rtpmanager: rename gstrtp* -> rtp*

This was done in 0.10 to avoid conflict with the rtp elements in
farsight, but the gst-prefixing is no longer needed in 0.11

diff --git a/docs/plugins/gst-plugins-good-plugins-docs.sgml b/docs/plugins/gst-plugins-good-plugins-docs.sgml
index 7d7fc9a..9a7e854 100644 (file)
--- a/docs/plugins/gst-plugins-good-plugins-docs.sgml
+++ b/docs/plugins/gst-plugins-good-plugins-docs.sgml
@@ -83,11 +83,6 @@
     <xi:include href="xml/element-gdkpixbufsink.xml" />
     <xi:include href="xml/element-goom.xml" />
     <xi:include href="xml/element-goom2k1.xml" />
-    <xi:include href="xml/element-gstrtpbin.xml" />
-    <xi:include href="xml/element-gstrtpjitterbuffer.xml" />
-    <xi:include href="xml/element-gstrtpptdemux.xml" />
-    <xi:include href="xml/element-gstrtpsession.xml" />
-    <xi:include href="xml/element-gstrtpssrcdemux.xml" />
     <xi:include href="xml/element-halaudiosink.xml" />
     <xi:include href="xml/element-halaudiosrc.xml" />
     <xi:include href="xml/element-hdv1394src.xml" />
@@ -145,6 +140,11 @@
     <xi:include href="xml/element-rtpj2kpay.xml" />
     <xi:include href="xml/element-rtpjpegpay.xml" />
     <xi:include href="xml/element-rtspsrc.xml" />
+    <xi:include href="xml/element-rtpbin.xml" />
+    <xi:include href="xml/element-rtpjitterbuffer.xml" />
+    <xi:include href="xml/element-rtpptdemux.xml" />
+    <xi:include href="xml/element-rtpsession.xml" />
+    <xi:include href="xml/element-rtpssrcdemux.xml" />
     <xi:include href="xml/element-shagadelictv.xml" />
     <xi:include href="xml/element-shapewipe.xml" />
     <xi:include href="xml/element-smokedec.xml" />
@@ -233,7 +233,7 @@
     <xi:include href="xml/plugin-pulseaudio.xml" />
     <xi:include href="xml/plugin-replaygain.xml" />
     <xi:include href="xml/plugin-rtp.xml" />
-    <xi:include href="xml/plugin-gstrtpmanager.xml" />
+    <xi:include href="xml/plugin-rtpmanager.xml" />
     <xi:include href="xml/plugin-rtsp.xml" />
     <xi:include href="xml/plugin-shapewipe.xml" />
     <xi:include href="xml/plugin-shout2send.xml" />

diff --git a/docs/plugins/gst-plugins-good-plugins-sections.txt b/docs/plugins/gst-plugins-good-plugins-sections.txt
index 0b21cb5..c92a571 100644 (file)
--- a/docs/plugins/gst-plugins-good-plugins-sections.txt
+++ b/docs/plugins/gst-plugins-good-plugins-sections.txt
@@ -957,8 +957,8 @@ GST_IS_GOOM_CLASS
 </SECTION>
 
 <SECTION>
-<FILE>element-gstrtpbin</FILE>
-<TITLE>gstrtpbin</TITLE>
+<FILE>element-rtpbin</FILE>
+<TITLE>rtpbin</TITLE>
 GstRtpBin
 <SUBSECTION Standard>
 GstRtpBinPrivate
@@ -972,8 +972,8 @@ GST_IS_RTP_BIN_CLASS
 </SECTION>
 
 <SECTION>
-<FILE>element-gstrtpjitterbuffer</FILE>
-<TITLE>gstrtpjitterbuffer</TITLE>
+<FILE>element-rtpjitterbuffer</FILE>
+<TITLE>rtpjitterbuffer</TITLE>
 GstRtpJitterBuffer
 <SUBSECTION Standard>
 GstRtpJitterBufferClass
@@ -987,8 +987,8 @@ GST_IS_RTP_JITTER_BUFFER_CLASS
 </SECTION>
 
