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.

1 files changed, 17 insertions, 17 deletions

diff --git a/gst/rtpmanager/gstrtpsession.c b/gst/rtpmanager/gstrtpsession.c
index 431098d9..3e33cf6a 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;
   }
 }