Move rtpmanager from -bad to -good.

35 files changed, 0 insertions, 15481 deletions

diff --git a/configure.ac b/configure.ac
index 18b55f44..a690126d 100644
--- a/configure.ac
+++ b/configure.ac
@@ -284,7 +284,6 @@ AG_GST_CHECK_PLUGIN(pcapparse)
 AG_GST_CHECK_PLUGIN(qtmux)
 AG_GST_CHECK_PLUGIN(rawparse)
 AG_GST_CHECK_PLUGIN(real)
-AG_GST_CHECK_PLUGIN(rtpmanager)
 AG_GST_CHECK_PLUGIN(rtpmux)
 AG_GST_CHECK_PLUGIN(scaletempo)
 AG_GST_CHECK_PLUGIN(sdp)
@@ -1696,7 +1695,6 @@ gst/pcapparse/Makefile
 gst/qtmux/Makefile
 gst/rawparse/Makefile
 gst/real/Makefile
-gst/rtpmanager/Makefile
 gst/rtpmux/Makefile
 gst/scaletempo/Makefile
 gst/sdp/Makefile
diff --git a/docs/plugins/Makefile.am b/docs/plugins/Makefile.am
index 7ac083ae..dc581fc1 100644
--- a/docs/plugins/Makefile.am
+++ b/docs/plugins/Makefile.am
@@ -142,11 +142,6 @@ EXTRA_HFILES = \
 	$(top_srcdir)/gst/pcapparse/gstpcapparse.h \
 	$(top_srcdir)/gst/rawparse/gstaudioparse.h \
 	$(top_srcdir)/gst/rawparse/gstvideoparse.h \
-	$(top_srcdir)/gst/rtpmanager/gstrtpbin.h \
-	$(top_srcdir)/gst/rtpmanager/gstrtpjitterbuffer.h \
-	$(top_srcdir)/gst/rtpmanager/gstrtpptdemux.h \
-	$(top_srcdir)/gst/rtpmanager/gstrtpsession.h \
-	$(top_srcdir)/gst/rtpmanager/gstrtpssrcdemux.h \
 	$(top_srcdir)/gst/rtpmux/gstrtpmux.h \
 	$(top_srcdir)/gst/rtpmux/gstrtpdtmfmux.h \
 	$(top_srcdir)/gst/scaletempo/gstscaletempo.h \
diff --git a/docs/plugins/gst-plugins-bad-plugins-docs.sgml b/docs/plugins/gst-plugins-bad-plugins-docs.sgml
index ef8eb36b..cf6fed4b 100644
--- a/docs/plugins/gst-plugins-bad-plugins-docs.sgml
+++ b/docs/plugins/gst-plugins-bad-plugins-docs.sgml
@@ -38,11 +38,6 @@
     <xi:include href="xml/element-dvdspu.xml" />
     <xi:include href="xml/element-festival.xml" />
     <xi:include href="xml/element-fpsdisplaysink.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-input-selector.xml" />
     <xi:include href="xml/element-ivorbisdec.xml" />
     <xi:include href="xml/element-jackaudiosrc.xml" />
@@ -122,7 +117,6 @@
     <xi:include href="xml/plugin-freeze.xml" />
     <xi:include href="xml/plugin-frei0r.xml" />
     <xi:include href="xml/plugin-gsm.xml" />
-    <xi:include href="xml/plugin-gstrtpmanager.xml" />
     <xi:include href="xml/plugin-h264parse.xml" />
     <xi:include href="xml/plugin-jack.xml" />
     <xi:include href="xml/plugin-kate.xml" />
diff --git a/docs/plugins/gst-plugins-bad-plugins-sections.txt b/docs/plugins/gst-plugins-bad-plugins-sections.txt
index c60e127c..45edb369 100644
--- a/docs/plugins/gst-plugins-bad-plugins-sections.txt
+++ b/docs/plugins/gst-plugins-bad-plugins-sections.txt
@@ -781,82 +781,6 @@ gst_rtp_session_set_ssrc
 </SECTION>
 
 <SECTION>
-<FILE>element-gstrtpbin</FILE>
-<TITLE>gstrtpbin</TITLE>
-GstRtpBin
-<SUBSECTION Standard>
-GstRtpBinPrivate
-GstRtpBinClass
-GST_RTP_BIN 
-GST_IS_RTP_BIN
-GST_TYPE_RTP_BIN
-gst_rtp_bin_get_type
-GST_RTP_BIN_CLASS
-GST_IS_RTP_BIN_CLASS
-</SECTION>
-
-<SECTION>
-<FILE>element-gstrtpjitterbuffer</FILE>
-<TITLE>gstrtpjitterbuffer</TITLE>
-GstRtpJitterBuffer
-<SUBSECTION Standard>
-GstRtpJitterBufferClass
-GstRtpJitterBufferPrivate
-GST_RTP_JITTER_BUFFER
-GST_IS_RTP_JITTER_BUFFER
-GST_TYPE_RTP_JITTER_BUFFER
-gst_rtp_jitter_buffer_get_type
-GST_RTP_JITTER_BUFFER_CLASS
-GST_IS_RTP_JITTER_BUFFER_CLASS
-</SECTION>
-
-<SECTION>
-<FILE>element-gstrtpptdemux</FILE>
-<TITLE>gstrtpptdemux</TITLE>
-GstRtpPtDemux
-<SUBSECTION Standard>
-GstRtpPtDemuxClass
-GstRtpPtDemuxPad
-GST_RTP_PT_DEMUX
-GST_IS_RTP_PT_DEMUX
-GST_TYPE_RTP_PT_DEMUX
-gst_rtp_pt_demux_get_type
-GST_RTP_PT_DEMUX_CLASS
-GST_IS_RTP_PT_DEMUX_CLASS
-</SECTION>
-
-<SECTION>
-<FILE>element-gstrtpsession</FILE>
-<TITLE>gstrtpsession</TITLE>
-GstRtpSession
-<SUBSECTION Standard>
-GstRtpSessionClass
-GstRtpSessionPrivate
-GST_RTP_SESSION
-GST_IS_RTP_SESSION
-GST_TYPE_RTP_SESSION
-gst_rtp_session_get_type
-GST_RTP_SESSION_CLASS
-GST_IS_RTP_SESSION_CLASS
-GST_RTP_SESSION_CAST
-</SECTION>
-
-<SECTION>
-<FILE>element-gstrtpssrcdemux</FILE>
-<TITLE>gstrtpssrcdemux</TITLE>
-GstRtpSsrcDemux
-<SUBSECTION Standard>
-GstRtpSsrcDemuxClass
-GstRtpSsrcDemuxPad
-GST_RTP_SSRC_DEMUX
-GST_IS_RTP_SSRC_DEMUX
-GST_TYPE_RTP_SSRC_DEMUX
-gst_rtp_ssrc_demux_get_type
-GST_RTP_SSRC_DEMUX_CLASS
-GST_IS_RTP_SSRC_DEMUX_CLASS
-</SECTION>
-
-<SECTION>
 <FILE>element-scaletempo</FILE>
 <TITLE>scaletempo</TITLE>
 GstScaletempo
diff --git a/docs/plugins/gst-plugins-bad-plugins.args b/docs/plugins/gst-plugins-bad-plugins.args
index f25241fe..4b259eab 100644
--- a/docs/plugins/gst-plugins-bad-plugins.args
+++ b/docs/plugins/gst-plugins-bad-plugins.args
@@ -17559,36 +17559,6 @@
 </ARG>
 
 <ARG>
-<NAME>GstRTPJitterBuffer::drop-on-latency</NAME>
-<TYPE>gboolean</TYPE>
-<RANGE></RANGE>
-<FLAGS>rw</FLAGS>
-<NICK>Drop buffers when maximum latency is reached</NICK>
-<BLURB>Tells the jitterbuffer to never exceed the given latency in size.</BLURB>
-<DEFAULT>FALSE</DEFAULT>
-</ARG>
-
-<ARG>
-<NAME>GstRTPJitterBuffer::latency</NAME>
-<TYPE>guint</TYPE>
-<RANGE></RANGE>
-<FLAGS>rw</FLAGS>
-<NICK>Buffer latency in ms</NICK>
-<BLURB>Amount of ms to buffer.</BLURB>
-<DEFAULT>200</DEFAULT>
-</ARG>
-
-<ARG>
-<NAME>GstRTPBin::latency</NAME>
-<TYPE>guint</TYPE>
-<RANGE></RANGE>
-<FLAGS>rw</FLAGS>
-<NICK>Buffer latency in ms</NICK>
-<BLURB>Default amount of ms to buffer in the jitterbuffers.</BLURB>
-<DEFAULT>200</DEFAULT>
-</ARG>
-
-<ARG>
 <NAME>GstOSXVideoSink::embed</NAME>
 <TYPE>gboolean</TYPE>
 <RANGE></RANGE>
@@ -18109,286 +18079,6 @@
 </ARG>
 
 <ARG>
-<NAME>GstRtpSession::ntp-ns-base</NAME>
-<TYPE>guint64</TYPE>
-<RANGE></RANGE>
-<FLAGS>rw</FLAGS>
-<NICK>NTP base time</NICK>
-<BLURB>The NTP base time corresponding to running_time 0.</BLURB>
-<DEFAULT>0</DEFAULT>
-</ARG>
-
-<ARG>
-<NAME>GstRtpSession::bandwidth</NAME>
-<TYPE>gdouble</TYPE>
-<RANGE>>= 0</RANGE>
-<FLAGS>rw</FLAGS>
-<NICK>Bandwidth</NICK>
-<BLURB>The bandwidth of the session.</BLURB>
-<DEFAULT>64000</DEFAULT>
-</ARG>
-
-<ARG>
-<NAME>GstRtpSession::num-active-sources</NAME>
-<TYPE>guint</TYPE>
-<RANGE></RANGE>
-<FLAGS>r</FLAGS>
-<NICK>Num Active Sources</NICK>
-<BLURB>The number of active sources in the session.</BLURB>
-<DEFAULT>0</DEFAULT>
-</ARG>
-
-<ARG>
-<NAME>GstRtpSession::num-sources</NAME>
-<TYPE>guint</TYPE>
-<RANGE></RANGE>
-<FLAGS>r</FLAGS>
-<NICK>Num Sources</NICK>
-<BLURB>The number of sources in the session.</BLURB>
-<DEFAULT>0</DEFAULT>
-</ARG>
-
-<ARG>
-<NAME>GstRtpSession::rtcp-fraction</NAME>
-<TYPE>gdouble</TYPE>
-<RANGE>>= 0</RANGE>
-<FLAGS>rw</FLAGS>
-<NICK>RTCP Fraction</NICK>
-<BLURB>The fraction of the bandwidth used for RTCP.</BLURB>
-<DEFAULT>3000</DEFAULT>
-</ARG>
-
-<ARG>
-<NAME>GstRtpSession::sdes-cname</NAME>
-<TYPE>gchar*</TYPE>
-<RANGE></RANGE>
-<FLAGS>rw</FLAGS>
-<NICK>SDES CNAME</NICK>
-<BLURB>The CNAME to put in SDES messages of this session.</BLURB>
-<DEFAULT>NULL</DEFAULT>
-</ARG>
-
-<ARG>
-<NAME>GstRtpSession::sdes-email</NAME>
-<TYPE>gchar*</TYPE>
-<RANGE></RANGE>
-<FLAGS>rw</FLAGS>
-<NICK>SDES EMAIL</NICK>
-<BLURB>The EMAIL to put in SDES messages of this session.</BLURB>
-<DEFAULT>NULL</DEFAULT>
-</ARG>
-
-<ARG>
-<NAME>GstRtpSession::sdes-location</NAME>
-<TYPE>gchar*</TYPE>
-<RANGE></RANGE>
-<FLAGS>rw</FLAGS>
-<NICK>SDES LOCATION</NICK>
-<BLURB>The LOCATION to put in SDES messages of this session.</BLURB>
-<DEFAULT>NULL</DEFAULT>
-</ARG>
-
-<ARG>
-<NAME>GstRtpSession::sdes-name</NAME>
-<TYPE>gchar*</TYPE>
-<RANGE></RANGE>
-<FLAGS>rw</FLAGS>
-<NICK>SDES NAME</NICK>
-<BLURB>The NAME to put in SDES messages of this session.</BLURB>
-<DEFAULT>NULL</DEFAULT>
-</ARG>
-
-<ARG>
-<NAME>GstRtpSession::sdes-note</NAME>
-<TYPE>gchar*</TYPE>
-<RANGE></RANGE>
-<FLAGS>rw</FLAGS>
-<NICK>SDES NOTE</NICK>
-<BLURB>The NOTE to put in SDES messages of this session.</BLURB>
-<DEFAULT>NULL</DEFAULT>
-</ARG>
-
-<ARG>
-<NAME>GstRtpSession::sdes-phone</NAME>
-<TYPE>gchar*</TYPE>
-<RANGE></RANGE>
-<FLAGS>rw</FLAGS>
-<NICK>SDES PHONE</NICK>
-<BLURB>The PHONE to put in SDES messages of this session.</BLURB>
-<DEFAULT>NULL</DEFAULT>
-</ARG>
-
-<ARG>
-<NAME>GstRtpSession::sdes-tool</NAME>
-<TYPE>gchar*</TYPE>
-<RANGE></RANGE>
-<FLAGS>rw</FLAGS>
-<NICK>SDES TOOL</NICK>
-<BLURB>The TOOL to put in SDES messages of this session.</BLURB>
-<DEFAULT>NULL</DEFAULT>
-</ARG>
-
-<ARG>
-<NAME>GstRtpSession::internal-session</NAME>
-<TYPE>RTPSession*</TYPE>
-<RANGE></RANGE>
-<FLAGS>r</FLAGS>
-<NICK>Internal Session</NICK>
-<BLURB>The internal RTPSession object.</BLURB>
-<DEFAULT></DEFAULT>
-</ARG>
-
-<ARG>
-<NAME>GstRtpSession::sdes</NAME>
-<TYPE>GstStructure*</TYPE>
-<RANGE></RANGE>
-<FLAGS>rw</FLAGS>
-<NICK>SDES</NICK>
-<BLURB>The SDES items of this session.</BLURB>
-<DEFAULT></DEFAULT>
-</ARG>
-
-<ARG>
-<NAME>GstRtpJitterBuffer::drop-on-latency</NAME>
-<TYPE>gboolean</TYPE>
-<RANGE></RANGE>
-<FLAGS>rw</FLAGS>
-<NICK>Drop buffers when maximum latency is reached</NICK>
-<BLURB>Tells the jitterbuffer to never exceed the given latency in size.</BLURB>
-<DEFAULT>FALSE</DEFAULT>
-</ARG>
-
-<ARG>
-<NAME>GstRtpJitterBuffer::latency</NAME>
-<TYPE>guint</TYPE>
-<RANGE></RANGE>
-<FLAGS>rw</FLAGS>
-<NICK>Buffer latency in ms</NICK>
-<BLURB>Amount of ms to buffer.</BLURB>
-<DEFAULT>200</DEFAULT>
-</ARG>
-
-<ARG>
-<NAME>GstRtpJitterBuffer::ts-offset</NAME>
-<TYPE>gint64</TYPE>
-<RANGE></RANGE>
-<FLAGS>rw</FLAGS>
-<NICK>Timestamp Offset</NICK>
-<BLURB>Adjust buffer timestamps with offset in nanoseconds.</BLURB>
-<DEFAULT>0</DEFAULT>
-</ARG>
-
-<ARG>
-<NAME>GstRtpJitterBuffer::do-lost</NAME>
-<TYPE>gboolean</TYPE>
-<RANGE></RANGE>
-<FLAGS>rw</FLAGS>
-<NICK>Do Lost</NICK>
-<BLURB>Send an event downstream when a packet is lost.</BLURB>
-<DEFAULT>FALSE</DEFAULT>
-</ARG>
-
-<ARG>
-<NAME>GstRtpBin::latency</NAME>
-<TYPE>guint</TYPE>
-<RANGE></RANGE>
-<FLAGS>rw</FLAGS>
-<NICK>Buffer latency in ms</NICK>
-<BLURB>Default amount of ms to buffer in the jitterbuffers.</BLURB>
-<DEFAULT>200</DEFAULT>
-</ARG>
-
-<ARG>
-<NAME>GstRtpBin::sdes-cname</NAME>
-<TYPE>gchar*</TYPE>
-<RANGE></RANGE>
-<FLAGS>rw</FLAGS>
-<NICK>SDES CNAME</NICK>
-<BLURB>The CNAME to put in SDES messages of this session.</BLURB>
-<DEFAULT>NULL</DEFAULT>
-</ARG>
-
-<ARG>
-<NAME>GstRtpBin::sdes-email</NAME>
-<TYPE>gchar*</TYPE>
-<RANGE></RANGE>
-<FLAGS>rw</FLAGS>
-<NICK>SDES EMAIL</NICK>
-<BLURB>The EMAIL to put in SDES messages of this session.</BLURB>
-<DEFAULT>NULL</DEFAULT>
-</ARG>
-
-<ARG>
-<NAME>GstRtpBin::sdes-location</NAME>
-<TYPE>gchar*</TYPE>
-<RANGE></RANGE>
-<FLAGS>rw</FLAGS>
-<NICK>SDES LOCATION</NICK>
-<BLURB>The LOCATION to put in SDES messages of this session.</BLURB>
-<DEFAULT>NULL</DEFAULT>
-</ARG>
-
-<ARG>
-<NAME>GstRtpBin::sdes-name</NAME>
-<TYPE>gchar*</TYPE>
-<RANGE></RANGE>
-<FLAGS>rw</FLAGS>
-<NICK>SDES NAME</NICK>
-<BLURB>The NAME to put in SDES messages of this session.</BLURB>
-<DEFAULT>NULL</DEFAULT>
-</ARG>
-
-<ARG>
-<NAME>GstRtpBin::sdes-note</NAME>
-<TYPE>gchar*</TYPE>
-<RANGE></RANGE>
-<FLAGS>rw</FLAGS>
-<NICK>SDES NOTE</NICK>
-<BLURB>The NOTE to put in SDES messages of this session.</BLURB>
-<DEFAULT>NULL</DEFAULT>
-</ARG>
-
-<ARG>
-<NAME>GstRtpBin::sdes-phone</NAME>
-<TYPE>gchar*</TYPE>
-<RANGE></RANGE>
-<FLAGS>rw</FLAGS>
-<NICK>SDES PHONE</NICK>
-<BLURB>The PHONE to put in SDES messages of this session.</BLURB>
-<DEFAULT>NULL</DEFAULT>
-</ARG>
-
-<ARG>
-<NAME>GstRtpBin::sdes-tool</NAME>
-<TYPE>gchar*</TYPE>
-<RANGE></RANGE>
-<FLAGS>rw</FLAGS>
-<NICK>SDES TOOL</NICK>
-<BLURB>The TOOL to put in SDES messages of this session.</BLURB>
-<DEFAULT>NULL</DEFAULT>
-</ARG>
-
-<ARG>
-<NAME>GstRtpBin::do-lost</NAME>
-<TYPE>gboolean</TYPE>
-<RANGE></RANGE>
-<FLAGS>rw</FLAGS>
-<NICK>Do Lost</NICK>
-<BLURB>Send an event downstream when a packet is lost.</BLURB>
-<DEFAULT>FALSE</DEFAULT>
-</ARG>
-
-<ARG>
-<NAME>GstRtpBin::sdes</NAME>
-<TYPE>GstStructure*</TYPE>
-<RANGE></RANGE>
-<FLAGS>rw</FLAGS>
-<NICK>SDES</NICK>
-<BLURB>The SDES items of this session.</BLURB>
-<DEFAULT></DEFAULT>
-</ARG>
-
-<ARG>
 <NAME>GstGioSrc::location</NAME>
 <TYPE>gchar*</TYPE>
 <RANGE></RANGE>
diff --git a/docs/plugins/gst-plugins-bad-plugins.hierarchy b/docs/plugins/gst-plugins-bad-plugins.hierarchy
index 2c755d23..10a65892 100644
--- a/docs/plugins/gst-plugins-bad-plugins.hierarchy
+++ b/docs/plugins/gst-plugins-bad-plugins.hierarchy
@@ -16,7 +16,6 @@ GObject
         GstFPSDisplaySink
         GstAutoConvert
         GstSDPDemux
-        GstRtpBin
       GstSignalProcessor
         http---calf-sourceforge-net-plugins-Compressor
         http---calf-sourceforge-net-plugins-Filter
@@ -466,10 +465,6 @@ GObject
       GstSpeed
       GstInputSelector
       GstOutputSelector
-      GstRtpJitterBuffer
-      GstRtpPtDemux
-      GstRtpSession
-      GstRtpSsrcDemux
       GstRealVideoDec
       GstRealAudioDec
       GstRawParse
diff --git a/docs/plugins/gst-plugins-bad-plugins.signals b/docs/plugins/gst-plugins-bad-plugins.signals
index 73f6b85e..0550d9b1 100644
--- a/docs/plugins/gst-plugins-bad-plugins.signals
+++ b/docs/plugins/gst-plugins-bad-plugins.signals
@@ -51,302 +51,6 @@ gint64  arg3
 </SIGNAL>
 
 <SIGNAL>
-<NAME>GstRtpBin::clear-pt-map</NAME>
-<RETURNS>void</RETURNS>
-<FLAGS>la</FLAGS>
-GstRtpBin *gstrtpbin
-</SIGNAL>
-
-<SIGNAL>
-<NAME>GstRtpBin::on-bye-ssrc</NAME>
-<RETURNS>void</RETURNS>
-<FLAGS>l</FLAGS>
-GstRtpBin *gstrtpbin
-guint  arg1
-guint  arg2
-</SIGNAL>
-
-<SIGNAL>
-<NAME>GstRtpBin::on-bye-timeout</NAME>
-<RETURNS>void</RETURNS>
-<FLAGS>l</FLAGS>
-GstRtpBin *gstrtpbin
-guint  arg1
-guint  arg2
-</SIGNAL>
-
-<SIGNAL>
-<NAME>GstRtpBin::on-new-ssrc</NAME>
-<RETURNS>void</RETURNS>
-<FLAGS>l</FLAGS>
-GstRtpBin *gstrtpbin
-guint  arg1
-guint  arg2
-</SIGNAL>
-
-<SIGNAL>
-<NAME>GstRtpBin::on-ssrc-active</NAME>
-<RETURNS>void</RETURNS>
-<FLAGS>l</FLAGS>
-GstRtpBin *gstrtpbin
-guint  arg1
-guint  arg2
-</SIGNAL>
-
-<SIGNAL>
-<NAME>GstRtpBin::on-ssrc-collision</NAME>
-<RETURNS>void</RETURNS>
-<FLAGS>l</FLAGS>
-GstRtpBin *gstrtpbin
-guint  arg1
-guint  arg2
-</SIGNAL>
-
-<SIGNAL>
-<NAME>GstRtpBin::on-ssrc-sdes</NAME>
-<RETURNS>void</RETURNS>
-<FLAGS>l</FLAGS>
-GstRtpBin *gstrtpbin
-guint  arg1
-guint  arg2
-</SIGNAL>
-
-<SIGNAL>
-<NAME>GstRtpBin::on-ssrc-validated</NAME>
-<RETURNS>void</RETURNS>
-<FLAGS>l</FLAGS>
-GstRtpBin *gstrtpbin
-guint  arg1
-guint  arg2
-</SIGNAL>
-
-<SIGNAL>
-<NAME>GstRtpBin::on-timeout</NAME>
-<RETURNS>void</RETURNS>
-<FLAGS>l</FLAGS>
-GstRtpBin *gstrtpbin
-guint  arg1
-guint  arg2
-</SIGNAL>
-
-<SIGNAL>
-<NAME>GstRtpBin::request-pt-map</NAME>
-<RETURNS>GstCaps*</RETURNS>
-<FLAGS>l</FLAGS>
-GstRtpBin *gstrtpbin
-guint  arg1
-guint  arg2
-</SIGNAL>
-
-<SIGNAL>
-<NAME>GstRtpBin::get-internal-session</NAME>
-<RETURNS>RTPSession*</RETURNS>
-<FLAGS>la</FLAGS>
-GstRtpBin *gstrtpbin
-guint  arg1
-</SIGNAL>
-
-<SIGNAL>
-<NAME>GstRtpBin::on-sender-timeout</NAME>
-<RETURNS>void</RETURNS>
-<FLAGS>l</FLAGS>
-GstRtpBin *gstrtpbin
-guint  arg1
-guint  arg2
-</SIGNAL>
-
-<SIGNAL>
-<NAME>GstRtpBin::reset-sync</NAME>
-<RETURNS>void</RETURNS>
-<FLAGS>la</FLAGS>
-GstRtpBin *gstrtpbin
-</SIGNAL>
-
-<SIGNAL>
-<NAME>GstRtpBin::on-npt-stop</NAME>
-<RETURNS>void</RETURNS>
-<FLAGS>l</FLAGS>
-GstRtpBin *gstrtpbin
-guint  arg1
-guint  arg2
-</SIGNAL>
-
-<SIGNAL>
-<NAME>GstRtpJitterBuffer::clear-pt-map</NAME>
-<RETURNS>void</RETURNS>
-<FLAGS>la</FLAGS>
-GstRtpJitterBuffer *gstrtpjitterbuffer
-</SIGNAL>
-
-<SIGNAL>
-<NAME>GstRtpJitterBuffer::request-pt-map</NAME>
-<RETURNS>GstCaps*</RETURNS>
-<FLAGS>l</FLAGS>
-GstRtpJitterBuffer *gstrtpjitterbuffer
-guint  arg1
-</SIGNAL>
-
-<SIGNAL>
-<NAME>GstRtpJitterBuffer::handle-sync</NAME>
-<RETURNS>void</RETURNS>
-<FLAGS>l</FLAGS>
-GstRtpJitterBuffer *gstrtpjitterbuffer
-GstStructure *arg1
-</SIGNAL>
-
-<SIGNAL>
-<NAME>GstRtpJitterBuffer::on-npt-stop</NAME>
-<RETURNS>void</RETURNS>
-<FLAGS>l</FLAGS>
-GstRtpJitterBuffer *gstrtpjitterbuffer
-</SIGNAL>
-
-<SIGNAL>
-<NAME>GstRtpPtDemux::clear-pt-map</NAME>
-<RETURNS>void</RETURNS>
-<FLAGS>la</FLAGS>
-GstRtpPtDemux *gstrtpptdemux
-</SIGNAL>
-
-<SIGNAL>
-<NAME>GstRtpPtDemux::new-payload-type</NAME>
-<RETURNS>void</RETURNS>
-<FLAGS>l</FLAGS>
-GstRtpPtDemux *gstrtpptdemux
-guint  arg1
-GstPad *arg2
-</SIGNAL>
-
-<SIGNAL>
-<NAME>GstRtpPtDemux::payload-type-change</NAME>
-<RETURNS>void</RETURNS>
-<FLAGS>l</FLAGS>
-GstRtpPtDemux *gstrtpptdemux
-guint  arg1
-</SIGNAL>
-
-<SIGNAL>
-<NAME>GstRtpPtDemux::request-pt-map</NAME>
-<RETURNS>GstCaps*</RETURNS>
-<FLAGS>l</FLAGS>
-GstRtpPtDemux *gstrtpptdemux
-guint  arg1
-</SIGNAL>
-
-<SIGNAL>
-<NAME>GstRtpSession::clear-pt-map</NAME>
-<RETURNS>void</RETURNS>
-<FLAGS>a</FLAGS>
-GstRtpSession *gstrtpsession
-</SIGNAL>
-
-<SIGNAL>
-<NAME>GstRtpSession::on-bye-ssrc</NAME>
-<RETURNS>void</RETURNS>
-<FLAGS>l</FLAGS>
-GstRtpSession *gstrtpsession
-guint  arg1
-</SIGNAL>
-
-<SIGNAL>
-<NAME>GstRtpSession::on-bye-timeout</NAME>
-<RETURNS>void</RETURNS>
-<FLAGS>l</FLAGS>
-GstRtpSession *gstrtpsession
-guint  arg1
-</SIGNAL>
-
-<SIGNAL>
-<NAME>GstRtpSession::on-new-ssrc</NAME>
-<RETURNS>void</RETURNS>
-<FLAGS>l</FLAGS>
-GstRtpSession *gstrtpsession
-guint  arg1
-</SIGNAL>
-
-<SIGNAL>
-<NAME>GstRtpSession::on-ssrc-active</NAME>
-<RETURNS>void</RETURNS>
-<FLAGS>l</FLAGS>
-GstRtpSession *gstrtpsession
-guint  arg1
-</SIGNAL>
-
-<SIGNAL>
-<NAME>GstRtpSession::on-ssrc-collision</NAME>
-<RETURNS>void</RETURNS>
-<FLAGS>l</FLAGS>
-GstRtpSession *gstrtpsession
-guint  arg1
-</SIGNAL>
-
-<SIGNAL>
-<NAME>GstRtpSession::on-ssrc-sdes</NAME>
-<RETURNS>void</RETURNS>
-<FLAGS>l</FLAGS>
-GstRtpSession *gstrtpsession
-guint  arg1
-</SIGNAL>
-
-<SIGNAL>
-<NAME>GstRtpSession::on-ssrc-validated</NAME>
-<RETURNS>void</RETURNS>
-<FLAGS>l</FLAGS>
-GstRtpSession *gstrtpsession
-guint  arg1
-</SIGNAL>
-
-<SIGNAL>
-<NAME>GstRtpSession::on-timeout</NAME>
-<RETURNS>void</RETURNS>
-<FLAGS>l</FLAGS>
-GstRtpSession *gstrtpsession
-guint  arg1
-</SIGNAL>
-
-<SIGNAL>
-<NAME>GstRtpSession::request-pt-map</NAME>
-<RETURNS>GstCaps*</RETURNS>
-<FLAGS>l</FLAGS>
-GstRtpSession *gstrtpsession
-guint  arg1
-</SIGNAL>
-
-<SIGNAL>
-<NAME>GstRtpSession::on-sender-timeout</NAME>
-<RETURNS>void</RETURNS>
-<FLAGS>l</FLAGS>
-GstRtpSession *gstrtpsession
-guint  arg1
-</SIGNAL>
-
-<SIGNAL>
-<NAME>GstRtpSsrcDemux::new-ssrc-pad</NAME>
-<RETURNS>void</RETURNS>
-<FLAGS>l</FLAGS>
-GstRtpSsrcDemux *gstrtpssrcdemux
-guint  arg1
-GstPad *arg2
-</SIGNAL>
-
-<SIGNAL>
-<NAME>GstRtpSsrcDemux::clear-ssrc</NAME>
-<RETURNS>void</RETURNS>
-<FLAGS>la</FLAGS>
-GstRtpSsrcDemux *gstrtpssrcdemux
-guint  arg1
-</SIGNAL>
-
-<SIGNAL>
-<NAME>GstRtpSsrcDemux::removed-ssrc-pad</NAME>
-<RETURNS>void</RETURNS>
-<FLAGS>l</FLAGS>
-GstRtpSsrcDemux *gstrtpssrcdemux
-guint  arg1
-GstPad *arg2
-</SIGNAL>
-
-<SIGNAL>
 <NAME>GstCDAudio::track-change</NAME>
 <RETURNS>void</RETURNS>
 <FLAGS>l</FLAGS>
diff --git a/docs/plugins/inspect/plugin-gstrtpmanager.xml b/docs/plugins/inspect/plugin-gstrtpmanager.xml
deleted file mode 100644
index 82e96115..00000000
--- a/docs/plugins/inspect/plugin-gstrtpmanager.xml
+++ /dev/null
@@ -1,190 +0,0 @@
-<plugin>
-  <name>gstrtpmanager</name>
-  <description>RTP session management plugin library</description>
-  <filename>../../gst/rtpmanager/.libs/libgstrtpmanager.so</filename>
-  <basename>libgstrtpmanager.so</basename>
-  <version>0.10.13.1</version>
-  <license>LGPL</license>
-  <source>gst-plugins-bad</source>
-  <package>GStreamer Bad Plug-ins git/prerelease</package>
-  <origin>http://gstreamer.freedesktop.org</origin>
-  <elements>
-    <element>
-      <name>gstrtpbin</name>
-      <longname>RTP Bin</longname>
-      <class>Filter/Network/RTP</class>
-      <description>Implement an RTP bin</description>
-      <author>Wim Taymans &lt;wim.taymans@gmail.com&gt;</author>
-      <pads>
-        <caps>
-          <name>send_rtp_src_%d</name>
-          <direction>source</direction>
-          <presence>sometimes</presence>
-          <details>application/x-rtp</details>
-        </caps>
-        <caps>
-          <name>send_rtcp_src_%d</name>
-          <direction>source</direction>
-          <presence>request</presence>
-          <details>application/x-rtcp</details>
-        </caps>
-        <caps>
-          <name>recv_rtp_src_%d_%d_%d</name>
-          <direction>source</direction>
-          <presence>sometimes</presence>
-          <details>application/x-rtp</details>
-        </caps>
-        <caps>
-          <name>send_rtp_sink_%d</name>
-          <direction>sink</direction>
-          <presence>request</presence>
-          <details>application/x-rtp</details>
-        </caps>
-        <caps>
-          <name>recv_rtcp_sink_%d</name>
-          <direction>sink</direction>
-          <presence>request</presence>
-          <details>application/x-rtcp</details>
-        </caps>
-        <caps>
-          <name>recv_rtp_sink_%d</name>
-          <direction>sink</direction>
-          <presence>request</presence>
-          <details>application/x-rtp</details>
-        </caps>
-      </pads>
-    </element>
-    <element>
-      <name>gstrtpjitterbuffer</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>
-      <author>Philippe Kalaf &lt;philippe.kalaf@collabora.co.uk&gt;, Wim Taymans &lt;wim.taymans@gmail.com&gt;</author>
-      <pads>
-        <caps>
-          <name>sink_rtcp</name>
-          <direction>sink</direction>
-          <presence>request</presence>
-          <details>application/x-rtcp</details>
-        </caps>
-        <caps>
-          <name>sink</name>
-          <direction>sink</direction>
-          <presence>always</presence>
-          <details>application/x-rtp, clock-rate=(int)[ 1, 2147483647 ]</details>
-        </caps>
-        <caps>
-          <name>src</name>
-          <direction>source</direction>
-          <presence>always</presence>
-          <details>application/x-rtp</details>
-        </caps>
-      </pads>
-    </element>
-    <element>
-      <name>gstrtpptdemux</name>
-      <longname>RTP Demux</longname>
-      <class>Demux/Network/RTP</class>
-      <description>Parses codec streams transmitted in the same RTP session</description>
-      <author>Kai Vehmanen &lt;kai.vehmanen@nokia.com&gt;</author>
-      <pads>
-        <caps>
-          <name>src_%d</name>
-          <direction>source</direction>
-          <presence>sometimes</presence>
-          <details>application/x-rtp, payload=(int)[ 0, 255 ]</details>
-        </caps>
-        <caps>
-          <name>sink</name>
-          <direction>sink</direction>
-          <presence>always</presence>
-          <details>application/x-rtp</details>
-        </caps>
-      </pads>
-    </element>
-    <element>
-      <name>gstrtpsession</name>
-      <longname>RTP Session</longname>
-      <class>Filter/Network/RTP</class>
-      <description>Implement an RTP session</description>
-      <author>Wim Taymans &lt;wim.taymans@gmail.com&gt;</author>
-      <pads>
-        <caps>
-          <name>send_rtcp_src</name>
-          <direction>source</direction>
-          <presence>request</presence>
-          <details>application/x-rtcp</details>
-        </caps>
-        <caps>
-          <name>send_rtp_src</name>
-          <direction>source</direction>
-          <presence>sometimes</presence>
-          <details>application/x-rtp</details>
-        </caps>
-        <caps>
-          <name>sync_src</name>
-          <direction>source</direction>
-          <presence>sometimes</presence>
-          <details>application/x-rtcp</details>
-        </caps>
-        <caps>
-          <name>recv_rtp_src</name>
-          <direction>source</direction>
-          <presence>sometimes</presence>
-          <details>application/x-rtp</details>
-        </caps>
-        <caps>
-          <name>send_rtp_sink</name>
-          <direction>sink</direction>
-          <presence>request</presence>
-          <details>application/x-rtp</details>
-        </caps>
-        <caps>
-          <name>recv_rtcp_sink</name>
-          <direction>sink</direction>
-          <presence>request</presence>
-          <details>application/x-rtcp</details>
-        </caps>
-        <caps>
-          <name>recv_rtp_sink</name>
-          <direction>sink</direction>
-          <presence>request</presence>
-          <details>application/x-rtp</details>
-        </caps>
-      </pads>
-    </element>
-    <element>
-      <name>gstrtpssrcdemux</name>
-      <longname>RTP SSRC Demux</longname>
-      <class>Demux/Network/RTP</class>
-      <description>Splits RTP streams based on the SSRC</description>
-      <author>Wim Taymans &lt;wim.taymans@gmail.com&gt;</author>
-      <pads>
-        <caps>
-          <name>rtcp_src_%d</name>
-          <direction>source</direction>
-          <presence>sometimes</presence>
-          <details>application/x-rtcp</details>
-        </caps>
-        <caps>
-          <name>src_%d</name>
-          <direction>source</direction>
-          <presence>sometimes</presence>
-          <details>application/x-rtp</details>
-        </caps>
-        <caps>
-          <name>rtcp_sink</name>
-          <direction>sink</direction>
-          <presence>always</presence>
-          <details>application/x-rtcp</details>
-        </caps>
-        <caps>
-          <name>sink</name>
-          <direction>sink</direction>
-          <presence>always</presence>
-          <details>application/x-rtp</details>
-        </caps>
-      </pads>
-    </element>
-  </elements>
-</plugin>
\ No newline at end of file
diff --git a/gst-plugins-bad.spec.in b/gst-plugins-bad.spec.in
index ea4a5d82..958a24d7 100644
--- a/gst-plugins-bad.spec.in
+++ b/gst-plugins-bad.spec.in
@@ -83,7 +83,6 @@ rm -rf $RPM_BUILD_ROOT
 %{_libdir}/gstreamer-%{majorminor}/libgstreal.so
 %{_libdir}/gstreamer-%{majorminor}/libgstmve.so
 %{_libdir}/gstreamer-%{majorminor}/libgstmpegvideoparse.so
-%{_libdir}/gstreamer-%{majorminor}/libgstrtpmanager.so
 %{_libdir}/gstreamer-%{majorminor}/libgstbayer.so
 %{_libdir}/gstreamer-%{majorminor}/libgstdvdspu.so
 %{_libdir}/gstreamer-%{majorminor}/libgstfestival.so
diff --git a/gst/rtpmanager/.gitignore b/gst/rtpmanager/.gitignore
deleted file mode 100644
index 0962d997..00000000
--- a/gst/rtpmanager/.gitignore
+++ /dev/null
@@ -1,2 +0,0 @@
-gstrtpbin-marshal.h
-gstrtpbin-marshal.c
diff --git a/gst/rtpmanager/Makefile.am b/gst/rtpmanager/Makefile.am
deleted file mode 100644
index 8080f303..00000000
--- a/gst/rtpmanager/Makefile.am
+++ /dev/null
@@ -1,49 +0,0 @@
-plugin_LTLIBRARIES = libgstrtpmanager.la
-
-glib_enum_define = GST_RTP_BIN
-glib_enum_prefix = gst_rtp_bin
-
-include $(top_srcdir)/common/glib-gen.mak
-
-built_sources = gstrtpbin-marshal.c
-built_headers = gstrtpbin-marshal.h
-
-BUILT_SOURCES = $(built_sources) $(built_headers)
-
-libgstrtpmanager_la_SOURCES = gstrtpmanager.c \
-			      gstrtpbin.c \
-			      gstrtpjitterbuffer.c \
-			      gstrtpptdemux.c \
-			      gstrtpssrcdemux.c \
-			      rtpjitterbuffer.c      \
-			      rtpsession.c      \
-			      rtpsource.c      \
-			      rtpstats.c      \
-			      gstrtpsession.c
-
-nodist_libgstrtpmanager_la_SOURCES = \
-      $(built_sources)
-
-noinst_HEADERS = gstrtpbin.h \
-		 gstrtpjitterbuffer.h \
-                 gstrtpptdemux.h \
-                 gstrtpssrcdemux.h \
-                 rtpjitterbuffer.h \
-		 rtpsession.h  \
-		 rtpsource.h  \
-		 rtpstats.h  \
-		 gstrtpsession.h
-
-libgstrtpmanager_la_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(GST_CFLAGS) \
-	$(ERROR_CFLAGS)
-libgstrtpmanager_la_LIBADD = $(GST_PLUGINS_BASE_LIBS) \
-	-lgstnetbuffer-@GST_MAJORMINOR@ -lgstrtp-@GST_MAJORMINOR@ \
-	$(GST_BASE_LIBS) $(GST_LIBS_LIBS)
-libgstrtpmanager_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
-libgstrtpmanager_la_LIBTOOLFLAGS = --tag=disable-static
-
-CLEANFILES = $(BUILT_SOURCES)
-
-EXTRA_DIST = gstrtpbin-marshal.list
-
-
diff --git a/gst/rtpmanager/gstrtpbin-marshal.list b/gst/rtpmanager/gstrtpbin-marshal.list
deleted file mode 100644
index ed73e43b..00000000
--- a/gst/rtpmanager/gstrtpbin-marshal.list
+++ /dev/null
@@ -1,8 +0,0 @@
-UINT:UINT
-BOXED:UINT
-BOXED:UINT,UINT
-OBJECT:UINT
-VOID:UINT,OBJECT
-VOID:UINT
-VOID:UINT,UINT
-VOID:OBJECT,OBJECT
diff --git a/gst/rtpmanager/gstrtpbin.c b/gst/rtpmanager/gstrtpbin.c
deleted file mode 100644
index c09b0ab9..00000000
--- a/gst/rtpmanager/gstrtpbin.c
+++ /dev/null
@@ -1,2458 +0,0 @@
-/* GStreamer
- * Copyright (C) <2007> Wim Taymans <wim.taymans@gmail.com>
- *
- * This library is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Library General Public
- * License as published by the Free Software Foundation; either
- * version 2 of the License, or (at your option) any later version.
- *
- * This library is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Library General Public License for more details.
- *
- * You should have received a copy of the GNU Library General Public
- * License along with this library; if not, write to the
- * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
- * Boston, MA 02111-1307, USA.
- */
-
-/**
- * SECTION:element-gstrtpbin
- * @see_also: gstrtpjitterbuffer, gstrtpsession, gstrtpptdemux, gstrtpssrcdemux
- *
- * 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.
- * 
- * #GstRtpBin is configured with a number of request pads that define the
- * functionality that is activated, similar to the #GstRtpSession element.
- * 
- * 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 gstrtpsession
- * manager and after being validated forwarded on #GstRtpsSrcDemux 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
- * #GstRtpsSrcDemux element. The #GstRtpsSrcDemux element will demux the packets based
- * on the payload type and will create a unique pad recv_rtp_src_%%d_%%d_%%d on
- * gstrtpbin with the session number, SSRC and payload type respectively as the pad
- * name.
- * 
- * 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.
- * 
- * If you want the session manager to generate and send RTCP packets, request
- * the send_rtcp_src_%%d pad with the session number in the pad name. Packet pushed
- * on this pad contain SR/RR RTCP reports that should be sent to all participants
- * in the session.
- * 
- * To use #GstRtpBin as a sender, request a send_rtp_sink_%%d pad, which will
- * automatically create a send_rtp_src_%%d pad. If the session number is not provided,
- * the pad from the lowest available session will be returned. 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_%%d pad after updating its internal state.
- * 
- * The session manager needs the clock-rate of the payload types it is handling
- * and will signal the #GstRtpSession::request-pt-map signal when it needs such a
- * mapping. One can clear the cached values with the #GstRtpSession::clear-pt-map
- * signal.
- * 
- * <refsect2>
- * <title>Example pipelines</title>
- * |[
- * gst-launch udpsrc port=5000 caps="application/x-rtp, ..." ! .recv_rtp_sink_0 \
- *     gstrtpbin ! rtptheoradepay ! theoradec ! xvimagesink
- * ]| Receive RTP data from port 5000 and send to the session 0 in gstrtpbin.
- * |[
- * gst-launch gstrtpbin name=rtpbin \
- *         v4l2src ! ffmpegcolorspace ! ffenc_h263 ! rtph263ppay ! rtpbin.send_rtp_sink_0 \
- *                   rtpbin.send_rtp_src_0 ! udpsink port=5000                            \
- *                   rtpbin.send_rtcp_src_0 ! udpsink port=5001 sync=false async=false    \
- *                   udpsrc port=5005 ! rtpbin.recv_rtcp_sink_0                           \
- *         audiotestsrc ! amrnbenc ! rtpamrpay ! rtpbin.send_rtp_sink_1                   \
- *                   rtpbin.send_rtp_src_1 ! udpsink port=5002                            \
- *                   rtpbin.send_rtcp_src_1 ! udpsink port=5003 sync=false async=false    \
- *                   udpsrc port=5007 ! rtpbin.recv_rtcp_sink_1
- * ]| Encode and payload H263 video captured from a v4l2src. Encode and payload AMR
- * audio generated from audiotestsrc. The video is sent to session 0 in rtpbin
- * and the audio is sent to session 1. Video packets are sent on UDP port 5000
- * and audio packets on port 5002. The video RTCP packets for session 0 are sent
- * on port 5001 and the audio RTCP packets for session 0 are sent on port 5003.
- * RTCP packets for session 0 are received on port 5005 and RTCP for session 1
- * is received on port 5007. Since RTCP packets from the sender should be sent
- * as soon as possible and do not participate in preroll, sync=false and 
- * async=false is configured on udpsink
- * |[
- * gst-launch -v gstrtpbin name=rtpbin                                          \
- *     udpsrc caps="application/x-rtp,media=(string)video,clock-rate=(int)90000,encoding-name=(string)H263-1998" \
- *             port=5000 ! rtpbin.recv_rtp_sink_0                                \
- *         rtpbin. ! rtph263pdepay ! ffdec_h263 ! xvimagesink                    \
- *      udpsrc port=5001 ! rtpbin.recv_rtcp_sink_0                               \
- *      rtpbin.send_rtcp_src_0 ! udpsink port=5005 sync=false async=false        \
- *     udpsrc caps="application/x-rtp,media=(string)audio,clock-rate=(int)8000,encoding-name=(string)AMR,encoding-params=(string)1,octet-align=(string)1" \
- *             port=5002 ! rtpbin.recv_rtp_sink_1                                \
- *         rtpbin. ! rtpamrdepay ! amrnbdec ! alsasink                           \
- *      udpsrc port=5003 ! rtpbin.recv_rtcp_sink_1                               \
- *      rtpbin.send_rtcp_src_1 ! udpsink port=5007 sync=false async=false
- * ]| Receive H263 on port 5000, send it through rtpbin in session 0, depayload,
- * decode and display the video.
- * Receive AMR on port 5002, send it through rtpbin in session 1, depayload,
- * decode and play the audio.
- * Receive server RTCP packets for session 0 on port 5001 and RTCP packets for
- * session 1 on port 5003. These packets will be used for session management and
- * synchronisation.
- * Send RTCP reports for session 0 on port 5005 and RTCP reports for session 1
- * on port 5007.