 <SECTION>
-<FILE>element-gstrtpptdemux</FILE>
-<TITLE>gstrtpptdemux</TITLE>
+<FILE>element-rtpptdemux</FILE>
+<TITLE>rtpptdemux</TITLE>
 GstRtpPtDemux
 <SUBSECTION Standard>
 GstRtpPtDemuxClass
@@ -1002,8 +1002,8 @@ GST_IS_RTP_PT_DEMUX_CLASS
 </SECTION>
 
 <SECTION>
-<FILE>element-gstrtpsession</FILE>
-<TITLE>gstrtpsession</TITLE>
+<FILE>element-rtpsession</FILE>
+<TITLE>rtpsession</TITLE>
 GstRtpSession
 <SUBSECTION Standard>
 GstRtpSessionClass
@@ -1018,8 +1018,8 @@ GST_RTP_SESSION_CAST
 </SECTION>
 
 <SECTION>
-<FILE>element-gstrtpssrcdemux</FILE>
-<TITLE>gstrtpssrcdemux</TITLE>
+<FILE>element-rtpssrcdemux</FILE>
+<TITLE>rtpssrcdemux</TITLE>
 GstRtpSsrcDemux
 <SUBSECTION Standard>
 GstRtpSsrcDemuxClass

diff --git a/docs/plugins/inspect/plugin-gstrtpmanager.xml b/docs/plugins/inspect/plugin-rtpmanager.xml
similarity index 96%
rename from docs/plugins/inspect/plugin-gstrtpmanager.xml
rename to docs/plugins/inspect/plugin-rtpmanager.xml
index b683112..56adfdc 100644 (file)
--- a/docs/plugins/inspect/plugin-gstrtpmanager.xml
+++ b/docs/plugins/inspect/plugin-rtpmanager.xml
@@ -1,5 +1,5 @@
 <plugin>
-  <name>gstrtpmanager</name>
+  <name>rtpmanager</name>
   <description>RTP session management plugin library</description>
   <filename>../../gst/rtpmanager/.libs/libgstrtpmanager.so</filename>
   <basename>libgstrtpmanager.so</basename>
@@ -10,7 +10,7 @@
   <origin>Unknown package origin</origin>
   <elements>
     <element>
-      <name>gstrtpbin</name>
+      <name>rtpbin</name>
       <longname>RTP Bin</longname>
       <class>Filter/Network/RTP</class>
       <description>Real-Time Transport Protocol bin</description>
@@ -55,7 +55,7 @@
       </pads>
     </element>
     <element>
-      <name>gstrtpjitterbuffer</name>
+      <name>rtpjitterbuffer</name>
       <longname>RTP packet jitter-buffer</longname>
       <class>Filter/Network/RTP</class>
       <description>A buffer that deals with network jitter and other transmission faults</description>
@@ -82,7 +82,7 @@
       </pads>
     </element>
     <element>
-      <name>gstrtpptdemux</name>
+      <name>rtpptdemux</name>
       <longname>RTP Demux</longname>
       <class>Demux/Network/RTP</class>
       <description>Parses codec streams transmitted in the same RTP session</description>
@@ -103,7 +103,7 @@
       </pads>
     </element>
     <element>
-      <name>gstrtpsession</name>
+      <name>rtpsession</name>
       <longname>RTP Session</longname>
       <class>Filter/Network/RTP</class>
       <description>Implement an RTP session</description>
@@ -154,7 +154,7 @@
       </pads>
     </element>
     <element>
-      <name>gstrtpssrcdemux</name>
+      <name>rtpssrcdemux</name>
       <longname>RTP SSRC Demux</longname>
       <class>Demux/Network/RTP</class>
       <description>Splits RTP streams based on the SSRC</description>

diff --git a/gst/rtpmanager/gstrtpbin.c b/gst/rtpmanager/gstrtpbin.c
index 5da3608..bc1259a 100644 (file)
--- a/gst/rtpmanager/gstrtpbin.c
+++ b/gst/rtpmanager/gstrtpbin.c
@@ -555,10 +555,10 @@ create_session (GstRtpBin * rtpbin, gint id)
   GstElement *session, *demux;
   GstState target;
 
