULL;
   }
 }
@@ -2313,7 +2313,7 @@ gst_rtp_session_release_pad (GstElement * element, GstPad * pad)
 wrong_pad:
   {
     GST_RTP_SESSION_UNLOCK (rtpsession);
-    g_warning ("gstrtpsession: asked to release an unknown pad");
+    g_warning ("rtpsession: asked to release an unknown pad");
     return;
   }
 }

gst/rtpmanager/gstrtpssrcdemux.c

@@ -20,9 +20,9 @@
  */
 
 /**
- * SECTION:element-gstrtpssrcdemux
+ * SECTION:element-rtpssrcdemux
  *
- * gstrtpssrcdemux acts as a demuxer for RTP packets based on the SSRC of the
+ * rtpssrcdemux 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.
  * 
@@ -32,7 +32,7 @@
  * <refsect2>
  * <title>Example pipelines</title>
  * |[
- * gst-launch-1.0 udpsrc caps="application/x-rtp" ! gstrtpssrcdemux ! fakesink
+ * gst-launch-1.0 udpsrc caps="application/x-rtp" ! rtpssrcdemux ! fakesink
  * ]| Takes an RTP stream and send the RTP packets with the first detected SSRC
  * to fakesink, discarding the other SSRCs.
  * </refsect2>