- * </refsect2>
- *
- * Last reviewed on 2007-08-30 (0.10.6)
- */
-
-#ifdef HAVE_CONFIG_H
-#include "config.h"
-#endif
-#include <string.h>
-
-#include <gst/rtp/gstrtpbuffer.h>
-#include <gst/rtp/gstrtcpbuffer.h>
-
-#include "gstrtpbin-marshal.h"
-#include "gstrtpbin.h"
-#include "rtpsession.h"
-#include "gstrtpsession.h"
-#include "gstrtpjitterbuffer.h"
-
-GST_DEBUG_CATEGORY_STATIC (gst_rtp_bin_debug);
-#define GST_CAT_DEFAULT gst_rtp_bin_debug
-
-/* elementfactory information */
-static const GstElementDetails rtpbin_details = GST_ELEMENT_DETAILS ("RTP Bin",
-    "Filter/Network/RTP",
-    "Implement an RTP bin",
-    "Wim Taymans <wim.taymans@gmail.com>");
-
-/* sink pads */
-static GstStaticPadTemplate rtpbin_recv_rtp_sink_template =
-GST_STATIC_PAD_TEMPLATE ("recv_rtp_sink_%d",
-    GST_PAD_SINK,
-    GST_PAD_REQUEST,
-    GST_STATIC_CAPS ("application/x-rtp")
-    );
-
-static GstStaticPadTemplate rtpbin_recv_rtcp_sink_template =
-GST_STATIC_PAD_TEMPLATE ("recv_rtcp_sink_%d",
-    GST_PAD_SINK,
-    GST_PAD_REQUEST,
-    GST_STATIC_CAPS ("application/x-rtcp")
-    );
-
-static GstStaticPadTemplate rtpbin_send_rtp_sink_template =
-GST_STATIC_PAD_TEMPLATE ("send_rtp_sink_%d",
-    GST_PAD_SINK,
-    GST_PAD_REQUEST,
-    GST_STATIC_CAPS ("application/x-rtp")
-    );
-
-/* src pads */
-static GstStaticPadTemplate rtpbin_recv_rtp_src_template =
-GST_STATIC_PAD_TEMPLATE ("recv_rtp_src_%d_%d_%d",
-    GST_PAD_SRC,
-    GST_PAD_SOMETIMES,
-    GST_STATIC_CAPS ("application/x-rtp")
-    );
-
-static GstStaticPadTemplate rtpbin_send_rtcp_src_template =
-GST_STATIC_PAD_TEMPLATE ("send_rtcp_src_%d",
-    GST_PAD_SRC,
-    GST_PAD_REQUEST,
-    GST_STATIC_CAPS ("application/x-rtcp")
-    );
-
-static GstStaticPadTemplate rtpbin_send_rtp_src_template =
-GST_STATIC_PAD_TEMPLATE ("send_rtp_src_%d",
-    GST_PAD_SRC,
-    GST_PAD_SOMETIMES,
-    GST_STATIC_CAPS ("application/x-rtp")
-    );
-
-#define GST_RTP_BIN_GET_PRIVATE(obj)  \
-   (G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_RTP_BIN, GstRtpBinPrivate))
-
-#define GST_RTP_BIN_LOCK(bin)   g_mutex_lock ((bin)->priv->bin_lock)
-#define GST_RTP_BIN_UNLOCK(bin) g_mutex_unlock ((bin)->priv->bin_lock)
-
-/* lock to protect dynamic callbacks, like pad-added and new ssrc. */
-#define GST_RTP_BIN_DYN_LOCK(bin)    g_mutex_lock ((bin)->priv->dyn_lock)
-#define GST_RTP_BIN_DYN_UNLOCK(bin)  g_mutex_unlock ((bin)->priv->dyn_lock)
-
-/* lock for shutdown */
-#define GST_RTP_BIN_SHUTDOWN_LOCK(bin,label)     \
-G_STMT_START {                                   \
-  if (g_atomic_int_get (&bin->priv->shutdown))   \
-    goto label;                                  \
-  GST_RTP_BIN_DYN_LOCK (bin);                    \
-  if (g_atomic_int_get (&bin->priv->shutdown)) { \
-    GST_RTP_BIN_DYN_UNLOCK (bin);                \
-    goto label;                                  \
-  }                                              \
-} G_STMT_END
-
-/* unlock for shutdown */
-#define GST_RTP_BIN_SHUTDOWN_UNLOCK(bin)         \
-  GST_RTP_BIN_DYN_UNLOCK (bin);                  \
-
-struct _GstRtpBinPrivate
-{
-  GMutex *bin_lock;
-
-  /* lock protecting dynamic adding/removing */
-  GMutex *dyn_lock;
-
-  /* the time when we went to playing */
-  GstClockTime ntp_ns_base;
-
-  /* if we are shutting down or not */
-  gint shutdown;
-};
-
-/* signals and args */
-enum
-{
-  SIGNAL_REQUEST_PT_MAP,
-  SIGNAL_CLEAR_PT_MAP,
-  SIGNAL_RESET_SYNC,
-  SIGNAL_GET_INTERNAL_SESSION,
-
-  SIGNAL_ON_NEW_SSRC,
-  SIGNAL_ON_SSRC_COLLISION,
-  SIGNAL_ON_SSRC_VALIDATED,
-  SIGNAL_ON_SSRC_ACTIVE,
-  SIGNAL_ON_SSRC_SDES,
-  SIGNAL_ON_BYE_SSRC,
-  SIGNAL_ON_BYE_TIMEOUT,
-  SIGNAL_ON_TIMEOUT,
-  SIGNAL_ON_SENDER_TIMEOUT,
-  SIGNAL_ON_NPT_STOP,
-  LAST_SIGNAL
-};
-
-#define DEFAULT_LATENCY_MS	     200
-#define DEFAULT_SDES                 NULL
-#define DEFAULT_DO_LOST              FALSE
-
-enum
-{
-  PROP_0,
-  PROP_LATENCY,
-  PROP_SDES,
-  PROP_DO_LOST,
-  PROP_LAST
-};
-
-/* helper objects */
-typedef struct _GstRtpBinSession GstRtpBinSession;
-typedef struct _GstRtpBinStream GstRtpBinStream;
-typedef struct _GstRtpBinClient GstRtpBinClient;
-
-static guint gst_rtp_bin_signals[LAST_SIGNAL] = { 0 };
-
-static GstCaps *pt_map_requested (GstElement * element, guint pt,
-    GstRtpBinSession * session);
-static void free_stream (GstRtpBinStream * stream);
-
-/* Manages the RTP stream for one SSRC.
- *
- * We pipe the stream (comming from the SSRC demuxer) into a jitterbuffer.
- * If we see an SDES RTCP packet that links multiple SSRCs together based on a
- * common CNAME, we create a GstRtpBinClient structure to group the SSRCs
- * together (see below).
- */
-struct _GstRtpBinStream
-{
-  /* the SSRC of this stream */
-  guint32 ssrc;
-
-  /* parent bin */
-  GstRtpBin *bin;
-
-  /* the session this SSRC belongs to */
-  GstRtpBinSession *session;
-
-  /* the jitterbuffer of the SSRC */
-  GstElement *buffer;
-  gulong buffer_handlesync_sig;
-  gulong buffer_ptreq_sig;
-  gulong buffer_ntpstop_sig;
-
-  /* the PT demuxer of the SSRC */
-  GstElement *demux;
-  gulong demux_newpad_sig;
-  gulong demux_padremoved_sig;
-  gulong demux_ptreq_sig;
-  gulong demux_pt_change_sig;
-
-  /* if we have calculated a valid unix_delta for this stream */
-  gboolean have_sync;
-  /* mapping to local RTP and NTP time */
-  gint64 unix_delta;
-};
-
-#define GST_RTP_SESSION_LOCK(sess)   g_mutex_lock ((sess)->lock)
-#define GST_RTP_SESSION_UNLOCK(sess) g_mutex_unlock ((sess)->lock)
-
-/* Manages the receiving end of the packets.
- *
- * There is one such structure for each RTP session (audio/video/...).
- * We get the RTP/RTCP packets and stuff them into the session manager. From
- * there they are pushed into an SSRC demuxer that splits the stream based on
- * SSRC. Each of the SSRC streams go into their own jitterbuffer (managed with
- * the GstRtpBinStream above).
- */
-struct _GstRtpBinSession
-{
-  /* session id */
-  gint id;
-  /* the parent bin */
-  GstRtpBin *bin;
-  /* the session element */
-  GstElement *session;
-  /* the SSRC demuxer */
-  GstElement *demux;
-  gulong demux_newpad_sig;
-  gulong demux_padremoved_sig;
-
-  GMutex *lock;
-
-  /* list of GstRtpBinStream */
-  GSList *streams;
-
-  /* mapping of payload type to caps */
-  GHashTable *ptmap;
-
-  /* the pads of the session */
-  GstPad *recv_rtp_sink;
-  GstPad *recv_rtp_sink_ghost;
-  GstPad *recv_rtp_src;
-  GstPad *recv_rtcp_sink;
-  GstPad *recv_rtcp_sink_ghost;
-  GstPad *sync_src;
-  GstPad *send_rtp_sink;
-  GstPad *send_rtp_sink_ghost;
-  GstPad *send_rtp_src;
-  GstPad *send_rtp_src_ghost;
-  GstPad *send_rtcp_src;
-  GstPad *send_rtcp_src_ghost;
-};
-
-/* Manages the RTP streams that come from one client and should therefore be
- * synchronized.
- */
-struct _GstRtpBinClient
-{
-  /* the common CNAME for the streams */
-  gchar *cname;
-  guint cname_len;
-
-  /* the streams */
-  guint nstreams;
-  GSList *streams;
-};
-
-/* find a session with the given id. Must be called with RTP_BIN_LOCK */
-static GstRtpBinSession *
-find_session_by_id (GstRtpBin * rtpbin, gint id)
-{
-  GSList *walk;
-
-  for (walk = rtpbin->sessions; walk; walk = g_slist_next (walk)) {
-    GstRtpBinSession *sess = (GstRtpBinSession *) walk->data;
-
-    if (sess->id == id)
-      return sess;
-  }
-  return NULL;
-}
-
-/* find a session with the given request pad. Must be called with RTP_BIN_LOCK */
-static GstRtpBinSession *
-find_session_by_pad (GstRtpBin * rtpbin, GstPad * pad)
-{
-  GSList *walk;
-
-  for (walk = rtpbin->sessions; walk; walk = g_slist_next (walk)) {
-    GstRtpBinSession *sess = (GstRtpBinSession *) walk->data;
-
-    if ((sess->recv_rtp_sink_ghost == pad) ||
-        (sess->recv_rtcp_sink_ghost == pad) ||
-        (sess->send_rtp_sink_ghost == pad)
-        || (sess->send_rtcp_src_ghost == pad))
-      return sess;
-  }
-  return NULL;
-}
-
-static void
-on_new_ssrc (GstElement * session, guint32 ssrc, GstRtpBinSession * sess)
-{
-  g_signal_emit (sess->bin, gst_rtp_bin_signals[SIGNAL_ON_NEW_SSRC], 0,
-      sess->id, ssrc);
-}
-
-static void
-on_ssrc_collision (GstElement * session, guint32 ssrc, GstRtpBinSession * sess)
-{
-  g_signal_emit (sess->bin, gst_rtp_bin_signals[SIGNAL_ON_SSRC_COLLISION], 0,
-      sess->id, ssrc);
-}
-
-static void
-on_ssrc_validated (GstElement * session, guint32 ssrc, GstRtpBinSession * sess)
-{
-  g_signal_emit (sess->bin, gst_rtp_bin_signals[SIGNAL_ON_SSRC_VALIDATED], 0,
-      sess->id, ssrc);
-}
-
-static void
-on_ssrc_active (GstElement * session, guint32 ssrc, GstRtpBinSession * sess)
-{
-  g_signal_emit (sess->bin, gst_rtp_bin_signals[SIGNAL_ON_SSRC_ACTIVE], 0,
-      sess->id, ssrc);
-}
-
-static void
-on_ssrc_sdes (GstElement * session, guint32 ssrc, GstRtpBinSession * sess)
-{
-  g_signal_emit (sess->bin, gst_rtp_bin_signals[SIGNAL_ON_SSRC_SDES], 0,
-      sess->id, ssrc);
-}
-
-static void
-on_bye_ssrc (GstElement * session, guint32 ssrc, GstRtpBinSession * sess)
-{
-  g_signal_emit (sess->bin, gst_rtp_bin_signals[SIGNAL_ON_BYE_SSRC], 0,
-      sess->id, ssrc);
-}
-
-static void
-on_bye_timeout (GstElement * session, guint32 ssrc, GstRtpBinSession * sess)
-{
-  g_signal_emit (sess->bin, gst_rtp_bin_signals[SIGNAL_ON_BYE_TIMEOUT], 0,
-      sess->id, ssrc);
-}
-
-static void
-on_timeout (GstElement * session, guint32 ssrc, GstRtpBinSession * sess)
-{
-  g_signal_emit (sess->bin, gst_rtp_bin_signals[SIGNAL_ON_TIMEOUT], 0,
-      sess->id, ssrc);
-}
-
-static void
-on_sender_timeout (GstElement * session, guint32 ssrc, GstRtpBinSession * sess)
-{
-  g_signal_emit (sess->bin, gst_rtp_bin_signals[SIGNAL_ON_SENDER_TIMEOUT], 0,
-      sess->id, ssrc);
-}
-
-static void
-on_npt_stop (GstElement * jbuf, GstRtpBinStream * stream)
-{
-  g_signal_emit (stream->bin, gst_rtp_bin_signals[SIGNAL_ON_NPT_STOP], 0,
-      stream->session->id, stream->ssrc);
-}
-
-/* must be called with the SESSION lock */
-static GstRtpBinStream *
-find_stream_by_ssrc (GstRtpBinSession * session, guint32 ssrc)
-{
-  GSList *walk;
-
-  for (walk = session->streams; walk; walk = g_slist_next (walk)) {
-    GstRtpBinStream *stream = (GstRtpBinStream *) walk->data;
-
-    if (stream->ssrc == ssrc)
-      return stream;
-  }
-  return NULL;
-}
-
-static void
-ssrc_demux_pad_removed (GstElement * element, guint ssrc, GstPad * pad,
-    GstRtpBinSession * session)
-{
-  GstRtpBinStream *stream = NULL;
-
-  GST_RTP_SESSION_LOCK (session);
-  if ((stream = find_stream_by_ssrc (session, ssrc)))
-    session->streams = g_slist_remove (session->streams, stream);
-  GST_RTP_SESSION_UNLOCK (session);
-
-  if (stream)
-    free_stream (stream);
-}
-
-/* create a session with the given id.  Must be called with RTP_BIN_LOCK */
-static GstRtpBinSession *
-create_session (GstRtpBin * rtpbin, gint id)
-{
-  GstRtpBinSession *sess;
-  GstElement *session, *demux;
-  GstState target;
-
-  if (!(session = gst_element_factory_make ("gstrtpsession", NULL)))
-    goto no_session;
-
-  if (!(demux = gst_element_factory_make ("gstrtpssrcdemux", NULL)))
-    goto no_demux;
-
-  sess = g_new0 (GstRtpBinSession, 1);
-  sess->lock = g_mutex_new ();
-  sess->id = id;
-  sess->bin = rtpbin;
-  sess->session = session;
-  sess->demux = demux;
-  sess->ptmap = g_hash_table_new_full (NULL, NULL, NULL,
-      (GDestroyNotify) gst_caps_unref);
-  rtpbin->sessions = g_slist_prepend (rtpbin->sessions, sess);
-
-  /* set NTP base or new session */
-  g_object_set (session, "ntp-ns-base", rtpbin->priv->ntp_ns_base, NULL);
-  /* configure SDES items */
-  GST_OBJECT_LOCK (rtpbin);
-  g_object_set (session, "sdes", rtpbin->sdes, NULL);
-  GST_OBJECT_UNLOCK (rtpbin);
-
-  /* provide clock_rate to the session manager when needed */
-  g_signal_connect (session, "request-pt-map",
-      (GCallback) pt_map_requested, sess);
-
-  g_signal_connect (sess->session, "on-new-ssrc",
-      (GCallback) on_new_ssrc, sess);
-  g_signal_connect (sess->session, "on-ssrc-collision",
-      (GCallback) on_ssrc_collision, sess);
-  g_signal_connect (sess->session, "on-ssrc-validated",
-      (GCallback) on_ssrc_validated, sess);
-  g_signal_connect (sess->session, "on-ssrc-active",
-      (GCallback) on_ssrc_active, sess);
-  g_signal_connect (sess->session, "on-ssrc-sdes",
-      (GCallback) on_ssrc_sdes, sess);
-  g_signal_connect (sess->session, "on-bye-ssrc",
-      (GCallback) on_bye_ssrc, sess);
-  g_signal_connect (sess->session, "on-bye-timeout",
-      (GCallback) on_bye_timeout, sess);
-  g_signal_connect (sess->session, "on-timeout", (GCallback) on_timeout, sess);
-  g_signal_connect (sess->session, "on-sender-timeout",
-      (GCallback) on_sender_timeout, sess);
-
-  gst_bin_add (GST_BIN_CAST (rtpbin), session);
-  gst_bin_add (GST_BIN_CAST (rtpbin), demux);
-
-  GST_OBJECT_LOCK (rtpbin);
-  target = GST_STATE_TARGET (rtpbin);
-  GST_OBJECT_UNLOCK (rtpbin);
-
-  /* change state only to what's needed */
-  gst_element_set_state (demux, target);
-  gst_element_set_state (session, target);
-
-  return sess;
-
-  /* ERRORS */
-no_session:
-  {
-    g_warning ("gstrtpbin: could not create gstrtpsession element");
-    return NULL;
-  }
-no_demux:
-  {
-    gst_object_unref (session);
-    g_warning ("gstrtpbin: could not create gstrtpssrcdemux element");
-    return NULL;
-  }
-}
-
-static void
-free_session (GstRtpBinSession * sess, GstRtpBin * bin)
-{
-  GST_DEBUG_OBJECT (bin, "freeing session %p", sess);
-
-  gst_element_set_state (sess->demux, GST_STATE_NULL);
-  gst_element_set_state (sess->session, GST_STATE_NULL);
-
-  if (sess->recv_rtp_sink != NULL) {
-    gst_element_release_request_pad (sess->session, sess->recv_rtp_sink);
-    gst_object_unref (sess->recv_rtp_sink);
-  }
-  if (sess->recv_rtp_src != NULL)
-    gst_object_unref (sess->recv_rtp_src);
-  if (sess->recv_rtcp_sink != NULL) {
-    gst_element_release_request_pad (sess->session, sess->recv_rtcp_sink);
-    gst_object_unref (sess->recv_rtcp_sink);
-  }
-  if (sess->sync_src != NULL)
-    gst_object_unref (sess->sync_src);
-  if (sess->send_rtp_sink != NULL) {
-    gst_element_release_request_pad (sess->session, sess->send_rtp_sink);
-    gst_object_unref (sess->send_rtp_sink);
-  }
-  if (sess->send_rtp_src != NULL)
-    gst_object_unref (sess->send_rtp_src);
-  if (sess->send_rtcp_src != NULL) {
-    gst_element_release_request_pad (sess->session, sess->send_rtcp_src);
-    gst_object_unref (sess->send_rtcp_src);
-  }
-
-  gst_bin_remove (GST_BIN_CAST (bin), sess->session);
-  gst_bin_remove (GST_BIN_CAST (bin), sess->demux);
-
-  g_slist_foreach (sess->streams, (GFunc) free_stream, NULL);
-  g_slist_free (sess->streams);
-
-  g_mutex_free (sess->lock);
-  g_hash_table_destroy (sess->ptmap);
-
-  g_free (sess);
-}
-
-/* get the payload type caps for the specific payload @pt in @session */
-static GstCaps *
-get_pt_map (GstRtpBinSession * session, guint pt)
-{
-  GstCaps *caps = NULL;
-  GstRtpBin *bin;
-  GValue ret = { 0 };
-  GValue args[3] = { {0}, {0}, {0} };
-
-  GST_DEBUG ("searching pt %d in cache", pt);
-
-  GST_RTP_SESSION_LOCK (session);
-
-  /* first look in the cache */
-  caps = g_hash_table_lookup (session->ptmap, GINT_TO_POINTER (pt));
-  if (caps) {
-    gst_caps_ref (caps);
-    goto done;
-  }
-
-  bin = session->bin;
-
-  GST_DEBUG ("emiting signal for pt %d in session %d", pt, session->id);
-
-  /* not in cache, send signal to request caps */
-  g_value_init (&args[0], GST_TYPE_ELEMENT);
-  g_value_set_object (&args[0], bin);
-  g_value_init (&args[1], G_TYPE_UINT);
-  g_value_set_uint (&args[1], session->id);
-  g_value_init (&args[2], G_TYPE_UINT);
-  g_value_set_uint (&args[2], pt);
-
-  g_value_init (&ret, GST_TYPE_CAPS);
-  g_value_set_boxed (&ret, NULL);
-
-  GST_RTP_SESSION_UNLOCK (session);
-
-  g_signal_emitv (args, gst_rtp_bin_signals[SIGNAL_REQUEST_PT_MAP], 0, &ret);
-
-  GST_RTP_SESSION_LOCK (session);
-
-  g_value_unset (&args[0]);
-  g_value_unset (&args[1]);
-  g_value_unset (&args[2]);
-
-  /* look in the cache again because we let the lock go */
-  caps = g_hash_table_lookup (session->ptmap, GINT_TO_POINTER (pt));
-  if (caps) {
-    gst_caps_ref (caps);
-    g_value_unset (&ret);
-    goto done;
-  }
-
-  caps = (GstCaps *) g_value_dup_boxed (&ret);
-  g_value_unset (&ret);
-  if (!caps)
-    goto no_caps;
-
-  GST_DEBUG ("caching pt %d as %" GST_PTR_FORMAT, pt, caps);
-
-  /* store in cache, take additional ref */
-  g_hash_table_insert (session->ptmap, GINT_TO_POINTER (pt),
-      gst_caps_ref (caps));
-
-done:
-  GST_RTP_SESSION_UNLOCK (session);
-
-  return caps;
-
-  /* ERRORS */
-no_caps:
-  {
-    GST_RTP_SESSION_UNLOCK (session);
-    GST_DEBUG ("no pt map could be obtained");
-    return NULL;
-  }
-}
-
-static gboolean
-return_true (gpointer key, gpointer value, gpointer user_data)
-{
-  return TRUE;
-}
-
-static void
-gst_rtp_bin_reset_sync (GstRtpBin * rtpbin)
-{
-  GSList *clients, *streams;
-
-  GST_DEBUG_OBJECT (rtpbin, "Reset sync on all clients");
-
-  GST_RTP_BIN_LOCK (rtpbin);
-  for (clients = rtpbin->clients; clients; clients = g_slist_next (clients)) {
-    GstRtpBinClient *client = (GstRtpBinClient *) clients->data;
-
-    /* reset sync on all streams for this client */
-    for (streams = client->streams; streams; streams = g_slist_next (streams)) {
-      GstRtpBinStream *stream = (GstRtpBinStream *) streams->data;
-
-      /* make use require a new SR packet for this stream before we attempt new
-       * lip-sync */
-      stream->have_sync = FALSE;
-      stream->unix_delta = 0;
-    }
-  }
-  GST_RTP_BIN_UNLOCK (rtpbin);
-}
-
-static void
-gst_rtp_bin_clear_pt_map (GstRtpBin * bin)
-{
-  GSList *sessions, *streams;
-
-  GST_RTP_BIN_LOCK (bin);
-  GST_DEBUG_OBJECT (bin, "clearing pt map");
-  for (sessions = bin->sessions; sessions; sessions = g_slist_next (sessions)) {
-    GstRtpBinSession *session = (GstRtpBinSession *) sessions->data;
-
-    GST_DEBUG_OBJECT (bin, "clearing session %p", session);
-    g_signal_emit_by_name (session->session, "clear-pt-map", NULL);
-
-    GST_RTP_SESSION_LOCK (session);
-    g_hash_table_foreach_remove (session->ptmap, return_true, NULL);
-
-    for (streams = session->streams; streams; streams = g_slist_next (streams)) {
-      GstRtpBinStream *stream = (GstRtpBinStream *) streams->data;
-
-      GST_DEBUG_OBJECT (bin, "clearing stream %p", stream);
-      g_signal_emit_by_name (stream->buffer, "clear-pt-map", NULL);
-      g_signal_emit_by_name (stream->demux, "clear-pt-map", NULL);
-    }
-    GST_RTP_SESSION_UNLOCK (session);
-  }
-  GST_RTP_BIN_UNLOCK (bin);
-
-  /* reset sync too */
-  gst_rtp_bin_reset_sync (bin);
-}
-
-static RTPSession *
-gst_rtp_bin_get_internal_session (GstRtpBin * bin, guint session_id)
-{
-  RTPSession *internal_session = NULL;
-  GstRtpBinSession *session;
-
-  GST_RTP_BIN_LOCK (bin);
-  GST_DEBUG_OBJECT (bin, "retrieving internal RTPSession object, index: %d",
-      session_id);
-  session = find_session_by_id (bin, (gint) session_id);
-  if (session) {
-    g_object_get (session->session, "internal-session", &internal_session,
-        NULL);
-  }
-  GST_RTP_BIN_UNLOCK (bin);
-
-  return internal_session;
-}
-
-static void
-gst_rtp_bin_propagate_property_to_jitterbuffer (GstRtpBin * bin,
-    const gchar * name, const GValue * value)
-{
-  GSList *sessions, *streams;
-
-  GST_RTP_BIN_LOCK (bin);
-  for (sessions = bin->sessions; sessions; sessions = g_slist_next (sessions)) {
-    GstRtpBinSession *session = (GstRtpBinSession *) sessions->data;
-
-    GST_RTP_SESSION_LOCK (session);
-    for (streams = session->streams; streams; streams = g_slist_next (streams)) {
-      GstRtpBinStream *stream = (GstRtpBinStream *) streams->data;
-
-      g_object_set_property (G_OBJECT (stream->buffer), name, value);
-    }
-    GST_RTP_SESSION_UNLOCK (session);
-  }
-  GST_RTP_BIN_UNLOCK (bin);
-}
-
-/* get a client with the given SDES name. Must be called with RTP_BIN_LOCK */
-static GstRtpBinClient *
-get_client (GstRtpBin * bin, guint8 len, guint8 * data, gboolean * created)
-{
-  GstRtpBinClient *result = NULL;
-  GSList *walk;
-
-  for (walk = bin->clients; walk; walk = g_slist_next (walk)) {
-    GstRtpBinClient *client = (GstRtpBinClient *) walk->data;
-
-    if (len != client->cname_len)
-      continue;
-
-    if (!strncmp ((gchar *) data, client->cname, client->cname_len)) {
-      GST_DEBUG_OBJECT (bin, "found existing client %p with CNAME %s", client,
-          client->cname);
-      result = client;
-      break;
-    }
-  }
-
-  /* nothing found, create one */
-  if (result == NULL) {
-    result = g_new0 (GstRtpBinClient, 1);
-    result->cname = g_strndup ((gchar *) data, len);
-    result->cname_len = len;
-    bin->clients = g_slist_prepend (bin->clients, result);
-    GST_DEBUG_OBJECT (bin, "created new client %p with CNAME %s", result,
-        result->cname);
-  }
-  return result;
-}
-
-static void
-free_client (GstRtpBinClient * client, GstRtpBin * bin)
-{
-  GST_DEBUG_OBJECT (bin, "freeing client %p", client);
-  g_slist_free (client->streams);
-  g_free (client->cname);
-  g_free (client);
-}
-
-/* associate a stream to the given CNAME. This will make sure all streams for
- * that CNAME are synchronized together.
- * Must be called with GST_RTP_BIN_LOCK */
-static void
-gst_rtp_bin_associate (GstRtpBin * bin, GstRtpBinStream * stream, guint8 len,
-    guint8 * data, guint64 last_unix, guint64 last_extrtptime,
-    guint64 clock_base, guint64 clock_base_time, guint clock_rate)
-{
-  GstRtpBinClient *client;
-  gboolean created;
-  GSList *walk;
-  guint64 local_unix;
-  guint64 local_rtp;
-
-  /* first find or create the CNAME */
-  client = get_client (bin, len, data, &created);
-
-  /* find stream in the client */
-  for (walk = client->streams; walk; walk = g_slist_next (walk)) {
-    GstRtpBinStream *ostream = (GstRtpBinStream *) walk->data;
-
-    if (ostream == stream)
-      break;
-  }
-  /* not found, add it to the list */
-  if (walk == NULL) {
-    GST_DEBUG_OBJECT (bin,
-        "new association of SSRC %08x with client %p with CNAME %s",
-        stream->ssrc, client, client->cname);
-    client->streams = g_slist_prepend (client->streams, stream);
-    client->nstreams++;
-  } else {
-    GST_DEBUG_OBJECT (bin,
-        "found association of SSRC %08x with client %p with CNAME %s",
-        stream->ssrc, client, client->cname);
-  }
-
-  /* take the extended rtptime we found in the SR packet and map it to the
-   * local rtptime. The local rtp time is used to construct timestamps on the
-   * buffers. */
-  local_rtp = last_extrtptime - clock_base;
-
-  GST_DEBUG_OBJECT (bin,
-      "base %" G_GUINT64_FORMAT ", extrtptime %" G_GUINT64_FORMAT
-      ", local RTP %" G_GUINT64_FORMAT ", clock-rate %d", clock_base,
-      last_extrtptime, local_rtp, clock_rate);
-
-  /* calculate local NTP time in gstreamer timestamp, we essentially perform the
-   * same conversion that a jitterbuffer would use to convert an rtp timestamp
-   * into a corresponding gstreamer timestamp. */
-  local_unix = gst_util_uint64_scale_int (local_rtp, GST_SECOND, clock_rate);
-  local_unix += clock_base_time;
-
-  /* calculate delta between server and receiver. last_unix is created by
-   * converting the ntptime in the last SR packet to a gstreamer timestamp. This
-   * delta expresses the difference to our timeline and the server timeline. */
-  stream->unix_delta = last_unix - local_unix;
-  stream->have_sync = TRUE;
-
-  GST_DEBUG_OBJECT (bin,
-      "local UNIX %" G_GUINT64_FORMAT ", remote UNIX %" G_GUINT64_FORMAT
-      ", delta %" G_GINT64_FORMAT, local_unix, last_unix, stream->unix_delta);
-
-  /* recalc inter stream playout offset, but only if there is more than one
-   * stream. */
-  if (client->nstreams > 1) {
-    gint64 min;
-
-    /* calculate the min of all deltas, ignoring streams that did not yet have a
-     * valid unix_delta because we did not yet receive an SR packet for those
-     * streams. 
-     * We calculate the mininum because we would like to only apply positive
-     * offsets to streams, delaying their playback instead of trying to speed up
-     * other streams (which might be imposible when we have to create negative
-     * latencies).
-     * The stream that has the smallest diff is selected as the reference stream,
-     * all other streams will have a positive offset to this difference. */
-    min = G_MAXINT64;
-    for (walk = client->streams; walk; walk = g_slist_next (walk)) {
-      GstRtpBinStream *ostream = (GstRtpBinStream *) walk->data;
-
-      if (!ostream->have_sync)
-        continue;
-
-      if (ostream->unix_delta < min)
-        min = ostream->unix_delta;
-    }
-
-    GST_DEBUG_OBJECT (bin, "client %p min delta %" G_GINT64_FORMAT, client,
-        min);
-
-    /* calculate offsets for each stream */
-    for (walk = client->streams; walk; walk = g_slist_next (walk)) {
-      GstRtpBinStream *ostream = (GstRtpBinStream *) walk->data;
-      gint64 ts_offset, prev_ts_offset;
-
-      /* ignore streams for which we didn't receive an SR packet yet, we
-       * can't synchronize them yet. We can however sync other streams just
-       * fine. */
-      if (!ostream->have_sync)
-        continue;
-
-      /* calculate offset to our reference stream, this should always give a
-       * positive number. */
-      ts_offset = ostream->unix_delta - min;
-
-      g_object_get (ostream->buffer, "ts-offset", &prev_ts_offset, NULL);
-
-      /* delta changed, see how much */
-      if (prev_ts_offset != ts_offset) {
-        gint64 diff;
-
-        if (prev_ts_offset > ts_offset)
-          diff = prev_ts_offset - ts_offset;
-        else
-          diff = ts_offset - prev_ts_offset;
-
-        GST_DEBUG_OBJECT (bin,
-            "ts-offset %" G_GUINT64_FORMAT ", prev %" G_GUINT64_FORMAT
-            ", diff: %" G_GINT64_FORMAT, ts_offset, prev_ts_offset, diff);
-
-        /* only change diff when it changed more than 4 milliseconds. This
-         * compensates for rounding errors in NTP to RTP timestamp
-         * conversions */
-        if (diff > 4 * GST_MSECOND && diff < (3 * GST_SECOND)) {
-          g_object_set (ostream->buffer, "ts-offset", ts_offset, NULL);
-        }
-      }
-      GST_DEBUG_OBJECT (bin, "stream SSRC %08x, delta %" G_GINT64_FORMAT,
-          ostream->ssrc, ts_offset);
-    }
-  }
-  return;
-}
-
-#define GST_RTCP_BUFFER_FOR_PACKETS(b,buffer,packet) \
-  for ((b) = gst_rtcp_buffer_get_first_packet ((buffer), (packet)); (b); \
-          (b) = gst_rtcp_packet_move_to_next ((packet)))
-
-#define GST_RTCP_SDES_FOR_ITEMS(b,packet) \
-  for ((b) = gst_rtcp_packet_sdes_first_item ((packet)); (b); \
-          (b) = gst_rtcp_packet_sdes_next_item ((packet)))
-
-#define GST_RTCP_SDES_FOR_ENTRIES(b,packet) \
-  for ((b) = gst_rtcp_packet_sdes_first_entry ((packet)); (b); \
-          (b) = gst_rtcp_packet_sdes_next_entry ((packet)))
-
-static void
-gst_rtp_bin_handle_sync (GstElement * jitterbuffer, GstStructure * s,
-    GstRtpBinStream * stream)
-{
-  GstRtpBin *bin;
-  GstRTCPPacket packet;
-  guint32 ssrc;
-  guint64 ntptime;
-  gboolean have_sr, have_sdes;
-  gboolean more;
-  guint64 clock_base;
-  guint64 clock_base_time;
-  guint clock_rate;
-  guint64 extrtptime;
-  GstBuffer *buffer;
-
-  bin = stream->bin;
-
-  GST_DEBUG_OBJECT (bin, "sync handler called");
-
-  /* get the last relation between the rtp timestamps and the gstreamer
-   * timestamps. We get this info directly from the jitterbuffer which
-   * constructs gstreamer timestamps from rtp timestamps and so it know exactly
-   * what the current situation is. */
-  clock_base = g_value_get_uint64 (gst_structure_get_value (s, "base-rtptime"));
-  clock_base_time =
-      g_value_get_uint64 (gst_structure_get_value (s, "base-time"));
-  clock_rate = g_value_get_uint (gst_structure_get_value (s, "clock-rate"));
-  extrtptime =
-      g_value_get_uint64 (gst_structure_get_value (s, "sr-ext-rtptime"));
-  buffer = gst_value_get_buffer (gst_structure_get_value (s, "sr-buffer"));
-
-  have_sr = FALSE;
-  have_sdes = FALSE;
-  GST_RTCP_BUFFER_FOR_PACKETS (more, buffer, &packet) {
-    /* first packet must be SR or RR or else the validate would have failed */
-    switch (gst_rtcp_packet_get_type (&packet)) {
-      case GST_RTCP_TYPE_SR:
-        /* only parse first. There is only supposed to be one SR in the packet
-         * but we will deal with malformed packets gracefully */
-        if (have_sr)
-          break;
-        /* get NTP and RTP times */
-        gst_rtcp_packet_sr_get_sender_info (&packet, &ssrc, &ntptime, NULL,
-            NULL, NULL);
-
-        GST_DEBUG_OBJECT (bin, "received sync packet from SSRC %08x", ssrc);
-        /* ignore SR that is not ours */
-        if (ssrc != stream->ssrc)
-          continue;
-
-        have_sr = TRUE;
-        break;
-      case GST_RTCP_TYPE_SDES:
-      {
-        gboolean more_items, more_entries;
-
-        /* only deal with first SDES, there is only supposed to be one SDES in
-         * the RTCP packet but we deal with bad packets gracefully. Also bail
-         * out if we have not seen an SR item yet. */
-        if (have_sdes || !have_sr)
-          break;
-
-        GST_RTCP_SDES_FOR_ITEMS (more_items, &packet) {
-          /* skip items that are not about the SSRC of the sender */
-          if (gst_rtcp_packet_sdes_get_ssrc (&packet) != ssrc)
-            continue;
-
-          /* find the CNAME entry */
-          GST_RTCP_SDES_FOR_ENTRIES (more_entries, &packet) {
-            GstRTCPSDESType type;
-            guint8 len;
-            guint8 *data;
-
-            gst_rtcp_packet_sdes_get_entry (&packet, &type, &len, &data);
-
-            if (type == GST_RTCP_SDES_CNAME) {
-              GST_RTP_BIN_LOCK (bin);
-              /* associate the stream to CNAME */
-              gst_rtp_bin_associate (bin, stream, len, data,
-                  gst_rtcp_ntp_to_unix (ntptime), extrtptime,
-                  clock_base, clock_base_time, clock_rate);
-              GST_RTP_BIN_UNLOCK (bin);
-            }
-          }
-        }
-        have_sdes = TRUE;
-        break;
-      }
-      default:
-        /* we can ignore these packets */
-        break;
-    }
-  }
-}
-
-/* create a new stream with @ssrc in @session. Must be called with
- * RTP_SESSION_LOCK. */
-static GstRtpBinStream *
-create_stream (GstRtpBinSession * session, guint32 ssrc)
-{
-  GstElement *buffer, *demux;
-  GstRtpBinStream *stream;
-  GstRtpBin *rtpbin;
-  GstState target;
-
-  if (!(buffer = gst_element_factory_make ("gstrtpjitterbuffer", NULL)))
-    goto no_jitterbuffer;
-
-  if (!(demux = gst_element_factory_make ("gstrtpptdemux", NULL)))
-    goto no_demux;
-
-  rtpbin = session->bin;
-
-  stream = g_new0 (GstRtpBinStream, 1);
-  stream->ssrc = ssrc;
-  stream->bin = rtpbin;
-  stream->session = session;
-  stream->buffer = buffer;
-  stream->demux = demux;
-  stream->have_sync = FALSE;
-  stream->unix_delta = 0;
-  session->streams = g_slist_prepend (session->streams, stream);
-
-  /* provide clock_rate to the jitterbuffer when needed */
-  stream->buffer_ptreq_sig = g_signal_connect (buffer, "request-pt-map",
-      (GCallback) pt_map_requested, session);
-  stream->buffer_ntpstop_sig = g_signal_connect (buffer, "on-npt-stop",
-      (GCallback) on_npt_stop, stream);
-
-  /* configure latency and packet lost */
-  g_object_set (buffer, "latency", rtpbin->latency, NULL);
-  g_object_set (buffer, "do-lost", rtpbin->do_lost, NULL);
-
-  gst_bin_add (GST_BIN_CAST (rtpbin), demux);
-  gst_bin_add (GST_BIN_CAST (rtpbin), buffer);
-
-  /* link stuff */
-  gst_element_link (buffer, demux);
-
-  GST_OBJECT_LOCK (rtpbin);
-  target = GST_STATE_TARGET (rtpbin);
-  GST_OBJECT_UNLOCK (rtpbin);
-
-  /* from sink to source */
-  gst_element_set_state (demux, target);
-  gst_element_set_state (buffer, target);
-
-  return stream;
-
-  /* ERRORS */
-no_jitterbuffer:
-  {
-    g_warning ("gstrtpbin: could not create gstrtpjitterbuffer element");
-    return NULL;
-  }
-no_demux:
-  {
-    gst_object_unref (buffer);
-    g_warning ("gstrtpbin: could not create gstrtpptdemux element");
-    return NULL;
-  }
-}
-
-static void
-free_stream (GstRtpBinStream * stream)
-{
-  GstRtpBinSession *session;
-
-  session = stream->session;
-
-  g_signal_handler_disconnect (stream->demux, stream->demux_newpad_sig);
-  g_signal_handler_disconnect (stream->demux, stream->demux_ptreq_sig);
-  g_signal_handler_disconnect (stream->buffer, stream->buffer_handlesync_sig);
-  g_signal_handler_disconnect (stream->buffer, stream->buffer_ptreq_sig);
-  g_signal_handler_disconnect (stream->buffer, stream->buffer_ntpstop_sig);
-
-  gst_element_set_state (stream->demux, GST_STATE_NULL);
-  gst_element_set_state (stream->buffer, GST_STATE_NULL);
-
-  /* now remove this signal, we need this while going to NULL because it to 
-   * do some cleanups */
-  g_signal_handler_disconnect (stream->demux, stream->demux_padremoved_sig);
-
-  gst_bin_remove (GST_BIN_CAST (session->bin), stream->buffer);
-  gst_bin_remove (GST_BIN_CAST (session->bin), stream->demux);
-
-  g_free (stream);
-}
-
-/* GObject vmethods */
-static void gst_rtp_bin_dispose (GObject * object);
-static void gst_rtp_bin_finalize (GObject * object);
-static void gst_rtp_bin_set_property (GObject * object, guint prop_id,
-    const GValue * value, GParamSpec * pspec);
-static void gst_rtp_bin_get_property (GObject * object, guint prop_id,
-    GValue * value, GParamSpec * pspec);
-
-/* GstElement vmethods */
-static GstStateChangeReturn gst_rtp_bin_change_state (GstElement * element,
-    GstStateChange transition);
-static GstPad *gst_rtp_bin_request_new_pad (GstElement * element,
-    GstPadTemplate * templ, const gchar * name);
-static void gst_rtp_bin_release_pad (GstElement * element, GstPad * pad);
-static void gst_rtp_bin_handle_message (GstBin * bin, GstMessage * message);
-static void gst_rtp_bin_clear_pt_map (GstRtpBin * bin);
-
-GST_BOILERPLATE (GstRtpBin, gst_rtp_bin, GstBin, GST_TYPE_BIN);
-
-static void
-gst_rtp_bin_base_init (gpointer klass)
-{
-  GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
-
-  /* sink pads */
-  gst_element_class_add_pad_template (element_class,
-      gst_static_pad_template_get (&rtpbin_recv_rtp_sink_template));
-  gst_element_class_add_pad_template (element_class,
-      gst_static_pad_template_get (&rtpbin_recv_rtcp_sink_template));
-  gst_element_class_add_pad_template (element_class,
-      gst_static_pad_template_get (&rtpbin_send_rtp_sink_template));
-
-  /* src pads */
-  gst_element_class_add_pad_template (element_class,
-      gst_static_pad_template_get (&rtpbin_recv_rtp_src_template));
-  gst_element_class_add_pad_template (element_class,
-      gst_static_pad_template_get (&rtpbin_send_rtcp_src_template));
-  gst_element_class_add_pad_template (element_class,
-      gst_static_pad_template_get (&rtpbin_send_rtp_src_template));
-
-  gst_element_class_set_details (element_class, &rtpbin_details);
-}
-
-static void
-gst_rtp_bin_class_init (GstRtpBinClass * klass)
-{
-  GObjectClass *gobject_class;
-  GstElementClass *gstelement_class;
-  GstBinClass *gstbin_class;
-
-  gobject_class = (GObjectClass *) klass;
-  gstelement_class = (GstElementClass *) klass;
-  gstbin_class = (GstBinClass *) klass;
-
-  g_type_class_add_private (klass, sizeof (GstRtpBinPrivate));
-
-  gobject_class->dispose = gst_rtp_bin_dispose;
-  gobject_class->finalize = gst_rtp_bin_finalize;
-  gobject_class->set_property = gst_rtp_bin_set_property;
-  gobject_class->get_property = gst_rtp_bin_get_property;
-
-  g_object_class_install_property (gobject_class, PROP_LATENCY,
-      g_param_spec_uint ("latency", "Buffer latency in ms",
-          "Default amount of ms to buffer in the jitterbuffers", 0,
-          G_MAXUINT, DEFAULT_LATENCY_MS, G_PARAM_READWRITE));
-
-  /**
-   * GstRtpBin::request-pt-map:
-   * @rtpbin: the object which received the signal
-   * @session: the session
-   * @pt: the pt
-   *
-   * Request the payload type as #GstCaps for @pt in @session.
-   */
-  gst_rtp_bin_signals[SIGNAL_REQUEST_PT_MAP] =
-      g_signal_new ("request-pt-map", G_TYPE_FROM_CLASS (klass),
-      G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass, request_pt_map),
-      NULL, NULL, gst_rtp_bin_marshal_BOXED__UINT_UINT, GST_TYPE_CAPS, 2,
-      G_TYPE_UINT, G_TYPE_UINT);
-  /**
-   * GstRtpBin::clear-pt-map:
-   * @rtpbin: the object which received the signal
-   *
-   * Clear all previously cached pt-mapping obtained with
-   * #GstRtpBin::request-pt-map.
-   */
-  gst_rtp_bin_signals[SIGNAL_CLEAR_PT_MAP] =
-      g_signal_new ("clear-pt-map", G_TYPE_FROM_CLASS (klass),
-      G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION, G_STRUCT_OFFSET (GstRtpBinClass,
-          clear_pt_map), NULL, NULL, g_cclosure_marshal_VOID__VOID, G_TYPE_NONE,
-      0, G_TYPE_NONE);
-  /**
-   * GstRtpBin::reset-sync:
-   * @rtpbin: the object which received the signal
-   *
-   * Reset all currently configured lip-sync parameters and require new SR
-   * packets for all streams before lip-sync is attempted again.
-   */
-  gst_rtp_bin_signals[SIGNAL_RESET_SYNC] =
-      g_signal_new ("reset-sync", G_TYPE_FROM_CLASS (klass),
-      G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION, G_STRUCT_OFFSET (GstRtpBinClass,
-          reset_sync), NULL, NULL, g_cclosure_marshal_VOID__VOID, G_TYPE_NONE,
-      0, G_TYPE_NONE);
-
-  /**
-   * GstRtpBin::get-internal-session:
-   * @rtpbin: the object which received the signal
-   * @id: the session id
-   *
-   * Request the internal RTPSession object as #GObject in session @id.
-   */
-  gst_rtp_bin_signals[SIGNAL_GET_INTERNAL_SESSION] =
-      g_signal_new ("get-internal-session", G_TYPE_FROM_CLASS (klass),
-      G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION, G_STRUCT_OFFSET (GstRtpBinClass,
-          get_internal_session), NULL, NULL, gst_rtp_bin_marshal_OBJECT__UINT,
-      RTP_TYPE_SESSION, 1, G_TYPE_UINT);
-
-  /**
-   * GstRtpBin::on-new-ssrc:
-   * @rtpbin: the object which received the signal
-   * @session: the session
-   * @ssrc: the SSRC 
-   *
-   * Notify of a new SSRC that entered @session.
-   */
-  gst_rtp_bin_signals[SIGNAL_ON_NEW_SSRC] =
-      g_signal_new ("on-new-ssrc", G_TYPE_FROM_CLASS (klass),
-      G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass, on_new_ssrc),
-      NULL, NULL, gst_rtp_bin_marshal_VOID__UINT_UINT, G_TYPE_NONE, 2,
-      G_TYPE_UINT, G_TYPE_UINT);
-  /**
-   * GstRtpBin::on-ssrc-collision:
-   * @rtpbin: the object which received the signal
-   * @session: the session
-   * @ssrc: the SSRC 
-   *
-   * Notify when we have an SSRC collision
-   */
-  gst_rtp_bin_signals[SIGNAL_ON_SSRC_COLLISION] =
-      g_signal_new ("on-ssrc-collision", G_TYPE_FROM_CLASS (klass),
-      G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass, on_ssrc_collision),
-      NULL, NULL, gst_rtp_bin_marshal_VOID__UINT_UINT, G_TYPE_NONE, 2,
-      G_TYPE_UINT, G_TYPE_UINT);
-  /**
-   * GstRtpBin::on-ssrc-validated:
-   * @rtpbin: the object which received the signal
-   * @session: the session
-   * @ssrc: the SSRC 
-   *
-   * Notify of a new SSRC that became validated.
-   */
-  gst_rtp_bin_signals[SIGNAL_ON_SSRC_VALIDATED] =
-      g_signal_new ("on-ssrc-validated", G_TYPE_FROM_CLASS (klass),
-      G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass, on_ssrc_validated),
-      NULL, NULL, gst_rtp_bin_marshal_VOID__UINT_UINT, G_TYPE_NONE, 2,
-      G_TYPE_UINT, G_TYPE_UINT);
-  /**
-   * GstRtpBin::on-ssrc-active:
-   * @rtpbin: the object which received the signal
-   * @session: the session
-   * @ssrc: the SSRC
-   *
-   * Notify of a SSRC that is active, i.e., sending RTCP.
-   */
-  gst_rtp_bin_signals[SIGNAL_ON_SSRC_ACTIVE] =
-      g_signal_new ("on-ssrc-active", G_TYPE_FROM_CLASS (klass),
-      G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass, on_ssrc_active),
-      NULL, NULL, gst_rtp_bin_marshal_VOID__UINT_UINT, G_TYPE_NONE, 2,
-      G_TYPE_UINT, G_TYPE_UINT);
-  /**
-   * GstRtpBin::on-ssrc-sdes:
-   * @rtpbin: the object which received the signal
-   * @session: the session
-   * @ssrc: the SSRC
-   *
-   * Notify of a SSRC that is active, i.e., sending RTCP.
-   */
-  gst_rtp_bin_signals[SIGNAL_ON_SSRC_SDES] =
-      g_signal_new ("on-ssrc-sdes", G_TYPE_FROM_CLASS (klass),
-      G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass, on_ssrc_sdes),
-      NULL, NULL, gst_rtp_bin_marshal_VOID__UINT_UINT, G_TYPE_NONE, 2,
-      G_TYPE_UINT, G_TYPE_UINT);
-
-  /**
-   * GstRtpBin::on-bye-ssrc:
-   * @rtpbin: the object which received the signal
-   * @session: the session
-   * @ssrc: the SSRC 
-   *
-   * Notify of an SSRC that became inactive because of a BYE packet.
-   */
-  gst_rtp_bin_signals[SIGNAL_ON_BYE_SSRC] =
-      g_signal_new ("on-bye-ssrc", G_TYPE_FROM_CLASS (klass),
-      G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass, on_bye_ssrc),
-      NULL, NULL, gst_rtp_bin_marshal_VOID__UINT_UINT, G_TYPE_NONE, 2,
-      G_TYPE_UINT, G_TYPE_UINT);
-  /**
-   * GstRtpBin::on-bye-timeout:
-   * @rtpbin: the object which received the signal
-   * @session: the session
-   * @ssrc: the SSRC 
-   *
-   * Notify of an SSRC that has timed out because of BYE
-   */
-  gst_rtp_bin_signals[SIGNAL_ON_BYE_TIMEOUT] =
-      g_signal_new ("on-bye-timeout", G_TYPE_FROM_CLASS (klass),
-      G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass, on_bye_timeout),
-      NULL, NULL, gst_rtp_bin_marshal_VOID__UINT_UINT, G_TYPE_NONE, 2,
-      G_TYPE_UINT, G_TYPE_UINT);
-  /**
-   * GstRtpBin::on-timeout:
-   * @rtpbin: the object which received the signal
-   * @session: the session
-   * @ssrc: the SSRC 
-   *
-   * Notify of an SSRC that has timed out
-   */
-  gst_rtp_bin_signals[SIGNAL_ON_TIMEOUT] =
-      g_signal_new ("on-timeout", G_TYPE_FROM_CLASS (klass),
-      G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass, on_timeout),
-      NULL, NULL, gst_rtp_bin_marshal_VOID__UINT_UINT, G_TYPE_NONE, 2,
-      G_TYPE_UINT, G_TYPE_UINT);
-  /**
-   * GstRtpBin::on-sender-timeout:
-   * @rtpbin: the object which received the signal
-   * @session: the session
-   * @ssrc: the SSRC 
-   *
-   * Notify of a sender SSRC that has timed out and became a receiver
-   */
-  gst_rtp_bin_signals[SIGNAL_ON_SENDER_TIMEOUT] =
-      g_signal_new ("on-sender-timeout", G_TYPE_FROM_CLASS (klass),
-      G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass, on_sender_timeout),
-      NULL, NULL, gst_rtp_bin_marshal_VOID__UINT_UINT, G_TYPE_NONE, 2,
-      G_TYPE_UINT, G_TYPE_UINT);
-
-  /**
-   * GstRtpBin::on-npt-stop:
-   * @rtpbin: the object which received the signal
-   * @session: the session
-   * @ssrc: the SSRC 
-   *
-   * Notify that SSRC sender has sent data up to the configured NPT stop time.