-  if (!(session = gst_element_factory_make ("gstrtpsession", NULL)))
+  if (!(session = gst_element_factory_make ("rtpsession", NULL)))
     goto no_session;
 
-  if (!(demux = gst_element_factory_make ("gstrtpssrcdemux", NULL)))
+  if (!(demux = gst_element_factory_make ("rtpssrcdemux", NULL)))
     goto no_demux;
 
   sess = g_new0 (GstRtpBinSession, 1);
@@ -615,13 +615,13 @@ create_session (GstRtpBin * rtpbin, gint id)
   /* ERRORS */
 no_session:
   {
-    g_warning ("gstrtpbin: could not create gstrtpsession element");
+    g_warning ("rtpbin: could not create gstrtpsession element");
     return NULL;
   }
 no_demux:
   {
     gst_object_unref (session);
-    g_warning ("gstrtpbin: could not create gstrtpssrcdemux element");
+    g_warning ("rtpbin: could not create gstrtpssrcdemux element");
     return NULL;
   }
 }
@@ -1403,11 +1403,11 @@ create_stream (GstRtpBinSession * session, guint32 ssrc)
 
   rtpbin = session->bin;
 
-  if (!(buffer = gst_element_factory_make ("gstrtpjitterbuffer", NULL)))
+  if (!(buffer = gst_element_factory_make ("rtpjitterbuffer", NULL)))
     goto no_jitterbuffer;
 
   if (!rtpbin->ignore_pt)
-    if (!(demux = gst_element_factory_make ("gstrtpptdemux", NULL)))
+    if (!(demux = gst_element_factory_make ("rtpptdemux", NULL)))
       goto no_demux;
 
 
@@ -1471,13 +1471,13 @@ create_stream (GstRtpBinSession * session, guint32 ssrc)
   /* ERRORS */
 no_jitterbuffer:
   {
-    g_warning ("gstrtpbin: could not create gstrtpjitterbuffer element");
+    g_warning ("rtpbin: could not create gstrtpjitterbuffer element");
     return NULL;
   }
 no_demux:
   {
     gst_object_unref (buffer);
-    g_warning ("gstrtpbin: could not create gstrtpptdemux element");
+    g_warning ("rtpbin: could not create gstrtpptdemux element");
     return NULL;
   }
 }
@@ -2646,7 +2646,7 @@ create_recv_rtp (GstRtpBin * rtpbin, GstPadTemplate * templ, const gchar * name)
   /* ERRORS */
 no_name:
   {
-    g_warning ("gstrtpbin: invalid name given");
+    g_warning ("rtpbin: invalid name given");
     return NULL;
   }
 create_error:
@@ -2656,12 +2656,12 @@ create_error:
   }
 pad_failed:
   {
-    g_warning ("gstrtpbin: failed to get session pad");
+    g_warning ("rtpbin: failed to get session pad");
     return NULL;
   }
 link_failed:
   {
-    g_warning ("gstrtpbin: failed to link pads");
+    g_warning ("rtpbin: failed to link pads");
     return NULL;
   }
 }
@@ -2757,7 +2757,7 @@ create_recv_rtcp (GstRtpBin * rtpbin, GstPadTemplate * templ,
   /* ERRORS */
 no_name:
   {
-    g_warning ("gstrtpbin: invalid name given");
+    g_warning ("rtpbin: invalid name given");
     return NULL;
   }
 create_error:
@@ -2767,12 +2767,12 @@ create_error:
   }
 pad_failed:
   {
-    g_warning ("gstrtpbin: failed to get session pad");
+    g_warning ("rtpbin: failed to get session pad");
     return NULL;
   }
 link_failed:
   {
-    g_warning ("gstrtpbin: failed to link pads");
+    g_warning ("rtpbin: failed to link pads");
     return NULL;
   }
 }
@@ -2858,7 +2858,7 @@ create_send_rtp (GstRtpBin * rtpbin, GstPadTemplate * templ, const gchar * name)
   /* ERRORS */
 no_name:
   {
-    g_warning ("gstrtpbin: invalid name given");
+    g_warning ("rtpbin: invalid name given");
     return NULL;
   }
 create_error:
@@ -2868,13 +2868,12 @@ create_error:
   }
 pad_failed:
   {
-    g_warning ("gstrtpbin: failed to get session pad for session %d", sessid);
+    g_warning ("rtpbin: failed to get session pad for session %d", sessid);
     return NULL;
   }
 no_srcpad:
   {
-    g_warning ("gstrtpbin: failed to get rtp source pad for session %d",
-        sessid);
+    g_warning ("rtpbin: failed to get rtp source pad for session %d", sessid);
     return NULL;
   }
 }
@@ -2944,17 +2943,17 @@ create_rtcp (GstRtpBin * rtpbin, GstPadTemplate * templ, const gchar * name)
   /* ERRORS */
 no_name:
   {
-    g_warning ("gstrtpbin: invalid name given");
+    g_warning ("rtpbin: invalid name given");
     return NULL;
   }
 no_session:
   {
-    g_warning ("gstrtpbin: session with id %d does not exist", sessid);
+    g_warning ("rtpbin: session with id %d does not exist", sessid);
     return NULL;
   }
 pad_failed:
   {
-    g_warning ("gstrtpbin: failed to get rtcp pad for session %d", sessid);
+    g_warning ("rtpbin: failed to get rtcp pad for session %d", sessid);
     return NULL;
   }
 }
@@ -3088,7 +3087,7 @@ wrong_template:
   {
     g_free (pad_name);
     GST_RTP_BIN_UNLOCK (rtpbin);
-    g_warning ("gstrtpbin: this is not our template");
+    g_warning ("rtpbin: this is not our template");
     return NULL;
   }
 }
@@ -3138,7 +3137,7 @@ gst_rtp_bin_release_pad (GstElement * element, GstPad * pad)
 unknown_pad:
   {
     GST_RTP_BIN_UNLOCK (rtpbin);
-    g_warning ("gstrtpbin: %s:%s is not one of our request pads",
+    g_warning ("rtpbin: %s:%s is not one of our request pads",
         GST_DEBUG_PAD_NAME (pad));
     return;
   }