-   */
-  gst_rtp_bin_signals[SIGNAL_ON_NPT_STOP] =
-      g_signal_new ("on-npt-stop", G_TYPE_FROM_CLASS (klass),
-      G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass, on_npt_stop),
-      NULL, NULL, gst_rtp_bin_marshal_VOID__UINT_UINT, G_TYPE_NONE, 2,
-      G_TYPE_UINT, G_TYPE_UINT);
-
-  g_object_class_install_property (gobject_class, PROP_SDES,
-      g_param_spec_boxed ("sdes", "SDES",
-          "The SDES items of this session",
-          GST_TYPE_STRUCTURE, G_PARAM_READWRITE));
-
-  g_object_class_install_property (gobject_class, PROP_DO_LOST,
-      g_param_spec_boolean ("do-lost", "Do Lost",
-          "Send an event downstream when a packet is lost", DEFAULT_DO_LOST,
-          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
-
-  gstelement_class->change_state = GST_DEBUG_FUNCPTR (gst_rtp_bin_change_state);
-  gstelement_class->request_new_pad =
-      GST_DEBUG_FUNCPTR (gst_rtp_bin_request_new_pad);
-  gstelement_class->release_pad = GST_DEBUG_FUNCPTR (gst_rtp_bin_release_pad);
-
-  gstbin_class->handle_message = GST_DEBUG_FUNCPTR (gst_rtp_bin_handle_message);
-
-  klass->clear_pt_map = GST_DEBUG_FUNCPTR (gst_rtp_bin_clear_pt_map);
-  klass->reset_sync = GST_DEBUG_FUNCPTR (gst_rtp_bin_reset_sync);
-  klass->get_internal_session =
-      GST_DEBUG_FUNCPTR (gst_rtp_bin_get_internal_session);
-
-  GST_DEBUG_CATEGORY_INIT (gst_rtp_bin_debug, "rtpbin", 0, "RTP bin");
-}
-
-static void
-gst_rtp_bin_init (GstRtpBin * rtpbin, GstRtpBinClass * klass)
-{
-  gchar *str;
-
-  rtpbin->priv = GST_RTP_BIN_GET_PRIVATE (rtpbin);
-  rtpbin->priv->bin_lock = g_mutex_new ();
-  rtpbin->priv->dyn_lock = g_mutex_new ();
-
-  rtpbin->latency = DEFAULT_LATENCY_MS;
-  rtpbin->do_lost = DEFAULT_DO_LOST;
-
-  /* some default SDES entries */
-  str = g_strdup_printf ("%s@%s", g_get_user_name (), g_get_host_name ());
-  rtpbin->sdes = gst_structure_new ("application/x-rtp-source-sdes",
-      "cname", G_TYPE_STRING, str,
-      "name", G_TYPE_STRING, g_get_real_name (),
-      "tool", G_TYPE_STRING, "GStreamer", NULL);
-  g_free (str);
-}
-
-static void
-gst_rtp_bin_dispose (GObject * object)
-{
-  GstRtpBin *rtpbin;
-
-  rtpbin = GST_RTP_BIN (object);
-
-  GST_DEBUG_OBJECT (object, "freeing sessions");
-  g_slist_foreach (rtpbin->sessions, (GFunc) free_session, rtpbin);
-  g_slist_free (rtpbin->sessions);
-  rtpbin->sessions = NULL;
-  GST_DEBUG_OBJECT (object, "freeing clients");
-  g_slist_foreach (rtpbin->clients, (GFunc) free_client, rtpbin);
-  g_slist_free (rtpbin->clients);
-  rtpbin->clients = NULL;
-
-  G_OBJECT_CLASS (parent_class)->dispose (object);
-}
-
-static void
-gst_rtp_bin_finalize (GObject * object)
-{
-  GstRtpBin *rtpbin;
-
-  rtpbin = GST_RTP_BIN (object);
-
-  if (rtpbin->sdes)
-    gst_structure_free (rtpbin->sdes);
-
-  g_mutex_free (rtpbin->priv->bin_lock);
-  g_mutex_free (rtpbin->priv->dyn_lock);
-
-  G_OBJECT_CLASS (parent_class)->finalize (object);
-}
-
-
-static void
-gst_rtp_bin_set_sdes_struct (GstRtpBin * bin, const GstStructure * sdes)
-{
-  GSList *item;
-
-  if (sdes == NULL)
-    return;
-
-  GST_RTP_BIN_LOCK (bin);
-
-  GST_OBJECT_LOCK (bin);
-  if (bin->sdes)
-    gst_structure_free (bin->sdes);
-  bin->sdes = gst_structure_copy (sdes);
-
-  /* store in all sessions */
-  for (item = bin->sessions; item; item = g_slist_next (item))
-    g_object_set (item->data, "sdes", sdes, NULL);
-  GST_OBJECT_UNLOCK (bin);
-
-  GST_RTP_BIN_UNLOCK (bin);
-}
-
-static GstStructure *
-gst_rtp_bin_get_sdes_struct (GstRtpBin * bin)
-{
-  GstStructure *result;
-
-  GST_OBJECT_LOCK (bin);
-  result = gst_structure_copy (bin->sdes);
-  GST_OBJECT_UNLOCK (bin);
-
-  return result;
-}
-
-static void
-gst_rtp_bin_set_property (GObject * object, guint prop_id,
-    const GValue * value, GParamSpec * pspec)
-{
-  GstRtpBin *rtpbin;
-
-  rtpbin = GST_RTP_BIN (object);
-
-  switch (prop_id) {
-    case PROP_LATENCY:
-      GST_RTP_BIN_LOCK (rtpbin);
-      rtpbin->latency = g_value_get_uint (value);
-      GST_RTP_BIN_UNLOCK (rtpbin);
-      /* propegate the property down to the jitterbuffer */
-      gst_rtp_bin_propagate_property_to_jitterbuffer (rtpbin, "latency", value);
-      break;
-    case PROP_SDES:
-      gst_rtp_bin_set_sdes_struct (rtpbin, g_value_get_boxed (value));
-      break;
-    case PROP_DO_LOST:
-      GST_RTP_BIN_LOCK (rtpbin);
-      rtpbin->do_lost = g_value_get_boolean (value);
-      GST_RTP_BIN_UNLOCK (rtpbin);
-      gst_rtp_bin_propagate_property_to_jitterbuffer (rtpbin, "do-lost", value);
-      break;
-    default:
-      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
-      break;
-  }
-}
-
-static void
-gst_rtp_bin_get_property (GObject * object, guint prop_id,
-    GValue * value, GParamSpec * pspec)
-{
-  GstRtpBin *rtpbin;
-
-  rtpbin = GST_RTP_BIN (object);
-
-  switch (prop_id) {
-    case PROP_LATENCY:
-      GST_RTP_BIN_LOCK (rtpbin);
-      g_value_set_uint (value, rtpbin->latency);
-      GST_RTP_BIN_UNLOCK (rtpbin);
-      break;
-    case PROP_SDES:
-      g_value_take_boxed (value, gst_rtp_bin_get_sdes_struct (rtpbin));
-      break;
-    case PROP_DO_LOST:
-      GST_RTP_BIN_LOCK (rtpbin);
-      g_value_set_boolean (value, rtpbin->do_lost);
-      GST_RTP_BIN_UNLOCK (rtpbin);
-      break;
-    default:
-      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
-      break;
-  }
-}
-
-static void
-gst_rtp_bin_handle_message (GstBin * bin, GstMessage * message)
-{
-  GstRtpBin *rtpbin;
-
-  rtpbin = GST_RTP_BIN (bin);
-
-  switch (GST_MESSAGE_TYPE (message)) {
-    case GST_MESSAGE_ELEMENT:
-    {
-      const GstStructure *s = gst_message_get_structure (message);
-
-      /* we change the structure name and add the session ID to it */
-      if (gst_structure_has_name (s, "application/x-rtp-source-sdes")) {
-        GSList *walk;
-
-        /* find the session, the message source has it */
-        for (walk = rtpbin->sessions; walk; walk = g_slist_next (walk)) {
-          GstRtpBinSession *sess = (GstRtpBinSession *) walk->data;
-
-          /* if we found the session, change message. else we exit the loop and
-           * leave the message unchanged */
-          if (GST_OBJECT_CAST (sess->session) == GST_MESSAGE_SRC (message)) {
-            message = gst_message_make_writable (message);
-            s = gst_message_get_structure (message);
-
-            gst_structure_set ((GstStructure *) s, "session", G_TYPE_UINT,
-                sess->id, NULL);
-            break;
-          }
-        }
-      }
-      /* fallthrough to forward the modified message to the parent */
-    }
-    default:
-    {
-      GST_BIN_CLASS (parent_class)->handle_message (bin, message);
-      break;
-    }
-  }
-}
-
-static void
-calc_ntp_ns_base (GstRtpBin * bin)
-{
-  GstClockTime now;
-  GTimeVal current;
-  GSList *walk;
-
-  /* get the current time and convert it to NTP time in nanoseconds */
-  g_get_current_time (&current);
-  now = GST_TIMEVAL_TO_TIME (current);
-  now += (2208988800LL * GST_SECOND);
-
-  GST_RTP_BIN_LOCK (bin);
-  bin->priv->ntp_ns_base = now;
-  for (walk = bin->sessions; walk; walk = g_slist_next (walk)) {
-    GstRtpBinSession *session = (GstRtpBinSession *) walk->data;
-
-    g_object_set (session->session, "ntp-ns-base", now, NULL);
-  }
-  GST_RTP_BIN_UNLOCK (bin);
-
-  return;
-}
-
-static GstStateChangeReturn
-gst_rtp_bin_change_state (GstElement * element, GstStateChange transition)
-{
-  GstStateChangeReturn res;
-  GstRtpBin *rtpbin;
-  GstRtpBinPrivate *priv;
-
-  rtpbin = GST_RTP_BIN (element);
-  priv = rtpbin->priv;
-
-  switch (transition) {
-    case GST_STATE_CHANGE_NULL_TO_READY:
-      break;
-    case GST_STATE_CHANGE_READY_TO_PAUSED:
-      GST_LOG_OBJECT (rtpbin, "clearing shutdown flag");
-      g_atomic_int_set (&priv->shutdown, 0);
-      break;
-    case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
-      calc_ntp_ns_base (rtpbin);
-      break;
-    case GST_STATE_CHANGE_PAUSED_TO_READY:
-      GST_LOG_OBJECT (rtpbin, "setting shutdown flag");
-      g_atomic_int_set (&priv->shutdown, 1);
-      /* wait for all callbacks to end by taking the lock. No new callbacks will
-       * be able to happen as we set the shutdown flag. */
-      GST_RTP_BIN_DYN_LOCK (rtpbin);
-      GST_LOG_OBJECT (rtpbin, "dynamic lock taken, we can continue shutdown");
-      GST_RTP_BIN_DYN_UNLOCK (rtpbin);
-      break;
-    default:
-      break;
-  }
-
-  res = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
-
-  switch (transition) {
-    case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
-      break;
-    case GST_STATE_CHANGE_PAUSED_TO_READY:
-      break;
-    case GST_STATE_CHANGE_READY_TO_NULL:
-      break;
-    default:
-      break;
-  }
-  return res;
-}
-
-/* a new pad (SSRC) was created in @session. This signal is emited from the
- * payload demuxer. */
-static void
-new_payload_found (GstElement * element, guint pt, GstPad * pad,
-    GstRtpBinStream * stream)
-{
-  GstRtpBin *rtpbin;
-  GstElementClass *klass;
-  GstPadTemplate *templ;
-  gchar *padname;
-  GstPad *gpad;
-
-  rtpbin = stream->bin;
-
-  GST_DEBUG ("new payload pad %d", pt);
-
-  GST_RTP_BIN_SHUTDOWN_LOCK (rtpbin, shutdown);
-
-  /* ghost the pad to the parent */
-  klass = GST_ELEMENT_GET_CLASS (rtpbin);
-  templ = gst_element_class_get_pad_template (klass, "recv_rtp_src_%d_%d_%d");
-  padname = g_strdup_printf ("recv_rtp_src_%d_%u_%d",
-      stream->session->id, stream->ssrc, pt);
-  gpad = gst_ghost_pad_new_from_template (padname, pad, templ);
-  g_free (padname);
-  g_object_set_data (G_OBJECT (pad), "GstRTPBin.ghostpad", gpad);
-
-  gst_pad_set_caps (gpad, GST_PAD_CAPS (pad));
-  gst_pad_set_active (gpad, TRUE);
-  gst_element_add_pad (GST_ELEMENT_CAST (rtpbin), gpad);
-  GST_RTP_BIN_SHUTDOWN_UNLOCK (rtpbin);
-
-  return;
-
-shutdown:
-  {
-    GST_DEBUG ("ignoring, we are shutting down");
-    return;
-  }
-}
-
-static void
-payload_pad_removed (GstElement * element, GstPad * pad,
-    GstRtpBinStream * stream)
-{
-  GstRtpBin *rtpbin;
-  GstPad *gpad;
-
-  rtpbin = stream->bin;
-
-  GST_DEBUG ("payload pad removed");
-
-  GST_RTP_BIN_DYN_LOCK (rtpbin);
-  if ((gpad = g_object_get_data (G_OBJECT (pad), "GstRTPBin.ghostpad"))) {
-    g_object_set_data (G_OBJECT (pad), "GstRTPBin.ghostpad", NULL);
-
-    gst_pad_set_active (gpad, FALSE);
-    gst_element_remove_pad (GST_ELEMENT_CAST (rtpbin), gpad);
-  }
-  GST_RTP_BIN_DYN_UNLOCK (rtpbin);
-}
-
-static GstCaps *
-pt_map_requested (GstElement * element, guint pt, GstRtpBinSession * session)
-{
-  GstRtpBin *rtpbin;
-  GstCaps *caps;
-
-  rtpbin = session->bin;
-
-  GST_DEBUG_OBJECT (rtpbin, "payload map requested for pt %d in session %d", pt,
-      session->id);
-
-  caps = get_pt_map (session, pt);
-  if (!caps)
-    goto no_caps;
-
-  return caps;
-
-  /* ERRORS */
-no_caps:
-  {
-    GST_DEBUG_OBJECT (rtpbin, "could not get caps");
-    return NULL;
-  }
-}
-
-/* emited when caps changed for the session */
-static void
-caps_changed (GstPad * pad, GParamSpec * pspec, GstRtpBinSession * session)
-{
-  GstRtpBin *bin;
-  GstCaps *caps;
-  gint payload;
-  const GstStructure *s;
-
-  bin = session->bin;
-
-  g_object_get (pad, "caps", &caps, NULL);
-
-  if (caps == NULL)
-    return;
-
-  GST_DEBUG_OBJECT (bin, "got caps %" GST_PTR_FORMAT, caps);
-
-  s = gst_caps_get_structure (caps, 0);
-
-  /* get payload, finish when it's not there */
-  if (!gst_structure_get_int (s, "payload", &payload))
-    return;
-
-  GST_RTP_SESSION_LOCK (session);
-  GST_DEBUG_OBJECT (bin, "insert caps for payload %d", payload);
-  g_hash_table_insert (session->ptmap, GINT_TO_POINTER (payload), caps);
-  GST_RTP_SESSION_UNLOCK (session);
-}
-
-/* a new pad (SSRC) was created in @session */
-static void
-new_ssrc_pad_found (GstElement * element, guint ssrc, GstPad * pad,
-    GstRtpBinSession * session)
-{
-  GstRtpBin *rtpbin;
-  GstRtpBinStream *stream;
-  GstPad *sinkpad, *srcpad;
-  gchar *padname;
-
-  rtpbin = session->bin;
-
-  GST_DEBUG_OBJECT (rtpbin, "new SSRC pad %08x, %s:%s", ssrc,
-      GST_DEBUG_PAD_NAME (pad));
-
-  GST_RTP_BIN_SHUTDOWN_LOCK (rtpbin, shutdown);
-
-  GST_RTP_SESSION_LOCK (session);
-
-  /* create new stream */
-  stream = create_stream (session, ssrc);
-  if (!stream)
-    goto no_stream;
-
-  /* get pad and link */
-  GST_DEBUG_OBJECT (rtpbin, "linking jitterbuffer RTP");
-  padname = g_strdup_printf ("src_%d", ssrc);
-  srcpad = gst_element_get_static_pad (element, padname);
-  g_free (padname);
-  sinkpad = gst_element_get_static_pad (stream->buffer, "sink");
-  gst_pad_link (srcpad, sinkpad);
-  gst_object_unref (sinkpad);
-  gst_object_unref (srcpad);
-
-  GST_DEBUG_OBJECT (rtpbin, "linking jitterbuffer RTCP");
-  padname = g_strdup_printf ("rtcp_src_%d", ssrc);
-  srcpad = gst_element_get_static_pad (element, padname);
-  g_free (padname);
-  sinkpad = gst_element_get_request_pad (stream->buffer, "sink_rtcp");
-  gst_pad_link (srcpad, sinkpad);
-  gst_object_unref (sinkpad);
-  gst_object_unref (srcpad);
-
-  /* connect to the RTCP sync signal from the jitterbuffer */
-  GST_DEBUG_OBJECT (rtpbin, "connecting sync signal");
-  stream->buffer_handlesync_sig = g_signal_connect (stream->buffer,
-      "handle-sync", (GCallback) gst_rtp_bin_handle_sync, stream);
-
-  /* connect to the new-pad signal of the payload demuxer, this will expose the
-   * new pad by ghosting it. */
-  stream->demux_newpad_sig = g_signal_connect (stream->demux,
-      "new-payload-type", (GCallback) new_payload_found, stream);
-  stream->demux_padremoved_sig = g_signal_connect (stream->demux,
-      "pad-removed", (GCallback) payload_pad_removed, stream);
-
-  /* connect to the request-pt-map signal. This signal will be emited by the
-   * demuxer so that it can apply a proper caps on the buffers for the
-   * depayloaders. */
-  stream->demux_ptreq_sig = g_signal_connect (stream->demux,
-      "request-pt-map", (GCallback) pt_map_requested, session);
-
-  GST_RTP_SESSION_UNLOCK (session);
-  GST_RTP_BIN_SHUTDOWN_UNLOCK (rtpbin);
-
-  return;
-
-  /* ERRORS */
-shutdown:
-  {
-    GST_DEBUG_OBJECT (rtpbin, "we are shutting down");
-    return;
-  }
-no_stream:
-  {
-    GST_RTP_SESSION_UNLOCK (session);
-    GST_RTP_BIN_SHUTDOWN_UNLOCK (rtpbin);
-    GST_DEBUG_OBJECT (rtpbin, "could not create stream");
-    return;
-  }
-}
-
-/* Create a pad for receiving RTP for the session in @name. Must be called with
- * RTP_BIN_LOCK.
- */
-static GstPad *
-create_recv_rtp (GstRtpBin * rtpbin, GstPadTemplate * templ, const gchar * name)
-{
-  GstPad *sinkdpad;
-  guint sessid;
-  GstRtpBinSession *session;
-  GstPadLinkReturn lres;
-
-  /* first get the session number */
-  if (name == NULL || sscanf (name, "recv_rtp_sink_%d", &sessid) != 1)
-    goto no_name;
-
-  GST_DEBUG_OBJECT (rtpbin, "finding session %d", sessid);
-
-  /* get or create session */
-  session = find_session_by_id (rtpbin, sessid);
-  if (!session) {
-    GST_DEBUG_OBJECT (rtpbin, "creating session %d", sessid);
-    /* create session now */
-    session = create_session (rtpbin, sessid);
-    if (session == NULL)
-      goto create_error;
-  }
-
-  /* check if pad was requested */
-  if (session->recv_rtp_sink_ghost != NULL)
-    return session->recv_rtp_sink_ghost;
-
-  GST_DEBUG_OBJECT (rtpbin, "getting RTP sink pad");
-  /* get recv_rtp pad and store */
-  session->recv_rtp_sink =
-      gst_element_get_request_pad (session->session, "recv_rtp_sink");
-  if (session->recv_rtp_sink == NULL)
-    goto pad_failed;
-
-  g_signal_connect (session->recv_rtp_sink, "notify::caps",
-      (GCallback) caps_changed, session);
-
-  GST_DEBUG_OBJECT (rtpbin, "getting RTP src pad");
-  /* get srcpad, link to SSRCDemux */
-  session->recv_rtp_src =
-      gst_element_get_static_pad (session->session, "recv_rtp_src");
-  if (session->recv_rtp_src == NULL)
-    goto pad_failed;
-
-  GST_DEBUG_OBJECT (rtpbin, "getting demuxer RTP sink pad");
-  sinkdpad = gst_element_get_static_pad (session->demux, "sink");
-  GST_DEBUG_OBJECT (rtpbin, "linking demuxer RTP sink pad");
-  lres = gst_pad_link (session->recv_rtp_src, sinkdpad);
-  gst_object_unref (sinkdpad);
-  if (lres != GST_PAD_LINK_OK)
-    goto link_failed;
-
-  /* connect to the new-ssrc-pad signal of the SSRC demuxer */
-  session->demux_newpad_sig = g_signal_connect (session->demux,
-      "new-ssrc-pad", (GCallback) new_ssrc_pad_found, session);
-  session->demux_padremoved_sig = g_signal_connect (session->demux,
-      "removed-ssrc-pad", (GCallback) ssrc_demux_pad_removed, session);
-
-  GST_DEBUG_OBJECT (rtpbin, "ghosting session sink pad");
-  session->recv_rtp_sink_ghost =
-      gst_ghost_pad_new_from_template (name, session->recv_rtp_sink, templ);
-  gst_pad_set_active (session->recv_rtp_sink_ghost, TRUE);
-  gst_element_add_pad (GST_ELEMENT_CAST (rtpbin), session->recv_rtp_sink_ghost);
-
-  return session->recv_rtp_sink_ghost;
-
-  /* ERRORS */
-no_name:
-  {
-    g_warning ("gstrtpbin: invalid name given");
-    return NULL;
-  }
-create_error:
-  {
-    /* create_session already warned */
-    return NULL;
-  }
-pad_failed:
-  {
-    g_warning ("gstrtpbin: failed to get session pad");
-    return NULL;
-  }
-link_failed:
-  {
-    g_warning ("gstrtpbin: failed to link pads");
-    return NULL;
-  }
-}
-
-static void
-remove_recv_rtp (GstRtpBin * rtpbin, GstRtpBinSession * session)
-{
-  if (session->demux_newpad_sig) {
-    g_signal_handler_disconnect (session->demux, session->demux_newpad_sig);
-    session->demux_newpad_sig = 0;
-  }
-  if (session->demux_padremoved_sig) {
-    g_signal_handler_disconnect (session->demux, session->demux_padremoved_sig);
-    session->demux_padremoved_sig = 0;
-  }
-  if (session->recv_rtp_src) {
-    gst_object_unref (session->recv_rtp_src);
-    session->recv_rtp_src = NULL;
-  }
-  if (session->recv_rtp_sink) {
-    gst_element_release_request_pad (session->session, session->recv_rtp_sink);
-    gst_object_unref (session->recv_rtp_sink);
-    session->recv_rtp_sink = NULL;
-  }
-  if (session->recv_rtp_sink_ghost) {
-    gst_pad_set_active (session->recv_rtp_sink_ghost, FALSE);
-    gst_element_remove_pad (GST_ELEMENT_CAST (rtpbin),
-        session->recv_rtp_sink_ghost);
-    session->recv_rtp_sink_ghost = NULL;
-  }
-}
-
-/* Create a pad for receiving RTCP for the session in @name. Must be called with
- * RTP_BIN_LOCK.
- */
-static GstPad *
-create_recv_rtcp (GstRtpBin * rtpbin, GstPadTemplate * templ,
-    const gchar * name)
-{
-  guint sessid;
-  GstRtpBinSession *session;
-  GstPad *sinkdpad;
-  GstPadLinkReturn lres;
-
-  /* first get the session number */
-  if (name == NULL || sscanf (name, "recv_rtcp_sink_%d", &sessid) != 1)
-    goto no_name;
-
-  GST_DEBUG_OBJECT (rtpbin, "finding session %d", sessid);
-
-  /* get or create the session */
-  session = find_session_by_id (rtpbin, sessid);
-  if (!session) {
-    GST_DEBUG_OBJECT (rtpbin, "creating session %d", sessid);
-    /* create session now */
-    session = create_session (rtpbin, sessid);
-    if (session == NULL)
-      goto create_error;
-  }
-
-  /* check if pad was requested */
-  if (session->recv_rtcp_sink_ghost != NULL)
-    return session->recv_rtcp_sink_ghost;
-
-  /* get recv_rtp pad and store */
-  GST_DEBUG_OBJECT (rtpbin, "getting RTCP sink pad");
-  session->recv_rtcp_sink =
-      gst_element_get_request_pad (session->session, "recv_rtcp_sink");
-  if (session->recv_rtcp_sink == NULL)
-    goto pad_failed;
-
-  /* get srcpad, link to SSRCDemux */
-  GST_DEBUG_OBJECT (rtpbin, "getting sync src pad");
-  session->sync_src = gst_element_get_static_pad (session->session, "sync_src");
-  if (session->sync_src == NULL)
-    goto pad_failed;
-
-  GST_DEBUG_OBJECT (rtpbin, "getting demuxer RTCP sink pad");
-  sinkdpad = gst_element_get_static_pad (session->demux, "rtcp_sink");
-  lres = gst_pad_link (session->sync_src, sinkdpad);
-  gst_object_unref (sinkdpad);
-  if (lres != GST_PAD_LINK_OK)
-    goto link_failed;
-
-  session->recv_rtcp_sink_ghost =
-      gst_ghost_pad_new_from_template (name, session->recv_rtcp_sink, templ);
-  gst_pad_set_active (session->recv_rtcp_sink_ghost, TRUE);
-  gst_element_add_pad (GST_ELEMENT_CAST (rtpbin),
-      session->recv_rtcp_sink_ghost);
-
-  return session->recv_rtcp_sink_ghost;
-
-  /* ERRORS */
-no_name:
-  {
-    g_warning ("gstrtpbin: invalid name given");
-    return NULL;
-  }
-create_error:
-  {
-    /* create_session already warned */
-    return NULL;
-  }
-pad_failed:
-  {
-    g_warning ("gstrtpbin: failed to get session pad");
-    return NULL;
-  }
-link_failed:
-  {
-    g_warning ("gstrtpbin: failed to link pads");
-    return NULL;
-  }
-}
-
-static void
-remove_recv_rtcp (GstRtpBin * rtpbin, GstRtpBinSession * session)
-{
-  if (session->recv_rtcp_sink_ghost) {
-    gst_pad_set_active (session->recv_rtcp_sink_ghost, FALSE);
-    gst_element_remove_pad (GST_ELEMENT_CAST (rtpbin),
-        session->recv_rtcp_sink_ghost);
-    session->recv_rtcp_sink_ghost = NULL;
-  }
-  if (session->sync_src) {
-    /* releasing the request pad should also unref the sync pad */
-    gst_object_unref (session->sync_src);
-    session->sync_src = NULL;
-  }
-  if (session->recv_rtcp_sink) {
-    gst_element_release_request_pad (session->session, session->recv_rtcp_sink);
-    gst_object_unref (session->recv_rtcp_sink);
-    session->recv_rtcp_sink = NULL;
-  }
-}
-
-/* Create a pad for sending RTP for the session in @name. Must be called with
- * RTP_BIN_LOCK.
- */
-static GstPad *
-create_send_rtp (GstRtpBin * rtpbin, GstPadTemplate * templ, const gchar * name)
-{
-  gchar *gname;
-  guint sessid;
-  GstRtpBinSession *session;
-  GstElementClass *klass;
-
-  /* first get the session number */
-  if (name == NULL || sscanf (name, "send_rtp_sink_%d", &sessid) != 1)
-    goto no_name;
-
-  /* get or create session */
-  session = find_session_by_id (rtpbin, sessid);
-  if (!session) {
-    /* create session now */
-    session = create_session (rtpbin, sessid);
-    if (session == NULL)
-      goto create_error;
-  }
-
-  /* check if pad was requested */
-  if (session->send_rtp_sink_ghost != NULL)
-    return session->send_rtp_sink_ghost;
-
-  /* get send_rtp pad and store */
-  session->send_rtp_sink =
-      gst_element_get_request_pad (session->session, "send_rtp_sink");
-  if (session->send_rtp_sink == NULL)
-    goto pad_failed;
-
-  session->send_rtp_sink_ghost =
-      gst_ghost_pad_new_from_template (name, session->send_rtp_sink, templ);
-  gst_pad_set_active (session->send_rtp_sink_ghost, TRUE);
-  gst_element_add_pad (GST_ELEMENT_CAST (rtpbin), session->send_rtp_sink_ghost);
-
-  /* get srcpad */
-  session->send_rtp_src =
-      gst_element_get_static_pad (session->session, "send_rtp_src");
-  if (session->send_rtp_src == NULL)
-    goto no_srcpad;
-
-  /* ghost the new source pad */
-  klass = GST_ELEMENT_GET_CLASS (rtpbin);
-  gname = g_strdup_printf ("send_rtp_src_%d", sessid);
-  templ = gst_element_class_get_pad_template (klass, "send_rtp_src_%d");
-  session->send_rtp_src_ghost =
-      gst_ghost_pad_new_from_template (gname, session->send_rtp_src, templ);
-  gst_pad_set_active (session->send_rtp_src_ghost, TRUE);
-  gst_element_add_pad (GST_ELEMENT_CAST (rtpbin), session->send_rtp_src_ghost);
-  g_free (gname);
-
-  return session->send_rtp_sink_ghost;
-
-  /* ERRORS */
-no_name:
-  {
-    g_warning ("gstrtpbin: invalid name given");
-    return NULL;
-  }
-create_error:
-  {
-    /* create_session already warned */
-    return NULL;
-  }
-pad_failed:
-  {
-    g_warning ("gstrtpbin: 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);
-    return NULL;
-  }
-}
-
-static void
-remove_send_rtp (GstRtpBin * rtpbin, GstRtpBinSession * session)
-{
-  if (session->send_rtp_src_ghost) {
-    gst_pad_set_active (session->send_rtp_src_ghost, FALSE);
-    gst_element_remove_pad (GST_ELEMENT_CAST (rtpbin),
-        session->send_rtp_src_ghost);
-    session->send_rtp_src_ghost = NULL;
-  }
-  if (session->send_rtp_src) {
-    gst_object_unref (session->send_rtp_src);
-    session->send_rtp_src = NULL;
-  }
-  if (session->send_rtp_sink) {
-    gst_element_release_request_pad (GST_ELEMENT_CAST (session->session),
-        session->send_rtp_sink);
-    gst_object_unref (session->send_rtp_sink);
-    session->send_rtp_sink = NULL;
-  }
-  if (session->send_rtp_sink_ghost) {
-    gst_pad_set_active (session->send_rtp_sink_ghost, FALSE);
-    gst_element_remove_pad (GST_ELEMENT_CAST (rtpbin),
-        session->send_rtp_sink_ghost);
-    session->send_rtp_sink_ghost = NULL;
-  }
-}
-
-/* Create a pad for sending RTCP for the session in @name. Must be called with
- * RTP_BIN_LOCK.
- */
-static GstPad *
-create_rtcp (GstRtpBin * rtpbin, GstPadTemplate * templ, const gchar * name)
-{
-  guint sessid;
-  GstRtpBinSession *session;
-
-  /* first get the session number */
-  if (name == NULL || sscanf (name, "send_rtcp_src_%d", &sessid) != 1)
-    goto no_name;
-
-  /* get or create session */
-  session = find_session_by_id (rtpbin, sessid);
-  if (!session)
-    goto no_session;
-
-  /* check if pad was requested */
-  if (session->send_rtcp_src_ghost != NULL)
-    return session->send_rtcp_src_ghost;
-
-  /* get rtcp_src pad and store */
-  session->send_rtcp_src =
-      gst_element_get_request_pad (session->session, "send_rtcp_src");
-  if (session->send_rtcp_src == NULL)
-    goto pad_failed;
-
-  session->send_rtcp_src_ghost =
-      gst_ghost_pad_new_from_template (name, session->send_rtcp_src, templ);
-  gst_pad_set_active (session->send_rtcp_src_ghost, TRUE);
-  gst_element_add_pad (GST_ELEMENT_CAST (rtpbin), session->send_rtcp_src_ghost);
-
-  return session->send_rtcp_src_ghost;
-
-  /* ERRORS */
-no_name:
-  {
-    g_warning ("gstrtpbin: invalid name given");
-    return NULL;
-  }
-no_session:
-  {
-    g_warning ("gstrtpbin: session with id %d does not exist", sessid);
-    return NULL;
-  }
-pad_failed:
-  {
-    g_warning ("gstrtpbin: failed to get rtcp pad for session %d", sessid);
-    return NULL;
-  }
-}
-
-static void
-remove_rtcp (GstRtpBin * rtpbin, GstRtpBinSession * session)
-{
-  if (session->send_rtcp_src_ghost) {
-    gst_pad_set_active (session->send_rtcp_src_ghost, FALSE);
-    gst_element_remove_pad (GST_ELEMENT_CAST (rtpbin),
-        session->send_rtcp_src_ghost);
-    session->send_rtcp_src_ghost = NULL;
-  }
-  if (session->send_rtcp_src) {
-    gst_element_release_request_pad (session->session, session->send_rtcp_src);
-    gst_object_unref (session->send_rtcp_src);
-    session->send_rtcp_src = NULL;
-  }
-}
-
-/* If the requested name is NULL we should create a name with
- * the session number assuming we want the lowest posible session
- * with a free pad like the template */
-static gchar *
-gst_rtp_bin_get_free_pad_name (GstElement * element, GstPadTemplate * templ)
-{
-  gboolean name_found = FALSE;
-  gint session = 0;
-  GstPad *pad = NULL;
-  GstIterator *pad_it = NULL;
-  gchar *pad_name = NULL;
-
-  GST_DEBUG_OBJECT (element, "find a free pad name for template");
-  while (!name_found) {
-    g_free (pad_name);
-    pad_name = g_strdup_printf (templ->name_template, session++);
-    pad_it = gst_element_iterate_pads (GST_ELEMENT (element));
-    name_found = TRUE;
-    while (gst_iterator_next (pad_it, (gpointer) & pad) == GST_ITERATOR_OK) {
-      gchar *name;
-
-      name = gst_pad_get_name (pad);
-      if (strcmp (name, pad_name) == 0)
-        name_found = FALSE;
-      g_free (name);
-    }
-    gst_iterator_free (pad_it);
-  }
-
-  GST_DEBUG_OBJECT (element, "free pad name found: '%s'", pad_name);
-  return pad_name;
-}
-
-/* 
- */
-static GstPad *
-gst_rtp_bin_request_new_pad (GstElement * element,
-    GstPadTemplate * templ, const gchar * name)
-{
-  GstRtpBin *rtpbin;
-  GstElementClass *klass;
-  GstPad *result;
-
-  gchar *pad_name = NULL;
-
-  g_return_val_if_fail (templ != NULL, NULL);
-  g_return_val_if_fail (GST_IS_RTP_BIN (element), NULL);
-
-  rtpbin = GST_RTP_BIN (element);
-  klass = GST_ELEMENT_GET_CLASS (element);
-
-  GST_RTP_BIN_LOCK (rtpbin);
-
-  if (name == NULL) {
-    /* use a free pad name */
-    pad_name = gst_rtp_bin_get_free_pad_name (element, templ);
-  } else {
-    /* use the provided name */
-    pad_name = g_strdup (name);
-  }
-
-  GST_DEBUG_OBJECT (rtpbin, "Trying to request a pad with name %s", pad_name);
-
-  /* figure out the template */
-  if (templ == gst_element_class_get_pad_template (klass, "recv_rtp_sink_%d")) {
-    result = create_recv_rtp (rtpbin, templ, pad_name);
-  } else if (templ == gst_element_class_get_pad_template (klass,
-          "recv_rtcp_sink_%d")) {
-    result = create_recv_rtcp (rtpbin, templ, pad_name);
-  } else if (templ == gst_element_class_get_pad_template (klass,
-          "send_rtp_sink_%d")) {
-    result = create_send_rtp (rtpbin, templ, pad_name);
-  } else if (templ == gst_element_class_get_pad_template (klass,
-          "send_rtcp_src_%d")) {
-    result = create_rtcp (rtpbin, templ, pad_name);
-  } else
-    goto wrong_template;
-
-  g_free (pad_name);
-  GST_RTP_BIN_UNLOCK (rtpbin);
-
-  return result;
-
-  /* ERRORS */
-wrong_template:
-  {
-    g_free (pad_name);
-    GST_RTP_BIN_UNLOCK (rtpbin);
-    g_warning ("gstrtpbin: this is not our template");
-    return NULL;
-  }
-}
-
-static void
-gst_rtp_bin_release_pad (GstElement * element, GstPad * pad)
-{
-  GstRtpBinSession *session;
-  GstRtpBin *rtpbin;
-
-  g_return_if_fail (GST_IS_GHOST_PAD (pad));
-  g_return_if_fail (GST_IS_RTP_BIN (element));
-
-  rtpbin = GST_RTP_BIN (element);
-
-  GST_RTP_BIN_LOCK (rtpbin);
-  GST_DEBUG_OBJECT (rtpbin, "Trying to release pad %s:%s",
-      GST_DEBUG_PAD_NAME (pad));
-
-  if (!(session = find_session_by_pad (rtpbin, pad)))
-    goto unknown_pad;
-
-  if (session->recv_rtp_sink_ghost == pad) {
-    remove_recv_rtp (rtpbin, session);
-  } else if (session->recv_rtcp_sink_ghost == pad) {
-    remove_recv_rtcp (rtpbin, session);
-  } else if (session->send_rtp_sink_ghost == pad) {
-    remove_send_rtp (rtpbin, session);
-  } else if (session->send_rtcp_src_ghost == pad) {
-    remove_rtcp (rtpbin, session);
-  }
-
-  /* no more request pads, free the complete session */
-  if (session->recv_rtp_sink_ghost == NULL
-      && session->recv_rtcp_sink_ghost == NULL
-      && session->send_rtp_sink_ghost == NULL
-      && session->send_rtcp_src_ghost == NULL) {
-    GST_DEBUG_OBJECT (rtpbin, "no more pads for session %p", session);
-    rtpbin->sessions = g_slist_remove (rtpbin->sessions, session);
-    free_session (session, rtpbin);
-  }
-  GST_RTP_BIN_UNLOCK (rtpbin);
-
-  return;
-
-  /* ERROR */
-unknown_pad:
-  {
-    GST_RTP_BIN_UNLOCK (rtpbin);
-    g_warning ("gstrtpbin: %s:%s is not one of our request pads",
-        GST_DEBUG_PAD_NAME (pad));
-    return;
-  }
-}
diff --git a/gst/rtpmanager/gstrtpbin.h b/gst/rtpmanager/gstrtpbin.h
deleted file mode 100644
index bed6ad02..00000000
--- a/gst/rtpmanager/gstrtpbin.h
+++ /dev/null
@@ -1,88 +0,0 @@
-/* GStreamer
- * Copyright (C) <2007> Wim Taymans <wim.taymans@gmail.com>
- *
- * This library is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Library General Public
- * License as published by the Free Software Foundation; either
- * version 2 of the License, or (at your option) any later version.
- *
- * This library is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Library General Public License for more details.
- *
- * You should have received a copy of the GNU Library General Public
- * License along with this library; if not, write to the
- * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
- * Boston, MA 02111-1307, USA.
- */
-
-#ifndef __GST_RTP_BIN_H__
-#define __GST_RTP_BIN_H__
-
-#include <gst/gst.h>
-
-#include "rtpsession.h"
-
-#define GST_TYPE_RTP_BIN \
-  (gst_rtp_bin_get_type())
-#define GST_RTP_BIN(obj) \
-  (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_RTP_BIN,GstRtpBin))
-#define GST_RTP_BIN_CLASS(klass) \
-  (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_RTP_BIN,GstRtpBinClass))
-#define GST_IS_RTP_BIN(obj) \
-  (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_RTP_BIN))
-#define GST_IS_RTP_BIN_CLASS(klass) \
-  (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_RTP_BIN))
-
-typedef struct _GstRtpBin GstRtpBin;
-typedef struct _GstRtpBinClass GstRtpBinClass;
-typedef struct _GstRtpBinPrivate GstRtpBinPrivate;
-
-struct _GstRtpBin {
-  GstBin         bin;
-
-  /*< private >*/
-  /* default latency for sessions */
-  guint           latency;
-  gboolean        do_lost;
-  /* a list of session */
-  GSList         *sessions;
-
-  /* a list of clients, these are streams with the same CNAME */
-  GSList         *clients;
-
-  /* the default SDES items for sessions */
-  GstStructure   *sdes;
-
-  /*< private >*/
-  GstRtpBinPrivate *priv;
-};
-
-struct _GstRtpBinClass {
-  GstBinClass  parent_class;
-
-  /* get the caps for pt */
-  GstCaps*    (*request_pt_map)       (GstRtpBin *rtpbin, guint session, guint pt);
-
-  /* action signals */
-  void        (*clear_pt_map)         (GstRtpBin *rtpbin);
-  void        (*reset_sync)           (GstRtpBin *rtpbin);
-  RTPSession* (*get_internal_session) (GstRtpBin *rtpbin, guint session_id);
-
-  /* session manager signals */
-  void     (*on_new_ssrc)       (GstRtpBin *rtpbin, guint session, guint32 ssrc);
-  void     (*on_ssrc_collision) (GstRtpBin *rtpbin, guint session, guint32 ssrc);
-  void     (*on_ssrc_validated) (GstRtpBin *rtpbin, guint session, guint32 ssrc);
-  void     (*on_ssrc_active)    (GstRtpBin *rtpbin, guint session, guint32 ssrc);
-  void     (*on_ssrc_sdes)      (GstRtpBin *rtpbin, guint session, guint32 ssrc);
-  void     (*on_bye_ssrc)       (GstRtpBin *rtpbin, guint session, guint32 ssrc);
-  void     (*on_bye_timeout)    (GstRtpBin *rtpbin, guint session, guint32 ssrc);
-  void     (*on_timeout)        (GstRtpBin *rtpbin, guint session, guint32 ssrc);
-  void     (*on_sender_timeout) (GstRtpBin *rtpbin, guint session, guint32 ssrc);
-  void     (*on_npt_stop)       (GstRtpBin *rtpbin, guint session, guint32 ssrc);
-};
-
-GType gst_rtp_bin_get_type (void);
-
-#endif /* __GST_RTP_BIN_H__ */
diff --git a/gst/rtpmanager/gstrtpjitterbuffer.c b/gst/rtpmanager/gstrtpjitterbuffer.c
deleted file mode 100644
index 55126054..00000000
--- a/gst/rtpmanager/gstrtpjitterbuffer.c
+++ /dev/null
@@ -1,1972 +0,0 @@
-/*
- * Farsight Voice+Video library
- *
- *  Copyright 2007 Collabora Ltd, 
- *  Copyright 2007 Nokia Corporation
- *   @author: Philippe Kalaf <philippe.kalaf@collabora.co.uk>.
- *  Copyright 2007 Wim Taymans <wim.taymans@gmail.com>
- *
- * This library is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Library General Public
- * License as published by the Free Software Foundation; either
- * version 2 of the License, or (at your option) any later version.
- *
- * This library is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Library General Public License for more details.
- *
- * You should have received a copy of the GNU Library General Public
- * License along with this library; if not, write to the
- * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
- * Boston, MA 02111-1307, USA.
- *
- */
-
-/**
- * SECTION:element-gstrtpjitterbuffer
- *
- * 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 #GstRtpJitterBuffer:latency property.
- * Packets arriving too late are considered to be lost packets.
- * 
- * This element acts as a live element and so adds #GstRtpJitterBuffer:latency
- * to the pipeline.
- * 
- * The element needs the clock-rate of the RTP payload in order to estimate the
- * delay. This information is obtained either from the caps on the sink pad or,
- * 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.
- * 
- * <refsect2>
- * <title>Example pipelines</title>
- * |[
- * gst-launch rtspsrc location=rtsp://192.168.1.133:8554/mpeg1or2AudioVideoTest ! gstrtpjitterbuffer ! 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.
- * </refsect2>
- *
- * Last reviewed on 2007-05-28 (0.10.5)
- */
-
-#ifdef HAVE_CONFIG_H
-#include "config.h"
-#endif
-
-#include <stdlib.h>
-#include <string.h>
-#include <gst/rtp/gstrtpbuffer.h>
-
-#include "gstrtpbin-marshal.h"
-
-#include "gstrtpjitterbuffer.h"
-#include "rtpjitterbuffer.h"
-#include "rtpstats.h"
-
-GST_DEBUG_CATEGORY (rtpjitterbuffer_debug);
-#define GST_CAT_DEFAULT (rtpjitterbuffer_debug)
-
-/* low and high threshold tell the queue when to start and stop buffering */
-#define LOW_THRESHOLD 0.2
-#define HIGH_THRESHOLD 0.8
-
-/* elementfactory information */
-static const GstElementDetails gst_rtp_jitter_buffer_details =
-GST_ELEMENT_DETAILS ("RTP packet jitter-buffer",
-    "Filter/Network/RTP",
-    "A buffer that deals with network jitter and other transmission faults",
-    "Philippe Kalaf <philippe.kalaf@collabora.co.uk>, "
-    "Wim Taymans <wim.taymans@gmail.com>");
-
-/* RTPJitterBuffer signals and args */
-enum
-{
-  SIGNAL_REQUEST_PT_MAP,
-  SIGNAL_CLEAR_PT_MAP,
-  SIGNAL_HANDLE_SYNC,
-  SIGNAL_ON_NPT_STOP,
-  LAST_SIGNAL
-};
-
-#define DEFAULT_LATENCY_MS      200
-#define DEFAULT_DROP_ON_LATENCY FALSE
-#define DEFAULT_TS_OFFSET       0
-#define DEFAULT_DO_LOST         FALSE
-
-enum
-{
-  PROP_0,
-  PROP_LATENCY,
-  PROP_DROP_ON_LATENCY,
-  PROP_TS_OFFSET,
-  PROP_DO_LOST,
-  PROP_LAST
-};
-
-#define JBUF_LOCK(priv)   (g_mutex_lock ((priv)->jbuf_lock))
-
-#define JBUF_LOCK_CHECK(priv,label) G_STMT_START {    \
-  JBUF_LOCK (priv);                                   \
-  if (G_UNLIKELY (priv->srcresult != GST_FLOW_OK))    \
-    goto label;                                       \
-} G_STMT_END
-
-#define JBUF_UNLOCK(priv) (g_mutex_unlock ((priv)->jbuf_lock))
-#define JBUF_WAIT(priv)   (g_cond_wait ((priv)->jbuf_cond, (priv)->jbuf_lock))
-
-#define JBUF_WAIT_CHECK(priv,label) G_STMT_START {    \
-  JBUF_WAIT(priv);                                    \
-  if (G_UNLIKELY (priv->srcresult != GST_FLOW_OK))    \
-    goto label;                                       \
-} G_STMT_END
-
-#define JBUF_SIGNAL(priv) (g_cond_signal ((priv)->jbuf_cond))
-
-struct _GstRtpJitterBufferPrivate
-{
-  GstPad *sinkpad, *srcpad;
-  GstPad *rtcpsinkpad;
-
-  RTPJitterBuffer *jbuf;
-  GMutex *jbuf_lock;
-  GCond *jbuf_cond;
-  gboolean waiting;
-  gboolean discont;
-
-  /* properties */
-  guint latency_ms;
-  gboolean drop_on_latency;
-  gint64 ts_offset;
-  gboolean do_lost;
-
-  /* the last seqnum we pushed out */
-  guint32 last_popped_seqnum;
-  /* the next expected seqnum we push */
-  guint32 next_seqnum;
-  /* last output time */
-  GstClockTime last_out_time;
-  /* the next expected seqnum we receive */
-  guint32 next_in_seqnum;
-
-  /* start and stop ranges */
-  GstClockTime npt_start;
-  GstClockTime npt_stop;
-  guint64 ext_timestamp;
-  guint64 last_elapsed;
-  guint64 estimated_eos;
-  GstClockID eos_id;
-  gboolean reached_npt_stop;
-
-  /* state */
-  gboolean eos;
-
-  /* clock rate and rtp timestamp offset */
-  gint last_pt;
-  gint32 clock_rate;
-  gint64 clock_base;
-  gint64 prev_ts_offset;
-
-  /* when we are shutting down */
-  GstFlowReturn srcresult;
-  gboolean blocked;
-
-  /* for sync */
-  GstSegment segment;
-  GstClockID clock_id;
-  gboolean unscheduled;
-  /* the latency of the upstream peer, we have to take this into account when
-   * synchronizing the buffers. */
-  GstClockTime peer_latency;
-
-  /* some accounting */
-  guint64 num_late;
-  guint64 num_duplicates;
-};
-
-#define GST_RTP_JITTER_BUFFER_GET_PRIVATE(o) \
-  (G_TYPE_INSTANCE_GET_PRIVATE ((o), GST_TYPE_RTP_JITTER_BUFFER, \
-                                GstRtpJitterBufferPrivate))
-
-static GstStaticPadTemplate gst_rtp_jitter_buffer_sink_template =
-GST_STATIC_PAD_TEMPLATE ("sink",
-    GST_PAD_SINK,
-    GST_PAD_ALWAYS,
-    GST_STATIC_CAPS ("application/x-rtp, "
-        "clock-rate = (int) [ 1, 2147483647 ]"
-        /* "payload = (int) , "
-         * "encoding-name = (string) "
-         */ )
-    );
-
-static GstStaticPadTemplate gst_rtp_jitter_buffer_sink_rtcp_template =
-GST_STATIC_PAD_TEMPLATE ("sink_rtcp",
-    GST_PAD_SINK,
-    GST_PAD_REQUEST,
-    GST_STATIC_CAPS ("application/x-rtcp")
-    );
-
-static GstStaticPadTemplate gst_rtp_jitter_buffer_src_template =
-GST_STATIC_PAD_TEMPLATE ("src",
-    GST_PAD_SRC,
-    GST_PAD_ALWAYS,
-    GST_STATIC_CAPS ("application/x-rtp"
-        /* "payload = (int) , "
-         * "clock-rate = (int) , "
-         * "encoding-name = (string) "
-         */ )
-    );
-
-static guint gst_rtp_jitter_buffer_signals[LAST_SIGNAL] = { 0 };
-
-GST_BOILERPLATE (GstRtpJitterBuffer, gst_rtp_jitter_buffer, GstElement,
-    GST_TYPE_ELEMENT);
-
-/* object overrides */
-static void gst_rtp_jitter_buffer_set_property (GObject * object,
-    guint prop_id, const GValue * value, GParamSpec * pspec);
-static void gst_rtp_jitter_buffer_get_property (GObject * object,
-    guint prop_id, GValue * value, GParamSpec * pspec);
-static void gst_rtp_jitter_buffer_finalize (GObject * object);
-
-/* element overrides */
-static GstStateChangeReturn gst_rtp_jitter_buffer_change_state (GstElement
-    * element, GstStateChange transition);
-static GstPad *gst_rtp_jitter_buffer_request_new_pad (GstElement * element,
-    GstPadTemplate * templ, const gchar * name);
-static void gst_rtp_jitter_buffer_release_pad (GstElement * element,
-    GstPad * pad);
-
-/* pad overrides */
-static GstCaps *gst_rtp_jitter_buffer_getcaps (GstPad * pad);
-static GList *gst_rtp_jitter_buffer_internal_links (GstPad * pad);
-
-/* sinkpad overrides */
-static gboolean gst_jitter_buffer_sink_setcaps (GstPad * pad, GstCaps * caps);
-static gboolean gst_rtp_jitter_buffer_sink_event (GstPad * pad,
-    GstEvent * event);
-static GstFlowReturn gst_rtp_jitter_buffer_chain (GstPad * pad,
-    GstBuffer * buffer);
-
-static gboolean gst_rtp_jitter_buffer_sink_rtcp_event (GstPad * pad,
-    GstEvent * event);
-static GstFlowReturn gst_rtp_jitter_buffer_chain_rtcp (GstPad * pad,
-    GstBuffer * buffer);
-
-/* srcpad overrides */
-static gboolean gst_rtp_jitter_buffer_src_event (GstPad * pad,
-    GstEvent * event);
-static gboolean
-gst_rtp_jitter_buffer_src_activate_push (GstPad * pad, gboolean active);
-static void gst_rtp_jitter_buffer_loop (GstRtpJitterBuffer * jitterbuffer);
-static gboolean gst_rtp_jitter_buffer_query (GstPad * pad, GstQuery * query);
-
-static void
-gst_rtp_jitter_buffer_clear_pt_map (GstRtpJitterBuffer * jitterbuffer);
-
-static void
-gst_rtp_jitter_buffer_base_init (gpointer klass)
-{
-  GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
-
-  gst_element_class_add_pad_template (element_class,
-      gst_static_pad_template_get (&gst_rtp_jitter_buffer_src_template));
-  gst_element_class_add_pad_template (element_class,
-      gst_static_pad_template_get (&gst_rtp_jitter_buffer_sink_template));
-  gst_element_class_add_pad_template (element_class,
-      gst_static_pad_template_get (&gst_rtp_jitter_buffer_sink_rtcp_template));
-
-  gst_element_class_set_details (element_class, &gst_rtp_jitter_buffer_details);
-}
-
-static void
-gst_rtp_jitter_buffer_class_init (GstRtpJitterBufferClass * klass)
-{
-  GObjectClass *gobject_class;
-  GstElementClass *gstelement_class;
-
-  gobject_class = (GObjectClass *) klass;
-  gstelement_class = (GstElementClass *) klass;
-
-  g_type_class_add_private (klass, sizeof (GstRtpJitterBufferPrivate));
-
-  gobject_class->finalize = GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_finalize);
-
-  gobject_class->set_property = gst_rtp_jitter_buffer_set_property;
-  gobject_class->get_property = gst_rtp_jitter_buffer_get_property;
-
-  /**
-   * GstRtpJitterBuffer::latency:
-   * 
-   * The maximum latency of the jitterbuffer. Packets will be kept in the buffer
-   * for at most this time.