diff --git a/gst/rtpmanager/gstrtpmanager.c b/gst/rtpmanager/gstrtpmanager.c
index f38a77a..6902bf5 100644 (file)
--- a/gst/rtpmanager/gstrtpmanager.c
+++ b/gst/rtpmanager/gstrtpmanager.c
@@ -30,23 +30,22 @@
 static gboolean
 plugin_init (GstPlugin * plugin)
 {
-  if (!gst_element_register (plugin, "gstrtpbin", GST_RANK_NONE,
-          GST_TYPE_RTP_BIN))
+  if (!gst_element_register (plugin, "rtpbin", GST_RANK_NONE, GST_TYPE_RTP_BIN))
     return FALSE;
 
-  if (!gst_element_register (plugin, "gstrtpjitterbuffer", GST_RANK_NONE,
+  if (!gst_element_register (plugin, "rtpjitterbuffer", GST_RANK_NONE,
           GST_TYPE_RTP_JITTER_BUFFER))
     return FALSE;
 
-  if (!gst_element_register (plugin, "gstrtpptdemux", GST_RANK_NONE,
+  if (!gst_element_register (plugin, "rtpptdemux", GST_RANK_NONE,
           GST_TYPE_RTP_PT_DEMUX))
     return FALSE;
 
-  if (!gst_element_register (plugin, "gstrtpsession", GST_RANK_NONE,
+  if (!gst_element_register (plugin, "rtpsession", GST_RANK_NONE,
           GST_TYPE_RTP_SESSION))
     return FALSE;
 
-  if (!gst_element_register (plugin, "gstrtpssrcdemux", GST_RANK_NONE,
+  if (!gst_element_register (plugin, "rtpssrcdemux", GST_RANK_NONE,
           GST_TYPE_RTP_SSRC_DEMUX))
     return FALSE;
 