-   */
-  g_object_class_install_property (gobject_class, PROP_LATENCY,
-      g_param_spec_uint ("latency", "Buffer latency in ms",
-          "Amount of ms to buffer", 0, G_MAXUINT, DEFAULT_LATENCY_MS,
-          G_PARAM_READWRITE));
-  /**
-   * GstRtpJitterBuffer::drop-on-latency:
-   * 
-   * Drop oldest buffers when the queue is completely filled. 
-   */
-  g_object_class_install_property (gobject_class, PROP_DROP_ON_LATENCY,
-      g_param_spec_boolean ("drop-on-latency",
-          "Drop buffers when maximum latency is reached",
-          "Tells the jitterbuffer to never exceed the given latency in size",
-          DEFAULT_DROP_ON_LATENCY, G_PARAM_READWRITE));
-  /**
-   * GstRtpJitterBuffer::ts-offset:
-   * 
-   * Adjust GStreamer output buffer timestamps in the jitterbuffer with offset.
-   * This is mainly used to ensure interstream synchronisation.
-   */
-  g_object_class_install_property (gobject_class, PROP_TS_OFFSET,
-      g_param_spec_int64 ("ts-offset", "Timestamp Offset",
-          "Adjust buffer timestamps with offset in nanoseconds", G_MININT64,
-          G_MAXINT64, DEFAULT_TS_OFFSET,
-          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
-
-  /**
-   * GstRtpJitterBuffer::do-lost:
-   * 
-   * Send out a GstRTPPacketLost event downstream when a packet is considered
-   * lost.
-   */
-  g_object_class_install_property (gobject_class, PROP_DO_LOST,
-      g_param_spec_boolean ("do-lost", "Do Lost",
-          "Send an event downstream when a packet is lost", DEFAULT_DO_LOST,
-          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
-  /**
-   * GstRtpJitterBuffer::request-pt-map:
-   * @buffer: the object which received the signal
-   * @pt: the pt
-   *
-   * Request the payload type as #GstCaps for @pt.
-   */
-  gst_rtp_jitter_buffer_signals[SIGNAL_REQUEST_PT_MAP] =
-      g_signal_new ("request-pt-map", G_TYPE_FROM_CLASS (klass),
-      G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpJitterBufferClass,
-          request_pt_map), NULL, NULL, gst_rtp_bin_marshal_BOXED__UINT,
-      GST_TYPE_CAPS, 1, G_TYPE_UINT);
-  /**
-   * GstRtpJitterBuffer::handle-sync:
-   * @buffer: the object which received the signal
-   * @struct: a GstStructure containing sync values.
-   *
-   * Be notified of new sync values.
-   */
-  gst_rtp_jitter_buffer_signals[SIGNAL_HANDLE_SYNC] =
-      g_signal_new ("handle-sync", G_TYPE_FROM_CLASS (klass),
-      G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpJitterBufferClass,
-          handle_sync), NULL, NULL, g_cclosure_marshal_VOID__BOXED,
-      G_TYPE_NONE, 1, GST_TYPE_STRUCTURE | G_SIGNAL_TYPE_STATIC_SCOPE);
-
-  /**
-   * GstRtpJitterBuffer::on-npt-stop
-   * @buffer: the object which received the signal
-   *
-   * Signal that the jitterbufer has pushed the RTP packet that corresponds to
-   * the npt-stop position.
-   */
-  gst_rtp_jitter_buffer_signals[SIGNAL_ON_NPT_STOP] =
-      g_signal_new ("on-npt-stop", G_TYPE_FROM_CLASS (klass),
-      G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpJitterBufferClass,
-          on_npt_stop), NULL, NULL, g_cclosure_marshal_VOID__VOID,
-      G_TYPE_NONE, 0, G_TYPE_NONE);
-
-  /**
-   * GstRtpJitterBuffer::clear-pt-map:
-   * @buffer: the object which received the signal
-   *
-   * Invalidate the clock-rate as obtained with the
-   * #GstRtpJitterBuffer::request-pt-map signal.
-   */
-  gst_rtp_jitter_buffer_signals[SIGNAL_CLEAR_PT_MAP] =
-      g_signal_new ("clear-pt-map", G_TYPE_FROM_CLASS (klass),
-      G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION,
-      G_STRUCT_OFFSET (GstRtpJitterBufferClass, clear_pt_map), NULL, NULL,
-      g_cclosure_marshal_VOID__VOID, G_TYPE_NONE, 0, G_TYPE_NONE);
-
-  gstelement_class->change_state =
-      GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_change_state);
-  gstelement_class->request_new_pad =
-      GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_request_new_pad);
-  gstelement_class->release_pad =
-      GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_release_pad);
-
-  klass->clear_pt_map = GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_clear_pt_map);
-
-  GST_DEBUG_CATEGORY_INIT
-      (rtpjitterbuffer_debug, "gstrtpjitterbuffer", 0, "RTP Jitter Buffer");
-}
-
-static void
-gst_rtp_jitter_buffer_init (GstRtpJitterBuffer * jitterbuffer,
-    GstRtpJitterBufferClass * klass)
-{
-  GstRtpJitterBufferPrivate *priv;
-
-  priv = GST_RTP_JITTER_BUFFER_GET_PRIVATE (jitterbuffer);
-  jitterbuffer->priv = priv;
-
-  priv->latency_ms = DEFAULT_LATENCY_MS;
-  priv->drop_on_latency = DEFAULT_DROP_ON_LATENCY;
-  priv->do_lost = DEFAULT_DO_LOST;
-
-  priv->jbuf = rtp_jitter_buffer_new ();
-  priv->jbuf_lock = g_mutex_new ();
-  priv->jbuf_cond = g_cond_new ();
-
-  priv->srcpad =
-      gst_pad_new_from_static_template (&gst_rtp_jitter_buffer_src_template,
-      "src");
-
-  gst_pad_set_activatepush_function (priv->srcpad,
-      GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_src_activate_push));
-  gst_pad_set_query_function (priv->srcpad,
-      GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_query));
-  gst_pad_set_getcaps_function (priv->srcpad,
-      GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_getcaps));
-  gst_pad_set_event_function (priv->srcpad,
-      GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_src_event));
-
-  priv->sinkpad =
-      gst_pad_new_from_static_template (&gst_rtp_jitter_buffer_sink_template,
-      "sink");
-
-  gst_pad_set_chain_function (priv->sinkpad,
-      GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_chain));
-  gst_pad_set_event_function (priv->sinkpad,
-      GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_sink_event));
-  gst_pad_set_setcaps_function (priv->sinkpad,
-      GST_DEBUG_FUNCPTR (gst_jitter_buffer_sink_setcaps));
-  gst_pad_set_getcaps_function (priv->sinkpad,
-      GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_getcaps));
-
-  gst_element_add_pad (GST_ELEMENT (jitterbuffer), priv->srcpad);
-  gst_element_add_pad (GST_ELEMENT (jitterbuffer), priv->sinkpad);
-}
-
-static void
-gst_rtp_jitter_buffer_finalize (GObject * object)
-{
-  GstRtpJitterBuffer *jitterbuffer;
-
-  jitterbuffer = GST_RTP_JITTER_BUFFER (object);
-
-  g_mutex_free (jitterbuffer->priv->jbuf_lock);
-  g_cond_free (jitterbuffer->priv->jbuf_cond);
-
-  g_object_unref (jitterbuffer->priv->jbuf);
-
-  G_OBJECT_CLASS (parent_class)->finalize (object);
-}
-
-static GList *
-gst_rtp_jitter_buffer_internal_links (GstPad * pad)
-{
-  GstRtpJitterBuffer *jitterbuffer;
-  GstRtpJitterBufferPrivate *priv;
-  GList *res = NULL;
-
-  jitterbuffer = GST_RTP_JITTER_BUFFER (gst_pad_get_parent (pad));
-  priv = jitterbuffer->priv;
-
-  if (pad == priv->sinkpad) {
-    res = g_list_prepend (res, priv->srcpad);
-  } else if (pad == priv->srcpad) {
-    res = g_list_prepend (res, priv->sinkpad);
-  } else if (pad == priv->rtcpsinkpad) {
-    res = NULL;
-  }
-
-  gst_object_unref (jitterbuffer);
-
-  return res;
-}
-
-static GstPad *
-create_rtcp_sink (GstRtpJitterBuffer * jitterbuffer)
-{
-  GstRtpJitterBufferPrivate *priv;
-
-  priv = jitterbuffer->priv;
-
-  GST_DEBUG_OBJECT (jitterbuffer, "creating RTCP sink pad");
-
-  priv->rtcpsinkpad =
-      gst_pad_new_from_static_template
-      (&gst_rtp_jitter_buffer_sink_rtcp_template, "sink_rtcp");
-  gst_pad_set_chain_function (priv->rtcpsinkpad,
-      gst_rtp_jitter_buffer_chain_rtcp);
-  gst_pad_set_event_function (priv->rtcpsinkpad,
-      (GstPadEventFunction) gst_rtp_jitter_buffer_sink_rtcp_event);
-  gst_pad_set_internal_link_function (priv->rtcpsinkpad,
-      gst_rtp_jitter_buffer_internal_links);
-  gst_pad_set_active (priv->rtcpsinkpad, TRUE);
-  gst_element_add_pad (GST_ELEMENT_CAST (jitterbuffer), priv->rtcpsinkpad);
-
-  return priv->rtcpsinkpad;
-}
-
-static void
-remove_rtcp_sink (GstRtpJitterBuffer * jitterbuffer)
-{
-  GstRtpJitterBufferPrivate *priv;
-
-  priv = jitterbuffer->priv;
-
-  GST_DEBUG_OBJECT (jitterbuffer, "removing RTCP sink pad");
-
-  gst_pad_set_active (priv->rtcpsinkpad, FALSE);
-
-  gst_element_remove_pad (GST_ELEMENT_CAST (jitterbuffer), priv->rtcpsinkpad);
-  priv->rtcpsinkpad = NULL;
-}
-
-static GstPad *
-gst_rtp_jitter_buffer_request_new_pad (GstElement * element,
-    GstPadTemplate * templ, const gchar * name)
-{
-  GstRtpJitterBuffer *jitterbuffer;
-  GstElementClass *klass;
-  GstPad *result;
-  GstRtpJitterBufferPrivate *priv;
-
-  g_return_val_if_fail (templ != NULL, NULL);
-  g_return_val_if_fail (GST_IS_RTP_JITTER_BUFFER (element), NULL);
-
-  jitterbuffer = GST_RTP_JITTER_BUFFER (element);
-  priv = jitterbuffer->priv;
-  klass = GST_ELEMENT_GET_CLASS (element);
-
-  GST_DEBUG_OBJECT (element, "requesting pad %s", GST_STR_NULL (name));
-
-  /* figure out the template */
-  if (templ == gst_element_class_get_pad_template (klass, "sink_rtcp")) {
-    if (priv->rtcpsinkpad != NULL)
-      goto exists;
-
-    result = create_rtcp_sink (jitterbuffer);
-  } else
-    goto wrong_template;
-
-  return result;
-
-  /* ERRORS */
-wrong_template:
-  {
-    g_warning ("gstrtpjitterbuffer: this is not our template");
-    return NULL;
-  }
-exists:
-  {
-    g_warning ("gstrtpjitterbuffer: pad already requested");
-    return NULL;
-  }
-}
-
-static void
-gst_rtp_jitter_buffer_release_pad (GstElement * element, GstPad * pad)
-{
-  GstRtpJitterBuffer *jitterbuffer;
-  GstRtpJitterBufferPrivate *priv;
-
-  g_return_if_fail (GST_IS_RTP_JITTER_BUFFER (element));
-  g_return_if_fail (GST_IS_PAD (pad));
-
-  jitterbuffer = GST_RTP_JITTER_BUFFER (element);
-  priv = jitterbuffer->priv;
-
-  GST_DEBUG_OBJECT (element, "releasing pad %s:%s", GST_DEBUG_PAD_NAME (pad));
-
-  if (priv->rtcpsinkpad == pad) {
-    remove_rtcp_sink (jitterbuffer);
-  } else
-    goto wrong_pad;
-
-  return;
-
-  /* ERRORS */
-wrong_pad:
-  {
-    g_warning ("gstjitterbuffer: asked to release an unknown pad");
-    return;
-  }
-}
-
-static void
-gst_rtp_jitter_buffer_clear_pt_map (GstRtpJitterBuffer * jitterbuffer)
-{
-  GstRtpJitterBufferPrivate *priv;
-
-  priv = jitterbuffer->priv;
-
-  /* this will trigger a new pt-map request signal, FIXME, do something better. */
-  priv->clock_rate = -1;
-}
-
-static GstCaps *
-gst_rtp_jitter_buffer_getcaps (GstPad * pad)
-{
-  GstRtpJitterBuffer *jitterbuffer;
-  GstRtpJitterBufferPrivate *priv;
-  GstPad *other;
-  GstCaps *caps;
-  const GstCaps *templ;
-
-  jitterbuffer = GST_RTP_JITTER_BUFFER (gst_pad_get_parent (pad));
-  priv = jitterbuffer->priv;
-
-  other = (pad == priv->srcpad ? priv->sinkpad : priv->srcpad);
-
-  caps = gst_pad_peer_get_caps (other);
-
-  templ = gst_pad_get_pad_template_caps (pad);
-  if (caps == NULL) {
-    GST_DEBUG_OBJECT (jitterbuffer, "copy template");
-    caps = gst_caps_copy (templ);
-  } else {
-    GstCaps *intersect;
-
-    GST_DEBUG_OBJECT (jitterbuffer, "intersect with template");
-
-    intersect = gst_caps_intersect (caps, templ);
-    gst_caps_unref (caps);
-
-    caps = intersect;
-  }
-  gst_object_unref (jitterbuffer);
-
-  return caps;
-}
-
-static gboolean
-gst_jitter_buffer_sink_parse_caps (GstRtpJitterBuffer * jitterbuffer,
-    GstCaps * caps)
-{
-  GstRtpJitterBufferPrivate *priv;
-  GstStructure *caps_struct;
-  guint val;
-  GstClockTime tval;
-
-  priv = jitterbuffer->priv;
-
-  /* first parse the caps */
-  caps_struct = gst_caps_get_structure (caps, 0);
-
-  GST_DEBUG_OBJECT (jitterbuffer, "got caps");
-
-  /* we need a clock-rate to convert the rtp timestamps to GStreamer time and to
-   * measure the amount of data in the buffer */
-  if (!gst_structure_get_int (caps_struct, "clock-rate", &priv->clock_rate))
-    goto error;
-
-  if (priv->clock_rate <= 0)
-    goto wrong_rate;
-
-  GST_DEBUG_OBJECT (jitterbuffer, "got clock-rate %d", priv->clock_rate);
-
-  /* The clock base is the RTP timestamp corrsponding to the npt-start value. We
-   * can use this to track the amount of time elapsed on the sender. */
-  if (gst_structure_get_uint (caps_struct, "clock-base", &val))
-    priv->clock_base = val;
-  else
-    priv->clock_base = -1;
-
-  priv->ext_timestamp = priv->clock_base;
-
-  GST_DEBUG_OBJECT (jitterbuffer, "got clock-base %" G_GINT64_FORMAT,
-      priv->clock_base);
-
-  if (gst_structure_get_uint (caps_struct, "seqnum-base", &val)) {
-    /* first expected seqnum, only update when we didn't have a previous base. */
-    if (priv->next_in_seqnum == -1)
-      priv->next_in_seqnum = val;
-    if (priv->next_seqnum == -1)
-      priv->next_seqnum = val;
-  }
-
-  GST_DEBUG_OBJECT (jitterbuffer, "got seqnum-base %d", priv->next_in_seqnum);
-
-  /* the start and stop times. The seqnum-base corresponds to the start time. We
-   * will keep track of the seqnums on the output and when we reach the one
-   * corresponding to npt-stop, we emit the npt-stop-reached signal */
-  if (gst_structure_get_clock_time (caps_struct, "npt-start", &tval))
-    priv->npt_start = tval;
-  else
-    priv->npt_start = 0;
-
-  if (gst_structure_get_clock_time (caps_struct, "npt-stop", &tval))
-    priv->npt_stop = tval;
-  else
-    priv->npt_stop = -1;
-
-  GST_DEBUG_OBJECT (jitterbuffer,
-      "npt start/stop: %" GST_TIME_FORMAT "-%" GST_TIME_FORMAT,
-      GST_TIME_ARGS (priv->npt_start), GST_TIME_ARGS (priv->npt_stop));
-
-  return TRUE;
-
-  /* ERRORS */
-error:
-  {
-    GST_DEBUG_OBJECT (jitterbuffer, "No clock-rate in caps!");
-    return FALSE;
-  }
-wrong_rate:
-  {
-    GST_DEBUG_OBJECT (jitterbuffer, "Invalid clock-rate %d", priv->clock_rate);
-    return FALSE;
-  }
-}
-
-static gboolean
-gst_jitter_buffer_sink_setcaps (GstPad * pad, GstCaps * caps)
-{
-  GstRtpJitterBuffer *jitterbuffer;
-  GstRtpJitterBufferPrivate *priv;
-  gboolean res;
-
-  jitterbuffer = GST_RTP_JITTER_BUFFER (gst_pad_get_parent (pad));
-  priv = jitterbuffer->priv;
-
-  res = gst_jitter_buffer_sink_parse_caps (jitterbuffer, caps);
-
-  /* set same caps on srcpad on success */
-  if (res)
-    gst_pad_set_caps (priv->srcpad, caps);
-
-  gst_object_unref (jitterbuffer);
-
-  return res;
-}
-
-static void
-gst_rtp_jitter_buffer_flush_start (GstRtpJitterBuffer * jitterbuffer)
-{
-  GstRtpJitterBufferPrivate *priv;
-
-  priv = jitterbuffer->priv;
-
-  JBUF_LOCK (priv);
-  /* mark ourselves as flushing */
-  priv->srcresult = GST_FLOW_WRONG_STATE;
-  GST_DEBUG_OBJECT (jitterbuffer, "Disabling pop on queue");
-  /* this unblocks any waiting pops on the src pad task */
-  JBUF_SIGNAL (priv);
-  /* unlock clock, we just unschedule, the entry will be released by the 
-   * locking streaming thread. */
-  if (priv->clock_id) {
-    gst_clock_id_unschedule (priv->clock_id);
-    priv->unscheduled = TRUE;
-  }
-  JBUF_UNLOCK (priv);
-}
-
-static void
-gst_rtp_jitter_buffer_flush_stop (GstRtpJitterBuffer * jitterbuffer)
-{
-  GstRtpJitterBufferPrivate *priv;
-
-  priv = jitterbuffer->priv;
-
-  JBUF_LOCK (priv);
-  GST_DEBUG_OBJECT (jitterbuffer, "Enabling pop on queue");
-  /* Mark as non flushing */
-  priv->srcresult = GST_FLOW_OK;
-  gst_segment_init (&priv->segment, GST_FORMAT_TIME);
-  priv->last_popped_seqnum = -1;
-  priv->last_out_time = -1;
-  priv->next_seqnum = -1;
-  priv->next_in_seqnum = -1;
-  priv->clock_rate = -1;
-  priv->eos = FALSE;
-  priv->estimated_eos = -1;
-  priv->last_elapsed = 0;
-  priv->reached_npt_stop = FALSE;
-  priv->ext_timestamp = -1;
-  GST_DEBUG_OBJECT (jitterbuffer, "flush and reset jitterbuffer");
-  rtp_jitter_buffer_flush (priv->jbuf);
-  rtp_jitter_buffer_reset_skew (priv->jbuf);
-  JBUF_UNLOCK (priv);
-}
-
-static gboolean
-gst_rtp_jitter_buffer_src_activate_push (GstPad * pad, gboolean active)
-{
-  gboolean result = TRUE;
-  GstRtpJitterBuffer *jitterbuffer = NULL;
-
-  jitterbuffer = GST_RTP_JITTER_BUFFER (gst_pad_get_parent (pad));
-
-  if (active) {
-    /* allow data processing */
-    gst_rtp_jitter_buffer_flush_stop (jitterbuffer);
-
-    /* start pushing out buffers */
-    GST_DEBUG_OBJECT (jitterbuffer, "Starting task on srcpad");
-    gst_pad_start_task (jitterbuffer->priv->srcpad,
-        (GstTaskFunction) gst_rtp_jitter_buffer_loop, jitterbuffer);
-  } else {
-    /* make sure all data processing stops ASAP */
-    gst_rtp_jitter_buffer_flush_start (jitterbuffer);
-
-    /* NOTE this will hardlock if the state change is called from the src pad
-     * task thread because we will _join() the thread. */
-    GST_DEBUG_OBJECT (jitterbuffer, "Stopping task on srcpad");
-    result = gst_pad_stop_task (pad);
-  }
-
-  gst_object_unref (jitterbuffer);
-
-  return result;
-}
-
-static GstStateChangeReturn
-gst_rtp_jitter_buffer_change_state (GstElement * element,
-    GstStateChange transition)
-{
-  GstRtpJitterBuffer *jitterbuffer;
-  GstRtpJitterBufferPrivate *priv;
-  GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
-
-  jitterbuffer = GST_RTP_JITTER_BUFFER (element);
-  priv = jitterbuffer->priv;
-
-  switch (transition) {
-    case GST_STATE_CHANGE_NULL_TO_READY:
-      break;
-    case GST_STATE_CHANGE_READY_TO_PAUSED:
-      JBUF_LOCK (priv);
-      /* reset negotiated values */
-      priv->clock_rate = -1;
-      priv->clock_base = -1;
-      priv->peer_latency = 0;
-      priv->last_pt = -1;
-      /* block until we go to PLAYING */
-      priv->blocked = TRUE;
-      /* reset skew detection initialy */
-      rtp_jitter_buffer_reset_skew (priv->jbuf);
-      JBUF_UNLOCK (priv);
-      break;
-    case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
-      JBUF_LOCK (priv);
-      /* unblock to allow streaming in PLAYING */
-      priv->blocked = FALSE;
-      JBUF_SIGNAL (priv);
-      JBUF_UNLOCK (priv);
-      break;
-    default:
-      break;
-  }
-
-  ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
-
-  switch (transition) {
-    case GST_STATE_CHANGE_READY_TO_PAUSED:
-      /* we are a live element because we sync to the clock, which we can only
-       * do in the PLAYING state */
-      if (ret != GST_STATE_CHANGE_FAILURE)
-        ret = GST_STATE_CHANGE_NO_PREROLL;
-      break;
-    case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
-      JBUF_LOCK (priv);
-      /* block to stop streaming when PAUSED */
-      priv->blocked = TRUE;
-      JBUF_UNLOCK (priv);
-      if (ret != GST_STATE_CHANGE_FAILURE)
-        ret = GST_STATE_CHANGE_NO_PREROLL;
-      break;
-    case GST_STATE_CHANGE_PAUSED_TO_READY:
-      break;
-    case GST_STATE_CHANGE_READY_TO_NULL:
-      break;
-    default:
-      break;
-  }
-
-  return ret;
-}
-
-static gboolean
-gst_rtp_jitter_buffer_src_event (GstPad * pad, GstEvent * event)
-{
-  gboolean ret = TRUE;
-  GstRtpJitterBuffer *jitterbuffer;
-  GstRtpJitterBufferPrivate *priv;
-
-  jitterbuffer = GST_RTP_JITTER_BUFFER (gst_pad_get_parent (pad));
-  priv = jitterbuffer->priv;
-
-  GST_DEBUG_OBJECT (jitterbuffer, "received %s", GST_EVENT_TYPE_NAME (event));
-
-  switch (GST_EVENT_TYPE (event)) {
-    default:
-      ret = gst_pad_push_event (priv->sinkpad, event);
-      break;
-  }
-  gst_object_unref (jitterbuffer);
-
-  return ret;
-}
-
-static gboolean
-gst_rtp_jitter_buffer_sink_event (GstPad * pad, GstEvent * event)
-{
-  gboolean ret = TRUE;
-  GstRtpJitterBuffer *jitterbuffer;
-  GstRtpJitterBufferPrivate *priv;
-
-  jitterbuffer = GST_RTP_JITTER_BUFFER (gst_pad_get_parent (pad));
-  priv = jitterbuffer->priv;
-
-  GST_DEBUG_OBJECT (jitterbuffer, "received %s", GST_EVENT_TYPE_NAME (event));
-
-  switch (GST_EVENT_TYPE (event)) {
-    case GST_EVENT_NEWSEGMENT:
-    {
-      GstFormat format;
-      gdouble rate, arate;
-      gint64 start, stop, time;
-      gboolean update;
-
-      gst_event_parse_new_segment_full (event, &update, &rate, &arate, &format,
-          &start, &stop, &time);
-
-      /* we need time for now */
-      if (format != GST_FORMAT_TIME)
-        goto newseg_wrong_format;
-
-      GST_DEBUG_OBJECT (jitterbuffer,
-          "newsegment: update %d, rate %g, arate %g, start %" GST_TIME_FORMAT
-          ", stop %" GST_TIME_FORMAT ", time %" GST_TIME_FORMAT,
-          update, rate, arate, GST_TIME_ARGS (start), GST_TIME_ARGS (stop),
-          GST_TIME_ARGS (time));
-
-      /* now configure the values, we need these to time the release of the
-       * buffers on the srcpad. */
-      gst_segment_set_newsegment_full (&priv->segment, update,
-          rate, arate, format, start, stop, time);
-
-      /* FIXME, push SEGMENT in the queue. Sorting order might be difficult. */
-      ret = gst_pad_push_event (priv->srcpad, event);
-      break;
-    }
-    case GST_EVENT_FLUSH_START:
-      gst_rtp_jitter_buffer_flush_start (jitterbuffer);
-      ret = gst_pad_push_event (priv->srcpad, event);
-      break;
-    case GST_EVENT_FLUSH_STOP:
-      ret = gst_pad_push_event (priv->srcpad, event);
-      ret = gst_rtp_jitter_buffer_src_activate_push (priv->srcpad, TRUE);
-      break;
-    case GST_EVENT_EOS:
-    {
-      /* push EOS in queue. We always push it at the head */
-      JBUF_LOCK (priv);
-      /* check for flushing, we need to discard the event and return FALSE when
-       * we are flushing */
-      ret = priv->srcresult == GST_FLOW_OK;
-      if (ret && !priv->eos) {
-        GST_DEBUG_OBJECT (jitterbuffer, "queuing EOS");
-        priv->eos = TRUE;
-        JBUF_SIGNAL (priv);
-      } else if (priv->eos) {
-        GST_DEBUG_OBJECT (jitterbuffer, "dropping EOS, we are already EOS");
-      } else {
-        GST_DEBUG_OBJECT (jitterbuffer, "dropping EOS, reason %s",
-            gst_flow_get_name (priv->srcresult));
-      }
-      JBUF_UNLOCK (priv);
-      gst_event_unref (event);
-      break;
-    }
-    default:
-      ret = gst_pad_push_event (priv->srcpad, event);
-      break;
-  }
-
-done:
-  gst_object_unref (jitterbuffer);
-
-  return ret;
-
-  /* ERRORS */
-newseg_wrong_format:
-  {
-    GST_DEBUG_OBJECT (jitterbuffer, "received non TIME newsegment");
-    ret = FALSE;
-    goto done;
-  }
-}
-
-static gboolean
-gst_rtp_jitter_buffer_sink_rtcp_event (GstPad * pad, GstEvent * event)
-{
-  GstRtpJitterBuffer *jitterbuffer;
-  GstRtpJitterBufferPrivate *priv;
-
-  jitterbuffer = GST_RTP_JITTER_BUFFER (gst_pad_get_parent (pad));
-  priv = jitterbuffer->priv;
-
-  GST_DEBUG_OBJECT (jitterbuffer, "received %s", GST_EVENT_TYPE_NAME (event));
-
-  switch (GST_EVENT_TYPE (event)) {
-    case GST_EVENT_FLUSH_START:
-      break;
-    case GST_EVENT_FLUSH_STOP:
-      break;
-    default:
-      break;
-  }
-  gst_event_unref (event);
-  gst_object_unref (jitterbuffer);
-
-  return TRUE;
-}
-
-static gboolean
-gst_rtp_jitter_buffer_get_clock_rate (GstRtpJitterBuffer * jitterbuffer,
-    guint8 pt)
-{
-  GValue ret = { 0 };
-  GValue args[2] = { {0}, {0} };
-  GstCaps *caps;
-  gboolean res;
-
-  g_value_init (&args[0], GST_TYPE_ELEMENT);
-  g_value_set_object (&args[0], jitterbuffer);
-  g_value_init (&args[1], G_TYPE_UINT);
-  g_value_set_uint (&args[1], pt);
-
-  g_value_init (&ret, GST_TYPE_CAPS);
-  g_value_set_boxed (&ret, NULL);
-
-  g_signal_emitv (args, gst_rtp_jitter_buffer_signals[SIGNAL_REQUEST_PT_MAP], 0,
-      &ret);
-
-  g_value_unset (&args[0]);
-  g_value_unset (&args[1]);
-  caps = (GstCaps *) g_value_dup_boxed (&ret);
-  g_value_unset (&ret);
-  if (!caps)
-    goto no_caps;
-
-  res = gst_jitter_buffer_sink_parse_caps (jitterbuffer, caps);
-
-  gst_caps_unref (caps);
-
-  return res;
-
-  /* ERRORS */
-no_caps:
-  {
-    GST_DEBUG_OBJECT (jitterbuffer, "could not get caps");
-    return FALSE;
-  }
-}
-
-static GstFlowReturn
-gst_rtp_jitter_buffer_chain (GstPad * pad, GstBuffer * buffer)
-{
-  GstRtpJitterBuffer *jitterbuffer;
-  GstRtpJitterBufferPrivate *priv;
-  guint16 seqnum;
-  GstFlowReturn ret = GST_FLOW_OK;
-  GstClockTime timestamp;
-  guint64 latency_ts;
-  gboolean tail;
-  guint8 pt;
-
-  jitterbuffer = GST_RTP_JITTER_BUFFER (gst_pad_get_parent (pad));
-
-  if (G_UNLIKELY (!gst_rtp_buffer_validate (buffer)))
-    goto invalid_buffer;
-
-  priv = jitterbuffer->priv;
-
-  pt = gst_rtp_buffer_get_payload_type (buffer);
-
-  if (G_UNLIKELY (priv->last_pt != pt)) {
-    GstCaps *caps;
-
-    GST_DEBUG_OBJECT (jitterbuffer, "pt changed from %u to %u", priv->last_pt,
-        pt);
-
-    priv->last_pt = pt;
-    /* reset clock-rate so that we get a new one */
-    priv->clock_rate = -1;
-    /* Try to get the clock-rate from the caps first if we can. If there are no
-     * caps we must fire the signal to get the clock-rate. */
-    if ((caps = GST_BUFFER_CAPS (buffer))) {
-      gst_jitter_buffer_sink_parse_caps (jitterbuffer, caps);
-    }
-  }
-
-  if (G_UNLIKELY (priv->clock_rate == -1)) {
-    /* no clock rate given on the caps, try to get one with the signal */
-    gst_rtp_jitter_buffer_get_clock_rate (jitterbuffer, pt);
-    if (G_UNLIKELY (priv->clock_rate == -1))
-      goto no_clock_rate;
-  }
-
-  /* take the timestamp of the buffer. This is the time when the packet was
-   * received and is used to calculate jitter and clock skew. We will adjust
-   * this timestamp with the smoothed value after processing it in the
-   * jitterbuffer. */
-  timestamp = GST_BUFFER_TIMESTAMP (buffer);
-  /* bring to running time */
-  timestamp = gst_segment_to_running_time (&priv->segment, GST_FORMAT_TIME,
-      timestamp);
-
-  seqnum = gst_rtp_buffer_get_seq (buffer);
-
-  GST_DEBUG_OBJECT (jitterbuffer,
-      "Received packet #%d at time %" GST_TIME_FORMAT, seqnum,
-      GST_TIME_ARGS (timestamp));
-
-  JBUF_LOCK_CHECK (priv, out_flushing);
-  /* don't accept more data on EOS */
-  if (G_UNLIKELY (priv->eos))
-    goto have_eos;
-
-  /* now check against our expected seqnum */
-  if (G_LIKELY (priv->next_in_seqnum != -1)) {
-    gint gap;
-    gboolean reset = FALSE;
-
-    gap = gst_rtp_buffer_compare_seqnum (priv->next_in_seqnum, seqnum);
-    if (G_UNLIKELY (gap != 0)) {
-      GST_DEBUG_OBJECT (jitterbuffer, "expected #%d, got #%d, gap of %d",
-          priv->next_in_seqnum, seqnum, gap);
-      /* priv->next_in_seqnum >= seqnum, this packet is too late or the
-       * sender might have been restarted with different seqnum. */
-      if (gap < -RTP_MAX_MISORDER) {
-        GST_DEBUG_OBJECT (jitterbuffer, "reset: buffer too old %d", gap);
-        reset = TRUE;
-      }
-      /* priv->next_in_seqnum < seqnum, this is a new packet */
-      else if (G_UNLIKELY (gap > RTP_MAX_DROPOUT)) {
-        GST_DEBUG_OBJECT (jitterbuffer, "reset: too many dropped packets %d",
-            gap);
-        reset = TRUE;
-      } else {
-        GST_DEBUG_OBJECT (jitterbuffer, "tolerable gap");
-      }
-    }
-    if (G_UNLIKELY (reset)) {
-      GST_DEBUG_OBJECT (jitterbuffer, "flush and reset jitterbuffer");
-      rtp_jitter_buffer_flush (priv->jbuf);
-      rtp_jitter_buffer_reset_skew (priv->jbuf);
-      priv->last_popped_seqnum = -1;
-      priv->next_seqnum = seqnum;
-    }
-  }
-  priv->next_in_seqnum = (seqnum + 1) & 0xffff;
-
-  /* let's check if this buffer is too late, we can only accept packets with
-   * bigger seqnum than the one we last pushed. */
-  if (G_LIKELY (priv->last_popped_seqnum != -1)) {
-    gint gap;
-
-    gap = gst_rtp_buffer_compare_seqnum (priv->last_popped_seqnum, seqnum);
-
-    /* priv->last_popped_seqnum >= seqnum, we're too late. */
-    if (G_UNLIKELY (gap <= 0))
-      goto too_late;
-  }
-
-  /* let's drop oldest packet if the queue is already full and drop-on-latency
-   * is set. We can only do this when there actually is a latency. When no
-   * latency is set, we just pump it in the queue and let the other end push it
-   * out as fast as possible. */
-  if (priv->latency_ms && priv->drop_on_latency) {
-    latency_ts =
-        gst_util_uint64_scale_int (priv->latency_ms, priv->clock_rate, 1000);
-
-    if (G_UNLIKELY (rtp_jitter_buffer_get_ts_diff (priv->jbuf) >= latency_ts)) {
-      GstBuffer *old_buf;
-
-      old_buf = rtp_jitter_buffer_pop (priv->jbuf);
-
-      GST_DEBUG_OBJECT (jitterbuffer, "Queue full, dropping old packet #%d",
-          gst_rtp_buffer_get_seq (old_buf));
-
-      gst_buffer_unref (old_buf);
-    }
-  }
-
-  /* we need to make the metadata writable before pushing it in the jitterbuffer
-   * because the jitterbuffer will update the timestamp */
-  buffer = gst_buffer_make_metadata_writable (buffer);
-
-  /* now insert the packet into the queue in sorted order. This function returns
-   * FALSE if a packet with the same seqnum was already in the queue, meaning we
-   * have a duplicate. */
-  if (G_UNLIKELY (!rtp_jitter_buffer_insert (priv->jbuf, buffer, timestamp,
-              priv->clock_rate, &tail)))
-    goto duplicate;
-
-  /* signal addition of new buffer when the _loop is waiting. */
-  if (priv->waiting)
-    JBUF_SIGNAL (priv);
-
-  /* let's unschedule and unblock any waiting buffers. We only want to do this
-   * when the tail buffer changed */
-  if (G_UNLIKELY (priv->clock_id && tail)) {
-    GST_DEBUG_OBJECT (jitterbuffer,
-        "Unscheduling waiting buffer, new tail buffer");
-    gst_clock_id_unschedule (priv->clock_id);
-    priv->unscheduled = TRUE;
-  }
-
-  GST_DEBUG_OBJECT (jitterbuffer, "Pushed packet #%d, now %d packets, tail: %d",
-      seqnum, rtp_jitter_buffer_num_packets (priv->jbuf), tail);
-
-finished:
-  JBUF_UNLOCK (priv);
-
-  gst_object_unref (jitterbuffer);
-
-  return ret;
-
-  /* ERRORS */
-invalid_buffer:
-  {
-    /* this is not fatal but should be filtered earlier */
-    GST_ELEMENT_WARNING (jitterbuffer, STREAM, DECODE, (NULL),
-        ("Received invalid RTP payload, dropping"));
-    gst_buffer_unref (buffer);
-    gst_object_unref (jitterbuffer);
-    return GST_FLOW_OK;
-  }
-no_clock_rate:
-  {
-    GST_WARNING_OBJECT (jitterbuffer,
-        "No clock-rate in caps!, dropping buffer");
-    gst_buffer_unref (buffer);
-    gst_object_unref (jitterbuffer);
-    return GST_FLOW_OK;
-  }
-out_flushing:
-  {
-    ret = priv->srcresult;
-    GST_DEBUG_OBJECT (jitterbuffer, "flushing %s", gst_flow_get_name (ret));
-    gst_buffer_unref (buffer);
-    goto finished;
-  }
-have_eos:
-  {
-    ret = GST_FLOW_UNEXPECTED;
-    GST_WARNING_OBJECT (jitterbuffer, "we are EOS, refusing buffer");
-    gst_buffer_unref (buffer);
-    goto finished;
-  }
-too_late:
-  {
-    GST_WARNING_OBJECT (jitterbuffer, "Packet #%d too late as #%d was already"
-        " popped, dropping", seqnum, priv->last_popped_seqnum);
-    priv->num_late++;
-    gst_buffer_unref (buffer);
-    goto finished;
-  }
-duplicate:
-  {
-    GST_WARNING_OBJECT (jitterbuffer, "Duplicate packet #%d detected, dropping",
-        seqnum);
-    priv->num_duplicates++;
-    gst_buffer_unref (buffer);
-    goto finished;
-  }
-}
-
-static GstClockTime
-apply_offset (GstRtpJitterBuffer * jitterbuffer, GstClockTime timestamp)
-{
-  GstRtpJitterBufferPrivate *priv;
-
-  priv = jitterbuffer->priv;
-
-  if (timestamp == -1)
-    return -1;
-
-  /* apply the timestamp offset */
-  timestamp += priv->ts_offset;
-
-  return timestamp;
-}
-
-static GstClockTime
-get_sync_time (GstRtpJitterBuffer * jitterbuffer, GstClockTime timestamp)
-{
-  GstClockTime result;
-  GstRtpJitterBufferPrivate *priv;
-
-  priv = jitterbuffer->priv;
-
-  result = timestamp + GST_ELEMENT_CAST (jitterbuffer)->base_time;
-  /* add latency, this includes our own latency and the peer latency. */
-  result += (priv->latency_ms * GST_MSECOND);
-  result += priv->peer_latency;
-
-  return result;
-}
-
-static gboolean
-eos_reached (GstClock * clock, GstClockTime time, GstClockID id,
-    GstRtpJitterBuffer * jitterbuffer)
-{
-  GstRtpJitterBufferPrivate *priv;
-
-  priv = jitterbuffer->priv;
-
-  JBUF_LOCK_CHECK (priv, flushing);
-  if (priv->waiting) {
-    GST_DEBUG_OBJECT (jitterbuffer, "got the NPT timeout");
-    priv->reached_npt_stop = TRUE;
-    JBUF_SIGNAL (priv);
-  }
-  JBUF_UNLOCK (priv);
-
-  return TRUE;
-
-  /* ERRORS */
-flushing:
-  {
-    JBUF_UNLOCK (priv);
-    return FALSE;
-  }
-}
-
-/**
- * This funcion will push out buffers on the source pad.
- *
- * For each pushed buffer, the seqnum is recorded, if the next buffer B has a
- * different seqnum (missing packets before B), this function will wait for the
- * missing packet to arrive up to the timestamp of buffer B.
- */
-static void
-gst_rtp_jitter_buffer_loop (GstRtpJitterBuffer * jitterbuffer)
-{
-  GstRtpJitterBufferPrivate *priv;
-  GstBuffer *outbuf;
-  GstFlowReturn result;
-  guint16 seqnum;
-  guint32 next_seqnum;
-  GstClockTime timestamp, out_time;
-  gboolean discont = FALSE;
-  gint gap;
-  GstClock *clock;
-  GstClockID id;
-  GstClockTime sync_time;
-
-  priv = jitterbuffer->priv;
-
-  JBUF_LOCK_CHECK (priv, flushing);
-again:
-  GST_DEBUG_OBJECT (jitterbuffer, "Peeking item");
-  while (TRUE) {
-    id = NULL;
-    /* always wait if we are blocked */
-    if (G_LIKELY (!priv->blocked)) {
-      /* if we have a packet, we can exit the loop and grab it */
-      if (rtp_jitter_buffer_num_packets (priv->jbuf) > 0)
-        break;
-      /* no packets but we are EOS, do eos logic */
-      if (G_UNLIKELY (priv->eos))
-        goto do_eos;
-      /* underrun, wait for packets or flushing now if we are expecting an EOS
-       * timeout, set the async timer for it too */
-      if (priv->estimated_eos != -1 && !priv->reached_npt_stop) {
-        sync_time = get_sync_time (jitterbuffer, priv->estimated_eos);
-
-        GST_OBJECT_LOCK (jitterbuffer);
-        clock = GST_ELEMENT_CLOCK (jitterbuffer);
-        if (clock) {
-          GST_DEBUG_OBJECT (jitterbuffer, "scheduling timeout");
-          id = gst_clock_new_single_shot_id (clock, sync_time);
-          gst_clock_id_wait_async (id, (GstClockCallback) eos_reached,
-              jitterbuffer);
-        }
-        GST_OBJECT_UNLOCK (jitterbuffer);
-      }
-    }
-    /* now we wait */
-    priv->waiting = TRUE;
-    JBUF_WAIT (priv);
-    priv->waiting = FALSE;
-
-    if (id) {
-      /* unschedule any pending async notifications we might have */
-      gst_clock_id_unschedule (id);
-      gst_clock_id_unref (id);
-    }
-    if (G_UNLIKELY (priv->srcresult != GST_FLOW_OK))
-      goto flushing;
-
-    if (id && priv->reached_npt_stop) {
-      goto do_npt_stop;
-    }
-  }
-
-  /* peek a buffer, we're just looking at the timestamp and the sequence number.
-   * If all is fine, we'll pop and push it. If the sequence number is wrong we
-   * wait on the timestamp. In the chain function we will unlock the wait when a
-   * new buffer is available. The peeked buffer is valid for as long as we hold
-   * the jitterbuffer lock. */
-  outbuf = rtp_jitter_buffer_peek (priv->jbuf);
-
-  /* get the seqnum and the next expected seqnum */
-  seqnum = gst_rtp_buffer_get_seq (outbuf);
-  next_seqnum = priv->next_seqnum;
-
-  /* get the timestamp, this is already corrected for clock skew by the
-   * jitterbuffer */
-  timestamp = GST_BUFFER_TIMESTAMP (outbuf);
-
-  GST_DEBUG_OBJECT (jitterbuffer,
-      "Peeked buffer #%d, expect #%d, timestamp %" GST_TIME_FORMAT
-      ", now %d left", seqnum, next_seqnum, GST_TIME_ARGS (timestamp),
-      rtp_jitter_buffer_num_packets (priv->jbuf));
-
-  /* apply our timestamp offset to the incomming buffer, this will be our output
-   * timestamp. */
-  out_time = apply_offset (jitterbuffer, timestamp);
-
-  /* get the gap between this and the previous packet. If we don't know the
-   * previous packet seqnum assume no gap. */
-  if (G_LIKELY (next_seqnum != -1)) {
-    gap = gst_rtp_buffer_compare_seqnum (next_seqnum, seqnum);
-
-    /* if we have a packet that we already pushed or considered dropped, pop it
-     * off and get the next packet */
-    if (G_UNLIKELY (gap < 0)) {
-      GST_DEBUG_OBJECT (jitterbuffer, "Old packet #%d, next #%d dropping",
-          seqnum, next_seqnum);
-      outbuf = rtp_jitter_buffer_pop (priv->jbuf);
-      gst_buffer_unref (outbuf);
-      goto again;
-    }
-  } else {
-    GST_DEBUG_OBJECT (jitterbuffer, "no next seqnum known, first packet");
-    gap = -1;
-  }
-
-  /* If we don't know what the next seqnum should be (== -1) we have to wait
-   * because it might be possible that we are not receiving this buffer in-order,
-   * a buffer with a lower seqnum could arrive later and we want to push that
-   * earlier buffer before this buffer then.
-   * If we know the expected seqnum, we can compare it to the current seqnum to
-   * determine if we have missing a packet. If we have a missing packet (which
-   * must be before this packet) we can wait for it until the deadline for this
-   * packet expires. */
-  if (G_UNLIKELY (gap != 0 && out_time != -1)) {
-    GstClockReturn ret;
-    GstClockTime duration = GST_CLOCK_TIME_NONE;
-
-    if (gap > 0) {
-      /* we have a gap */
-      GST_WARNING_OBJECT (jitterbuffer,
-          "Sequence number GAP detected: expected %d instead of %d (%d missing)",
-          next_seqnum, seqnum, gap);
-
-      if (priv->last_out_time != -1) {
-        GST_DEBUG_OBJECT (jitterbuffer,
-            "out_time %" GST_TIME_FORMAT ", last %" GST_TIME_FORMAT,
-            GST_TIME_ARGS (out_time), GST_TIME_ARGS (priv->last_out_time));
-        /* interpolate between the current time and the last time based on
-         * number of packets we are missing, this is the estimated duration
-         * for the missing packet based on equidistant packet spacing. Also make
-         * sure we never go negative. */
-        if (out_time > priv->last_out_time)
-          duration = (out_time - priv->last_out_time) / (gap + 1);
-        else
-          goto lost;
-
-        GST_DEBUG_OBJECT (jitterbuffer, "duration %" GST_TIME_FORMAT,
-            GST_TIME_ARGS (duration));
-        /* add this duration to the timestamp of the last packet we pushed */
-        out_time = (priv->last_out_time + duration);
-      }
-    } else {
-      /* we don't know what the next_seqnum should be, wait for the last
-       * possible moment to push this buffer, maybe we get an earlier seqnum
-       * while we wait */
-      GST_DEBUG_OBJECT (jitterbuffer, "First buffer %d, do sync", seqnum);
-    }
-
-    GST_OBJECT_LOCK (jitterbuffer);
-    clock = GST_ELEMENT_CLOCK (jitterbuffer);
-    if (!clock) {
-      GST_OBJECT_UNLOCK (jitterbuffer);
-      /* let's just push if there is no clock */
-      goto push_buffer;
-    }
-
-    GST_DEBUG_OBJECT (jitterbuffer, "sync to timestamp %" GST_TIME_FORMAT,
-        GST_TIME_ARGS (out_time));
-
-    /* prepare for sync against clock */
-    sync_time = get_sync_time (jitterbuffer, out_time);
-
-    /* create an entry for the clock */
-    id = priv->clock_id = gst_clock_new_single_shot_id (clock, sync_time);
-    priv->unscheduled = FALSE;
-    GST_OBJECT_UNLOCK (jitterbuffer);
-
-    /* release the lock so that the other end can push stuff or unlock */
-    JBUF_UNLOCK (priv);
-
-    ret = gst_clock_id_wait (id, NULL);
-
-    JBUF_LOCK (priv);
-    /* and free the entry */
-    gst_clock_id_unref (id);
-    priv->clock_id = NULL;
-
-    /* at this point, the clock could have been unlocked by a timeout, a new
-     * tail element was added to the queue or because we are shutting down. Check
-     * for shutdown first. */
-    if G_UNLIKELY
-      ((priv->srcresult != GST_FLOW_OK))
-          goto flushing;
-
-    /* if we got unscheduled and we are not flushing, it's because a new tail
-     * element became available in the queue or we flushed the queue.
-     * Grab it and try to push or sync. */
-    if (ret == GST_CLOCK_UNSCHEDULED || priv->unscheduled) {
-      GST_DEBUG_OBJECT (jitterbuffer,
-          "Wait got unscheduled, will retry to push with new buffer");
-      goto again;
-    }
-
-  lost:
-    /* we now timed out, this means we lost a packet or finished synchronizing
-     * on the first buffer. */
-    if (gap > 0) {
-      GstEvent *event;
-
-      /* we had a gap and thus we lost a packet. Create an event for this.  */
-      GST_DEBUG_OBJECT (jitterbuffer, "Packet #%d lost", next_seqnum);
-      priv->num_late++;
-      discont = TRUE;
-
-      /* update our expected next packet */
-      priv->last_popped_seqnum = next_seqnum;
-      priv->last_out_time = out_time;
-      priv->next_seqnum = (next_seqnum + 1) & 0xffff;
-
-      if (priv->do_lost) {
-        /* create paket lost event */
-        event = gst_event_new_custom (GST_EVENT_CUSTOM_DOWNSTREAM,
-            gst_structure_new ("GstRTPPacketLost",
-                "seqnum", G_TYPE_UINT, (guint) next_seqnum,
-                "timestamp", G_TYPE_UINT64, out_time,
-                "duration", G_TYPE_UINT64, duration, NULL));
-
-        JBUF_UNLOCK (priv);
-        gst_pad_push_event (priv->srcpad, event);
-        JBUF_LOCK_CHECK (priv, flushing);
-      }
-      /* look for next packet */
-      goto again;
-    }
-
-    /* there was no known gap,just the first packet, exit the loop and push */
-    GST_DEBUG_OBJECT (jitterbuffer, "First packet #%d synced", seqnum);
-
-    /* get new timestamp, latency might have changed */
-    out_time = apply_offset (jitterbuffer, timestamp);
-  }
-push_buffer:
-
-  /* when we get here we are ready to pop and push the buffer */
-  outbuf = rtp_jitter_buffer_pop (priv->jbuf);
-
-  if (G_UNLIKELY (discont || priv->discont)) {
-    /* set DISCONT flag when we missed a packet. We pushed the buffer writable
-     * into the jitterbuffer so we can modify now. */
-    GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
-    priv->discont = FALSE;
-  }
-
-  /* apply timestamp with offset to buffer now */
-  GST_BUFFER_TIMESTAMP (outbuf) = out_time;
-
-  /* update the elapsed time when we need to check against the npt stop time. */
-  if (priv->npt_stop != -1 && priv->ext_timestamp != -1
-      && priv->clock_base != -1) {
-    guint64 ext_time, elapsed, estimated;
-    guint32 rtp_time;
-
-    rtp_time = gst_rtp_buffer_get_timestamp (outbuf);
-
-    GST_LOG_OBJECT (jitterbuffer, "rtp %" G_GUINT32_FORMAT ", ext %"
-        G_GUINT64_FORMAT, rtp_time, priv->ext_timestamp);
-
-    if (rtp_time < priv->ext_timestamp) {
-      ext_time = priv->ext_timestamp;
-    } else {
-      ext_time = gst_rtp_buffer_ext_timestamp (&priv->ext_timestamp, rtp_time);
-    }
-
-    if (ext_time > priv->clock_base)
-      elapsed = ext_time - priv->clock_base;
-    else
-      elapsed = 0;
-
-    elapsed = gst_util_uint64_scale_int (elapsed, GST_SECOND, priv->clock_rate);
-
-    if (elapsed > priv->last_elapsed) {
-      guint64 left;
-
-      priv->last_elapsed = elapsed;
-
-      left = priv->npt_stop - priv->npt_start;
-
-      if (elapsed > 0)
-        estimated = gst_util_uint64_scale (out_time, left, elapsed);
-      else
-        estimated = -1;
-
-      GST_LOG_OBJECT (jitterbuffer, "elapsed %" GST_TIME_FORMAT ", estimated %"
-          GST_TIME_FORMAT, GST_TIME_ARGS (elapsed), GST_TIME_ARGS (estimated));
-
-      priv->estimated_eos = estimated;
-    }
-  }
-
-  /* now we are ready to push the buffer. Save the seqnum and release the lock
-   * so the other end can push stuff in the queue again. */
-  priv->last_popped_seqnum = seqnum;
-  priv->last_out_time = out_time;
-  priv->next_seqnum = (seqnum + 1) & 0xffff;
-  JBUF_UNLOCK (priv);
-
-  /* push buffer */
-  GST_DEBUG_OBJECT (jitterbuffer,
-      "Pushing buffer %d, timestamp %" GST_TIME_FORMAT, seqnum,
-      GST_TIME_ARGS (out_time));
-  result = gst_pad_push (priv->srcpad, outbuf);
-  if (G_UNLIKELY (result != GST_FLOW_OK))
-    goto pause;
-
-  return;
-
-  /* ERRORS */
-do_eos:
-  {
-    /* store result, we are flushing now */
-    GST_DEBUG_OBJECT (jitterbuffer, "We are EOS, pushing EOS downstream");
-    priv->srcresult = GST_FLOW_UNEXPECTED;
-    gst_pad_pause_task (priv->srcpad);
-    JBUF_UNLOCK (priv);
-    gst_pad_push_event (priv->srcpad, gst_event_new_eos ());
-    return;
-  }
-do_npt_stop:
-  {
-    /* store result, we are flushing now */
-    GST_DEBUG_OBJECT (jitterbuffer, "We reached the NPT stop");
-    JBUF_UNLOCK (priv);
-
-    g_signal_emit (jitterbuffer,
-        gst_rtp_jitter_buffer_signals[SIGNAL_ON_NPT_STOP], 0, NULL);
-    return;
-  }
-flushing:
-  {
-    GST_DEBUG_OBJECT (jitterbuffer, "we are flushing");
-    gst_pad_pause_task (priv->srcpad);
-    JBUF_UNLOCK (priv);
-    return;
-  }
-pause:
-  {
-    GST_DEBUG_OBJECT (jitterbuffer, "pausing task, reason %s",
-        gst_flow_get_name (result));
-
-    JBUF_LOCK (priv);
-    /* store result */
-    priv->srcresult = result;
-    /* we don't post errors or anything because upstream will do that for us
-     * when we pass the return value upstream. */
-    gst_pad_pause_task (priv->srcpad);
-    JBUF_UNLOCK (priv);
-    return;
-  }
-}
-
-static GstFlowReturn
-gst_rtp_jitter_buffer_chain_rtcp (GstPad * pad, GstBuffer * buffer)
-{
-  GstRtpJitterBuffer *jitterbuffer;
-  GstRtpJitterBufferPrivate *priv;
-  GstFlowReturn ret = GST_FLOW_OK;
-  guint64 base_rtptime, timestamp;
-  guint32 clock_rate;
-  guint64 last_rtptime;
-  guint32 ssrc;
-  GstRTCPPacket packet;
-  guint64 ext_rtptime, diff;
-  guint32 rtptime;
-  gboolean drop = FALSE;
-
-  jitterbuffer = GST_RTP_JITTER_BUFFER (gst_pad_get_parent (pad));
-
-  if (G_UNLIKELY (!gst_rtcp_buffer_validate (buffer)))
-    goto invalid_buffer;
-
-  priv = jitterbuffer->priv;
-
-  if (!gst_rtcp_buffer_get_first_packet (buffer, &packet))
-    goto invalid_buffer;
-
-  /* first packet must be SR or RR or else the validate would have failed */
-  switch (gst_rtcp_packet_get_type (&packet)) {
-    case GST_RTCP_TYPE_SR:
-      gst_rtcp_packet_sr_get_sender_info (&packet, &ssrc, NULL, &rtptime,
-          NULL, NULL);
-      break;
-    default:
-      goto ignore_buffer;
-  }
-
-  GST_DEBUG_OBJECT (jitterbuffer, "received RTCP of SSRC %08x", ssrc);
-
-  JBUF_LOCK (priv);
-  /* convert the RTP timestamp to our extended timestamp, using the same offset
-   * we used in the jitterbuffer */
-  ext_rtptime = priv->jbuf->ext_rtptime;
-  ext_rtptime = gst_rtp_buffer_ext_timestamp (&ext_rtptime, rtptime);
-
-  /* get the last values from the jitterbuffer */
-  rtp_jitter_buffer_get_sync (priv->jbuf, &base_rtptime, &timestamp,
-      &clock_rate, &last_rtptime);
-
-  GST_DEBUG_OBJECT (jitterbuffer, "ext SR %" G_GUINT64_FORMAT ", base %"
-      G_GUINT64_FORMAT ", clock-rate %" G_GUINT32_FORMAT,
-      ext_rtptime, base_rtptime, clock_rate);
-
-  if (base_rtptime == -1 || clock_rate == -1 || timestamp == -1) {
-    GST_DEBUG_OBJECT (jitterbuffer, "dropping, no RTP values");
-    drop = TRUE;
-  } else {
-    /* we can't accept anything that happened before we did the last resync */
-    if (base_rtptime > ext_rtptime) {
-      GST_DEBUG_OBJECT (jitterbuffer, "dropping, older than base time");
-      drop = TRUE;
-    } else {
-      /* the SR RTP timestamp must be something close to what we last observed
-       * in the jitterbuffer */
-      if (ext_rtptime > last_rtptime) {
-        /* check how far ahead it is to our RTP timestamps */
-        diff = ext_rtptime - last_rtptime;
-        /* if bigger than 1 second, we drop it */
-        if (diff > clock_rate) {
-          GST_DEBUG_OBJECT (jitterbuffer, "dropping, too far ahead");
-          drop = TRUE;
-        }
-        GST_DEBUG_OBJECT (jitterbuffer, "ext last %" G_GUINT64_FORMAT ", diff %"
-            G_GUINT64_FORMAT, last_rtptime, diff);
-      }
-    }
-  }
-  JBUF_UNLOCK (priv);
-
-  if (!drop) {
-    GstStructure *s;
-
-    s = gst_structure_new ("application/x-rtp-sync",
-        "base-rtptime", G_TYPE_UINT64, base_rtptime,
-        "base-time", G_TYPE_UINT64, timestamp,
-        "clock-rate", G_TYPE_UINT, clock_rate,
-        "sr-ext-rtptime", G_TYPE_UINT64, ext_rtptime,
-        "sr-buffer", GST_TYPE_BUFFER, buffer, NULL);
-
-    GST_DEBUG_OBJECT (jitterbuffer, "signaling sync");
-    g_signal_emit (jitterbuffer,
-        gst_rtp_jitter_buffer_signals[SIGNAL_HANDLE_SYNC], 0, s);
-    gst_structure_free (s);
-  } else {
-    GST_DEBUG_OBJECT (jitterbuffer, "dropping RTCP packet");
-    ret = GST_FLOW_OK;
-  }
-
-done:
-  gst_buffer_unref (buffer);
-  gst_object_unref (jitterbuffer);
-
-  return ret;
-
-invalid_buffer:
-  {
-    /* this is not fatal but should be filtered earlier */
-    GST_ELEMENT_WARNING (jitterbuffer, STREAM, DECODE, (NULL),
-        ("Received invalid RTCP payload, dropping"));
-    ret = GST_FLOW_OK;
-    goto done;
-  }
-ignore_buffer:
-  {
-    GST_DEBUG_OBJECT (jitterbuffer, "ignoring RTCP packet");
-    ret = GST_FLOW_OK;
-    goto done;
-  }
-}
-
-static gboolean
-gst_rtp_jitter_buffer_query (GstPad * pad, GstQuery * query)
-{
-  GstRtpJitterBuffer *jitterbuffer;
-  GstRtpJitterBufferPrivate *priv;
-  gboolean res = FALSE;
-
-  jitterbuffer = GST_RTP_JITTER_BUFFER (gst_pad_get_parent (pad));
-  priv = jitterbuffer->priv;
-
-  switch (GST_QUERY_TYPE (query)) {
-    case GST_QUERY_LATENCY:
-    {
-      /* We need to send the query upstream and add the returned latency to our
-       * own */
-      GstClockTime min_latency, max_latency;
-      gboolean us_live;
-      GstClockTime our_latency;
-
-      if ((res = gst_pad_peer_query (priv->sinkpad, query))) {
-        gst_query_parse_latency (query, &us_live, &min_latency, &max_latency);
-
-        GST_DEBUG_OBJECT (jitterbuffer, "Peer latency: min %"
-            GST_TIME_FORMAT " max %" GST_TIME_FORMAT,
-            GST_TIME_ARGS (min_latency), GST_TIME_ARGS (max_latency));
-
-        /* store this so that we can safely sync on the peer buffers. */
-        JBUF_LOCK (priv);
-        priv->peer_latency = min_latency;
-        our_latency = ((guint64) priv->latency_ms) * GST_MSECOND;
-        JBUF_UNLOCK (priv);
-
-        GST_DEBUG_OBJECT (jitterbuffer, "Our latency: %" GST_TIME_FORMAT,
-            GST_TIME_ARGS (our_latency));
-
-        /* we add some latency but can buffer an infinite amount of time */
-        min_latency += our_latency;
-        max_latency = -1;
-
-        GST_DEBUG_OBJECT (jitterbuffer, "Calculated total latency : min %"
-            GST_TIME_FORMAT " max %" GST_TIME_FORMAT,
-            GST_TIME_ARGS (min_latency), GST_TIME_ARGS (max_latency));
-
-        gst_query_set_latency (query, TRUE, min_latency, max_latency);
-      }
-      break;
-    }
-    default:
-      res = gst_pad_query_default (pad, query);
-      break;
-  }
-
-  gst_object_unref (jitterbuffer);
-
-  return res;
-}
-
-static void
-gst_rtp_jitter_buffer_set_property (GObject * object,
-    guint prop_id, const GValue * value, GParamSpec * pspec)
-{
-  GstRtpJitterBuffer *jitterbuffer;
-  GstRtpJitterBufferPrivate *priv;
-
-  jitterbuffer = GST_RTP_JITTER_BUFFER (object);
-  priv = jitterbuffer->priv;
-
-  switch (prop_id) {
-    case PROP_LATENCY:
-    {
-      guint new_latency, old_latency;
-
-      new_latency = g_value_get_uint (value);
-
-      JBUF_LOCK (priv);
-      old_latency = priv->latency_ms;
-      priv->latency_ms = new_latency;
-      JBUF_UNLOCK (priv);
-
-      /* post message if latency changed, this will inform the parent pipeline
-       * that a latency reconfiguration is possible/needed. */
-      if (new_latency != old_latency) {
-        GST_DEBUG_OBJECT (jitterbuffer, "latency changed to: %" GST_TIME_FORMAT,
-            GST_TIME_ARGS (new_latency * GST_MSECOND));
-
-        gst_element_post_message (GST_ELEMENT_CAST (jitterbuffer),
-            gst_message_new_latency (GST_OBJECT_CAST (jitterbuffer)));
-      }
-      break;
-    }
-    case PROP_DROP_ON_LATENCY:
-      JBUF_LOCK (priv);
-      priv->drop_on_latency = g_value_get_boolean (value);
-      JBUF_UNLOCK (priv);
-      break;
-    case PROP_TS_OFFSET:
-      JBUF_LOCK (priv);
-      priv->ts_offset = g_value_get_int64 (value);
-      /* FIXME, we don't really have a method for signaling a timestamp
-       * DISCONT without also making this a data discont. */
-      /* priv->discont = TRUE; */
-      JBUF_UNLOCK (priv);
-      break;
-    case PROP_DO_LOST:
-      JBUF_LOCK (priv);
-      priv->do_lost = g_value_get_boolean (value);
-      JBUF_UNLOCK (priv);
-      break;
-    default:
-      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
-      break;
-  }
-}
-
-static void
-gst_rtp_jitter_buffer_get_property (GObject * object,
-    guint prop_id, GValue * value, GParamSpec * pspec)
-{
-  GstRtpJitterBuffer *jitterbuffer;
-  GstRtpJitterBufferPrivate *priv;
-
-  jitterbuffer = GST_RTP_JITTER_BUFFER (object);
-  priv = jitterbuffer->priv;
-
-  switch (prop_id) {
-    case PROP_LATENCY:
-      JBUF_LOCK (priv);
-      g_value_set_uint (value, priv->latency_ms);
-      JBUF_UNLOCK (priv);
-      break;
-    case PROP_DROP_ON_LATENCY:
-      JBUF_LOCK (priv);
-      g_value_set_boolean (value, priv->drop_on_latency);
-      JBUF_UNLOCK (priv);
-      break;
-    case PROP_TS_OFFSET:
-      JBUF_LOCK (priv);
-      g_value_set_int64 (value, priv->ts_offset);
-      JBUF_UNLOCK (priv);
-      break;
-    case PROP_DO_LOST:
-      JBUF_LOCK (priv);
-      g_value_set_boolean (value, priv->do_lost);
-      JBUF_UNLOCK (priv);
-      break;
-    default:
-      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
-      break;
-  }
-}
diff --git a/gst/rtpmanager/gstrtpjitterbuffer.h b/gst/rtpmanager/gstrtpjitterbuffer.h
deleted file mode 100644
index 6d7610e5..00000000
--- a/gst/rtpmanager/gstrtpjitterbuffer.h
+++ /dev/null
@@ -1,88 +0,0 @@
-/*
- * Farsight Voice+Video library
- *
- *  Copyright 2007 Collabora Ltd, 
- *  Copyright 2007 Nokia Corporation
- *   @author: Philippe Kalaf <philippe.kalaf@collabora.co.uk>.
- *  Copyright 2007 Wim Taymans <wim.taymans@gmail.com>
- *
- * This library is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Library General Public
- * License as published by the Free Software Foundation; either
- * version 2 of the License, or (at your option) any later version.
- *
- * This library is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Library General Public License for more details.
- *
- * You should have received a copy of the GNU Library General Public
- * License along with this library; if not, write to the
- * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
- * Boston, MA 02111-1307, USA.
- *
- */
-
-#ifndef __GST_RTP_JITTER_BUFFER_H__
-#define __GST_RTP_JITTER_BUFFER_H__
-
-#include <gst/gst.h>
-#include <gst/rtp/gstrtpbuffer.h>
-
-G_BEGIN_DECLS
-
-/* #define's don't like whitespacey bits */
-#define GST_TYPE_RTP_JITTER_BUFFER \
-  (gst_rtp_jitter_buffer_get_type())
-#define GST_RTP_JITTER_BUFFER(obj) \
-  (G_TYPE_CHECK_INSTANCE_CAST((obj), \
-  GST_TYPE_RTP_JITTER_BUFFER,GstRtpJitterBuffer))
-#define GST_RTP_JITTER_BUFFER_CLASS(klass) \
-  (G_TYPE_CHECK_CLASS_CAST((klass), \
-  GST_TYPE_RTP_JITTER_BUFFER,GstRtpJitterBufferClass))
-#define GST_IS_RTP_JITTER_BUFFER(obj) \
-  (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_RTP_JITTER_BUFFER))
-#define GST_IS_RTP_JITTER_BUFFER_CLASS(obj) \
-  (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_RTP_JITTER_BUFFER))
-
-typedef struct _GstRtpJitterBuffer GstRtpJitterBuffer;
-typedef struct _GstRtpJitterBufferClass GstRtpJitterBufferClass;
-typedef struct _GstRtpJitterBufferPrivate GstRtpJitterBufferPrivate;
-
-/**
- * GstRtpJitterBuffer:
- *
- * Opaque jitterbuffer structure.
- */
-struct _GstRtpJitterBuffer
-{
-  GstElement parent;
-
-  /*< private >*/
-  GstRtpJitterBufferPrivate *priv;
-
-  gpointer _gst_reserved[GST_PADDING];
-};
-
-struct _GstRtpJitterBufferClass
-{
-  GstElementClass parent_class;
-
-  /* signals */
-  GstCaps* (*request_pt_map)   (GstRtpJitterBuffer *buffer, guint pt);
-
-  void     (*handle_sync)      (GstRtpJitterBuffer *buffer, GstStructure *s);
-  void     (*on_npt_stop)      (GstRtpJitterBuffer *buffer);
-
-  /* actions */
-  void     (*clear_pt_map)   (GstRtpJitterBuffer *buffer);
-
-  /*< private > */
-  gpointer _gst_reserved[GST_PADDING];
-};
-
-GType gst_rtp_jitter_buffer_get_type (void);
-
-G_END_DECLS
-
-#endif /* __GST_RTP_JITTER_BUFFER_H__ */
diff --git a/gst/rtpmanager/gstrtpmanager.c b/gst/rtpmanager/gstrtpmanager.c
deleted file mode 100644
index f38a77a8..00000000
--- a/gst/rtpmanager/gstrtpmanager.c
+++ /dev/null
@@ -1,60 +0,0 @@
-/* GStreamer
- * Copyright (C) <2007> Wim Taymans <wim.taymans@gmail.com>
- *
- * This library is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Library General Public
- * License as published by the Free Software Foundation; either
- * version 2 of the License, or (at your option) any later version.
- *
- * This library is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Library General Public License for more details.
- *
- * You should have received a copy of the GNU Library General Public
- * License along with this library; if not, write to the
- * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
- * Boston, MA 02111-1307, USA.
- */
-
-#ifdef HAVE_CONFIG_H
-#include "config.h"
-#endif
-
-#include "gstrtpbin.h"
-#include "gstrtpjitterbuffer.h"
-#include "gstrtpptdemux.h"
-#include "gstrtpsession.h"
-#include "gstrtpssrcdemux.h"
-
-static gboolean
-plugin_init (GstPlugin * plugin)
-{
-  if (!gst_element_register (plugin, "gstrtpbin", GST_RANK_NONE,
-          GST_TYPE_RTP_BIN))
-    return FALSE;
-
-  if (!gst_element_register (plugin, "gstrtpjitterbuffer", GST_RANK_NONE,
-          GST_TYPE_RTP_JITTER_BUFFER))
-    return FALSE;
-
-  if (!gst_element_register (plugin, "gstrtpptdemux", GST_RANK_NONE,
-          GST_TYPE_RTP_PT_DEMUX))
-    return FALSE;
-
-  if (!gst_element_register (plugin, "gstrtpsession", GST_RANK_NONE,
-          GST_TYPE_RTP_SESSION))
-    return FALSE;
-
-  if (!gst_element_register (plugin, "gstrtpssrcdemux", GST_RANK_NONE,
-          GST_TYPE_RTP_SSRC_DEMUX))
-    return FALSE;
-
-  return TRUE;
-}
-
-GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
-    GST_VERSION_MINOR,
-    "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
deleted file mode 100644
index 6e34705e..00000000
--- a/gst/rtpmanager/gstrtpptdemux.c
+++ /dev/null
@@ -1,492 +0,0 @@
-/* 
- * RTP Demux element
- *
- * Copyright (C) 2005 Nokia Corporation.
- * @author Kai Vehmanen <kai.vehmanen@nokia.com>
- *
- * Loosely based on GStreamer gstdecodebin
- * Copyright (C) <2004> Wim Taymans <wim.taymans@gmail.com>
- *
- * This library is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Library General Public
- * License as published by the Free Software Foundation; either
- * version 2 of the License, or (at your option) any later version.
- *
- * This library is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Library General Public License for more details.
- *
- * You should have received a copy of the GNU Library General Public
- * License along with this library; if not, write to the
- * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
- * Boston, MA 02111-1307, USA.
- */
-
-/**
- * SECTION:element-gstrtpptdemux
- *
- * 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.
- * 
- * For each payload type that is detected, a new pad will be created and the
- * #GstRtpPtDemux::new-payload-type signal will be emitted. When the payload for
- * the RTP stream changes, the #GstRtpPtDemux::payload-type-change signal will be
- * emitted.
- * 
- * The element will try to set complete and unique application/x-rtp caps on the
- * outgoing buffers and pads based on the result of the
- * #GstRtpPtDemux::request-pt-map signal.
- * 
- * <refsect2>
- * <title>Example pipelines</title>
- * |[
- * gst-launch udpsrc caps="application/x-rtp" ! gstrtpptdemux ! fakesink
- * ]| Takes an RTP stream and send the RTP packets with the first detected
- * payload type to fakesink, discarding the other payload types.
- * </refsect2>
- *
- * Last reviewed on 2007-05-28 (0.10.5)
- */
-
-/*
- * Contributors:
- * Andre Moreira Magalhaes <andre.magalhaes@indt.org.br>
- */
-/*
- * Status:
- *  - works with the test_rtpdemux.c tool
- *
- * Check:
- *  - is emitting a signal enough, or should we
- *    use GstEvent to notify downstream elements
- *    of the new packet... no?
- *
- * Notes:
- *  - emits event both for new PTs, and whenever
- *    a PT is changed
- */
-
-#ifdef HAVE_CONFIG_H
-#include "config.h"
-#endif
-
-#include <string.h>
-#include <gst/gst.h>
-#include <gst/rtp/gstrtpbuffer.h>
-
-#include "gstrtpbin-marshal.h"
-#include "gstrtpptdemux.h"
-
-/* generic templates */
-static GstStaticPadTemplate rtp_pt_demux_sink_template =
-GST_STATIC_PAD_TEMPLATE ("sink",
-    GST_PAD_SINK,
-    GST_PAD_ALWAYS,
-    GST_STATIC_CAPS ("application/x-rtp")
-    );
-
-static GstStaticPadTemplate rtp_pt_demux_src_template =
-GST_STATIC_PAD_TEMPLATE ("src_%d",
-    GST_PAD_SRC,
-    GST_PAD_SOMETIMES,
-    GST_STATIC_CAPS ("application/x-rtp, " "payload = (int) [ 0, 255 ]")
-    );
-
-GST_DEBUG_CATEGORY_STATIC (gst_rtp_pt_demux_debug);
-#define GST_CAT_DEFAULT gst_rtp_pt_demux_debug
-
-/**
- * Item for storing GstPad<->pt pairs.
- */
-struct _GstRtpPtDemuxPad
-{
-  GstPad *pad;        /**< pointer to the actual pad */
-  gint pt;             /**< RTP payload-type attached to pad */
-  gboolean newcaps;
-};
-
-/* signals */
-enum
-{
-  SIGNAL_REQUEST_PT_MAP,
-  SIGNAL_NEW_PAYLOAD_TYPE,
-  SIGNAL_PAYLOAD_TYPE_CHANGE,
-  SIGNAL_CLEAR_PT_MAP,
-  LAST_SIGNAL
-};
-
-GST_BOILERPLATE (GstRtpPtDemux, gst_rtp_pt_demux, GstElement, GST_TYPE_ELEMENT);
-
-static void gst_rtp_pt_demux_finalize (GObject * object);
-
-static void gst_rtp_pt_demux_release (GstRtpPtDemux * ptdemux);
-static gboolean gst_rtp_pt_demux_setup (GstRtpPtDemux * ptdemux);
-
-static GstFlowReturn gst_rtp_pt_demux_chain (GstPad * pad, GstBuffer * buf);
-static GstStateChangeReturn gst_rtp_pt_demux_change_state (GstElement * element,
-    GstStateChange transition);
-static void gst_rtp_pt_demux_clear_pt_map (GstRtpPtDemux * rtpdemux);
-
-static GstRtpPtDemuxPad *find_pad_for_pt (GstRtpPtDemux * rtpdemux, guint8 pt);
-
-static guint gst_rtp_pt_demux_signals[LAST_SIGNAL] = { 0 };
-
-static GstElementDetails gst_rtp_pt_demux_details = {
-  "RTP Demux",
-  "Demux/Network/RTP",
-  "Parses codec streams transmitted in the same RTP session",
-  "Kai Vehmanen <kai.vehmanen@nokia.com>"
-};
-
-static void
-gst_rtp_pt_demux_base_init (gpointer g_class)
-{
-  GstElementClass *gstelement_klass = GST_ELEMENT_CLASS (g_class);
-
-  gst_element_class_add_pad_template (gstelement_klass,
-      gst_static_pad_template_get (&rtp_pt_demux_sink_template));
-  gst_element_class_add_pad_template (gstelement_klass,
-      gst_static_pad_template_get (&rtp_pt_demux_src_template));
-
-  gst_element_class_set_details (gstelement_klass, &gst_rtp_pt_demux_details);
-}
-
-static void
-gst_rtp_pt_demux_class_init (GstRtpPtDemuxClass * klass)
-{
-  GObjectClass *gobject_klass;
-  GstElementClass *gstelement_klass;
-
-  gobject_klass = (GObjectClass *) klass;
-  gstelement_klass = (GstElementClass *) klass;
-
-  /**
-   * GstRtpPtDemux::request-pt-map:
-   * @demux: the object which received the signal
-   * @pt: the payload type
-   *
-   * Request the payload type as #GstCaps for @pt.
-   */
-  gst_rtp_pt_demux_signals[SIGNAL_REQUEST_PT_MAP] =
-      g_signal_new ("request-pt-map", G_TYPE_FROM_CLASS (klass),
-      G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpPtDemuxClass, request_pt_map),
-      NULL, NULL, gst_rtp_bin_marshal_BOXED__UINT, GST_TYPE_CAPS, 1,
-      G_TYPE_UINT);
-
-  /**
-   * GstRtpPtDemux::new-payload-type:
-   * @demux: the object which received the signal
-   * @pt: the payload type
-   * @pad: the pad with the new payload
-   *
-   * Emited when a new payload type pad has been created in @demux.
-   */
-  gst_rtp_pt_demux_signals[SIGNAL_NEW_PAYLOAD_TYPE] =
-      g_signal_new ("new-payload-type", G_TYPE_FROM_CLASS (klass),
-      G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpPtDemuxClass, new_payload_type),
-      NULL, NULL, gst_rtp_bin_marshal_VOID__UINT_OBJECT, G_TYPE_NONE, 2,
-      G_TYPE_UINT, GST_TYPE_PAD);
-
-  /**
-   * GstRtpPtDemux::payload-type-change:
-   * @demux: the object which received the signal
-   * @pt: the new payload type
-   *
-   * Emited when the payload type changed.
-   */
-  gst_rtp_pt_demux_signals[SIGNAL_PAYLOAD_TYPE_CHANGE] =
-      g_signal_new ("payload-type-change", G_TYPE_FROM_CLASS (klass),
-      G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpPtDemuxClass,
-          payload_type_change), NULL, NULL, g_cclosure_marshal_VOID__UINT,
-      G_TYPE_NONE, 1, G_TYPE_UINT);
-
-  /**
-   * GstRtpPtDemux::clear-pt-map:
-   * @demux: the object which received the signal
-   *
-   * The application can call this signal to instruct the element to discard the
-   * currently cached payload type map.
-   */
-  gst_rtp_pt_demux_signals[SIGNAL_CLEAR_PT_MAP] =
-      g_signal_new ("clear-pt-map", G_TYPE_FROM_CLASS (klass),
-      G_SIGNAL_ACTION | G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpPtDemuxClass,
-          clear_pt_map), NULL, NULL, g_cclosure_marshal_VOID__VOID,
-      G_TYPE_NONE, 0, G_TYPE_NONE);
-
-  gobject_klass->finalize = GST_DEBUG_FUNCPTR (gst_rtp_pt_demux_finalize);
-
-  gstelement_klass->change_state =
-      GST_DEBUG_FUNCPTR (gst_rtp_pt_demux_change_state);
-
-  klass->clear_pt_map = GST_DEBUG_FUNCPTR (gst_rtp_pt_demux_clear_pt_map);
-
-  GST_DEBUG_CATEGORY_INIT (gst_rtp_pt_demux_debug,
-      "rtpptdemux", 0, "RTP codec demuxer");
-}
-
-static void
-gst_rtp_pt_demux_init (GstRtpPtDemux * ptdemux, GstRtpPtDemuxClass * g_class)
-{
-  GstElementClass *klass = GST_ELEMENT_GET_CLASS (ptdemux);
-
-  ptdemux->sink =
-      gst_pad_new_from_template (gst_element_class_get_pad_template (klass,
-          "sink"), "sink");
-  g_assert (ptdemux->sink != NULL);
-
-  gst_pad_set_chain_function (ptdemux->sink, gst_rtp_pt_demux_chain);
-
-  gst_element_add_pad (GST_ELEMENT (ptdemux), ptdemux->sink);
-}
-
-static void
-gst_rtp_pt_demux_finalize (GObject * object)
-{
-  gst_rtp_pt_demux_release (GST_RTP_PT_DEMUX (object));
-
-  G_OBJECT_CLASS (parent_class)->finalize (object);
-}
-
-static GstCaps *
-gst_rtp_pt_demux_get_caps (GstRtpPtDemux * rtpdemux, guint pt)
-{
-  GstCaps *caps;
-  GValue ret = { 0 };
-  GValue args[2] = { {0}, {0} };
-
-  /* figure out the caps */
-  g_value_init (&args[0], GST_TYPE_ELEMENT);
-  g_value_set_object (&args[0], rtpdemux);
-  g_value_init (&args[1], G_TYPE_UINT);
-  g_value_set_uint (&args[1], pt);
-
-  g_value_init (&ret, GST_TYPE_CAPS);
-  g_value_set_boxed (&ret, NULL);
-
-  g_signal_emitv (args, gst_rtp_pt_demux_signals[SIGNAL_REQUEST_PT_MAP], 0,
-      &ret);
-
-  g_value_unset (&args[0]);
-  g_value_unset (&args[1]);
-  caps = g_value_dup_boxed (&ret);
-  g_value_unset (&ret);
-  if (caps == NULL) {
-    caps = GST_PAD_CAPS (rtpdemux->sink);
-    if (caps)
-      gst_caps_ref (caps);
-  }
-
-  GST_DEBUG ("pt %d, got caps %" GST_PTR_FORMAT, pt, caps);
-
-  return caps;
-}
-
-static void
-gst_rtp_pt_demux_clear_pt_map (GstRtpPtDemux * rtpdemux)
-{
-  GSList *walk;
-
-  GST_OBJECT_LOCK (rtpdemux);
-  GST_DEBUG ("clearing pt map");
-  for (walk = rtpdemux->srcpads; walk; walk = g_slist_next (walk)) {
-    GstRtpPtDemuxPad *pad = walk->data;
-
-    pad->newcaps = TRUE;
-  }
-  GST_OBJECT_UNLOCK (rtpdemux);
-}
-
-static GstFlowReturn
-gst_rtp_pt_demux_chain (GstPad * pad, GstBuffer * buf)
-{
-  GstFlowReturn ret = GST_FLOW_OK;
-  GstRtpPtDemux *rtpdemux;
-  GstElement *element = GST_ELEMENT (GST_OBJECT_PARENT (pad));
-  guint8 pt;
-  GstPad *srcpad;
-  GstRtpPtDemuxPad *rtpdemuxpad;
-  GstCaps *caps;
-
-  rtpdemux = GST_RTP_PT_DEMUX (GST_OBJECT_PARENT (pad));
-
-  if (!gst_rtp_buffer_validate (buf))
-    goto invalid_buffer;
-
-  pt = gst_rtp_buffer_get_payload_type (buf);
-
-  GST_DEBUG_OBJECT (rtpdemux, "received buffer for pt %d", pt);
-
-  rtpdemuxpad = find_pad_for_pt (rtpdemux, pt);
-  if (rtpdemuxpad == NULL) {
-    /* new PT, create a src pad */
-    GstElementClass *klass;
-    GstPadTemplate *templ;
-    gchar *padname;
-
-    klass = GST_ELEMENT_GET_CLASS (rtpdemux);
-    templ = gst_element_class_get_pad_template (klass, "src_%d");
-    padname = g_strdup_printf ("src_%d", pt);
-    srcpad = gst_pad_new_from_template (templ, padname);
-    gst_pad_use_fixed_caps (srcpad);
-    g_free (padname);
-
-    caps = gst_rtp_pt_demux_get_caps (rtpdemux, pt);
-    if (!caps)
-      goto no_caps;
-
-    caps = gst_caps_make_writable (caps);
-    gst_caps_set_simple (caps, "payload", G_TYPE_INT, pt, NULL);
-    gst_pad_set_caps (srcpad, caps);
-    gst_caps_unref (caps);
-
-    GST_DEBUG ("Adding pt=%d to the list.", pt);
-    rtpdemuxpad = g_new0 (GstRtpPtDemuxPad, 1);
-    rtpdemuxpad->pt = pt;
-    rtpdemuxpad->newcaps = FALSE;
-    rtpdemuxpad->pad = srcpad;
-    GST_OBJECT_LOCK (rtpdemux);
-    rtpdemux->srcpads = g_slist_append (rtpdemux->srcpads, rtpdemuxpad);
-    GST_OBJECT_UNLOCK (rtpdemux);
-
-    gst_pad_set_active (srcpad, TRUE);
-    gst_element_add_pad (element, srcpad);
-
-    GST_DEBUG ("emitting new-payload-type for pt %d", pt);
-    g_signal_emit (G_OBJECT (rtpdemux),
-        gst_rtp_pt_demux_signals[SIGNAL_NEW_PAYLOAD_TYPE], 0, pt, srcpad);
-  }
-
-  srcpad = rtpdemuxpad->pad;
-
-  if (pt != rtpdemux->last_pt) {
-    gint emit_pt = pt;
-
-    /* our own signal with an extra flag that this is the only pad */
-    rtpdemux->last_pt = pt;
-    GST_DEBUG ("emitting payload-type-changed for pt %d", emit_pt);
-    g_signal_emit (G_OBJECT (rtpdemux),
-        gst_rtp_pt_demux_signals[SIGNAL_PAYLOAD_TYPE_CHANGE], 0, emit_pt);
-  }
-
-  if (rtpdemuxpad->newcaps) {
-    GST_DEBUG ("need new caps");
-    caps = gst_rtp_pt_demux_get_caps (rtpdemux, pt);
-    if (!caps)
-      goto no_caps;
-
-    caps = gst_caps_make_writable (caps);
-    gst_caps_set_simple (caps, "payload", G_TYPE_INT, pt, NULL);
-    gst_pad_set_caps (srcpad, caps);
-    gst_caps_unref (caps);
-    rtpdemuxpad->newcaps = FALSE;
-  }
-
-  gst_buffer_set_caps (buf, GST_PAD_CAPS (srcpad));
-
-  /* push to srcpad */
-  ret = gst_pad_push (srcpad, buf);
-
-  return ret;
-
-  /* ERRORS */
-invalid_buffer:
-  {
-    /* this is fatal and should be filtered earlier */
-    GST_ELEMENT_ERROR (rtpdemux, STREAM, DECODE, (NULL),
-        ("Dropping invalid RTP payload"));
-    gst_buffer_unref (buf);
-    return GST_FLOW_ERROR;
-  }
-no_caps:
-  {
-    GST_ELEMENT_ERROR (rtpdemux, STREAM, DECODE, (NULL),
-        ("Could not get caps for payload"));
-    gst_buffer_unref (buf);
-    return GST_FLOW_ERROR;
-  }
-}
-
-static GstRtpPtDemuxPad *
-find_pad_for_pt (GstRtpPtDemux * rtpdemux, guint8 pt)
-{
-  GstRtpPtDemuxPad *respad = NULL;
-  GSList *walk;
-
-  for (walk = rtpdemux->srcpads; walk; walk = g_slist_next (walk)) {
-    GstRtpPtDemuxPad *pad = walk->data;
-
-    if (pad->pt == pt) {
-      respad = pad;
-      break;
-    }
-  }
-  return respad;
-}
-
-/**
- * Reserves resources for the object.
- */
-static gboolean
-gst_rtp_pt_demux_setup (GstRtpPtDemux * ptdemux)
-{
-  ptdemux->srcpads = NULL;
-  ptdemux->last_pt = 0xFFFF;
-
-  return TRUE;
-}
-
-/**
- * Free resources for the object.
- */
-static void
-gst_rtp_pt_demux_release (GstRtpPtDemux * ptdemux)
-{
-  GSList *walk;
-
-  for (walk = ptdemux->srcpads; walk; walk = g_slist_next (walk)) {
-    GstRtpPtDemuxPad *pad = walk->data;
-
-    gst_pad_set_active (pad->pad, FALSE);
-    gst_element_remove_pad (GST_ELEMENT_CAST (ptdemux), pad->pad);
-    g_free (pad);
-  }
-  g_slist_free (ptdemux->srcpads);
-  ptdemux->srcpads = NULL;
-}
-
-static GstStateChangeReturn
-gst_rtp_pt_demux_change_state (GstElement * element, GstStateChange transition)
-{
-  GstStateChangeReturn ret;
-  GstRtpPtDemux *ptdemux;
-
-  ptdemux = GST_RTP_PT_DEMUX (element);
-
-  switch (transition) {
-    case GST_STATE_CHANGE_NULL_TO_READY:
-      if (gst_rtp_pt_demux_setup (ptdemux) != TRUE)
-        ret = GST_STATE_CHANGE_FAILURE;
-      break;
-    case GST_STATE_CHANGE_READY_TO_PAUSED:
-    case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
-    default:
-      break;
-  }
-
-  ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
-
-  switch (transition) {
-    case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
-    case GST_STATE_CHANGE_PAUSED_TO_READY:
-      break;
-    case GST_STATE_CHANGE_READY_TO_NULL:
-      gst_rtp_pt_demux_release (ptdemux);
-      break;
-    default:
-      break;
-  }
-
-  return ret;
-}
diff --git a/gst/rtpmanager/gstrtpptdemux.h b/gst/rtpmanager/gstrtpptdemux.h
deleted file mode 100644
index 028c97d4..00000000
--- a/gst/rtpmanager/gstrtpptdemux.h
+++ /dev/null
@@ -1,62 +0,0 @@
-/* GStreamer
- * Copyright (C) <2007> Wim Taymans <wim.taymans@gmail.com>
- *
- * This library is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Library General Public
- * License as published by the Free Software Foundation; either
- * version 2 of the License, or (at your option) any later version.
- *
- * This library is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Library General Public License for more details.
- *
- * You should have received a copy of the GNU Library General Public
- * License along with this library; if not, write to the
- * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
- * Boston, MA 02111-1307, USA.
- */
-
-#ifndef __GST_RTP_PT_DEMUX_H__
-#define __GST_RTP_PT_DEMUX_H__
-
-#include <gst/gst.h>
-
-#define GST_TYPE_RTP_PT_DEMUX            (gst_rtp_pt_demux_get_type())
-#define GST_RTP_PT_DEMUX(obj)            (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_RTP_PT_DEMUX,GstRtpPtDemux))
-#define GST_RTP_PT_DEMUX_CLASS(klass)    (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_RTP_PT_DEMUX,GstRtpPtDemuxClass))
-#define GST_IS_RTP_PT_DEMUX(obj)         (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_RTP_PT_DEMUX))
-#define GST_IS_RTP_PT_DEMUX_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_RTP_PT_DEMUX))
-
-typedef struct _GstRtpPtDemux GstRtpPtDemux;
-typedef struct _GstRtpPtDemuxClass GstRtpPtDemuxClass;
-typedef struct _GstRtpPtDemuxPad GstRtpPtDemuxPad;
-
-struct _GstRtpPtDemux
-{
-  GstElement parent;  /**< parent class */
-
-  GstPad *sink;       /**< the sink pad */
-  guint16 last_pt;    /**< pt of the last packet 0xFFFF if none */
-  GSList *srcpads;    /**< a linked list of GstRtpPtDemuxPad objects */
-};
-
-struct _GstRtpPtDemuxClass
-{
-  GstElementClass parent_class;
-
-  /* get the caps for pt */
-  GstCaps* (*request_pt_map)      (GstRtpPtDemux *demux, guint pt);
-
-  /* signal emmited when a new PT is found from the incoming stream */
-  void     (*new_payload_type)    (GstRtpPtDemux *demux, guint pt, GstPad * pad);
-
-  /* signal emitted when the payload type changes */
-  void     (*payload_type_change) (GstRtpPtDemux *demux, guint pt);
-
-  void     (*clear_pt_map)        (GstRtpPtDemux *demux);
-};
-
-GType gst_rtp_pt_demux_get_type (void);
-
-#endif /* __GST_RTP_PT_DEMUX_H__ */
diff --git a/gst/rtpmanager/gstrtpsession.c b/gst/rtpmanager/gstrtpsession.c
deleted file mode 100644
index dcddb689..00000000
--- a/gst/rtpmanager/gstrtpsession.c
+++ /dev/null
@@ -1,1940 +0,0 @@
-/* GStreamer
- * Copyright (C) <2007> Wim Taymans <wim.taymans@gmail.com>
- *
- * This library is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Library General Public
- * License as published by the Free Software Foundation; either
- * version 2 of the License, or (at your option) any later version.
- *
- * This library is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Library General Public License for more details.
- *
- * You should have received a copy of the GNU Library General Public
- * License along with this library; if not, write to the
- * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
- * Boston, MA 02111-1307, USA.
- */
-
-/**
- * SECTION:element-gstrtpsession
- * @see_also: gstrtpjitterbuffer, gstrtpbin, gstrtpptdemux, gstrtpssrcdemux
- *
- * 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.
- * Based on what REQUEST pads are requested from the session manager, specific
- * functionality can be activated.
- * 
- * The session manager currently implements RFC 3550 including:
- * <itemizedlist>
- *   <listitem>
- *     <para>RTP packet validation based on consecutive sequence numbers.</para>
- *   </listitem>
- *   <listitem>
- *     <para>Maintainance of the SSRC participant database.</para>
- *   </listitem>
- *   <listitem>
- *     <para>Keeping per participant statistics based on received RTCP packets.</para>
- *   </listitem>
- *   <listitem>
- *     <para>Scheduling of RR/SR RTCP packets.</para>
- *   </listitem>
- * </itemizedlist>
- * 
- * 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.
- * 
- * 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.
- * 
- * 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
- * so that they can be used to perform inter-stream synchronisation when needed.
- * 
- * If you want the session manager to generate and send RTCP packets, request
- * the send_rtcp_src pad. Packet pushed on this pad contain SR/RR RTCP reports
- * that should be sent to all participants in the session.
- * 
- * 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.
- * 
- * The session manager needs the clock-rate of the payload types it is handling
- * and will signal the #GstRtpSession::request-pt-map signal when it needs such a
- * mapping. One can clear the cached values with the #GstRtpSession::clear-pt-map
- * signal.
- * 
- * <refsect2>
- * <title>Example pipelines</title>
- * |[
- * gst-launch udpsrc port=5000 caps="application/x-rtp, ..." ! .recv_rtp_sink gstrtpsession .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 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
- * ]| Receive theora RTP packets from port 5000 and send them to the depayloader,
- * decoder and display. Receive RTCP packets from port 5001 and process them in
- * the session manager.
- * 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 videotestsrc ! theoraenc ! rtptheorapay ! .send_rtp_sink gstrtpsession .send_rtp_src ! udpsink port=5000
- * ]| Send theora RTP packets through the session manager and out on UDP port
- * 5000.
- * |[
- * gst-launch videotestsrc ! theoraenc ! rtptheorapay ! .send_rtp_sink gstrtpsession 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
- * 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.