@@ -55,6 +54,6 @@ plugin_init (GstPlugin * plugin)
 
 GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
     GST_VERSION_MINOR,
-    "gstrtpmanager",
+    "rtpmanager",
     "RTP session management plugin library",
     plugin_init, VERSION, "LGPL", GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN)

diff --git a/tests/check/elements/rtpbin.c b/tests/check/elements/rtpbin.c
index 3f41ca7..f4393d7 100644 (file)
--- a/tests/check/elements/rtpbin.c
+++ b/tests/check/elements/rtpbin.c
@@ -29,7 +29,7 @@ GST_START_TEST (test_cleanup_send)
   GObject *session;
   gint count = 2;
 
-  rtpbin = gst_element_factory_make ("gstrtpbin", "rtpbin");
+  rtpbin = gst_element_factory_make ("rtpbin", "rtpbin");
 
   while (count--) {
     /* request session 0 */
@@ -231,7 +231,7 @@ GST_START_TEST (test_cleanup_recv)
 
   init_data (&data);
 
-  rtpbin = gst_element_factory_make ("gstrtpbin", "rtpbin");
+  rtpbin = gst_element_factory_make ("rtpbin", "rtpbin");
 
   g_signal_connect (rtpbin, "pad-added", (GCallback) pad_added_cb, &data);
   g_signal_connect (rtpbin, "pad-removed", (GCallback) pad_removed_cb, &data);
@@ -306,7 +306,7 @@ GST_START_TEST (test_cleanup_recv2)
 
   init_data (&data);
 
-  rtpbin = gst_element_factory_make ("gstrtpbin", "rtpbin");
+  rtpbin = gst_element_factory_make ("rtpbin", "rtpbin");
 
   g_signal_connect (rtpbin, "pad-added", (GCallback) pad_added_cb, &data);
   g_signal_connect (rtpbin, "pad-removed", (GCallback) pad_removed_cb, &data);
@@ -382,7 +382,7 @@ GST_START_TEST (test_request_pad_by_template_name)
   GstElement *rtpbin;
   GstPad *rtp_sink1, *rtp_sink2, *rtp_sink3;
 
-  rtpbin = gst_element_factory_make ("gstrtpbin", "rtpbin");
+  rtpbin = gst_element_factory_make ("rtpbin", "rtpbin");
   rtp_sink1 = gst_element_get_request_pad (rtpbin, "recv_rtp_sink_%u");
   fail_unless (rtp_sink1 != NULL);
   fail_unless_equals_string (GST_PAD_NAME (rtp_sink1), "recv_rtp_sink_0");
@@ -417,7 +417,7 @@ GST_END_TEST;
 static Suite *
 gstrtpbin_suite (void)
 {
-  Suite *s = suite_create ("gstrtpbin");
+  Suite *s = suite_create ("rtpbin");
   TCase *tc_chain = tcase_create ("general");
 
   suite_add_tcase (s, tc_chain);

diff --git a/tests/examples/rtp/client-PCMA.c b/tests/examples/rtp/client-PCMA.c
index 885b826..6b6989e 100644 (file)
--- a/tests/examples/rtp/client-PCMA.c
+++ b/tests/examples/rtp/client-PCMA.c
@@ -184,7 +184,7 @@ main (int argc, char *argv[])
   g_assert (res == TRUE);
 
   /* the rtpbin element */
-  rtpbin = gst_element_factory_make ("gstrtpbin", "rtpbin");
+  rtpbin = gst_element_factory_make ("rtpbin", "rtpbin");
   g_assert (rtpbin);
 
   gst_bin_add (GST_BIN (pipeline), rtpbin);

diff --git a/tests/examples/rtp/client-PCMA.py b/tests/examples/rtp/client-PCMA.py
index f6230f9..5830b10 100755 (executable)
--- a/tests/examples/rtp/client-PCMA.py
+++ b/tests/examples/rtp/client-PCMA.py
@@ -35,7 +35,7 @@ RTP_RECV_PORT = 5002
 RTCP_RECV_PORT = 5003
 RTCP_SEND_PORT = 5007 
 