- * </refsect2>
- *
- * Last reviewed on 2007-05-28 (0.10.5)
- */
-
-#ifdef HAVE_CONFIG_H
-#include "config.h"
-#endif
-
-#include <gst/rtp/gstrtpbuffer.h>
-
-#include "gstrtpbin-marshal.h"
-#include "gstrtpsession.h"
-#include "rtpsession.h"
-
-GST_DEBUG_CATEGORY_STATIC (gst_rtp_session_debug);
-#define GST_CAT_DEFAULT gst_rtp_session_debug
-
-/* elementfactory information */
-static const GstElementDetails rtpsession_details =
-GST_ELEMENT_DETAILS ("RTP Session",
-    "Filter/Network/RTP",
-    "Implement an RTP session",
-    "Wim Taymans <wim.taymans@gmail.com>");
-
-/* sink pads */
-static GstStaticPadTemplate rtpsession_recv_rtp_sink_template =
-GST_STATIC_PAD_TEMPLATE ("recv_rtp_sink",
-    GST_PAD_SINK,
-    GST_PAD_REQUEST,
-    GST_STATIC_CAPS ("application/x-rtp")
-    );
-
-static GstStaticPadTemplate rtpsession_recv_rtcp_sink_template =
-GST_STATIC_PAD_TEMPLATE ("recv_rtcp_sink",
-    GST_PAD_SINK,
-    GST_PAD_REQUEST,
-    GST_STATIC_CAPS ("application/x-rtcp")
-    );
-
-static GstStaticPadTemplate rtpsession_send_rtp_sink_template =
-GST_STATIC_PAD_TEMPLATE ("send_rtp_sink",
-    GST_PAD_SINK,
-    GST_PAD_REQUEST,
-    GST_STATIC_CAPS ("application/x-rtp")
-    );
-
-/* src pads */
-static GstStaticPadTemplate rtpsession_recv_rtp_src_template =
-GST_STATIC_PAD_TEMPLATE ("recv_rtp_src",
-    GST_PAD_SRC,
-    GST_PAD_SOMETIMES,
-    GST_STATIC_CAPS ("application/x-rtp")
-    );
-
-static GstStaticPadTemplate rtpsession_sync_src_template =
-GST_STATIC_PAD_TEMPLATE ("sync_src",
-    GST_PAD_SRC,
-    GST_PAD_SOMETIMES,
-    GST_STATIC_CAPS ("application/x-rtcp")
-    );
-
-static GstStaticPadTemplate rtpsession_send_rtp_src_template =
-GST_STATIC_PAD_TEMPLATE ("send_rtp_src",
-    GST_PAD_SRC,
-    GST_PAD_SOMETIMES,
-    GST_STATIC_CAPS ("application/x-rtp")
-    );
-
-static GstStaticPadTemplate rtpsession_send_rtcp_src_template =
-GST_STATIC_PAD_TEMPLATE ("send_rtcp_src",
-    GST_PAD_SRC,
-    GST_PAD_REQUEST,
-    GST_STATIC_CAPS ("application/x-rtcp")
-    );
-
-/* signals and args */
-enum
-{
-  SIGNAL_REQUEST_PT_MAP,
-  SIGNAL_CLEAR_PT_MAP,
-
-  SIGNAL_ON_NEW_SSRC,
-  SIGNAL_ON_SSRC_COLLISION,
-  SIGNAL_ON_SSRC_VALIDATED,
-  SIGNAL_ON_SSRC_ACTIVE,
-  SIGNAL_ON_SSRC_SDES,
-  SIGNAL_ON_BYE_SSRC,
-  SIGNAL_ON_BYE_TIMEOUT,
-  SIGNAL_ON_TIMEOUT,
-  SIGNAL_ON_SENDER_TIMEOUT,
-  LAST_SIGNAL
-};
-
-#define DEFAULT_NTP_NS_BASE          0
-#define DEFAULT_BANDWIDTH            RTP_STATS_BANDWIDTH
-#define DEFAULT_RTCP_FRACTION        RTP_STATS_RTCP_BANDWIDTH
-#define DEFAULT_SDES                 NULL
-#define DEFAULT_NUM_SOURCES          0
-#define DEFAULT_NUM_ACTIVE_SOURCES   0
-
-enum
-{
-  PROP_0,
-  PROP_NTP_NS_BASE,
-  PROP_BANDWIDTH,
-  PROP_RTCP_FRACTION,
-  PROP_SDES,
-  PROP_NUM_SOURCES,
-  PROP_NUM_ACTIVE_SOURCES,
-  PROP_INTERNAL_SESSION,
-  PROP_LAST
-};
-
-#define GST_RTP_SESSION_GET_PRIVATE(obj)  \
-	   (G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_RTP_SESSION, GstRtpSessionPrivate))
-
-#define GST_RTP_SESSION_LOCK(sess)   g_mutex_lock ((sess)->priv->lock)
-#define GST_RTP_SESSION_UNLOCK(sess) g_mutex_unlock ((sess)->priv->lock)
-
-struct _GstRtpSessionPrivate
-{
-  GMutex *lock;
-  GstClock *sysclock;
-
-  RTPSession *session;
-
-  /* thread for sending out RTCP */
-  GstClockID id;
-  gboolean stop_thread;
-  GThread *thread;
-  gboolean thread_stopped;
-
-  /* caps mapping */
-  GHashTable *ptmap;
-
-  /* NTP base time */
-  guint64 ntpnsbase;
-};
-
-/* callbacks to handle actions from the session manager */
-static GstFlowReturn gst_rtp_session_process_rtp (RTPSession * sess,
-    RTPSource * src, GstBuffer * buffer, gpointer user_data);
-static GstFlowReturn gst_rtp_session_send_rtp (RTPSession * sess,
-    RTPSource * src, gpointer data, gpointer user_data);
-static GstFlowReturn gst_rtp_session_send_rtcp (RTPSession * sess,
-    RTPSource * src, GstBuffer * buffer, gboolean eos, gpointer user_data);
-static GstFlowReturn gst_rtp_session_sync_rtcp (RTPSession * sess,
-    RTPSource * src, GstBuffer * buffer, gpointer user_data);
-static gint gst_rtp_session_clock_rate (RTPSession * sess, guint8 payload,
-    gpointer user_data);
-static void gst_rtp_session_reconsider (RTPSession * sess, gpointer user_data);
-
-static RTPSessionCallbacks callbacks = {
-  gst_rtp_session_process_rtp,
-  gst_rtp_session_send_rtp,
-  gst_rtp_session_sync_rtcp,
-  gst_rtp_session_send_rtcp,
-  gst_rtp_session_clock_rate,
-  gst_rtp_session_reconsider
-};
-
-/* GObject vmethods */
-static void gst_rtp_session_finalize (GObject * object);
-static void gst_rtp_session_set_property (GObject * object, guint prop_id,
-    const GValue * value, GParamSpec * pspec);
-static void gst_rtp_session_get_property (GObject * object, guint prop_id,
-    GValue * value, GParamSpec * pspec);
-
-/* GstElement vmethods */
-static GstStateChangeReturn gst_rtp_session_change_state (GstElement * element,
-    GstStateChange transition);
-static GstPad *gst_rtp_session_request_new_pad (GstElement * element,
-    GstPadTemplate * templ, const gchar * name);
-static void gst_rtp_session_release_pad (GstElement * element, GstPad * pad);
-
-static void gst_rtp_session_clear_pt_map (GstRtpSession * rtpsession);
-
-static guint gst_rtp_session_signals[LAST_SIGNAL] = { 0 };
-
-static void
-on_new_ssrc (RTPSession * session, RTPSource * src, GstRtpSession * sess)
-{
-  g_signal_emit (sess, gst_rtp_session_signals[SIGNAL_ON_NEW_SSRC], 0,
-      src->ssrc);
-}
-
-static void
-on_ssrc_collision (RTPSession * session, RTPSource * src, GstRtpSession * sess)
-{
-  g_signal_emit (sess, gst_rtp_session_signals[SIGNAL_ON_SSRC_COLLISION], 0,
-      src->ssrc);
-}
-
-static void
-on_ssrc_validated (RTPSession * session, RTPSource * src, GstRtpSession * sess)
-{
-  g_signal_emit (sess, gst_rtp_session_signals[SIGNAL_ON_SSRC_VALIDATED], 0,
-      src->ssrc);
-}
-
-static void
-on_ssrc_active (RTPSession * session, RTPSource * src, GstRtpSession * sess)
-{
-  g_signal_emit (sess, gst_rtp_session_signals[SIGNAL_ON_SSRC_ACTIVE], 0,
-      src->ssrc);
-}
-
-static void
-on_ssrc_sdes (RTPSession * session, RTPSource * src, GstRtpSession * sess)
-{
-  GstStructure *s;
-  GstMessage *m;
-
-  /* convert the new SDES info into a message */
-  RTP_SESSION_LOCK (session);
-  g_object_get (src, "sdes", &s, NULL);
-  RTP_SESSION_UNLOCK (session);
-
-  m = gst_message_new_custom (GST_MESSAGE_ELEMENT, GST_OBJECT (sess), s);
-  gst_element_post_message (GST_ELEMENT_CAST (sess), m);
-
-  g_signal_emit (sess, gst_rtp_session_signals[SIGNAL_ON_SSRC_SDES], 0,
-      src->ssrc);
-}
-
-static void
-on_bye_ssrc (RTPSession * session, RTPSource * src, GstRtpSession * sess)
-{
-  g_signal_emit (sess, gst_rtp_session_signals[SIGNAL_ON_BYE_SSRC], 0,
-      src->ssrc);
-}
-
-static void
-on_bye_timeout (RTPSession * session, RTPSource * src, GstRtpSession * sess)
-{
-  g_signal_emit (sess, gst_rtp_session_signals[SIGNAL_ON_BYE_TIMEOUT], 0,
-      src->ssrc);
-}
-
-static void
-on_timeout (RTPSession * session, RTPSource * src, GstRtpSession * sess)
-{
-  g_signal_emit (sess, gst_rtp_session_signals[SIGNAL_ON_TIMEOUT], 0,
-      src->ssrc);
-}
-
-static void
-on_sender_timeout (RTPSession * session, RTPSource * src, GstRtpSession * sess)
-{
-  g_signal_emit (sess, gst_rtp_session_signals[SIGNAL_ON_SENDER_TIMEOUT], 0,
-      src->ssrc);
-}
-
-GST_BOILERPLATE (GstRtpSession, gst_rtp_session, GstElement, GST_TYPE_ELEMENT);
-
-static void
-gst_rtp_session_base_init (gpointer klass)
-{
-  GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
-
-  /* sink pads */
-  gst_element_class_add_pad_template (element_class,
-      gst_static_pad_template_get (&rtpsession_recv_rtp_sink_template));
-  gst_element_class_add_pad_template (element_class,
-      gst_static_pad_template_get (&rtpsession_recv_rtcp_sink_template));
-  gst_element_class_add_pad_template (element_class,
-      gst_static_pad_template_get (&rtpsession_send_rtp_sink_template));
-
-  /* src pads */
-  gst_element_class_add_pad_template (element_class,
-      gst_static_pad_template_get (&rtpsession_recv_rtp_src_template));
-  gst_element_class_add_pad_template (element_class,
-      gst_static_pad_template_get (&rtpsession_sync_src_template));
-  gst_element_class_add_pad_template (element_class,
-      gst_static_pad_template_get (&rtpsession_send_rtp_src_template));
-  gst_element_class_add_pad_template (element_class,
-      gst_static_pad_template_get (&rtpsession_send_rtcp_src_template));
-
-  gst_element_class_set_details (element_class, &rtpsession_details);
-}
-
-static void
-gst_rtp_session_class_init (GstRtpSessionClass * klass)
-{
-  GObjectClass *gobject_class;
-  GstElementClass *gstelement_class;
-
-  gobject_class = (GObjectClass *) klass;
-  gstelement_class = (GstElementClass *) klass;
-
-  g_type_class_add_private (klass, sizeof (GstRtpSessionPrivate));
-
-  gobject_class->finalize = gst_rtp_session_finalize;
-  gobject_class->set_property = gst_rtp_session_set_property;
-  gobject_class->get_property = gst_rtp_session_get_property;
-
-  /**
-   * GstRtpSession::request-pt-map:
-   * @sess: the object which received the signal
-   * @pt: the pt
-   *
-   * Request the payload type as #GstCaps for @pt.
-   */
-  gst_rtp_session_signals[SIGNAL_REQUEST_PT_MAP] =
-      g_signal_new ("request-pt-map", G_TYPE_FROM_CLASS (klass),
-      G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpSessionClass, request_pt_map),
-      NULL, NULL, gst_rtp_bin_marshal_BOXED__UINT, GST_TYPE_CAPS, 1,
-      G_TYPE_UINT);
-  /**
-   * GstRtpSession::clear-pt-map:
-   * @sess: the object which received the signal
-   *
-   * Clear the cached pt-maps requested with #GstRtpSession::request-pt-map.
-   */
-  gst_rtp_session_signals[SIGNAL_CLEAR_PT_MAP] =
-      g_signal_new ("clear-pt-map", G_TYPE_FROM_CLASS (klass),
-      G_SIGNAL_ACTION, G_STRUCT_OFFSET (GstRtpSessionClass, clear_pt_map),
-      NULL, NULL, g_cclosure_marshal_VOID__VOID, G_TYPE_NONE, 0, G_TYPE_NONE);
-
-  /**
-   * GstRtpSession::on-new-ssrc:
-   * @sess: the object which received the signal
-   * @ssrc: the SSRC 
-   *
-   * Notify of a new SSRC that entered @session.
-   */
-  gst_rtp_session_signals[SIGNAL_ON_NEW_SSRC] =
-      g_signal_new ("on-new-ssrc", G_TYPE_FROM_CLASS (klass),
-      G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpSessionClass, on_new_ssrc),
-      NULL, NULL, g_cclosure_marshal_VOID__UINT, G_TYPE_NONE, 1, G_TYPE_UINT);
-  /**
-   * GstRtpSession::on-ssrc_collision:
-   * @sess: the object which received the signal
-   * @ssrc: the SSRC 
-   *
-   * Notify when we have an SSRC collision
-   */
-  gst_rtp_session_signals[SIGNAL_ON_SSRC_COLLISION] =
-      g_signal_new ("on-ssrc-collision", G_TYPE_FROM_CLASS (klass),
-      G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpSessionClass,
-          on_ssrc_collision), NULL, NULL, g_cclosure_marshal_VOID__UINT,
-      G_TYPE_NONE, 1, G_TYPE_UINT);
-  /**
-   * GstRtpSession::on-ssrc_validated:
-   * @sess: the object which received the signal
-   * @ssrc: the SSRC 
-   *
-   * Notify of a new SSRC that became validated.
-   */
-  gst_rtp_session_signals[SIGNAL_ON_SSRC_VALIDATED] =
-      g_signal_new ("on-ssrc-validated", G_TYPE_FROM_CLASS (klass),
-      G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpSessionClass,
-          on_ssrc_validated), NULL, NULL, g_cclosure_marshal_VOID__UINT,
-      G_TYPE_NONE, 1, G_TYPE_UINT);
-  /**
-   * GstRtpSession::on-ssrc_active:
-   * @sess: the object which received the signal
-   * @ssrc: the SSRC
-   *
-   * Notify of a SSRC that is active, i.e., sending RTCP.
-   */
-  gst_rtp_session_signals[SIGNAL_ON_SSRC_ACTIVE] =
-      g_signal_new ("on-ssrc-active", G_TYPE_FROM_CLASS (klass),
-      G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpSessionClass,
-          on_ssrc_active), NULL, NULL, g_cclosure_marshal_VOID__UINT,
-      G_TYPE_NONE, 1, G_TYPE_UINT);
-  /**
-   * GstRtpSession::on-ssrc-sdes:
-   * @session: the object which received the signal
-   * @src: the SSRC
-   *
-   * Notify that a new SDES was received for SSRC.
-   */
-  gst_rtp_session_signals[SIGNAL_ON_SSRC_SDES] =
-      g_signal_new ("on-ssrc-sdes", G_TYPE_FROM_CLASS (klass),
-      G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpSessionClass, on_ssrc_sdes),
-      NULL, NULL, g_cclosure_marshal_VOID__UINT, G_TYPE_NONE, 1, G_TYPE_UINT);
-
-  /**
-   * GstRtpSession::on-bye-ssrc:
-   * @sess: the object which received the signal
-   * @ssrc: the SSRC 
-   *
-   * Notify of an SSRC that became inactive because of a BYE packet.
-   */
-  gst_rtp_session_signals[SIGNAL_ON_BYE_SSRC] =
-      g_signal_new ("on-bye-ssrc", G_TYPE_FROM_CLASS (klass),
-      G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpSessionClass, on_bye_ssrc),
-      NULL, NULL, g_cclosure_marshal_VOID__UINT, G_TYPE_NONE, 1, G_TYPE_UINT);
-  /**
-   * GstRtpSession::on-bye-timeout:
-   * @sess: the object which received the signal
-   * @ssrc: the SSRC 
-   *
-   * Notify of an SSRC that has timed out because of BYE
-   */
-  gst_rtp_session_signals[SIGNAL_ON_BYE_TIMEOUT] =
-      g_signal_new ("on-bye-timeout", G_TYPE_FROM_CLASS (klass),
-      G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpSessionClass, on_bye_timeout),
-      NULL, NULL, g_cclosure_marshal_VOID__UINT, G_TYPE_NONE, 1, G_TYPE_UINT);
-  /**
-   * GstRtpSession::on-timeout:
-   * @sess: the object which received the signal
-   * @ssrc: the SSRC 
-   *
-   * Notify of an SSRC that has timed out
-   */
-  gst_rtp_session_signals[SIGNAL_ON_TIMEOUT] =
-      g_signal_new ("on-timeout", G_TYPE_FROM_CLASS (klass),
-      G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpSessionClass, on_timeout),
-      NULL, NULL, g_cclosure_marshal_VOID__UINT, G_TYPE_NONE, 1, G_TYPE_UINT);
-  /**
-   * GstRtpSession::on-sender-timeout:
-   * @sess: the object which received the signal
-   * @ssrc: the SSRC 
-   *
-   * Notify of a sender SSRC that has timed out and became a receiver
-   */
-  gst_rtp_session_signals[SIGNAL_ON_SENDER_TIMEOUT] =
-      g_signal_new ("on-sender-timeout", G_TYPE_FROM_CLASS (klass),
-      G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpSessionClass,
-          on_sender_timeout), NULL, NULL, g_cclosure_marshal_VOID__UINT,
-      G_TYPE_NONE, 1, G_TYPE_UINT);
-
-  g_object_class_install_property (gobject_class, PROP_NTP_NS_BASE,
-      g_param_spec_uint64 ("ntp-ns-base", "NTP base time",
-          "The NTP base time corresponding to running_time 0", 0,
-          G_MAXUINT64, DEFAULT_NTP_NS_BASE, G_PARAM_READWRITE));
-
-  g_object_class_install_property (gobject_class, PROP_BANDWIDTH,
-      g_param_spec_double ("bandwidth", "Bandwidth",
-          "The bandwidth of the session",
-          0.0, G_MAXDOUBLE, DEFAULT_BANDWIDTH, G_PARAM_READWRITE));
-
-  g_object_class_install_property (gobject_class, PROP_RTCP_FRACTION,
-      g_param_spec_double ("rtcp-fraction", "RTCP Fraction",
-          "The fraction of the bandwidth used for RTCP",
-          0.0, G_MAXDOUBLE, DEFAULT_RTCP_FRACTION, G_PARAM_READWRITE));
-
-  g_object_class_install_property (gobject_class, PROP_SDES,
-      g_param_spec_boxed ("sdes", "SDES",
-          "The SDES items of this session",
-          GST_TYPE_STRUCTURE, G_PARAM_READWRITE));
-
-  g_object_class_install_property (gobject_class, PROP_NUM_SOURCES,
-      g_param_spec_uint ("num-sources", "Num Sources",
-          "The number of sources in the session", 0, G_MAXUINT,
-          DEFAULT_NUM_SOURCES, G_PARAM_READABLE));
-
-  g_object_class_install_property (gobject_class, PROP_NUM_ACTIVE_SOURCES,
-      g_param_spec_uint ("num-active-sources", "Num Active Sources",
-          "The number of active sources in the session", 0, G_MAXUINT,
-          DEFAULT_NUM_ACTIVE_SOURCES, G_PARAM_READABLE));
-
-  g_object_class_install_property (gobject_class, PROP_INTERNAL_SESSION,
-      g_param_spec_object ("internal-session", "Internal Session",
-          "The internal RTPSession object", RTP_TYPE_SESSION,
-          G_PARAM_READABLE));
-
-  gstelement_class->change_state =
-      GST_DEBUG_FUNCPTR (gst_rtp_session_change_state);
-  gstelement_class->request_new_pad =
-      GST_DEBUG_FUNCPTR (gst_rtp_session_request_new_pad);
-  gstelement_class->release_pad =
-      GST_DEBUG_FUNCPTR (gst_rtp_session_release_pad);
-
-  klass->clear_pt_map = GST_DEBUG_FUNCPTR (gst_rtp_session_clear_pt_map);
-
-  GST_DEBUG_CATEGORY_INIT (gst_rtp_session_debug,
-      "rtpsession", 0, "RTP Session");
-}
-
-static void
-gst_rtp_session_init (GstRtpSession * rtpsession, GstRtpSessionClass * klass)
-{
-  rtpsession->priv = GST_RTP_SESSION_GET_PRIVATE (rtpsession);
-  rtpsession->priv->lock = g_mutex_new ();
-  rtpsession->priv->sysclock = gst_system_clock_obtain ();
-  rtpsession->priv->session = rtp_session_new ();
-
-  /* configure callbacks */
-  rtp_session_set_callbacks (rtpsession->priv->session, &callbacks, rtpsession);
-  /* configure signals */
-  g_signal_connect (rtpsession->priv->session, "on-new-ssrc",
-      (GCallback) on_new_ssrc, rtpsession);
-  g_signal_connect (rtpsession->priv->session, "on-ssrc-collision",
-      (GCallback) on_ssrc_collision, rtpsession);
-  g_signal_connect (rtpsession->priv->session, "on-ssrc-validated",
-      (GCallback) on_ssrc_validated, rtpsession);
-  g_signal_connect (rtpsession->priv->session, "on-ssrc-active",
-      (GCallback) on_ssrc_active, rtpsession);
-  g_signal_connect (rtpsession->priv->session, "on-ssrc-sdes",
-      (GCallback) on_ssrc_sdes, rtpsession);
-  g_signal_connect (rtpsession->priv->session, "on-bye-ssrc",
-      (GCallback) on_bye_ssrc, rtpsession);
-  g_signal_connect (rtpsession->priv->session, "on-bye-timeout",
-      (GCallback) on_bye_timeout, rtpsession);
-  g_signal_connect (rtpsession->priv->session, "on-timeout",
-      (GCallback) on_timeout, rtpsession);
-  g_signal_connect (rtpsession->priv->session, "on-sender-timeout",
-      (GCallback) on_sender_timeout, rtpsession);
-  rtpsession->priv->ptmap = g_hash_table_new_full (NULL, NULL, NULL,
-      (GDestroyNotify) gst_caps_unref);
-
-  gst_segment_init (&rtpsession->recv_rtp_seg, GST_FORMAT_UNDEFINED);
-  gst_segment_init (&rtpsession->send_rtp_seg, GST_FORMAT_UNDEFINED);
-
-  rtpsession->priv->thread_stopped = TRUE;
-}
-
-static void
-gst_rtp_session_finalize (GObject * object)
-{
-  GstRtpSession *rtpsession;
-
-  rtpsession = GST_RTP_SESSION (object);
-
-  if (rtpsession->recv_rtp_sink != NULL)
-    gst_object_unref (rtpsession->recv_rtp_sink);
-  if (rtpsession->recv_rtcp_sink != NULL)
-    gst_object_unref (rtpsession->recv_rtcp_sink);
-  if (rtpsession->send_rtp_sink != NULL)
-    gst_object_unref (rtpsession->send_rtp_sink);
-  if (rtpsession->send_rtcp_src != NULL)
-    gst_object_unref (rtpsession->send_rtcp_src);
-
-  g_hash_table_destroy (rtpsession->priv->ptmap);
-  g_mutex_free (rtpsession->priv->lock);
-  g_object_unref (rtpsession->priv->sysclock);
-  g_object_unref (rtpsession->priv->session);
-
-  G_OBJECT_CLASS (parent_class)->finalize (object);
-}
-
-static void
-gst_rtp_session_set_property (GObject * object, guint prop_id,
-    const GValue * value, GParamSpec * pspec)
-{
-  GstRtpSession *rtpsession;
-  GstRtpSessionPrivate *priv;
-
-  rtpsession = GST_RTP_SESSION (object);
-  priv = rtpsession->priv;
-
-  switch (prop_id) {
-    case PROP_NTP_NS_BASE:
-      GST_OBJECT_LOCK (rtpsession);
-      priv->ntpnsbase = g_value_get_uint64 (value);
-      GST_DEBUG_OBJECT (rtpsession, "setting NTP base to %" GST_TIME_FORMAT,
-          GST_TIME_ARGS (priv->ntpnsbase));
-      GST_OBJECT_UNLOCK (rtpsession);
-      break;
-    case PROP_BANDWIDTH:
-      rtp_session_set_bandwidth (priv->session, g_value_get_double (value));
-      break;
-    case PROP_RTCP_FRACTION:
-      rtp_session_set_rtcp_fraction (priv->session, g_value_get_double (value));
-      break;
-    case PROP_SDES:
-      rtp_session_set_sdes_struct (priv->session, g_value_get_boxed (value));
-      break;
-    default:
-      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
-      break;
-  }
-}
-
-static void
-gst_rtp_session_get_property (GObject * object, guint prop_id,
-    GValue * value, GParamSpec * pspec)
-{
-  GstRtpSession *rtpsession;
-  GstRtpSessionPrivate *priv;
-
-  rtpsession = GST_RTP_SESSION (object);
-  priv = rtpsession->priv;
-
-  switch (prop_id) {
-    case PROP_NTP_NS_BASE:
-      GST_OBJECT_LOCK (rtpsession);
-      g_value_set_uint64 (value, priv->ntpnsbase);
-      GST_OBJECT_UNLOCK (rtpsession);
-      break;
-    case PROP_BANDWIDTH:
-      g_value_set_double (value, rtp_session_get_bandwidth (priv->session));
-      break;
-    case PROP_RTCP_FRACTION:
-      g_value_set_double (value, rtp_session_get_rtcp_fraction (priv->session));
-      break;
-    case PROP_SDES:
-      g_value_take_boxed (value, rtp_session_get_sdes_struct (priv->session));
-      break;
-    case PROP_NUM_SOURCES:
-      g_value_set_uint (value, rtp_session_get_num_sources (priv->session));
-      break;
-    case PROP_NUM_ACTIVE_SOURCES:
-      g_value_set_uint (value,
-          rtp_session_get_num_active_sources (priv->session));
-      break;
-    case PROP_INTERNAL_SESSION:
-      g_value_set_object (value, priv->session);
-      break;
-    default:
-      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
-      break;
-  }
-}
-
-static void
-get_current_times (GstRtpSession * rtpsession,
-    GstClockTime * running_time, guint64 * ntpnstime)
-{
-  guint64 ntpns;
-  GstClock *clock;
-  GstClockTime base_time, ntpnsbase, rt;
-
-  GST_OBJECT_LOCK (rtpsession);
-  if ((clock = GST_ELEMENT_CLOCK (rtpsession))) {
-    base_time = GST_ELEMENT_CAST (rtpsession)->base_time;
-    ntpnsbase = rtpsession->priv->ntpnsbase;
-    gst_object_ref (clock);
-    GST_OBJECT_UNLOCK (rtpsession);
-
-    /* get current clock time and convert to running time */
-    rt = gst_clock_get_time (clock) - base_time;
-    /* add NTP base offset to get NTP ns time */
-    ntpns = rt + ntpnsbase;
-
-    gst_object_unref (clock);
-  } else {
-    GST_OBJECT_UNLOCK (rtpsession);
-    rt = -1;
-    ntpns = -1;
-  }
-  if (running_time)
-    *running_time = rt;
-  if (ntpnstime)
-    *ntpnstime = ntpns;
-}
-
-static void
-rtcp_thread (GstRtpSession * rtpsession)
-{
-  GstClockID id;
-  GstClockTime current_time;
-  GstClockTime next_timeout;
-  guint64 ntpnstime;
-
-  GST_DEBUG_OBJECT (rtpsession, "entering RTCP thread");
-
-  GST_RTP_SESSION_LOCK (rtpsession);
-
-  current_time = gst_clock_get_time (rtpsession->priv->sysclock);
-
-  while (!rtpsession->priv->stop_thread) {
-    GstClockReturn res;
-
-    /* get initial estimate */
-    next_timeout =
-        rtp_session_next_timeout (rtpsession->priv->session, current_time);
-
-    GST_DEBUG_OBJECT (rtpsession, "next check time %" GST_TIME_FORMAT,
-        GST_TIME_ARGS (next_timeout));
-
-    /* leave if no more timeouts, the session ended */
-    if (next_timeout == GST_CLOCK_TIME_NONE)
-      break;
-
-    id = rtpsession->priv->id =
-        gst_clock_new_single_shot_id (rtpsession->priv->sysclock, next_timeout);
-    GST_RTP_SESSION_UNLOCK (rtpsession);
-
-    res = gst_clock_id_wait (id, NULL);
-
-    GST_RTP_SESSION_LOCK (rtpsession);
-    gst_clock_id_unref (id);
-    rtpsession->priv->id = NULL;
-
-    if (rtpsession->priv->stop_thread)
-      break;
-
-    /* update current time */
-    current_time = gst_clock_get_time (rtpsession->priv->sysclock);
-
-    /* get current NTP time */
-    get_current_times (rtpsession, NULL, &ntpnstime);
-
-    /* we get unlocked because we need to perform reconsideration, don't perform
-     * the timeout but get a new reporting estimate. */
-    GST_DEBUG_OBJECT (rtpsession, "unlocked %d, current %" GST_TIME_FORMAT,
-        res, GST_TIME_ARGS (current_time));
-
-    /* perform actions, we ignore result. Release lock because it might push. */
-    GST_RTP_SESSION_UNLOCK (rtpsession);
-    rtp_session_on_timeout (rtpsession->priv->session, current_time, ntpnstime);
-    GST_RTP_SESSION_LOCK (rtpsession);
-  }
-  /* mark the thread as stopped now */
-  rtpsession->priv->thread_stopped = TRUE;
-  GST_RTP_SESSION_UNLOCK (rtpsession);
-
-  GST_DEBUG_OBJECT (rtpsession, "leaving RTCP thread");
-}
-
-static gboolean
-start_rtcp_thread (GstRtpSession * rtpsession)
-{
-  GError *error = NULL;
-  gboolean res;
-
-  GST_DEBUG_OBJECT (rtpsession, "starting RTCP thread");
-
-  GST_RTP_SESSION_LOCK (rtpsession);
-  rtpsession->priv->stop_thread = FALSE;
-  if (rtpsession->priv->thread_stopped) {
-    /* if the thread stopped, and we still have a handle to the thread, join it
-     * now. We can safely join with the lock held, the thread will not take it
-     * anymore. */
-    if (rtpsession->priv->thread)
-      g_thread_join (rtpsession->priv->thread);
-    /* only create a new thread if the old one was stopped. Otherwise we can
-     * just reuse the currently running one. */
-    rtpsession->priv->thread =
-        g_thread_create ((GThreadFunc) rtcp_thread, rtpsession, TRUE, &error);
-    rtpsession->priv->thread_stopped = FALSE;
-  }
-  GST_RTP_SESSION_UNLOCK (rtpsession);
-
-  if (error != NULL) {
-    res = FALSE;
-    GST_DEBUG_OBJECT (rtpsession, "failed to start thread, %s", error->message);
-    g_error_free (error);
-  } else {
-    res = TRUE;
-  }
-  return res;
-}
-
-static void
-stop_rtcp_thread (GstRtpSession * rtpsession)
-{
-  GST_DEBUG_OBJECT (rtpsession, "stopping RTCP thread");
-
-  GST_RTP_SESSION_LOCK (rtpsession);
-  rtpsession->priv->stop_thread = TRUE;
-  if (rtpsession->priv->id)
-    gst_clock_id_unschedule (rtpsession->priv->id);
-  GST_RTP_SESSION_UNLOCK (rtpsession);
-}
-
-static void
-join_rtcp_thread (GstRtpSession * rtpsession)
-{
-  GST_RTP_SESSION_LOCK (rtpsession);
-  /* don't try to join when we have no thread */
-  if (rtpsession->priv->thread != NULL) {
-    GST_DEBUG_OBJECT (rtpsession, "joining RTCP thread");
-    GST_RTP_SESSION_UNLOCK (rtpsession);
-
-    g_thread_join (rtpsession->priv->thread);
-
-    GST_RTP_SESSION_LOCK (rtpsession);
-    /* after the join, take the lock and clear the thread structure. The caller
-     * is supposed to not concurrently call start and join. */
-    rtpsession->priv->thread = NULL;
-  }
-  GST_RTP_SESSION_UNLOCK (rtpsession);
-}
-
-static GstStateChangeReturn
-gst_rtp_session_change_state (GstElement * element, GstStateChange transition)
-{
-  GstStateChangeReturn res;
-  GstRtpSession *rtpsession;
-  GstRtpSessionPrivate *priv;
-
-  rtpsession = GST_RTP_SESSION (element);
-  priv = rtpsession->priv;
-
-  switch (transition) {
-    case GST_STATE_CHANGE_NULL_TO_READY:
-      break;
-    case GST_STATE_CHANGE_READY_TO_PAUSED:
-      break;
-    case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
-      break;
-    case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
-    case GST_STATE_CHANGE_PAUSED_TO_READY:
-      /* no need to join yet, we might want to continue later. Also, the
-       * dataflow could block downstream so that a join could just block
-       * forever. */
-      stop_rtcp_thread (rtpsession);
-      break;
-    default:
-      break;
-  }
-
-  res = parent_class->change_state (element, transition);
-
-  switch (transition) {
-    case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
-      if (!start_rtcp_thread (rtpsession))
-        goto failed_thread;
-      break;
-    case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
-      break;
-    case GST_STATE_CHANGE_PAUSED_TO_READY:
-      /* downstream is now releasing the dataflow and we can join. */
-      join_rtcp_thread (rtpsession);
-      break;
-    case GST_STATE_CHANGE_READY_TO_NULL:
-      break;
-    default:
-      break;
-  }
-  return res;
-
-  /* ERRORS */
-failed_thread:
-  {
-    return GST_STATE_CHANGE_FAILURE;
-  }
-}
-
-static gboolean
-return_true (gpointer key, gpointer value, gpointer user_data)
-{
-  return TRUE;
-}
-
-static void
-gst_rtp_session_clear_pt_map (GstRtpSession * rtpsession)
-{
-  g_hash_table_foreach_remove (rtpsession->priv->ptmap, return_true, NULL);
-}
-
-/* called when the session manager has an RTP packet or a list of packets
- * ready for further processing */
-static GstFlowReturn
-gst_rtp_session_process_rtp (RTPSession * sess, RTPSource * src,
-    GstBuffer * buffer, gpointer user_data)
-{
-  GstFlowReturn result;
-  GstRtpSession *rtpsession;
-  GstRtpSessionPrivate *priv;
-
-  rtpsession = GST_RTP_SESSION (user_data);
-  priv = rtpsession->priv;
-
-  if (rtpsession->recv_rtp_src) {
-    GST_LOG_OBJECT (rtpsession, "pushing received RTP packet");
-    result = gst_pad_push (rtpsession->recv_rtp_src, buffer);
-  } else {
-    GST_DEBUG_OBJECT (rtpsession, "dropping received RTP packet");
-    gst_buffer_unref (buffer);
-    result = GST_FLOW_OK;
-  }
-  return result;
-}
-
-/* called when the session manager has an RTP packet ready for further
- * sending */
-static GstFlowReturn
-gst_rtp_session_send_rtp (RTPSession * sess, RTPSource * src,
-    gpointer data, gpointer user_data)
-{
-  GstFlowReturn result;
-  GstRtpSession *rtpsession;
-  GstRtpSessionPrivate *priv;
-
-  rtpsession = GST_RTP_SESSION (user_data);
-  priv = rtpsession->priv;
-
-  if (rtpsession->send_rtp_src) {
-    if (GST_IS_BUFFER (data)) {
-      GST_LOG_OBJECT (rtpsession, "sending RTP packet");
-      result = gst_pad_push (rtpsession->send_rtp_src, GST_BUFFER_CAST (data));
-    } else {
-      GST_LOG_OBJECT (rtpsession, "sending RTP list");
-      result = gst_pad_push_list (rtpsession->send_rtp_src,
-          GST_BUFFER_LIST_CAST (data));
-    }
-  } else {
-    gst_mini_object_unref (GST_MINI_OBJECT_CAST (data));
-    result = GST_FLOW_OK;
-  }
-  return result;
-}
-
-/* called when the session manager has an RTCP packet ready for further
- * sending. The eos flag is set when an EOS event should be sent downstream as
- * well. */
-static GstFlowReturn
-gst_rtp_session_send_rtcp (RTPSession * sess, RTPSource * src,
-    GstBuffer * buffer, gboolean eos, gpointer user_data)
-{
-  GstFlowReturn result;
-  GstRtpSession *rtpsession;
-  GstRtpSessionPrivate *priv;
-
-  rtpsession = GST_RTP_SESSION (user_data);
-  priv = rtpsession->priv;
-
-  if (rtpsession->send_rtcp_src) {
-    GstCaps *caps;
-
-    /* set rtcp caps on output pad */
-    if (!(caps = GST_PAD_CAPS (rtpsession->send_rtcp_src))) {
-      caps = gst_caps_new_simple ("application/x-rtcp", NULL);
-      gst_pad_set_caps (rtpsession->send_rtcp_src, caps);
-      gst_caps_unref (caps);
-    }
-    gst_buffer_set_caps (buffer, caps);
-    GST_LOG_OBJECT (rtpsession, "sending RTCP");
-    result = gst_pad_push (rtpsession->send_rtcp_src, buffer);
-
-    /* we have to send EOS after this packet */
-    if (eos) {
-      GST_LOG_OBJECT (rtpsession, "sending EOS");
-      gst_pad_push_event (rtpsession->send_rtcp_src, gst_event_new_eos ());
-    }
-  } else {
-    GST_DEBUG_OBJECT (rtpsession, "not sending RTCP, no output pad");
-    gst_buffer_unref (buffer);
-    result = GST_FLOW_OK;
-  }
-  return result;
-}
-
-/* called when the session manager has an SR RTCP packet ready for handling
- * inter stream synchronisation */
-static GstFlowReturn
-gst_rtp_session_sync_rtcp (RTPSession * sess,
-    RTPSource * src, GstBuffer * buffer, gpointer user_data)
-{
-  GstFlowReturn result;
-  GstRtpSession *rtpsession;
-  GstRtpSessionPrivate *priv;
-
-  rtpsession = GST_RTP_SESSION (user_data);
-  priv = rtpsession->priv;
-
-  if (rtpsession->sync_src) {
-    GstCaps *caps;
-
-    /* set rtcp caps on output pad */
-    if (!(caps = GST_PAD_CAPS (rtpsession->sync_src))) {
-      caps = gst_caps_new_simple ("application/x-rtcp", NULL);
-      gst_pad_set_caps (rtpsession->sync_src, caps);
-      gst_caps_unref (caps);
-    }
-    gst_buffer_set_caps (buffer, caps);
-    GST_LOG_OBJECT (rtpsession, "sending Sync RTCP");
-    result = gst_pad_push (rtpsession->sync_src, buffer);
-  } else {
-    GST_DEBUG_OBJECT (rtpsession, "not sending Sync RTCP, no output pad");
-    gst_buffer_unref (buffer);
-    result = GST_FLOW_OK;
-  }
-  return result;
-}
-
-static void
-gst_rtp_session_cache_caps (GstRtpSession * rtpsession, GstCaps * caps)
-{
-  GstRtpSessionPrivate *priv;
-  const GstStructure *s;
-  gint payload;
-
-  priv = rtpsession->priv;
-
-  GST_DEBUG_OBJECT (rtpsession, "parsing caps");
-
-  s = gst_caps_get_structure (caps, 0);
-  if (!gst_structure_get_int (s, "payload", &payload))
-    return;
-
-  if (g_hash_table_lookup (priv->ptmap, GINT_TO_POINTER (payload)))
-    return;
-
-  g_hash_table_insert (priv->ptmap, GINT_TO_POINTER (payload),
-      gst_caps_ref (caps));
-}
-
-/* called when the session manager needs the clock rate */
-static gint
-gst_rtp_session_clock_rate (RTPSession * sess, guint8 payload,
-    gpointer user_data)
-{
-  gint ipayload, result = -1;
-  GstRtpSession *rtpsession;
-  GstRtpSessionPrivate *priv;
-  GValue ret = { 0 };
-  GValue args[2] = { {0}, {0} };
-  GstCaps *caps;
-  const GstStructure *s;
-
-  rtpsession = GST_RTP_SESSION_CAST (user_data);
-  priv = rtpsession->priv;
-
-  GST_RTP_SESSION_LOCK (rtpsession);
-  ipayload = payload;           /* make compiler happy */
-  caps = g_hash_table_lookup (priv->ptmap, GINT_TO_POINTER (ipayload));
-  if (caps) {
-    gst_caps_ref (caps);
-    goto found;
-  }
-
-  /* not found in the cache, try to get it with a signal */
-  g_value_init (&args[0], GST_TYPE_ELEMENT);
-  g_value_set_object (&args[0], rtpsession);
-  g_value_init (&args[1], G_TYPE_UINT);
-  g_value_set_uint (&args[1], payload);
-
-  g_value_init (&ret, GST_TYPE_CAPS);
-  g_value_set_boxed (&ret, NULL);
-
-  g_signal_emitv (args, gst_rtp_session_signals[SIGNAL_REQUEST_PT_MAP], 0,
-      &ret);
-
-  g_value_unset (&args[0]);
-  g_value_unset (&args[1]);
-  caps = (GstCaps *) g_value_dup_boxed (&ret);
-  g_value_unset (&ret);
-  if (!caps)
-    goto no_caps;
-
-  gst_rtp_session_cache_caps (rtpsession, caps);
-
-found:
-  s = gst_caps_get_structure (caps, 0);
-  if (!gst_structure_get_int (s, "clock-rate", &result))
-    goto no_clock_rate;
-
-  gst_caps_unref (caps);
-
-  GST_DEBUG_OBJECT (rtpsession, "parsed clock-rate %d", result);
-
-done:
-  GST_RTP_SESSION_UNLOCK (rtpsession);
-
-  return result;
-
-  /* ERRORS */
-no_caps:
-  {
-    GST_DEBUG_OBJECT (rtpsession, "could not get caps");
-    goto done;
-  }
-no_clock_rate:
-  {
-    gst_caps_unref (caps);
-    GST_DEBUG_OBJECT (rtpsession, "No clock-rate in caps!");
-    goto done;
-  }
-}
-
-/* called when the session manager asks us to reconsider the timeout */
-static void
-gst_rtp_session_reconsider (RTPSession * sess, gpointer user_data)
-{
-  GstRtpSession *rtpsession;
-
-  rtpsession = GST_RTP_SESSION_CAST (user_data);
-
-  GST_RTP_SESSION_LOCK (rtpsession);
-  GST_DEBUG_OBJECT (rtpsession, "unlock timer for reconsideration");
-  if (rtpsession->priv->id)
-    gst_clock_id_unschedule (rtpsession->priv->id);
-  GST_RTP_SESSION_UNLOCK (rtpsession);
-}
-
-static gboolean
-gst_rtp_session_event_recv_rtp_sink (GstPad * pad, GstEvent * event)
-{
-  GstRtpSession *rtpsession;
-  GstRtpSessionPrivate *priv;
-  gboolean ret = FALSE;
-
-  rtpsession = GST_RTP_SESSION (gst_pad_get_parent (pad));
-  priv = rtpsession->priv;
-
-  GST_DEBUG_OBJECT (rtpsession, "received event %s",
-      GST_EVENT_TYPE_NAME (event));
-
-  switch (GST_EVENT_TYPE (event)) {
-    case GST_EVENT_FLUSH_STOP:
-      gst_segment_init (&rtpsession->recv_rtp_seg, GST_FORMAT_UNDEFINED);
-      ret = gst_pad_push_event (rtpsession->recv_rtp_src, event);
-      break;
-    case GST_EVENT_NEWSEGMENT:
-    {
-      gboolean update;
-      gdouble rate, arate;
-      GstFormat format;
-      gint64 start, stop, time;
-      GstSegment *segment;
-
-      segment = &rtpsession->recv_rtp_seg;
-
-      /* the newsegment event is needed to convert the RTP timestamp to
-       * running_time, which is needed to generate a mapping from RTP to NTP
-       * timestamps in SR reports */
-      gst_event_parse_new_segment_full (event, &update, &rate, &arate, &format,
-          &start, &stop, &time);
-
-      GST_DEBUG_OBJECT (rtpsession,
-          "configured NEWSEGMENT update %d, rate %lf, applied rate %lf, "
-          "format GST_FORMAT_TIME, "
-          "%" GST_TIME_FORMAT " -- %" GST_TIME_FORMAT
-          ", time %" GST_TIME_FORMAT ", accum %" GST_TIME_FORMAT,
-          update, rate, arate, GST_TIME_ARGS (segment->start),
-          GST_TIME_ARGS (segment->stop), GST_TIME_ARGS (segment->time),
-          GST_TIME_ARGS (segment->accum));
-
-      gst_segment_set_newsegment_full (segment, update, rate,
-          arate, format, start, stop, time);
-
-      /* push event forward */
-      ret = gst_pad_push_event (rtpsession->recv_rtp_src, event);
-      break;
-    }
-    default:
-      ret = gst_pad_push_event (rtpsession->recv_rtp_src, event);
-      break;
-  }
-  gst_object_unref (rtpsession);
-
-  return ret;
-
-}
-
-static GList *
-gst_rtp_session_internal_links (GstPad * pad)
-{
-  GstRtpSession *rtpsession;
-  GstRtpSessionPrivate *priv;
-  GList *res = NULL;
-
-  rtpsession = GST_RTP_SESSION (gst_pad_get_parent (pad));
-  priv = rtpsession->priv;
-
-  if (pad == rtpsession->recv_rtp_src) {
-    res = g_list_prepend (res, rtpsession->recv_rtp_sink);
-  } else if (pad == rtpsession->recv_rtp_sink) {
-    res = g_list_prepend (res, rtpsession->recv_rtp_src);
-  } else if (pad == rtpsession->send_rtp_src) {
-    res = g_list_prepend (res, rtpsession->send_rtp_sink);
-  } else if (pad == rtpsession->send_rtp_sink) {
-    res = g_list_prepend (res, rtpsession->send_rtp_src);
-  }
-
-  gst_object_unref (rtpsession);
-
-  return res;
-}
-
-static gboolean
-gst_rtp_session_sink_setcaps (GstPad * pad, GstCaps * caps)
-{
-  GstRtpSession *rtpsession;
-  GstRtpSessionPrivate *priv;
-
-  rtpsession = GST_RTP_SESSION (gst_pad_get_parent (pad));
-  priv = rtpsession->priv;
-
-  GST_RTP_SESSION_LOCK (rtpsession);
-  gst_rtp_session_cache_caps (rtpsession, caps);
-  GST_RTP_SESSION_UNLOCK (rtpsession);
-
-  gst_object_unref (rtpsession);
-
-  return TRUE;
-}
-
-/* receive a packet from a sender, send it to the RTP session manager and
- * forward the packet on the rtp_src pad
- */
-static GstFlowReturn
-gst_rtp_session_chain_recv_rtp (GstPad * pad, GstBuffer * buffer)
-{
-  GstRtpSession *rtpsession;
-  GstRtpSessionPrivate *priv;
-  GstFlowReturn ret;
-  GstClockTime current_time, running_time;
-  guint64 ntpnstime;
-  GstClockTime timestamp;
-
-  rtpsession = GST_RTP_SESSION (gst_pad_get_parent (pad));
-  priv = rtpsession->priv;
-
-  GST_LOG_OBJECT (rtpsession, "received RTP packet");
-
-  /* get NTP time when this packet was captured, this depends on the timestamp. */
-  timestamp = GST_BUFFER_TIMESTAMP (buffer);
-  if (GST_CLOCK_TIME_IS_VALID (timestamp)) {
-    /* convert to running time using the segment values */
-    running_time =
-        gst_segment_to_running_time (&rtpsession->recv_rtp_seg, GST_FORMAT_TIME,
-        timestamp);
-    /* add constant to convert running time to NTP time */
-    ntpnstime = running_time + priv->ntpnsbase;
-  } else {
-    get_current_times (rtpsession, &running_time, &ntpnstime);
-  }
-  current_time = gst_clock_get_time (priv->sysclock);
-
-  ret = rtp_session_process_rtp (priv->session, buffer, current_time,
-      running_time, ntpnstime);
-  if (ret != GST_FLOW_OK)
-    goto push_error;
-
-done:
-  gst_object_unref (rtpsession);
-
-  return ret;
-
-  /* ERRORS */
-push_error:
-  {
-    GST_DEBUG_OBJECT (rtpsession, "process returned %s",
-        gst_flow_get_name (ret));
-    goto done;
-  }
-}
-
-static gboolean
-gst_rtp_session_event_recv_rtcp_sink (GstPad * pad, GstEvent * event)
-{
-  GstRtpSession *rtpsession;
-  GstRtpSessionPrivate *priv;
-  gboolean ret = FALSE;
-
-  rtpsession = GST_RTP_SESSION (gst_pad_get_parent (pad));
-  priv = rtpsession->priv;
-
-  GST_DEBUG_OBJECT (rtpsession, "received event %s",
-      GST_EVENT_TYPE_NAME (event));
-
-  switch (GST_EVENT_TYPE (event)) {
-    default:
-      ret = gst_pad_push_event (rtpsession->sync_src, event);
-      break;
-  }
-  gst_object_unref (rtpsession);
-
-  return ret;
-}
-
-/* Receive an RTCP packet from a sender, send it to the RTP session manager and
- * forward the SR packets to the sync_src pad.