-#gst-launch -v gstrtpbin name=rtpbin                                                \
+#gst-launch -v rtpbin name=rtpbin                                                \
 #       udpsrc caps=$AUDIO_CAPS port=$RTP_RECV_PORT ! rtpbin.recv_rtp_sink_0              \
 #             rtpbin. ! rtppcmadepay ! alawdec ! audioconvert ! audioresample ! autoaudiosink \
 #           udpsrc port=$RTCP_RECV_PORT ! rtpbin.recv_rtcp_sink_0                              \
@@ -84,7 +84,7 @@ pipeline.add(audiodepay, audiodec, audioconv, audiores, audiosink)
 res = gst.element_link_many(audiodepay, audiodec, audioconv, audiores, audiosink)
 
 # the rtpbin element
-rtpbin = gst.element_factory_make('gstrtpbin', 'rtpbin') 
+rtpbin = gst.element_factory_make('rtpbin', 'rtpbin')
 
 pipeline.add(rtpbin)
 

diff --git a/tests/examples/rtp/server-alsasrc-PCMA.c b/tests/examples/rtp/server-alsasrc-PCMA.c
index d61ec30..8564799 100644 (file)
--- a/tests/examples/rtp/server-alsasrc-PCMA.c
+++ b/tests/examples/rtp/server-alsasrc-PCMA.c
@@ -104,7 +104,7 @@ print_stats (GstElement * rtpbin)
 
 /* build a pipeline equivalent to:
  *
- * gst-launch -v gstrtpbin name=rtpbin \
+ * gst-launch -v rtpbin name=rtpbin \
  *    $AUDIO_SRC ! audioconvert ! audioresample ! $AUDIO_ENC ! $AUDIO_PAY ! rtpbin.send_rtp_sink_0  \
  *           rtpbin.send_rtp_src_0 ! udpsink port=5002 host=$DEST                      \
  *           rtpbin.send_rtcp_src_0 ! udpsink port=5003 host=$DEST sync=false async=false \
@@ -150,7 +150,7 @@ main (int argc, char *argv[])
   }
 
   /* the rtpbin element */
-  rtpbin = gst_element_factory_make ("gstrtpbin", "rtpbin");
+  rtpbin = gst_element_factory_make ("rtpbin", "rtpbin");
   g_assert (rtpbin);
 
   gst_bin_add (GST_BIN (pipeline), rtpbin);

diff --git a/tests/examples/rtp/server-alsasrc-PCMA.py b/tests/examples/rtp/server-alsasrc-PCMA.py
index 099d79e..2caf0b9 100755 (executable)
--- a/tests/examples/rtp/server-alsasrc-PCMA.py
+++ b/tests/examples/rtp/server-alsasrc-PCMA.py
@@ -4,7 +4,7 @@ import gobject, pygst
 pygst.require("0.10")
 import gst 
 
-#gst-launch -v gstrtpbin name=rtpbin audiotestsrc ! audioconvert ! alawenc ! rtppcmapay ! rtpbin.send_rtp_sink_0 \
+#gst-launch -v rtpbin name=rtpbin audiotestsrc ! audioconvert ! alawenc ! rtppcmapay ! rtpbin.send_rtp_sink_0 \
 #                rtpbin.send_rtp_src_0 ! udpsink port=10000 host=xxx.xxx.xxx.xxx \
 #                rtpbin.send_rtcp_src_0 ! udpsink port=10001 host=xxx.xxx.xxx.xxx sync=false async=false \
 #                udpsrc port=10002 ! rtpbin.recv_rtcp_sink_0 
@@ -37,7 +37,7 @@ pipeline.add(audiosrc, audioconv, audiores, audioenc, audiopay)
 res = gst.element_link_many(audiosrc, audioconv, audiores, audioenc, audiopay)
 
 # the rtpbin element
-rtpbin = gst.element_factory_make('gstrtpbin', 'rtpbin')
+rtpbin = gst.element_factory_make('rtpbin', 'rtpbin')
 
 pipeline.add(rtpbin)