- */
-static GstFlowReturn
-gst_rtp_session_chain_recv_rtcp (GstPad * pad, GstBuffer * buffer)
-{
-  GstRtpSession *rtpsession;
-  GstRtpSessionPrivate *priv;
-  GstClockTime current_time;
-  GstFlowReturn ret;
-
-  rtpsession = GST_RTP_SESSION (gst_pad_get_parent (pad));
-  priv = rtpsession->priv;
-
-  GST_LOG_OBJECT (rtpsession, "received RTCP packet");
-
-  current_time = gst_clock_get_time (priv->sysclock);
-  ret = rtp_session_process_rtcp (priv->session, buffer, current_time);
-
-  gst_object_unref (rtpsession);
-
-  return GST_FLOW_OK;
-}
-
-static gboolean
-gst_rtp_session_query_send_rtcp_src (GstPad * pad, GstQuery * query)
-{
-  GstRtpSession *rtpsession;
-  GstRtpSessionPrivate *priv;
-  gboolean ret = FALSE;
-
-  rtpsession = GST_RTP_SESSION (gst_pad_get_parent (pad));
-  priv = rtpsession->priv;
-
-  GST_DEBUG_OBJECT (rtpsession, "received QUERY");
-
-  switch (GST_QUERY_TYPE (query)) {
-    case GST_QUERY_LATENCY:
-      ret = TRUE;
-      /* use the defaults for the latency query. */
-      gst_query_set_latency (query, FALSE, 0, -1);
-      break;
-    default:
-      /* other queries simply fail for now */
-      break;
-  }
-
-  gst_object_unref (rtpsession);
-
-  return ret;
-}
-
-static gboolean
-gst_rtp_session_event_send_rtcp_src (GstPad * pad, GstEvent * event)
-{
-  GstRtpSession *rtpsession;
-  GstRtpSessionPrivate *priv;
-  gboolean ret;
-
-  rtpsession = GST_RTP_SESSION (gst_pad_get_parent (pad));
-  priv = rtpsession->priv;
-
-  GST_DEBUG_OBJECT (rtpsession, "received EVENT");
-
-  switch (GST_EVENT_TYPE (event)) {
-    case GST_EVENT_SEEK:
-    case GST_EVENT_LATENCY:
-      gst_event_unref (event);
-      ret = TRUE;
-      break;
-    default:
-      /* other events simply fail for now */
-      gst_event_unref (event);
-      ret = FALSE;
-      break;
-  }
-
-  gst_object_unref (rtpsession);
-
-  return ret;
-}
-
-
-static gboolean
-gst_rtp_session_event_send_rtp_sink (GstPad * pad, GstEvent * event)
-{
-  GstRtpSession *rtpsession;
-  GstRtpSessionPrivate *priv;
-  gboolean ret = FALSE;
-
-  rtpsession = GST_RTP_SESSION (gst_pad_get_parent (pad));
-  priv = rtpsession->priv;
-
-  GST_DEBUG_OBJECT (rtpsession, "received event");
-
-  switch (GST_EVENT_TYPE (event)) {
-    case GST_EVENT_FLUSH_STOP:
-      gst_segment_init (&rtpsession->send_rtp_seg, GST_FORMAT_UNDEFINED);
-      ret = gst_pad_push_event (rtpsession->send_rtp_src, event);
-      break;
-    case GST_EVENT_NEWSEGMENT:{
-      gboolean update;
-      gdouble rate, arate;
-      GstFormat format;
-      gint64 start, stop, time;
-      GstSegment *segment;
-
-      segment = &rtpsession->send_rtp_seg;
-
-      /* the newsegment event is needed to convert the RTP timestamp to
-       * running_time, which is needed to generate a mapping from RTP to NTP
-       * timestamps in SR reports */
-      gst_event_parse_new_segment_full (event, &update, &rate, &arate, &format,
-          &start, &stop, &time);
-
-      GST_DEBUG_OBJECT (rtpsession,
-          "configured NEWSEGMENT update %d, rate %lf, applied rate %lf, "
-          "format GST_FORMAT_TIME, "
-          "%" GST_TIME_FORMAT " -- %" GST_TIME_FORMAT
-          ", time %" GST_TIME_FORMAT ", accum %" GST_TIME_FORMAT,
-          update, rate, arate, GST_TIME_ARGS (segment->start),
-          GST_TIME_ARGS (segment->stop), GST_TIME_ARGS (segment->time),
-          GST_TIME_ARGS (segment->accum));
-
-      gst_segment_set_newsegment_full (segment, update, rate,
-          arate, format, start, stop, time);
-
-      /* push event forward */
-      ret = gst_pad_push_event (rtpsession->send_rtp_src, event);
-      break;
-    }
-    case GST_EVENT_EOS:{
-      GstClockTime current_time;
-
-      /* push downstream FIXME, we are not supposed to leave the session just
-       * because we stop sending. */
-      ret = gst_pad_push_event (rtpsession->send_rtp_src, event);
-      current_time = gst_clock_get_time (rtpsession->priv->sysclock);
-      GST_DEBUG_OBJECT (rtpsession, "scheduling BYE message");
-      rtp_session_schedule_bye (rtpsession->priv->session, "End of stream",
-          current_time);
-      break;
-    }
-    default:
-      ret = gst_pad_push_event (rtpsession->send_rtp_src, event);
-      break;
-  }
-  gst_object_unref (rtpsession);
-
-  return ret;
-}
-
-static GstCaps *
-gst_rtp_session_getcaps_send_rtp (GstPad * pad)
-{
-  GstRtpSession *rtpsession;
-  GstRtpSessionPrivate *priv;
-  GstCaps *result;
-  GstStructure *s1, *s2;
-  guint ssrc;
-
-  rtpsession = GST_RTP_SESSION (gst_pad_get_parent (pad));
-  priv = rtpsession->priv;
-
-  ssrc = rtp_session_get_internal_ssrc (priv->session);
-
-  /* we can basically accept anything but we prefer to receive packets with our
-   * internal SSRC so that we don't have to patch it. Create a structure with
-   * the SSRC and another one without. */
-  s1 = gst_structure_new ("application/x-rtp", "ssrc", G_TYPE_UINT, ssrc, NULL);
-  s2 = gst_structure_new ("application/x-rtp", NULL);
-
-  result = gst_caps_new_full (s1, s2, NULL);
-
-  GST_DEBUG_OBJECT (rtpsession, "getting caps %" GST_PTR_FORMAT, result);
-
-  gst_object_unref (rtpsession);
-
-  return result;
-}
-
-static gboolean
-gst_rtp_session_setcaps_send_rtp (GstPad * pad, GstCaps * caps)
-{
-  GstRtpSession *rtpsession;
-  GstRtpSessionPrivate *priv;
-  GstStructure *s = gst_caps_get_structure (caps, 0);
-  guint ssrc;
-
-  rtpsession = GST_RTP_SESSION (gst_pad_get_parent (pad));
-  priv = rtpsession->priv;
-
-  if (gst_structure_get_uint (s, "ssrc", &ssrc)) {
-    GST_DEBUG_OBJECT (rtpsession, "setting internal SSRC to %08x", ssrc);
-    rtp_session_set_internal_ssrc (priv->session, ssrc);
-  }
-
-  gst_object_unref (rtpsession);
-
-  return TRUE;
-}
-
-/* Recieve an RTP packet or a list of packets to be send to the receivers,
- * send to RTP session manager and forward to send_rtp_src.
- */
-static GstFlowReturn
-gst_rtp_session_chain_send_rtp_common (GstPad * pad, gpointer data,
-    gboolean is_list)
-{
-  GstRtpSession *rtpsession;
-  GstRtpSessionPrivate *priv;
-  GstFlowReturn ret;
-  GstClockTime timestamp;
-  GstClockTime current_time;
-  guint64 ntpnstime;
-
-  rtpsession = GST_RTP_SESSION (gst_pad_get_parent (pad));
-  priv = rtpsession->priv;
-
-  GST_LOG_OBJECT (rtpsession, "received RTP %s", is_list ? "list" : "packet");
-
-  /* get NTP time when this packet was captured, this depends on the timestamp. */
-  if (is_list) {
-    GstBuffer *buffer = NULL;
-
-    /* All groups in an list have the same timestamp.
-     * So, just take it from the first group. */
-    buffer = gst_buffer_list_get (GST_BUFFER_LIST_CAST (data), 0, 0);
-    if (buffer)
-      timestamp = GST_BUFFER_TIMESTAMP (buffer);
-    else
-      timestamp = -1;
-  } else {
-    timestamp = GST_BUFFER_TIMESTAMP (GST_BUFFER_CAST (data));
-  }
-  if (GST_CLOCK_TIME_IS_VALID (timestamp)) {
-    /* convert to running time using the segment start value. */
-    ntpnstime =
-        gst_segment_to_running_time (&rtpsession->send_rtp_seg, GST_FORMAT_TIME,
-        timestamp);
-    /* convert to NTP time by adding the NTP base */
-    ntpnstime += priv->ntpnsbase;
-  } else {
-    /* no timestamp, we could take the current running_time and convert it to
-     * NTP time. */
-    ntpnstime = -1;
-  }
-
-  current_time = gst_clock_get_time (priv->sysclock);
-  ret =
-      rtp_session_send_rtp (priv->session, data, is_list, current_time,
-      ntpnstime);
-  if (ret != GST_FLOW_OK)
-    goto push_error;
-
-done:
-  gst_object_unref (rtpsession);
-
-  return ret;
-
-  /* ERRORS */
-push_error:
-  {
-    GST_DEBUG_OBJECT (rtpsession, "process returned %s",
-        gst_flow_get_name (ret));
-    goto done;
-  }
-}
-
-static GstFlowReturn
-gst_rtp_session_chain_send_rtp (GstPad * pad, GstBuffer * buffer)
-{
-  return gst_rtp_session_chain_send_rtp_common (pad, buffer, FALSE);
-}
-
-static GstFlowReturn
-gst_rtp_session_chain_send_rtp_list (GstPad * pad, GstBufferList * list)
-{
-  return gst_rtp_session_chain_send_rtp_common (pad, list, TRUE);
-}
-
-/* Create sinkpad to receive RTP packets from senders. This will also create a
- * srcpad for the RTP packets.
- */
-static GstPad *
-create_recv_rtp_sink (GstRtpSession * rtpsession)
-{
-  GST_DEBUG_OBJECT (rtpsession, "creating RTP sink pad");
-
-  rtpsession->recv_rtp_sink =
-      gst_pad_new_from_static_template (&rtpsession_recv_rtp_sink_template,
-      "recv_rtp_sink");
-  gst_pad_set_chain_function (rtpsession->recv_rtp_sink,
-      gst_rtp_session_chain_recv_rtp);
-  gst_pad_set_event_function (rtpsession->recv_rtp_sink,
-      (GstPadEventFunction) gst_rtp_session_event_recv_rtp_sink);
-  gst_pad_set_setcaps_function (rtpsession->recv_rtp_sink,
-      gst_rtp_session_sink_setcaps);
-  gst_pad_set_internal_link_function (rtpsession->recv_rtp_sink,
-      gst_rtp_session_internal_links);
-  gst_pad_set_active (rtpsession->recv_rtp_sink, TRUE);
-  gst_element_add_pad (GST_ELEMENT_CAST (rtpsession),
-      rtpsession->recv_rtp_sink);
-
-  GST_DEBUG_OBJECT (rtpsession, "creating RTP src pad");
-  rtpsession->recv_rtp_src =
-      gst_pad_new_from_static_template (&rtpsession_recv_rtp_src_template,
-      "recv_rtp_src");
-  gst_pad_set_internal_link_function (rtpsession->recv_rtp_src,
-      gst_rtp_session_internal_links);
-  gst_pad_use_fixed_caps (rtpsession->recv_rtp_src);
-  gst_pad_set_active (rtpsession->recv_rtp_src, TRUE);
-  gst_element_add_pad (GST_ELEMENT_CAST (rtpsession), rtpsession->recv_rtp_src);
-
-  return rtpsession->recv_rtp_sink;
-}
-
-/* Remove sinkpad to receive RTP packets from senders. This will also remove
- * the srcpad for the RTP packets.
- */
-static void
-remove_recv_rtp_sink (GstRtpSession * rtpsession)
-{
-  GST_DEBUG_OBJECT (rtpsession, "removing RTP sink pad");
-
-  /* deactivate from source to sink */
-  gst_pad_set_active (rtpsession->recv_rtp_src, FALSE);
-  gst_pad_set_active (rtpsession->recv_rtp_sink, FALSE);
-
-  /* remove pads */
-  gst_element_remove_pad (GST_ELEMENT_CAST (rtpsession),
-      rtpsession->recv_rtp_sink);
-  rtpsession->recv_rtp_sink = NULL;
-
-  GST_DEBUG_OBJECT (rtpsession, "removing RTP src pad");
-  gst_element_remove_pad (GST_ELEMENT_CAST (rtpsession),
-      rtpsession->recv_rtp_src);
-  rtpsession->recv_rtp_src = NULL;
-}
-
-/* Create a sinkpad to receive RTCP messages from senders, this will also create a
- * sync_src pad for the SR packets.
- */
-static GstPad *
-create_recv_rtcp_sink (GstRtpSession * rtpsession)
-{
-  GST_DEBUG_OBJECT (rtpsession, "creating RTCP sink pad");
-
-  rtpsession->recv_rtcp_sink =
-      gst_pad_new_from_static_template (&rtpsession_recv_rtcp_sink_template,
-      "recv_rtcp_sink");
-  gst_pad_set_chain_function (rtpsession->recv_rtcp_sink,
-      gst_rtp_session_chain_recv_rtcp);
-  gst_pad_set_event_function (rtpsession->recv_rtcp_sink,
-      (GstPadEventFunction) gst_rtp_session_event_recv_rtcp_sink);
-  gst_pad_set_internal_link_function (rtpsession->recv_rtcp_sink,
-      gst_rtp_session_internal_links);
-  gst_pad_set_active (rtpsession->recv_rtcp_sink, TRUE);
-  gst_element_add_pad (GST_ELEMENT_CAST (rtpsession),
-      rtpsession->recv_rtcp_sink);
-
-  GST_DEBUG_OBJECT (rtpsession, "creating sync src pad");
-  rtpsession->sync_src =
-      gst_pad_new_from_static_template (&rtpsession_sync_src_template,
-      "sync_src");
-  gst_pad_set_internal_link_function (rtpsession->sync_src,
-      gst_rtp_session_internal_links);
-  gst_pad_use_fixed_caps (rtpsession->sync_src);
-  gst_pad_set_active (rtpsession->sync_src, TRUE);
-  gst_element_add_pad (GST_ELEMENT_CAST (rtpsession), rtpsession->sync_src);
-
-  return rtpsession->recv_rtcp_sink;
-}
-
-static void
-remove_recv_rtcp_sink (GstRtpSession * rtpsession)
-{
-  GST_DEBUG_OBJECT (rtpsession, "removing RTCP sink pad");
-
-  gst_pad_set_active (rtpsession->sync_src, FALSE);
-  gst_pad_set_active (rtpsession->recv_rtcp_sink, FALSE);
-
-  gst_element_remove_pad (GST_ELEMENT_CAST (rtpsession),
-      rtpsession->recv_rtcp_sink);
-  rtpsession->recv_rtcp_sink = NULL;
-
-  GST_DEBUG_OBJECT (rtpsession, "removing sync src pad");
-  gst_element_remove_pad (GST_ELEMENT_CAST (rtpsession), rtpsession->sync_src);
-  rtpsession->sync_src = NULL;
-}
-
-/* Create a sinkpad to receive RTP packets for receivers. This will also create a
- * send_rtp_src pad.
- */
-static GstPad *
-create_send_rtp_sink (GstRtpSession * rtpsession)
-{
-  GST_DEBUG_OBJECT (rtpsession, "creating pad");
-
-  rtpsession->send_rtp_sink =
-      gst_pad_new_from_static_template (&rtpsession_send_rtp_sink_template,
-      "send_rtp_sink");
-  gst_pad_set_chain_function (rtpsession->send_rtp_sink,
-      gst_rtp_session_chain_send_rtp);
-  gst_pad_set_chain_list_function (rtpsession->send_rtp_sink,
-      gst_rtp_session_chain_send_rtp_list);
-  gst_pad_set_getcaps_function (rtpsession->send_rtp_sink,
-      gst_rtp_session_getcaps_send_rtp);
-  gst_pad_set_setcaps_function (rtpsession->send_rtp_sink,
-      gst_rtp_session_setcaps_send_rtp);
-  gst_pad_set_event_function (rtpsession->send_rtp_sink,
-      (GstPadEventFunction) gst_rtp_session_event_send_rtp_sink);
-  gst_pad_set_internal_link_function (rtpsession->send_rtp_sink,
-      gst_rtp_session_internal_links);
-  gst_pad_set_active (rtpsession->send_rtp_sink, TRUE);
-  gst_element_add_pad (GST_ELEMENT_CAST (rtpsession),
-      rtpsession->send_rtp_sink);
-
-  rtpsession->send_rtp_src =
-      gst_pad_new_from_static_template (&rtpsession_send_rtp_src_template,
-      "send_rtp_src");
-  gst_pad_set_internal_link_function (rtpsession->send_rtp_src,
-      gst_rtp_session_internal_links);
-  gst_pad_set_active (rtpsession->send_rtp_src, TRUE);
-  gst_element_add_pad (GST_ELEMENT_CAST (rtpsession), rtpsession->send_rtp_src);
-
-  return rtpsession->send_rtp_sink;
-}
-
-static void
-remove_send_rtp_sink (GstRtpSession * rtpsession)
-{
-  GST_DEBUG_OBJECT (rtpsession, "removing pad");
-
-  gst_pad_set_active (rtpsession->send_rtp_src, FALSE);
-  gst_pad_set_active (rtpsession->send_rtp_sink, FALSE);
-
-  gst_element_remove_pad (GST_ELEMENT_CAST (rtpsession),
-      rtpsession->send_rtp_sink);
-  rtpsession->send_rtp_sink = NULL;
-
-  gst_element_remove_pad (GST_ELEMENT_CAST (rtpsession),
-      rtpsession->send_rtp_src);
-  rtpsession->send_rtp_src = NULL;
-}
-
-/* Create a srcpad with the RTCP packets to send out.
- * This pad will be driven by the RTP session manager when it wants to send out
- * RTCP packets.
- */
-static GstPad *
-create_send_rtcp_src (GstRtpSession * rtpsession)
-{
-  GST_DEBUG_OBJECT (rtpsession, "creating pad");
-
-  rtpsession->send_rtcp_src =
-      gst_pad_new_from_static_template (&rtpsession_send_rtcp_src_template,
-      "send_rtcp_src");
-  gst_pad_use_fixed_caps (rtpsession->send_rtcp_src);
-  gst_pad_set_active (rtpsession->send_rtcp_src, TRUE);
-  gst_pad_set_internal_link_function (rtpsession->send_rtcp_src,
-      gst_rtp_session_internal_links);
-  gst_pad_set_query_function (rtpsession->send_rtcp_src,
-      gst_rtp_session_query_send_rtcp_src);
-  gst_pad_set_event_function (rtpsession->send_rtcp_src,
-      gst_rtp_session_event_send_rtcp_src);
-  gst_element_add_pad (GST_ELEMENT_CAST (rtpsession),
-      rtpsession->send_rtcp_src);
-
-  return rtpsession->send_rtcp_src;
-}
-
-static void
-remove_send_rtcp_src (GstRtpSession * rtpsession)
-{
-  GST_DEBUG_OBJECT (rtpsession, "removing pad");
-
-  gst_pad_set_active (rtpsession->send_rtcp_src, FALSE);
-
-  gst_element_remove_pad (GST_ELEMENT_CAST (rtpsession),
-      rtpsession->send_rtcp_src);
-  rtpsession->send_rtcp_src = NULL;
-}
-
-static GstPad *
-gst_rtp_session_request_new_pad (GstElement * element,
-    GstPadTemplate * templ, const gchar * name)
-{
-  GstRtpSession *rtpsession;
-  GstElementClass *klass;
-  GstPad *result;
-
-  g_return_val_if_fail (templ != NULL, NULL);
-  g_return_val_if_fail (GST_IS_RTP_SESSION (element), NULL);
-
-  rtpsession = GST_RTP_SESSION (element);
-  klass = GST_ELEMENT_GET_CLASS (element);
-
-  GST_DEBUG_OBJECT (element, "requesting pad %s", GST_STR_NULL (name));
-
-  GST_RTP_SESSION_LOCK (rtpsession);
-
-  /* figure out the template */
-  if (templ == gst_element_class_get_pad_template (klass, "recv_rtp_sink")) {
-    if (rtpsession->recv_rtp_sink != NULL)
-      goto exists;
-
-    result = create_recv_rtp_sink (rtpsession);
-  } else if (templ == gst_element_class_get_pad_template (klass,
-          "recv_rtcp_sink")) {
-    if (rtpsession->recv_rtcp_sink != NULL)
-      goto exists;
-
-    result = create_recv_rtcp_sink (rtpsession);
-  } else if (templ == gst_element_class_get_pad_template (klass,
-          "send_rtp_sink")) {
-    if (rtpsession->send_rtp_sink != NULL)
-      goto exists;
-
-    result = create_send_rtp_sink (rtpsession);
-  } else if (templ == gst_element_class_get_pad_template (klass,
-          "send_rtcp_src")) {
-    if (rtpsession->send_rtcp_src != NULL)
-      goto exists;
-
-    result = create_send_rtcp_src (rtpsession);
-  } else
-    goto wrong_template;
-
-  GST_RTP_SESSION_UNLOCK (rtpsession);
-
-  return result;
-
-  /* ERRORS */
-wrong_template:
-  {
-    GST_RTP_SESSION_UNLOCK (rtpsession);
-    g_warning ("gstrtpsession: this is not our template");
-    return NULL;
-  }
-exists:
-  {
-    GST_RTP_SESSION_UNLOCK (rtpsession);
-    g_warning ("gstrtpsession: pad already requested");
-    return NULL;
-  }
-}
-
-static void
-gst_rtp_session_release_pad (GstElement * element, GstPad * pad)
-{
-  GstRtpSession *rtpsession;
-
-  g_return_if_fail (GST_IS_RTP_SESSION (element));
-  g_return_if_fail (GST_IS_PAD (pad));
-
-  rtpsession = GST_RTP_SESSION (element);
-
-  GST_DEBUG_OBJECT (element, "releasing pad %s:%s", GST_DEBUG_PAD_NAME (pad));
-
-  GST_RTP_SESSION_LOCK (rtpsession);
-
-  if (rtpsession->recv_rtp_sink == pad) {
-    remove_recv_rtp_sink (rtpsession);
-  } else if (rtpsession->recv_rtcp_sink == pad) {
-    remove_recv_rtcp_sink (rtpsession);
-  } else if (rtpsession->send_rtp_sink == pad) {
-    remove_send_rtp_sink (rtpsession);
-  } else if (rtpsession->send_rtcp_src == pad) {
-    remove_send_rtcp_src (rtpsession);
-  } else
-    goto wrong_pad;
-
-  GST_RTP_SESSION_UNLOCK (rtpsession);
-
-  return;
-
-  /* ERRORS */
-wrong_pad:
-  {
-    GST_RTP_SESSION_UNLOCK (rtpsession);
-    g_warning ("gstrtpsession: asked to release an unknown pad");
-    return;
-  }
-}
diff --git a/gst/rtpmanager/gstrtpsession.h b/gst/rtpmanager/gstrtpsession.h
deleted file mode 100644
index 9481a1c2..00000000
--- a/gst/rtpmanager/gstrtpsession.h
+++ /dev/null
@@ -1,81 +0,0 @@
-/* GStreamer
- * Copyright (C) <2007> Wim Taymans <wim.taymans@gmail.com>
- *
- * This library is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Library General Public
- * License as published by the Free Software Foundation; either
- * version 2 of the License, or (at your option) any later version.
- *
- * This library is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Library General Public License for more details.
- *
- * You should have received a copy of the GNU Library General Public
- * License along with this library; if not, write to the
- * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
- * Boston, MA 02111-1307, USA.
- */
-
-#ifndef __GST_RTP_SESSION_H__
-#define __GST_RTP_SESSION_H__
-
-#include <gst/gst.h>
-
-#define GST_TYPE_RTP_SESSION \
-  (gst_rtp_session_get_type())
-#define GST_RTP_SESSION(obj) \
-  (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_RTP_SESSION,GstRtpSession))
-#define GST_RTP_SESSION_CLASS(klass) \
-  (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_RTP_SESSION,GstRtpSessionClass))
-#define GST_IS_RTP_SESSION(obj) \
-  (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_RTP_SESSION))
-#define GST_IS_RTP_SESSION_CLASS(klass) \
-  (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_RTP_SESSION))
-#define GST_RTP_SESSION_CAST(obj) ((GstRtpSession *)(obj))
-
-typedef struct _GstRtpSession GstRtpSession;
-typedef struct _GstRtpSessionClass GstRtpSessionClass;
-typedef struct _GstRtpSessionPrivate GstRtpSessionPrivate;
-
-struct _GstRtpSession {
-  GstElement     element;
-
-  /*< private >*/
-  GstPad        *recv_rtp_sink;
-  GstSegment     recv_rtp_seg;
-  GstPad        *recv_rtcp_sink;
-  GstPad        *send_rtp_sink;
-  GstSegment     send_rtp_seg;
-
-  GstPad        *recv_rtp_src;
-  GstPad        *sync_src;
-  GstPad        *send_rtp_src;
-  GstPad        *send_rtcp_src;
-
-  GstRtpSessionPrivate *priv;
-};
-
-struct _GstRtpSessionClass {
-  GstElementClass parent_class;
-
-  /* signals */
-  GstCaps* (*request_pt_map) (GstRtpSession *sess, guint pt);
-  void     (*clear_pt_map)   (GstRtpSession *sess);
-
-  void     (*on_new_ssrc)       (GstRtpSession *sess, guint32 ssrc);
-  void     (*on_ssrc_collision) (GstRtpSession *sess, guint32 ssrc);
-  void     (*on_ssrc_validated) (GstRtpSession *sess, guint32 ssrc);
-  void     (*on_ssrc_active)    (GstRtpSession *sess, guint32 ssrc);
-  void     (*on_ssrc_sdes)      (GstRtpSession *sess, guint32 ssrc);
-  void     (*on_bye_ssrc)       (GstRtpSession *sess, guint32 ssrc);
-  void     (*on_bye_timeout)    (GstRtpSession *sess, guint32 ssrc);
-  void     (*on_timeout)        (GstRtpSession *sess, guint32 ssrc);
-  void     (*on_sender_timeout) (GstRtpSession *sess, guint32 ssrc);
-};
-
-GType gst_rtp_session_get_type (void);
-
-void gst_rtp_session_set_ssrc (GstRtpSession *sess, guint32 ssrc);
-
-#endif /* __GST_RTP_SESSION_H__ */
diff --git a/gst/rtpmanager/gstrtpssrcdemux.c b/gst/rtpmanager/gstrtpssrcdemux.c
deleted file mode 100644
index 6a305d8e..00000000
--- a/gst/rtpmanager/gstrtpssrcdemux.c
+++ /dev/null
@@ -1,722 +0,0 @@
-/* GStreamer
- * Copyright (C) <2007> Wim Taymans <wim.taymans@gmail.com>
- *
- * RTP SSRC demuxer
- *
- * This library is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Library General Public
- * License as published by the Free Software Foundation; either
- * version 2 of the License, or (at your option) any later version.
- *
- * This library is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Library General Public License for more details.
- *
- * You should have received a copy of the GNU Library General Public
- * License along with this library; if not, write to the
- * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
- * Boston, MA 02111-1307, USA.
- */
-
-/**
- * SECTION:element-gstrtpssrcdemux
- *
- * 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.
- * 
- * For each SSRC that is detected, a new pad will be created and the
- * #GstRtpSsrcDemux::new-ssrc-pad signal will be emitted. 
- * 
- * <refsect2>
- * <title>Example pipelines</title>
- * |[
- * gst-launch udpsrc caps="application/x-rtp" ! gstrtpssrcdemux ! fakesink
- * ]| Takes an RTP stream and send the RTP packets with the first detected SSRC
- * to fakesink, discarding the other SSRCs.
- * </refsect2>
- *
- * Last reviewed on 2007-05-28 (0.10.5)
- */
-
-#ifdef HAVE_CONFIG_H
-#include "config.h"
-#endif
-
-#include <string.h>
-#include <gst/rtp/gstrtpbuffer.h>
-#include <gst/rtp/gstrtcpbuffer.h>
-
-#include "gstrtpbin-marshal.h"
-#include "gstrtpssrcdemux.h"
-
-GST_DEBUG_CATEGORY_STATIC (gst_rtp_ssrc_demux_debug);
-#define GST_CAT_DEFAULT gst_rtp_ssrc_demux_debug
-
-/* generic templates */
-static GstStaticPadTemplate rtp_ssrc_demux_sink_template =
-GST_STATIC_PAD_TEMPLATE ("sink",
-    GST_PAD_SINK,
-    GST_PAD_ALWAYS,
-    GST_STATIC_CAPS ("application/x-rtp")
-    );
-
-static GstStaticPadTemplate rtp_ssrc_demux_rtcp_sink_template =
-GST_STATIC_PAD_TEMPLATE ("rtcp_sink",
-    GST_PAD_SINK,
-    GST_PAD_ALWAYS,
-    GST_STATIC_CAPS ("application/x-rtcp")
-    );
-
-static GstStaticPadTemplate rtp_ssrc_demux_src_template =
-GST_STATIC_PAD_TEMPLATE ("src_%d",
-    GST_PAD_SRC,
-    GST_PAD_SOMETIMES,
-    GST_STATIC_CAPS ("application/x-rtp")
-    );
-
-static GstStaticPadTemplate rtp_ssrc_demux_rtcp_src_template =
-GST_STATIC_PAD_TEMPLATE ("rtcp_src_%d",
-    GST_PAD_SRC,
-    GST_PAD_SOMETIMES,
-    GST_STATIC_CAPS ("application/x-rtcp")
-    );
-
-static GstElementDetails gst_rtp_ssrc_demux_details = {
-  "RTP SSRC Demux",
-  "Demux/Network/RTP",
-  "Splits RTP streams based on the SSRC",
-  "Wim Taymans <wim.taymans@gmail.com>"
-};
-
-#define GST_PAD_LOCK(obj)   (g_mutex_lock ((obj)->padlock))
-#define GST_PAD_UNLOCK(obj) (g_mutex_unlock ((obj)->padlock))
-
-/* signals */
-enum
-{
-  SIGNAL_NEW_SSRC_PAD,
-  SIGNAL_REMOVED_SSRC_PAD,
-  SIGNAL_CLEAR_SSRC,
-  LAST_SIGNAL
-};
-
-GST_BOILERPLATE (GstRtpSsrcDemux, gst_rtp_ssrc_demux, GstElement,
-    GST_TYPE_ELEMENT);
-
-
-/* GObject vmethods */
-static void gst_rtp_ssrc_demux_dispose (GObject * object);
-static void gst_rtp_ssrc_demux_finalize (GObject * object);
-
-/* GstElement vmethods */
-static GstStateChangeReturn gst_rtp_ssrc_demux_change_state (GstElement *
-    element, GstStateChange transition);
-
-static void gst_rtp_ssrc_demux_clear_ssrc (GstRtpSsrcDemux * demux,
-    guint32 ssrc);
-
-/* sinkpad stuff */
-static GstFlowReturn gst_rtp_ssrc_demux_chain (GstPad * pad, GstBuffer * buf);
-static gboolean gst_rtp_ssrc_demux_sink_event (GstPad * pad, GstEvent * event);
-
-static GstFlowReturn gst_rtp_ssrc_demux_rtcp_chain (GstPad * pad,
-    GstBuffer * buf);
-static gboolean gst_rtp_ssrc_demux_rtcp_sink_event (GstPad * pad,
-    GstEvent * event);
-
-/* srcpad stuff */
-static gboolean gst_rtp_ssrc_demux_src_event (GstPad * pad, GstEvent * event);
-static GList *gst_rtp_ssrc_demux_internal_links (GstPad * pad);
-static gboolean gst_rtp_ssrc_demux_src_query (GstPad * pad, GstQuery * query);
-
-static guint gst_rtp_ssrc_demux_signals[LAST_SIGNAL] = { 0 };
-
-/*
- * Item for storing GstPad <-> SSRC pairs.
- */
-struct _GstRtpSsrcDemuxPad
-{
-  guint32 ssrc;
-  GstPad *rtp_pad;
-  GstCaps *caps;
-  GstPad *rtcp_pad;
-};
-
-/* find a src pad for a given SSRC, returns NULL if the SSRC was not found
- */
-static GstRtpSsrcDemuxPad *
-find_demux_pad_for_ssrc (GstRtpSsrcDemux * demux, guint32 ssrc)
-{
-  GSList *walk;
-
-  for (walk = demux->srcpads; walk; walk = g_slist_next (walk)) {
-    GstRtpSsrcDemuxPad *pad = (GstRtpSsrcDemuxPad *) walk->data;
-
-    if (pad->ssrc == ssrc)
-      return pad;
-  }
-  return NULL;
-}
-
-/* with PAD_LOCK */
-static GstRtpSsrcDemuxPad *
-create_demux_pad_for_ssrc (GstRtpSsrcDemux * demux, guint32 ssrc,
-    GstClockTime timestamp)
-{
-  GstPad *rtp_pad, *rtcp_pad;
-  GstElementClass *klass;
-  GstPadTemplate *templ;
-  gchar *padname;
-  GstRtpSsrcDemuxPad *demuxpad;
-
-  GST_DEBUG_OBJECT (demux, "creating pad for SSRC %08x", ssrc);
-
-  klass = GST_ELEMENT_GET_CLASS (demux);
-  templ = gst_element_class_get_pad_template (klass, "src_%d");
-  padname = g_strdup_printf ("src_%d", ssrc);
-  rtp_pad = gst_pad_new_from_template (templ, padname);
-  g_free (padname);
-
-  templ = gst_element_class_get_pad_template (klass, "rtcp_src_%d");
-  padname = g_strdup_printf ("rtcp_src_%d", ssrc);
-  rtcp_pad = gst_pad_new_from_template (templ, padname);
-  g_free (padname);
-
-  /* we use the first timestamp received to calculate the difference between
-   * timestamps on all streams */
-  GST_DEBUG_OBJECT (demux, "SSRC %08x, first timestamp %" GST_TIME_FORMAT,
-      ssrc, GST_TIME_ARGS (timestamp));
-
-  /* wrap in structure and add to list */
-  demuxpad = g_new0 (GstRtpSsrcDemuxPad, 1);
-  demuxpad->ssrc = ssrc;
-  demuxpad->rtp_pad = rtp_pad;
-  demuxpad->rtcp_pad = rtcp_pad;
-
-  GST_DEBUG_OBJECT (demux, "first timestamp %" GST_TIME_FORMAT,
-      GST_TIME_ARGS (timestamp));
-
-  gst_pad_set_element_private (rtp_pad, demuxpad);
-  gst_pad_set_element_private (rtcp_pad, demuxpad);
-
-  demux->srcpads = g_slist_prepend (demux->srcpads, demuxpad);
-
-  /* copy caps from input */
-  gst_pad_set_caps (rtp_pad, GST_PAD_CAPS (demux->rtp_sink));
-  gst_pad_use_fixed_caps (rtp_pad);
-  gst_pad_set_caps (rtcp_pad, GST_PAD_CAPS (demux->rtcp_sink));
-  gst_pad_use_fixed_caps (rtcp_pad);
-
-  gst_pad_set_event_function (rtp_pad, gst_rtp_ssrc_demux_src_event);
-  gst_pad_set_query_function (rtp_pad, gst_rtp_ssrc_demux_src_query);
-  gst_pad_set_internal_link_function (rtp_pad,
-      gst_rtp_ssrc_demux_internal_links);
-  gst_pad_set_active (rtp_pad, TRUE);
-
-  gst_pad_set_internal_link_function (rtcp_pad,
-      gst_rtp_ssrc_demux_internal_links);
-  gst_pad_set_active (rtcp_pad, TRUE);
-
-  gst_element_add_pad (GST_ELEMENT_CAST (demux), rtp_pad);
-  gst_element_add_pad (GST_ELEMENT_CAST (demux), rtcp_pad);
-
-  g_signal_emit (G_OBJECT (demux),
-      gst_rtp_ssrc_demux_signals[SIGNAL_NEW_SSRC_PAD], 0, ssrc, rtp_pad);
-
-  return demuxpad;
-}
-
-static void
-gst_rtp_ssrc_demux_base_init (gpointer g_class)
-{
-  GstElementClass *gstelement_klass = GST_ELEMENT_CLASS (g_class);
-
-  gst_element_class_add_pad_template (gstelement_klass,
-      gst_static_pad_template_get (&rtp_ssrc_demux_sink_template));
-  gst_element_class_add_pad_template (gstelement_klass,
-      gst_static_pad_template_get (&rtp_ssrc_demux_rtcp_sink_template));
-  gst_element_class_add_pad_template (gstelement_klass,
-      gst_static_pad_template_get (&rtp_ssrc_demux_src_template));
-  gst_element_class_add_pad_template (gstelement_klass,
-      gst_static_pad_template_get (&rtp_ssrc_demux_rtcp_src_template));
-
-  gst_element_class_set_details (gstelement_klass, &gst_rtp_ssrc_demux_details);
-}
-
-static void
-gst_rtp_ssrc_demux_class_init (GstRtpSsrcDemuxClass * klass)
-{
-  GObjectClass *gobject_klass;
-  GstElementClass *gstelement_klass;
-  GstRtpSsrcDemuxClass *gstrtpssrcdemux_klass;
-
-  gobject_klass = (GObjectClass *) klass;
-  gstelement_klass = (GstElementClass *) klass;
-  gstrtpssrcdemux_klass = (GstRtpSsrcDemuxClass *) klass;
-
-  gobject_klass->dispose = GST_DEBUG_FUNCPTR (gst_rtp_ssrc_demux_dispose);
-  gobject_klass->finalize = GST_DEBUG_FUNCPTR (gst_rtp_ssrc_demux_finalize);
-
-  /**
-   * GstRtpSsrcDemux::new-ssrc-pad:
-   * @demux: the object which received the signal
-   * @ssrc: the SSRC of the pad
-   * @pad: the new pad.
-   *
-   * Emited when a new SSRC pad has been created.
-   */
-  gst_rtp_ssrc_demux_signals[SIGNAL_NEW_SSRC_PAD] =
-      g_signal_new ("new-ssrc-pad",
-      G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
-      G_STRUCT_OFFSET (GstRtpSsrcDemuxClass, new_ssrc_pad),
-      NULL, NULL, gst_rtp_bin_marshal_VOID__UINT_OBJECT,
-      G_TYPE_NONE, 2, G_TYPE_UINT, GST_TYPE_PAD);
-
-  /**
-   * GstRtpSsrcDemux::removed-ssrc-pad:
-   * @demux: the object which received the signal
-   * @ssrc: the SSRC of the pad
-   * @pad: the removed pad.
-   *
-   * Emited when a SSRC pad has been removed.
-   */
-  gst_rtp_ssrc_demux_signals[SIGNAL_REMOVED_SSRC_PAD] =
-      g_signal_new ("removed-ssrc-pad",
-      G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
-      G_STRUCT_OFFSET (GstRtpSsrcDemuxClass, removed_ssrc_pad),
-      NULL, NULL, gst_rtp_bin_marshal_VOID__UINT_OBJECT,
-      G_TYPE_NONE, 2, G_TYPE_UINT, GST_TYPE_PAD);
-
-  /**
-   * GstRtpSsrcDemux::clear-ssrc:
-   * @demux: the object which received the signal
-   * @ssrc: the SSRC of the pad
-   *
-   * Action signal to remove the pad for SSRC.
-   */
-  gst_rtp_ssrc_demux_signals[SIGNAL_CLEAR_SSRC] =
-      g_signal_new ("clear-ssrc",
-      G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION,
-      G_STRUCT_OFFSET (GstRtpSsrcDemuxClass, clear_ssrc),
-      NULL, NULL, gst_rtp_bin_marshal_VOID__UINT, G_TYPE_NONE, 1, G_TYPE_UINT);
-
-  gstelement_klass->change_state =
-      GST_DEBUG_FUNCPTR (gst_rtp_ssrc_demux_change_state);
-  gstrtpssrcdemux_klass->clear_ssrc =
-      GST_DEBUG_FUNCPTR (gst_rtp_ssrc_demux_clear_ssrc);
-
-  GST_DEBUG_CATEGORY_INIT (gst_rtp_ssrc_demux_debug,
-      "rtpssrcdemux", 0, "RTP SSRC demuxer");
-}
-
-static void
-gst_rtp_ssrc_demux_init (GstRtpSsrcDemux * demux,
-    GstRtpSsrcDemuxClass * g_class)
-{
-  GstElementClass *klass = GST_ELEMENT_GET_CLASS (demux);
-
-  demux->rtp_sink =
-      gst_pad_new_from_template (gst_element_class_get_pad_template (klass,
-          "sink"), "sink");
-  gst_pad_set_chain_function (demux->rtp_sink, gst_rtp_ssrc_demux_chain);
-  gst_pad_set_event_function (demux->rtp_sink, gst_rtp_ssrc_demux_sink_event);
-  gst_element_add_pad (GST_ELEMENT_CAST (demux), demux->rtp_sink);
-
-  demux->rtcp_sink =
-      gst_pad_new_from_template (gst_element_class_get_pad_template (klass,
-          "rtcp_sink"), "rtcp_sink");
-  gst_pad_set_chain_function (demux->rtcp_sink, gst_rtp_ssrc_demux_rtcp_chain);
-  gst_pad_set_event_function (demux->rtcp_sink,
-      gst_rtp_ssrc_demux_rtcp_sink_event);
-  gst_element_add_pad (GST_ELEMENT_CAST (demux), demux->rtcp_sink);
-
-  demux->padlock = g_mutex_new ();
-
-  gst_segment_init (&demux->segment, GST_FORMAT_UNDEFINED);
-}
-
-static void
-gst_rtp_ssrc_demux_reset (GstRtpSsrcDemux * demux)
-{
-  GSList *walk;
-
-  for (walk = demux->srcpads; walk; walk = g_slist_next (walk)) {
-    GstRtpSsrcDemuxPad *dpad = (GstRtpSsrcDemuxPad *) walk->data;
-
-    gst_pad_set_active (dpad->rtp_pad, FALSE);
-    gst_pad_set_active (dpad->rtcp_pad, FALSE);
-
-    gst_element_remove_pad (GST_ELEMENT_CAST (demux), dpad->rtp_pad);
-    gst_element_remove_pad (GST_ELEMENT_CAST (demux), dpad->rtcp_pad);
-    g_free (dpad);
-  }
-  g_slist_free (demux->srcpads);
-  demux->srcpads = NULL;
-}
-
-static void
-gst_rtp_ssrc_demux_dispose (GObject * object)
-{
-  GstRtpSsrcDemux *demux;
-
-  demux = GST_RTP_SSRC_DEMUX (object);
-
-  gst_rtp_ssrc_demux_reset (demux);
-
-  G_OBJECT_CLASS (parent_class)->dispose (object);
-}
-
-static void
-gst_rtp_ssrc_demux_finalize (GObject * object)
-{
-  GstRtpSsrcDemux *demux;
-
-  demux = GST_RTP_SSRC_DEMUX (object);
-  g_mutex_free (demux->padlock);
-
-  G_OBJECT_CLASS (parent_class)->finalize (object);
-}
-
-static void
-gst_rtp_ssrc_demux_clear_ssrc (GstRtpSsrcDemux * demux, guint32 ssrc)
-{
-  GstRtpSsrcDemuxPad *dpad;
-
-  GST_PAD_LOCK (demux);
-  dpad = find_demux_pad_for_ssrc (demux, ssrc);
-  if (dpad != NULL)
-    goto unknown_pad;
-
-  GST_DEBUG_OBJECT (demux, "clearing pad for SSRC %08x", ssrc);
-
-  demux->srcpads = g_slist_remove (demux->srcpads, dpad);
-  GST_PAD_UNLOCK (demux);
-
-  gst_pad_set_active (dpad->rtp_pad, FALSE);
-  gst_pad_set_active (dpad->rtcp_pad, FALSE);
-
-  g_signal_emit (G_OBJECT (demux),
-      gst_rtp_ssrc_demux_signals[SIGNAL_REMOVED_SSRC_PAD], 0, ssrc,
-      dpad->rtp_pad);
-
-  gst_element_remove_pad (GST_ELEMENT_CAST (demux), dpad->rtp_pad);
-  gst_element_remove_pad (GST_ELEMENT_CAST (demux), dpad->rtcp_pad);
-
-  g_free (dpad);
-
-  return;
-
-  /* ERRORS */
-unknown_pad:
-  {
-    g_warning ("unknown SSRC %08x", ssrc);
-    return;
-  }
-}
-
-static gboolean
-gst_rtp_ssrc_demux_sink_event (GstPad * pad, GstEvent * event)
-{
-  GstRtpSsrcDemux *demux;
-  gboolean res = FALSE;
-
-  demux = GST_RTP_SSRC_DEMUX (gst_pad_get_parent (pad));
-
-  switch (GST_EVENT_TYPE (event)) {
-    case GST_EVENT_FLUSH_STOP:
-      gst_segment_init (&demux->segment, GST_FORMAT_UNDEFINED);
-    case GST_EVENT_NEWSEGMENT:
-    default:
-    {
-      GSList *walk;
-
-      res = TRUE;
-      GST_PAD_LOCK (demux);
-      for (walk = demux->srcpads; walk; walk = g_slist_next (walk)) {
-        GstRtpSsrcDemuxPad *pad = (GstRtpSsrcDemuxPad *) walk->data;
-
-        gst_event_ref (event);
-        res &= gst_pad_push_event (pad->rtp_pad, event);
-      }
-      GST_PAD_UNLOCK (demux);
-      gst_event_unref (event);
-      break;
-    }
-  }
-
-  gst_object_unref (demux);
-  return res;
-}
-
-static gboolean
-gst_rtp_ssrc_demux_rtcp_sink_event (GstPad * pad, GstEvent * event)
-{
-  GstRtpSsrcDemux *demux;
-  gboolean res = FALSE;
-
-  demux = GST_RTP_SSRC_DEMUX (gst_pad_get_parent (pad));
-
-  switch (GST_EVENT_TYPE (event)) {
-    case GST_EVENT_NEWSEGMENT:
-    default:
-    {
-      GSList *walk;
-
-      res = TRUE;
-      GST_PAD_LOCK (demux);
-      for (walk = demux->srcpads; walk; walk = g_slist_next (walk)) {
-        GstRtpSsrcDemuxPad *pad = (GstRtpSsrcDemuxPad *) walk->data;
-
-        gst_event_ref (event);
-        res &= gst_pad_push_event (pad->rtcp_pad, event);
-      }
-      GST_PAD_UNLOCK (demux);
-      gst_event_unref (event);
-      break;
-    }
-  }
-  gst_object_unref (demux);
-  return res;
-}
-
-static GstFlowReturn
-gst_rtp_ssrc_demux_chain (GstPad * pad, GstBuffer * buf)
-{
-  GstFlowReturn ret;
-  GstRtpSsrcDemux *demux;
-  guint32 ssrc;
-  GstRtpSsrcDemuxPad *dpad;
-
-  demux = GST_RTP_SSRC_DEMUX (GST_OBJECT_PARENT (pad));
-
-  if (!gst_rtp_buffer_validate (buf))
-    goto invalid_payload;
-
-  ssrc = gst_rtp_buffer_get_ssrc (buf);
-
-  GST_DEBUG_OBJECT (demux, "received buffer of SSRC %08x", ssrc);
-
-  GST_PAD_LOCK (demux);
-  dpad = find_demux_pad_for_ssrc (demux, ssrc);
-  if (dpad == NULL) {
-    if (!(dpad =
-            create_demux_pad_for_ssrc (demux, ssrc,
-                GST_BUFFER_TIMESTAMP (buf))))
-      goto create_failed;
-  }
-  GST_PAD_UNLOCK (demux);
-
-  /* push to srcpad */
-  ret = gst_pad_push (dpad->rtp_pad, buf);
-
-  return ret;
-
-  /* ERRORS */
-invalid_payload:
-  {
-    /* this is fatal and should be filtered earlier */
-    GST_ELEMENT_ERROR (demux, STREAM, DECODE, (NULL),
-        ("Dropping invalid RTP payload"));
-    gst_buffer_unref (buf);
-    return GST_FLOW_ERROR;
-  }
-create_failed:
-  {
-    GST_ELEMENT_ERROR (demux, STREAM, DECODE, (NULL),
-        ("Could not create new pad"));
-    GST_PAD_UNLOCK (demux);
-    gst_buffer_unref (buf);
-    return GST_FLOW_ERROR;
-  }
-}
-
-static GstFlowReturn
-gst_rtp_ssrc_demux_rtcp_chain (GstPad * pad, GstBuffer * buf)
-{
-  GstFlowReturn ret;
-  GstRtpSsrcDemux *demux;
-  guint32 ssrc;
-  GstRtpSsrcDemuxPad *dpad;
-  GstRTCPPacket packet;
-
-  demux = GST_RTP_SSRC_DEMUX (GST_OBJECT_PARENT (pad));
-
-  if (!gst_rtcp_buffer_validate (buf))
-    goto invalid_rtcp;
-
-  if (!gst_rtcp_buffer_get_first_packet (buf, &packet))
-    goto invalid_rtcp;
-
-  /* first packet must be SR or RR or else the validate would have failed */
-  switch (gst_rtcp_packet_get_type (&packet)) {
-    case GST_RTCP_TYPE_SR:
-      /* get the ssrc so that we can route it to the right source pad */
-      gst_rtcp_packet_sr_get_sender_info (&packet, &ssrc, NULL, NULL, NULL,
-          NULL);
-      break;
-    default:
-      goto unexpected_rtcp;
-  }
-
-  GST_DEBUG_OBJECT (demux, "received RTCP of SSRC %08x", ssrc);
-
-  GST_PAD_LOCK (demux);
-  dpad = find_demux_pad_for_ssrc (demux, ssrc);
-  if (dpad == NULL) {
-    GST_DEBUG_OBJECT (demux, "creating pad for SSRC %08x", ssrc);
-    if (!(dpad = create_demux_pad_for_ssrc (demux, ssrc, -1)))
-      goto create_failed;
-  }
-  GST_PAD_UNLOCK (demux);
-
-  /* push to srcpad */
-  ret = gst_pad_push (dpad->rtcp_pad, buf);
-
-  return ret;
-
-  /* ERRORS */
-invalid_rtcp:
-  {
-    /* this is fatal and should be filtered earlier */
-    GST_ELEMENT_ERROR (demux, STREAM, DECODE, (NULL),
-        ("Dropping invalid RTCP packet"));
-    gst_buffer_unref (buf);
-    return GST_FLOW_ERROR;
-  }
-unexpected_rtcp:
-  {
-    GST_DEBUG_OBJECT (demux, "dropping unexpected RTCP packet");
-    gst_buffer_unref (buf);
-    return GST_FLOW_OK;
-  }
-create_failed:
-  {
-    GST_ELEMENT_ERROR (demux, STREAM, DECODE, (NULL),
-        ("Could not create new pad"));
-    GST_PAD_UNLOCK (demux);
-    gst_buffer_unref (buf);
-    return GST_FLOW_ERROR;
-  }
-}
-
-static gboolean
-gst_rtp_ssrc_demux_src_event (GstPad * pad, GstEvent * event)
-{
-  GstRtpSsrcDemux *demux;
-  gboolean res = FALSE;
-
-  demux = GST_RTP_SSRC_DEMUX (gst_pad_get_parent (pad));
-
-  switch (GST_EVENT_TYPE (event)) {
-    case GST_EVENT_SEEK:
-    default:
-      res = gst_pad_event_default (pad, event);
-      break;
-  }
-  gst_object_unref (demux);
-  return res;
-}
-
-static GList *
-gst_rtp_ssrc_demux_internal_links (GstPad * pad)
-{
-  GstRtpSsrcDemux *demux;
-  GList *res = NULL;
-  GSList *walk;
-
-  demux = GST_RTP_SSRC_DEMUX (gst_pad_get_parent (pad));
-
-  GST_PAD_LOCK (demux);
-  for (walk = demux->srcpads; walk; walk = g_slist_next (walk)) {
-    GstRtpSsrcDemuxPad *dpad = (GstRtpSsrcDemuxPad *) walk->data;
-
-    if (pad == demux->rtp_sink) {
-      res = g_list_prepend (res, dpad->rtp_pad);
-    } else if (pad == demux->rtcp_sink) {
-      res = g_list_prepend (res, dpad->rtcp_pad);
-    } else if (pad == dpad->rtp_pad) {
-      res = g_list_prepend (res, demux->rtp_sink);
-      break;
-    } else if (pad == dpad->rtcp_pad) {
-      res = g_list_prepend (res, demux->rtcp_sink);
-      break;
-    }
-  }
-  GST_PAD_UNLOCK (demux);
-
-  gst_object_unref (demux);
-  return res;
-}
-
-static gboolean
-gst_rtp_ssrc_demux_src_query (GstPad * pad, GstQuery * query)
-{
-  GstRtpSsrcDemux *demux;
-  gboolean res = FALSE;
-
-  demux = GST_RTP_SSRC_DEMUX (gst_pad_get_parent (pad));
-
-  switch (GST_QUERY_TYPE (query)) {
-    case GST_QUERY_LATENCY:
-    {
-
-      if ((res = gst_pad_peer_query (demux->rtp_sink, query))) {
-        gboolean live;
-        GstClockTime min_latency, max_latency;
-        GstRtpSsrcDemuxPad *demuxpad;
-
-        demuxpad = gst_pad_get_element_private (pad);
-
-        gst_query_parse_latency (query, &live, &min_latency, &max_latency);
-
-        GST_DEBUG_OBJECT (demux, "peer min latency %" GST_TIME_FORMAT,
-            GST_TIME_ARGS (min_latency));
-
-        GST_DEBUG_OBJECT (demux, "latency for SSRC %08x", demuxpad->ssrc);
-
-        gst_query_set_latency (query, live, min_latency, max_latency);
-      }
-      break;
-    }
-    default:
-      res = gst_pad_query_default (pad, query);
-      break;
-  }
-  gst_object_unref (demux);
-
-  return res;
-}
-
-static GstStateChangeReturn
-gst_rtp_ssrc_demux_change_state (GstElement * element,
-    GstStateChange transition)
-{
-  GstStateChangeReturn ret;
-  GstRtpSsrcDemux *demux;
-
-  demux = GST_RTP_SSRC_DEMUX (element);
-
-  switch (transition) {
-    case GST_STATE_CHANGE_NULL_TO_READY:
-    case GST_STATE_CHANGE_READY_TO_PAUSED:
-    case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
-    default:
-      break;
-  }
-
-  ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
-
-  switch (transition) {
-    case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
-      break;
-    case GST_STATE_CHANGE_PAUSED_TO_READY:
-      gst_rtp_ssrc_demux_reset (demux);
-      break;
-    case GST_STATE_CHANGE_READY_TO_NULL:
-    default:
-      break;
-  }
-  return ret;
-}
diff --git a/gst/rtpmanager/gstrtpssrcdemux.h b/gst/rtpmanager/gstrtpssrcdemux.h
deleted file mode 100644
index d5a13caf..00000000
--- a/gst/rtpmanager/gstrtpssrcdemux.h
+++ /dev/null
@@ -1,62 +0,0 @@
-/* GStreamer
- * Copyright (C) <2007> Wim Taymans <wim.taymans@gmail.com>
- *
- * This library is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Library General Public
- * License as published by the Free Software Foundation; either
- * version 2 of the License, or (at your option) any later version.
- *
- * This library is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Library General Public License for more details.
- *
- * You should have received a copy of the GNU Library General Public
- * License along with this library; if not, write to the
- * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
- * Boston, MA 02111-1307, USA.
- */
-
-#ifndef __GST_RTP_SSRC_DEMUX_H__
-#define __GST_RTP_SSRC_DEMUX_H__
-
-#include <gst/gst.h>
-
-#define GST_TYPE_RTP_SSRC_DEMUX            (gst_rtp_ssrc_demux_get_type())
-#define GST_RTP_SSRC_DEMUX(obj)            (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_RTP_SSRC_DEMUX,GstRtpSsrcDemux))
-#define GST_RTP_SSRC_DEMUX_CLASS(klass)    (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_RTP_SSRC_DEMUX,GstRtpSsrcDemuxClass))
-#define GST_IS_RTP_SSRC_DEMUX(obj)         (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_RTP_SSRC_DEMUX))
-#define GST_IS_RTP_SSRC_DEMUX_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_RTP_SSRC_DEMUX))
-
-typedef struct _GstRtpSsrcDemux GstRtpSsrcDemux;
-typedef struct _GstRtpSsrcDemuxClass GstRtpSsrcDemuxClass;
-typedef struct _GstRtpSsrcDemuxPad GstRtpSsrcDemuxPad;
-
-struct _GstRtpSsrcDemux
-{
-  GstElement parent;
-
-  GstSegment   segment;
-
-  GstPad *rtp_sink;
-  GstPad *rtcp_sink;
-
-  GMutex *padlock;
-  GSList *srcpads;
-};
-
-struct _GstRtpSsrcDemuxClass
-{
-  GstElementClass parent_class;
-
-  /* signals */
-  void (*new_ssrc_pad)     (GstRtpSsrcDemux *demux, guint32 ssrc, GstPad *pad);
-  void (*removed_ssrc_pad) (GstRtpSsrcDemux *demux, guint32 ssrc, GstPad *pad);
-
-  /* actions */
-  void (*clear_ssrc)       (GstRtpSsrcDemux *demux, guint32 ssrc);
-};
-
-GType gst_rtp_ssrc_demux_get_type (void);
-
-#endif /* __GST_RTP_SSRC_DEMUX_H__ */
diff --git a/gst/rtpmanager/rtpjitterbuffer.c b/gst/rtpmanager/rtpjitterbuffer.c
deleted file mode 100644
index 123d26f0..00000000
--- a/gst/rtpmanager/rtpjitterbuffer.c
+++ /dev/null
@@ -1,593 +0,0 @@
-/* GStreamer
- * Copyright (C) <2007> Wim Taymans <wim.taymans@gmail.com>
- *
- * This library is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Library General Public
- * License as published by the Free Software Foundation; either
- * version 2 of the License, or (at your option) any later version.
- *
- * This library is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Library General Public License for more details.
- *
- * You should have received a copy of the GNU Library General Public
- * License along with this library; if not, write to the
- * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
- * Boston, MA 02111-1307, USA.
- */
-#include <string.h>
-#include <stdlib.h>
-
-#include <gst/rtp/gstrtpbuffer.h>
-#include <gst/rtp/gstrtcpbuffer.h>
-
-#include "rtpjitterbuffer.h"
-
-GST_DEBUG_CATEGORY_STATIC (rtp_jitter_buffer_debug);
-#define GST_CAT_DEFAULT rtp_jitter_buffer_debug
-
-#define MAX_WINDOW	RTP_JITTER_BUFFER_MAX_WINDOW
-#define MAX_TIME	(2 * GST_SECOND)
-
-/* signals and args */
-enum
-{
-  LAST_SIGNAL
-};
-
-enum
-{
-  PROP_0
-};
-
-/* GObject vmethods */
-static void rtp_jitter_buffer_finalize (GObject * object);
-
-/* static guint rtp_jitter_buffer_signals[LAST_SIGNAL] = { 0 }; */
-
-G_DEFINE_TYPE (RTPJitterBuffer, rtp_jitter_buffer, G_TYPE_OBJECT);
-
-static void
-rtp_jitter_buffer_class_init (RTPJitterBufferClass * klass)
-{
-  GObjectClass *gobject_class;
-
-  gobject_class = (GObjectClass *) klass;
-
-  gobject_class->finalize = rtp_jitter_buffer_finalize;
-
-  GST_DEBUG_CATEGORY_INIT (rtp_jitter_buffer_debug, "rtpjitterbuffer", 0,
-      "RTP Jitter Buffer");
-}
-
-static void
-rtp_jitter_buffer_init (RTPJitterBuffer * jbuf)
-{
-  jbuf->packets = g_queue_new ();
-
-  rtp_jitter_buffer_reset_skew (jbuf);
-}
-
-static void
-rtp_jitter_buffer_finalize (GObject * object)
-{
-  RTPJitterBuffer *jbuf;
-
-  jbuf = RTP_JITTER_BUFFER_CAST (object);
-
-  rtp_jitter_buffer_flush (jbuf);
-  g_queue_free (jbuf->packets);
-
-  G_OBJECT_CLASS (rtp_jitter_buffer_parent_class)->finalize (object);
-}
-
-/**
- * rtp_jitter_buffer_new:
- *
- * Create an #RTPJitterBuffer.
- *
- * Returns: a new #RTPJitterBuffer. Use g_object_unref() after usage.
- */
-RTPJitterBuffer *
-rtp_jitter_buffer_new (void)
-{
-  RTPJitterBuffer *jbuf;
-
-  jbuf = g_object_new (RTP_TYPE_JITTER_BUFFER, NULL);
-
-  return jbuf;
-}
-
-void
-rtp_jitter_buffer_reset_skew (RTPJitterBuffer * jbuf)
-{
-  jbuf->base_time = -1;
-  jbuf->base_rtptime = -1;
-  jbuf->base_extrtp = -1;
-  jbuf->clock_rate = -1;
-  jbuf->ext_rtptime = -1;
-  jbuf->last_rtptime = -1;
-  jbuf->window_pos = 0;
-  jbuf->window_filling = TRUE;
-  jbuf->window_min = 0;
-  jbuf->skew = 0;
-  jbuf->prev_send_diff = -1;
-  jbuf->prev_out_time = -1;
-  GST_DEBUG ("reset skew correction");
-}
-
-/* For the clock skew we use a windowed low point averaging algorithm as can be
- * found in http://www.grame.fr/pub/TR-050601.pdf. The idea is that the jitter is
- * composed of:
- *
- *  J = N + n
- *
- *   N   : a constant network delay.
- *   n   : random added noise. The noise is concentrated around 0
- *
- * In the receiver we can track the elapsed time at the sender with:
- *
- *  send_diff(i) = (Tsi - Ts0);
- *
- *   Tsi : The time at the sender at packet i
- *   Ts0 : The time at the sender at the first packet
- *
- * This is the difference between the RTP timestamp in the first received packet
- * and the current packet.
- *
- * At the receiver we have to deal with the jitter introduced by the network.
- *
- *  recv_diff(i) = (Tri - Tr0)
- *
- *   Tri : The time at the receiver at packet i
- *   Tr0 : The time at the receiver at the first packet
- *
- * Both of these values contain a jitter Ji, a jitter for packet i, so we can
- * write:
- *
- *  recv_diff(i) = (Cri + D + ni) - (Cr0 + D + n0))
- *
- *    Cri    : The time of the clock at the receiver for packet i
- *    D + ni : The jitter when receiving packet i
- *
- * We see that the network delay is irrelevant here as we can elliminate D:
- *
- *  recv_diff(i) = (Cri + ni) - (Cr0 + n0))
- *
- * The drift is now expressed as:
- *
- *  Drift(i) = recv_diff(i) - send_diff(i);
- *
- * We now keep the W latest values of Drift and find the minimum (this is the
- * one with the lowest network jitter and thus the one which is least affected
- * by it). We average this lowest value to smooth out the resulting network skew.
- *
- * Both the window and the weighting used for averaging influence the accuracy
- * of the drift estimation. Finding the correct parameters turns out to be a
- * compromise between accuracy and inertia. 
- *
- * We use a 2 second window or up to 512 data points, which is statistically big
- * enough to catch spikes (FIXME, detect spikes).
- * We also use a rather large weighting factor (125) to smoothly adapt. During
- * startup, when filling the window, we use a parabolic weighting factor, the
- * more the window is filled, the faster we move to the detected possible skew.
- *
- * Returns: @time adjusted with the clock skew.
- */
-static GstClockTime
-calculate_skew (RTPJitterBuffer * jbuf, guint32 rtptime, GstClockTime time,
-    guint32 clock_rate)
-{
-  guint64 ext_rtptime;
-  guint64 send_diff, recv_diff;
-  gint64 delta;
-  gint64 old;
-  gint pos, i;
-  GstClockTime gstrtptime, out_time;
-
-  ext_rtptime = gst_rtp_buffer_ext_timestamp (&jbuf->ext_rtptime, rtptime);
-
-  gstrtptime = gst_util_uint64_scale_int (ext_rtptime, GST_SECOND, clock_rate);
-
-  /* keep track of the last extended rtptime */
-  jbuf->last_rtptime = ext_rtptime;
-
-  if (jbuf->clock_rate != clock_rate) {
-    GST_WARNING ("Clock rate changed from %" G_GUINT32_FORMAT " to %"
-        G_GUINT32_FORMAT, jbuf->clock_rate, clock_rate);
-    jbuf->base_time = -1;
-    jbuf->base_rtptime = -1;
-    jbuf->clock_rate = clock_rate;
-    jbuf->prev_out_time = -1;
-    jbuf->prev_send_diff = -1;
-  }
-
-  /* first time, lock on to time and gstrtptime */
-  if (G_UNLIKELY (jbuf->base_time == -1)) {
-    jbuf->base_time = time;
-    jbuf->prev_out_time = -1;
-    GST_DEBUG ("Taking new base time %" GST_TIME_FORMAT, GST_TIME_ARGS (time));
-  }
-  if (G_UNLIKELY (jbuf->base_rtptime == -1)) {
-    jbuf->base_rtptime = gstrtptime;
-    jbuf->base_extrtp = ext_rtptime;
-    jbuf->prev_send_diff = -1;
-    GST_DEBUG ("Taking new base rtptime %" GST_TIME_FORMAT,
-        GST_TIME_ARGS (gstrtptime));
-  }
-
-  if (G_LIKELY (gstrtptime >= jbuf->base_rtptime))
-    send_diff = gstrtptime - jbuf->base_rtptime;
-  else {
-    /* elapsed time at sender, timestamps can go backwards and thus be smaller
-     * than our base time, take a new base time in that case. */
-    GST_WARNING ("backward timestamps at server, taking new base time");
-    jbuf->base_time = time;
-    jbuf->base_rtptime = gstrtptime;
-    jbuf->base_extrtp = ext_rtptime;
-    jbuf->prev_out_time = -1;
-    jbuf->prev_send_diff = -1;
-    send_diff = 0;
-  }
-
-  GST_DEBUG ("extrtp %" G_GUINT64_FORMAT ", gstrtp %" GST_TIME_FORMAT ", base %"
-      GST_TIME_FORMAT ", send_diff %" GST_TIME_FORMAT, ext_rtptime,
-      GST_TIME_ARGS (gstrtptime), GST_TIME_ARGS (jbuf->base_rtptime),
-      GST_TIME_ARGS (send_diff));
-
-  /* we don't have an arrival timestamp so we can't do skew detection. we
-   * should still apply a timestamp based on RTP timestamp and base_time */
-  if (time == -1 || jbuf->base_time == -1)
-    goto no_skew;
-
-  /* elapsed time at receiver, includes the jitter */
-  recv_diff = time - jbuf->base_time;
-
-  GST_DEBUG ("time %" GST_TIME_FORMAT ", base %" GST_TIME_FORMAT ", recv_diff %"
-      GST_TIME_FORMAT, GST_TIME_ARGS (time), GST_TIME_ARGS (jbuf->base_time),
-      GST_TIME_ARGS (recv_diff));
-
-  /* measure the diff */
-  delta = ((gint64) recv_diff) - ((gint64) send_diff);
-
-  /* if the difference between the sender timeline and the receiver timeline
-   * changed too quickly we have to resync because the server likely restarted
-   * its timestamps. */
-  if (ABS (delta - jbuf->skew) > GST_SECOND) {
-    GST_WARNING ("delta %" GST_TIME_FORMAT " too big, reset skew",
-        GST_TIME_ARGS (delta - jbuf->skew));
-    jbuf->base_time = time;
-    jbuf->base_rtptime = gstrtptime;
-    jbuf->base_extrtp = ext_rtptime;
-    jbuf->prev_out_time = -1;
-    jbuf->prev_send_diff = -1;
-    send_diff = 0;
-    delta = 0;
-  }
-
-  pos = jbuf->window_pos;
-
-  if (G_UNLIKELY (jbuf->window_filling)) {
-    /* we are filling the window */
-    GST_DEBUG ("filling %d, delta %" G_GINT64_FORMAT, pos, delta);
-    jbuf->window[pos++] = delta;
-    /* calc the min delta we observed */
-    if (G_UNLIKELY (pos == 1 || delta < jbuf->window_min))
-      jbuf->window_min = delta;
-
-    if (G_UNLIKELY (send_diff >= MAX_TIME || pos >= MAX_WINDOW)) {
-      jbuf->window_size = pos;
-
-      /* window filled */
-      GST_DEBUG ("min %" G_GINT64_FORMAT, jbuf->window_min);
-
-      /* the skew is now the min */
-      jbuf->skew = jbuf->window_min;
-      jbuf->window_filling = FALSE;
-    } else {
-      gint perc_time, perc_window, perc;
-
-      /* figure out how much we filled the window, this depends on the amount of
-       * time we have or the max number of points we keep. */
-      perc_time = send_diff * 100 / MAX_TIME;
-      perc_window = pos * 100 / MAX_WINDOW;
-      perc = MAX (perc_time, perc_window);
-
-      /* make a parabolic function, the closer we get to the MAX, the more value
-       * we give to the scaling factor of the new value */
-      perc = perc * perc;
-
-      /* quickly go to the min value when we are filling up, slowly when we are
-       * just starting because we're not sure it's a good value yet. */
-      jbuf->skew =
-          (perc * jbuf->window_min + ((10000 - perc) * jbuf->skew)) / 10000;
-      jbuf->window_size = pos + 1;
-    }
-  } else {
-    /* pick old value and store new value. We keep the previous value in order
-     * to quickly check if the min of the window changed */
-    old = jbuf->window[pos];
-    jbuf->window[pos++] = delta;
-
-    if (G_UNLIKELY (delta <= jbuf->window_min)) {
-      /* if the new value we inserted is smaller or equal to the current min,
-       * it becomes the new min */
-      jbuf->window_min = delta;
-    } else if (G_UNLIKELY (old == jbuf->window_min)) {
-      gint64 min = G_MAXINT64;
-
-      /* if we removed the old min, we have to find a new min */
-      for (i = 0; i < jbuf->window_size; i++) {
-        /* we found another value equal to the old min, we can stop searching now */
-        if (jbuf->window[i] == old) {
-          min = old;
-          break;
-        }
-        if (jbuf->window[i] < min)
-          min = jbuf->window[i];
-      }
-      jbuf->window_min = min;
-    }
-    /* average the min values */
-    jbuf->skew = (jbuf->window_min + (124 * jbuf->skew)) / 125;
-    GST_DEBUG ("delta %" G_GINT64_FORMAT ", new min: %" G_GINT64_FORMAT,
-        delta, jbuf->window_min);
-  }
-  /* wrap around in the window */
-  if (G_UNLIKELY (pos >= jbuf->window_size))
-    pos = 0;
-  jbuf->window_pos = pos;
-
-no_skew:
-  /* the output time is defined as the base timestamp plus the RTP time
-   * adjusted for the clock skew .*/
-  if (jbuf->base_time != -1) {
-    out_time = jbuf->base_time + send_diff + jbuf->skew;
-    /* check if timestamps are not going backwards, we can only check this if we
-     * have a previous out time and a previous send_diff */
-    if (G_LIKELY (jbuf->prev_out_time != -1 && jbuf->prev_send_diff != -1)) {
-      /* now check for backwards timestamps */
-      if (G_UNLIKELY (
-              /* if the server timestamps went up and the out_time backwards */
-              (send_diff > jbuf->prev_send_diff
-                  && out_time < jbuf->prev_out_time) ||
-              /* if the server timestamps went backwards and the out_time forwards */
-              (send_diff < jbuf->prev_send_diff
-                  && out_time > jbuf->prev_out_time) ||
-              /* if the server timestamps did not change */
-              send_diff == jbuf->prev_send_diff)) {
-        GST_DEBUG ("backwards timestamps, using previous time");
-        out_time = jbuf->prev_out_time;
-      }
-    }
-  } else
-    out_time = -1;
-
-  jbuf->prev_out_time = out_time;
-  jbuf->prev_send_diff = send_diff;
-
-  GST_DEBUG ("skew %" G_GINT64_FORMAT ", out %" GST_TIME_FORMAT,
-      jbuf->skew, GST_TIME_ARGS (out_time));
-
-  return out_time;
-}
-
-/**
- * rtp_jitter_buffer_insert:
- * @jbuf: an #RTPJitterBuffer
- * @buf: a buffer
- * @time: a running_time when this buffer was received in nanoseconds
- * @clock_rate: the clock-rate of the payload of @buf
- * @tail: TRUE when the tail element changed.
- *
- * Inserts @buf into the packet queue of @jbuf. The sequence number of the
- * packet will be used to sort the packets. This function takes ownerhip of
- * @buf when the function returns %TRUE.
- * @buf should have writable metadata when calling this function.
- *
- * Returns: %FALSE if a packet with the same number already existed.
- */
-gboolean
-rtp_jitter_buffer_insert (RTPJitterBuffer * jbuf, GstBuffer * buf,
-    GstClockTime time, guint32 clock_rate, gboolean * tail)
-{
-  GList *list;
-  guint32 rtptime;
-  guint16 seqnum;
-
-  g_return_val_if_fail (jbuf != NULL, FALSE);
-  g_return_val_if_fail (buf != NULL, FALSE);
-
-  seqnum = gst_rtp_buffer_get_seq (buf);
-
-  /* loop the list to skip strictly smaller seqnum buffers */
-  for (list = jbuf->packets->head; list; list = g_list_next (list)) {
-    guint16 qseq;
-    gint gap;
-
-    qseq = gst_rtp_buffer_get_seq (GST_BUFFER_CAST (list->data));
-
-    /* compare the new seqnum to the one in the buffer */
-    gap = gst_rtp_buffer_compare_seqnum (seqnum, qseq);
-
-    /* we hit a packet with the same seqnum, notify a duplicate */
-    if (G_UNLIKELY (gap == 0))
-      goto duplicate;
-
-    /* seqnum > qseq, we can stop looking */
-    if (G_LIKELY (gap < 0))
-      break;
-  }
-
-  /* do skew calculation by measuring the difference between rtptime and the
-   * receive time, this function will retimestamp @buf with the skew corrected
-   * running time. */
-  rtptime = gst_rtp_buffer_get_timestamp (buf);
-  time = calculate_skew (jbuf, rtptime, time, clock_rate);
-  GST_BUFFER_TIMESTAMP (buf) = time;
-
-  /* It's more likely that the packet was inserted in the front of the buffer */
-  if (G_LIKELY (list))
-    g_queue_insert_before (jbuf->packets, list, buf);
-  else
-    g_queue_push_tail (jbuf->packets, buf);
-
-  /* tail was changed when we did not find a previous packet, we set the return
-   * flag when requested. */
-  if (G_LIKELY (tail))
-    *tail = (list == NULL);
-
-  return TRUE;
-
-  /* ERRORS */
-duplicate:
-  {
-    GST_WARNING ("duplicate packet %d found", (gint) seqnum);
-    return FALSE;
-  }
-}
-
-/**
- * rtp_jitter_buffer_pop:
- * @jbuf: an #RTPJitterBuffer
- *
- * Pops the oldest buffer from the packet queue of @jbuf. The popped buffer will
- * have its timestamp adjusted with the incomming running_time and the detected
- * clock skew.
- *
- * Returns: a #GstBuffer or %NULL when there was no packet in the queue.
- */
-GstBuffer *
-rtp_jitter_buffer_pop (RTPJitterBuffer * jbuf)
-{
-  GstBuffer *buf;
-
-  g_return_val_if_fail (jbuf != NULL, FALSE);
-
-  buf = g_queue_pop_tail (jbuf->packets);
-
-  return buf;
-}
-
-/**
- * rtp_jitter_buffer_peek:
- * @jbuf: an #RTPJitterBuffer
- *
- * Peek the oldest buffer from the packet queue of @jbuf. Register a callback
- * with rtp_jitter_buffer_set_tail_changed() to be notified when an older packet
- * was inserted in the queue.
- *
- * Returns: a #GstBuffer or %NULL when there was no packet in the queue.
- */
-GstBuffer *
-rtp_jitter_buffer_peek (RTPJitterBuffer * jbuf)
-{
-  GstBuffer *buf;
-
-  g_return_val_if_fail (jbuf != NULL, FALSE);
-
-  buf = g_queue_peek_tail (jbuf->packets);
-
-  return buf;
-}
-
-/**
- * rtp_jitter_buffer_flush:
- * @jbuf: an #RTPJitterBuffer
- *
- * Flush all packets from the jitterbuffer.
- */
-void
-rtp_jitter_buffer_flush (RTPJitterBuffer * jbuf)
-{
-  GstBuffer *buffer;
-
-  g_return_if_fail (jbuf != NULL);
-
-  while ((buffer = g_queue_pop_head (jbuf->packets)))
-    gst_buffer_unref (buffer);
-}
-
-/**
- * rtp_jitter_buffer_num_packets:
- * @jbuf: an #RTPJitterBuffer
- *
- * Get the number of packets currently in "jbuf.
- *
- * Returns: The number of packets in @jbuf.
- */
-guint
-rtp_jitter_buffer_num_packets (RTPJitterBuffer * jbuf)
-{
-  g_return_val_if_fail (jbuf != NULL, 0);
-
-  return jbuf->packets->length;
-}
-
-/**
- * rtp_jitter_buffer_get_ts_diff:
- * @jbuf: an #RTPJitterBuffer
- *
- * Get the difference between the timestamps of first and last packet in the
- * jitterbuffer.
- *
- * Returns: The difference expressed in the timestamp units of the packets.
- */
-guint32
-rtp_jitter_buffer_get_ts_diff (RTPJitterBuffer * jbuf)
-{
-  guint64 high_ts, low_ts;
-  GstBuffer *high_buf, *low_buf;
-  guint32 result;
-
-  g_return_val_if_fail (jbuf != NULL, 0);
-
-  high_buf = g_queue_peek_head (jbuf->packets);
-  low_buf = g_queue_peek_tail (jbuf->packets);
-
-  if (!high_buf || !low_buf || high_buf == low_buf)
-    return 0;
-
-  high_ts = gst_rtp_buffer_get_timestamp (high_buf);
-  low_ts = gst_rtp_buffer_get_timestamp (low_buf);
-
-  /* it needs to work if ts wraps */
-  if (high_ts >= low_ts) {
-    result = (guint32) (high_ts - low_ts);
-  } else {
-    result = (guint32) (high_ts + G_MAXUINT32 + 1 - low_ts);
-  }
-  return result;
-}
-
-/**
- * rtp_jitter_buffer_get_sync:
- * @jbuf: an #RTPJitterBuffer
- * @rtptime: result RTP time
- * @timestamp: result GStreamer timestamp
- * @clock_rate: clock-rate of @rtptime
- * @last_rtptime: last seen rtptime.
- *
- * Calculates the relation between the RTP timestamp and the GStreamer timestamp
- * used for constructing timestamps.
- *
- * For extended RTP timestamp @rtptime with a clock-rate of @clock_rate,
- * the GStreamer timestamp is currently @timestamp.
- *
- * The last seen extended RTP timestamp with clock-rate @clock-rate is returned in
- * @last_rtptime.
- */
-void
-rtp_jitter_buffer_get_sync (RTPJitterBuffer * jbuf, guint64 * rtptime,
-    guint64 * timestamp, guint32 * clock_rate, guint64 * last_rtptime)
-{
-  if (rtptime)
-    *rtptime = jbuf->base_extrtp;
-  if (timestamp)
-    *timestamp = jbuf->base_time + jbuf->skew;
-  if (clock_rate)
-    *clock_rate = jbuf->clock_rate;
-  if (last_rtptime)
-    *last_rtptime = jbuf->last_rtptime;
-}
diff --git a/gst/rtpmanager/rtpjitterbuffer.h b/gst/rtpmanager/rtpjitterbuffer.h
deleted file mode 100644
index ff1a16b0..00000000
--- a/gst/rtpmanager/rtpjitterbuffer.h
+++ /dev/null
@@ -1,101 +0,0 @@
-/* GStreamer
- * Copyright (C) <2007> Wim Taymans <wim.taymans@gmail.com>
- *
- * This library is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Library General Public
- * License as published by the Free Software Foundation; either
- * version 2 of the License, or (at your option) any later version.
- *
- * This library is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Library General Public License for more details.
- *
- * You should have received a copy of the GNU Library General Public
- * License along with this library; if not, write to the
- * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
- * Boston, MA 02111-1307, USA.
- */
-
-#ifndef __RTP_JITTER_BUFFER_H__
-#define __RTP_JITTER_BUFFER_H__
-
-#include <gst/gst.h>
-#include <gst/rtp/gstrtcpbuffer.h>
-
-typedef struct _RTPJitterBuffer RTPJitterBuffer;
-typedef struct _RTPJitterBufferClass RTPJitterBufferClass;
-
-#define RTP_TYPE_JITTER_BUFFER             (rtp_jitter_buffer_get_type())
-#define RTP_JITTER_BUFFER(src)             (G_TYPE_CHECK_INSTANCE_CAST((src),RTP_TYPE_JITTER_BUFFER,RTPJitterBuffer))
-#define RTP_JITTER_BUFFER_CLASS(klass)     (G_TYPE_CHECK_CLASS_CAST((klass),RTP_TYPE_JITTER_BUFFER,RTPJitterBufferClass))
-#define RTP_IS_JITTER_BUFFER(src)          (G_TYPE_CHECK_INSTANCE_TYPE((src),RTP_TYPE_JITTER_BUFFER))
-#define RTP_IS_JITTER_BUFFER_CLASS(klass)  (G_TYPE_CHECK_CLASS_TYPE((klass),RTP_TYPE_JITTER_BUFFER))
-#define RTP_JITTER_BUFFER_CAST(src)        ((RTPJitterBuffer *)(src))
-
-/**
- * RTPTailChanged:
- * @jbuf: an #RTPJitterBuffer
- * @user_data: user data specified when registering
- *
- * This callback will be called when the tail buffer of @jbuf changed.
- */
-typedef void (*RTPTailChanged) (RTPJitterBuffer *jbuf, gpointer user_data);
-
-#define RTP_JITTER_BUFFER_MAX_WINDOW 512
-/**
- * RTPJitterBuffer:
- *
- * A JitterBuffer in the #RTPSession
- */
-struct _RTPJitterBuffer {
-  GObject        object;
-
-  GQueue        *packets;
-
-  /* for calculating skew */
-  GstClockTime   base_time;
-  GstClockTime   base_rtptime;
-  guint32        clock_rate;
-  GstClockTime   base_extrtp;
-  GstClockTime   prev_out_time;
-  guint64        ext_rtptime;
-  guint64        last_rtptime;
-  gint64         window[RTP_JITTER_BUFFER_MAX_WINDOW];
-  guint          window_pos;
-  guint          window_size;
-  gboolean       window_filling;
-  gint64         window_min;
-  gint64         skew;
-  gint64         prev_send_diff;
-};
-
-struct _RTPJitterBufferClass {
-  GObjectClass   parent_class;
-};
-
-GType rtp_jitter_buffer_get_type (void);
-
-/* managing lifetime */
-RTPJitterBuffer*      rtp_jitter_buffer_new              (void);
-
-void                  rtp_jitter_buffer_reset_skew       (RTPJitterBuffer *jbuf);
-
-gboolean              rtp_jitter_buffer_insert           (RTPJitterBuffer *jbuf, GstBuffer *buf,
-		                                          GstClockTime time,
-		                                          guint32 clock_rate,
-		                                          gboolean *tail);
-GstBuffer *           rtp_jitter_buffer_peek             (RTPJitterBuffer *jbuf);
-GstBuffer *           rtp_jitter_buffer_pop              (RTPJitterBuffer *jbuf);
-
-void                  rtp_jitter_buffer_flush            (RTPJitterBuffer *jbuf);
-
-guint                 rtp_jitter_buffer_num_packets      (RTPJitterBuffer *jbuf);
-guint32               rtp_jitter_buffer_get_ts_diff      (RTPJitterBuffer *jbuf);
-
-void                  rtp_jitter_buffer_get_sync         (RTPJitterBuffer *jbuf, guint64 *rtptime,
-                                                          guint64 *timestamp, guint32 *clock_rate,
-							  guint64 *last_rtptime);
-
-
-#endif /* __RTP_JITTER_BUFFER_H__ */
diff --git a/gst/rtpmanager/rtpsession.c b/gst/rtpmanager/rtpsession.c
deleted file mode 100644
index 3e17ec12..00000000
--- a/gst/rtpmanager/rtpsession.c
+++ /dev/null
@@ -1,2521 +0,0 @@
-/* GStreamer
- * Copyright (C) <2007> Wim Taymans <wim.taymans@gmail.com>
- *
- * This library is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Library General Public
- * License as published by the Free Software Foundation; either
- * version 2 of the License, or (at your option) any later version.
- *
- * This library is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Library General Public License for more details.
- *
- * You should have received a copy of the GNU Library General Public
- * License along with this library; if not, write to the
- * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
- * Boston, MA 02111-1307, USA.
- */
-
-#include <string.h>
-
-#include <gst/rtp/gstrtpbuffer.h>
-#include <gst/rtp/gstrtcpbuffer.h>
-#include <gst/netbuffer/gstnetbuffer.h>
-
-#include "gstrtpbin-marshal.h"
-#include "rtpsession.h"
-
-GST_DEBUG_CATEGORY_STATIC (rtp_session_debug);
-#define GST_CAT_DEFAULT rtp_session_debug
-
-/* signals and args */
-enum
-{
-  SIGNAL_GET_SOURCE_BY_SSRC,
-  SIGNAL_ON_NEW_SSRC,
-  SIGNAL_ON_SSRC_COLLISION,
-  SIGNAL_ON_SSRC_VALIDATED,
-  SIGNAL_ON_SSRC_ACTIVE,
-  SIGNAL_ON_SSRC_SDES,
-  SIGNAL_ON_BYE_SSRC,
-  SIGNAL_ON_BYE_TIMEOUT,
-  SIGNAL_ON_TIMEOUT,
-  SIGNAL_ON_SENDER_TIMEOUT,
-  LAST_SIGNAL
-};
-
-#define DEFAULT_INTERNAL_SOURCE      NULL
-#define DEFAULT_BANDWIDTH            RTP_STATS_BANDWIDTH
-#define DEFAULT_RTCP_FRACTION        RTP_STATS_RTCP_BANDWIDTH
-#define DEFAULT_RTCP_MTU             1400
-#define DEFAULT_SDES                 NULL
-#define DEFAULT_NUM_SOURCES          0
-#define DEFAULT_NUM_ACTIVE_SOURCES   0
-#define DEFAULT_SOURCES              NULL
-
-enum
-{
-  PROP_0,
-  PROP_INTERNAL_SSRC,
-  PROP_INTERNAL_SOURCE,
-  PROP_BANDWIDTH,
-  PROP_RTCP_FRACTION,
-  PROP_RTCP_MTU,
-  PROP_SDES,
-  PROP_NUM_SOURCES,
-  PROP_NUM_ACTIVE_SOURCES,
-  PROP_SOURCES,
-  PROP_LAST
-};
-
-/* update average packet size, we keep this scaled by 16 to keep enough
- * precision. */
-#define UPDATE_AVG(avg, val)    	\
-  if ((avg) == 0)			\
-   (avg) = (val) << 4;          	\
-  else 					\
-   (avg) = ((val) + (15 * (avg))) >> 4;
-
-/* The number RTCP intervals after which to timeout entries in the
- * collision table
- */
-#define RTCP_INTERVAL_COLLISION_TIMEOUT 10
-
-/* GObject vmethods */
-static void rtp_session_finalize (GObject * object);
-static void rtp_session_set_property (GObject * object, guint prop_id,
-    const GValue * value, GParamSpec * pspec);
-static void rtp_session_get_property (GObject * object, guint prop_id,
-    GValue * value, GParamSpec * pspec);
-
-static guint rtp_session_signals[LAST_SIGNAL] = { 0 };
-
-G_DEFINE_TYPE (RTPSession, rtp_session, G_TYPE_OBJECT);
-
-static RTPSource *obtain_source (RTPSession * sess, guint32 ssrc,
-    gboolean * created, RTPArrivalStats * arrival, gboolean rtp);
-static GstFlowReturn rtp_session_schedule_bye_locked (RTPSession * sess,
-    const gchar * reason, GstClockTime current_time);
-static GstClockTime calculate_rtcp_interval (RTPSession * sess,
-    gboolean deterministic, gboolean first);
-
-static void
-rtp_session_class_init (RTPSessionClass * klass)
-{
-  GObjectClass *gobject_class;
-
-  gobject_class = (GObjectClass *) klass;
-
-  gobject_class->finalize = rtp_session_finalize;
-  gobject_class->set_property = rtp_session_set_property;
-  gobject_class->get_property = rtp_session_get_property;
-
-  /**
-   * RTPSession::get-source-by-ssrc:
-   * @session: the object which received the signal
-   * @ssrc: the SSRC of the RTPSource
-   *
-   * Request the #RTPSource object with SSRC @ssrc in @session.
-   */
-  rtp_session_signals[SIGNAL_GET_SOURCE_BY_SSRC] =
-      g_signal_new ("get-source-by-ssrc", G_TYPE_FROM_CLASS (klass),
-      G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION, G_STRUCT_OFFSET (RTPSessionClass,
-          get_source_by_ssrc), NULL, NULL, gst_rtp_bin_marshal_OBJECT__UINT,
-      RTP_TYPE_SOURCE, 1, G_TYPE_UINT);
-
-  /**
-   * RTPSession::on-new-ssrc:
-   * @session: the object which received the signal
-   * @src: the new RTPSource
-   *
-   * Notify of a new SSRC that entered @session.
-   */
-  rtp_session_signals[SIGNAL_ON_NEW_SSRC] =
-      g_signal_new ("on-new-ssrc", G_TYPE_FROM_CLASS (klass),
-      G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_new_ssrc),
-      NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
-      RTP_TYPE_SOURCE);
-  /**
-   * RTPSession::on-ssrc-collision:
-   * @session: the object which received the signal
-   * @src: the #RTPSource that caused a collision
-   *
-   * Notify when we have an SSRC collision
-   */
-  rtp_session_signals[SIGNAL_ON_SSRC_COLLISION] =
-      g_signal_new ("on-ssrc-collision", G_TYPE_FROM_CLASS (klass),
-      G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_ssrc_collision),
-      NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
-      RTP_TYPE_SOURCE);
-  /**
-   * RTPSession::on-ssrc-validated:
-   * @session: the object which received the signal
-   * @src: the new validated RTPSource
-   *
-   * Notify of a new SSRC that became validated.
-   */
-  rtp_session_signals[SIGNAL_ON_SSRC_VALIDATED] =
-      g_signal_new ("on-ssrc-validated", G_TYPE_FROM_CLASS (klass),
-      G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_ssrc_validated),
-      NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
-      RTP_TYPE_SOURCE);
-  /**
-   * RTPSession::on-ssrc-active:
-   * @session: the object which received the signal
-   * @src: the active RTPSource
-   *
-   * Notify of a SSRC that is active, i.e., sending RTCP.
-   */
-  rtp_session_signals[SIGNAL_ON_SSRC_ACTIVE] =
-      g_signal_new ("on-ssrc-active", G_TYPE_FROM_CLASS (klass),
-      G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_ssrc_active),
-      NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
-      RTP_TYPE_SOURCE);
-  /**
-   * RTPSession::on-ssrc-sdes:
-   * @session: the object which received the signal
-   * @src: the RTPSource
-   *
-   * Notify that a new SDES was received for SSRC.
-   */
-  rtp_session_signals[SIGNAL_ON_SSRC_SDES] =
-      g_signal_new ("on-ssrc-sdes", G_TYPE_FROM_CLASS (klass),
-      G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_ssrc_sdes),
-      NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
-      RTP_TYPE_SOURCE);
-  /**
-   * RTPSession::on-bye-ssrc:
-   * @session: the object which received the signal
-   * @src: the RTPSource that went away
-   *
-   * Notify of an SSRC that became inactive because of a BYE packet.
-   */
-  rtp_session_signals[SIGNAL_ON_BYE_SSRC] =
-      g_signal_new ("on-bye-ssrc", G_TYPE_FROM_CLASS (klass),
-      G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_bye_ssrc),
-      NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
-      RTP_TYPE_SOURCE);
-  /**
-   * RTPSession::on-bye-timeout:
-   * @session: the object which received the signal
-   * @src: the RTPSource that timed out
-   *
-   * Notify of an SSRC that has timed out because of BYE
-   */
-  rtp_session_signals[SIGNAL_ON_BYE_TIMEOUT] =
-      g_signal_new ("on-bye-timeout", G_TYPE_FROM_CLASS (klass),
-      G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_bye_timeout),
-      NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
-      RTP_TYPE_SOURCE);
-  /**
-   * RTPSession::on-timeout:
-   * @session: the object which received the signal
-   * @src: the RTPSource that timed out
-   *
-   * Notify of an SSRC that has timed out
-   */
-  rtp_session_signals[SIGNAL_ON_TIMEOUT] =
-      g_signal_new ("on-timeout", G_TYPE_FROM_CLASS (klass),
-      G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_timeout),
-      NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
-      RTP_TYPE_SOURCE);
-  /**
-   * RTPSession::on-sender-timeout:
-   * @session: the object which received the signal
-   * @src: the RTPSource that timed out
-   *
-   * Notify of an SSRC that was a sender but timed out and became a receiver.
-   */
-  rtp_session_signals[SIGNAL_ON_SENDER_TIMEOUT] =
-      g_signal_new ("on-sender-timeout", G_TYPE_FROM_CLASS (klass),
-      G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_sender_timeout),
-      NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
-      RTP_TYPE_SOURCE);
-
-  g_object_class_install_property (gobject_class, PROP_INTERNAL_SSRC,
-      g_param_spec_uint ("internal-ssrc", "Internal SSRC",
-          "The internal SSRC used for the session",
-          0, G_MAXUINT, 0, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
-
-  g_object_class_install_property (gobject_class, PROP_INTERNAL_SOURCE,
-      g_param_spec_object ("internal-source", "Internal Source",
-          "The internal source element of the session",
-          RTP_TYPE_SOURCE, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
-
-  g_object_class_install_property (gobject_class, PROP_BANDWIDTH,
-      g_param_spec_double ("bandwidth", "Bandwidth",
-          "The bandwidth of the session",
-          0.0, G_MAXDOUBLE, DEFAULT_BANDWIDTH,
-          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
-
-  g_object_class_install_property (gobject_class, PROP_RTCP_FRACTION,
-      g_param_spec_double ("rtcp-fraction", "RTCP Fraction",
-          "The fraction of the bandwidth used for RTCP",
-          0.0, G_MAXDOUBLE, DEFAULT_RTCP_FRACTION,
-          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
-
-  g_object_class_install_property (gobject_class, PROP_RTCP_MTU,
-      g_param_spec_uint ("rtcp-mtu", "RTCP MTU",
-          "The maximum size of the RTCP packets",
-          16, G_MAXINT16, DEFAULT_RTCP_MTU,
-          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
-
-  g_object_class_install_property (gobject_class, PROP_SDES,
-      g_param_spec_boxed ("sdes", "SDES",
-          "The SDES items of this session",
-          GST_TYPE_STRUCTURE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
-
-  g_object_class_install_property (gobject_class, PROP_NUM_SOURCES,
-      g_param_spec_uint ("num-sources", "Num Sources",
-          "The number of sources in the session", 0, G_MAXUINT,
-          DEFAULT_NUM_SOURCES, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
-
-  g_object_class_install_property (gobject_class, PROP_NUM_ACTIVE_SOURCES,
-      g_param_spec_uint ("num-active-sources", "Num Active Sources",
-          "The number of active sources in the session", 0, G_MAXUINT,
-          DEFAULT_NUM_ACTIVE_SOURCES,
-          G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
-  /**
-   * RTPSource::sources
-   *
-   * Get a GValue Array of all sources in the session.
-   *
-   * <example>
-   * <title>Getting the #RTPSources of a session
-   * <programlisting>
-   * {
-   *   GValueArray *arr;
-   *   GValue *val;
-   *   guint i;
-   *
-   *   g_object_get (sess, "sources", &arr, NULL);
-   *
-   *   for (i = 0; i < arr->n_values; i++) {
-   *     RTPSource *source;
-   *
-   *     val = g_value_array_get_nth (arr, i);
-   *     source = g_value_get_object (val);
-   *   }
-   *   g_value_array_free (arr);
-   * }
-   * </programlisting>
-   * </example>
-   */
-  g_object_class_install_property (gobject_class, PROP_SOURCES,
-      g_param_spec_boxed ("sources", "Sources",
-          "An array of all known sources in the session",
-          G_TYPE_VALUE_ARRAY, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
-
-  klass->get_source_by_ssrc =
-      GST_DEBUG_FUNCPTR (rtp_session_get_source_by_ssrc);
-
-  GST_DEBUG_CATEGORY_INIT (rtp_session_debug, "rtpsession", 0, "RTP Session");
-}
-
-static void
-rtp_session_init (RTPSession * sess)
-{
-  gint i;
-  gchar *str;
-
-  sess->lock = g_mutex_new ();
-  sess->key = g_random_int ();
-  sess->mask_idx = 0;
-  sess->mask = 0;
-
-  for (i = 0; i < 32; i++) {
-    sess->ssrcs[i] =
-        g_hash_table_new_full (NULL, NULL, NULL,
-        (GDestroyNotify) g_object_unref);
-  }
-  sess->cnames = g_hash_table_new_full (NULL, NULL, g_free, NULL);
-
-  rtp_stats_init_defaults (&sess->stats);
-
-  /* create an active SSRC for this session manager */
-  sess->source = rtp_session_create_source (sess);
-  sess->source->validated = TRUE;
-  sess->source->internal = TRUE;
-  sess->stats.active_sources++;
-
-  /* default UDP header length */
-  sess->header_len = 28;
-  sess->mtu = DEFAULT_RTCP_MTU;
-
-  /* some default SDES entries */
-  str = g_strdup_printf ("%s@%s", g_get_user_name (), g_get_host_name ());
-  rtp_source_set_sdes_string (sess->source, GST_RTCP_SDES_CNAME, str);
-  g_free (str);
-
-  rtp_source_set_sdes_string (sess->source, GST_RTCP_SDES_NAME,
-      g_get_real_name ());
-  rtp_source_set_sdes_string (sess->source, GST_RTCP_SDES_TOOL, "GStreamer");
-
-  sess->first_rtcp = TRUE;
-
-  GST_DEBUG ("%p: session using SSRC: %08x", sess, sess->source->ssrc);
-}
-
-static void
-rtp_session_finalize (GObject * object)
-{
-  RTPSession *sess;
-  gint i;
-
-  sess = RTP_SESSION_CAST (object);
-
-  g_mutex_free (sess->lock);
-  for (i = 0; i < 32; i++)
-    g_hash_table_destroy (sess->ssrcs[i]);
-
-  g_list_foreach (sess->conflicting_addresses, (GFunc) g_free, NULL);
-  g_list_free (sess->conflicting_addresses);
-
-  g_free (sess->bye_reason);
-
-  g_hash_table_destroy (sess->cnames);
-  g_object_unref (sess->source);
-
-  G_OBJECT_CLASS (rtp_session_parent_class)->finalize (object);
-}
-
-static void
-copy_source (gpointer key, RTPSource * source, GValueArray * arr)
-{
-  GValue value = { 0 };
-
-  g_value_init (&value, RTP_TYPE_SOURCE);
-  g_value_take_object (&value, source);
-  /* copies the value */
-  g_value_array_append (arr, &value);
-}
-
-static GValueArray *
-rtp_session_create_sources (RTPSession * sess)
-{
-  GValueArray *res;
-  guint size;
-
-  RTP_SESSION_LOCK (sess);
-  /* get number of elements in the table */
-  size = g_hash_table_size (sess->ssrcs[sess->mask_idx]);
-  /* create the result value array */
-  res = g_value_array_new (size);
-
-  /* and copy all values into the array */
-  g_hash_table_foreach (sess->ssrcs[sess->mask_idx], (GHFunc) copy_source, res);
-  RTP_SESSION_UNLOCK (sess);
-
-  return res;
-}
-
-static void
-rtp_session_set_property (GObject * object, guint prop_id,
-    const GValue * value, GParamSpec * pspec)
-{
-  RTPSession *sess;
-
-  sess = RTP_SESSION (object);
-
-  switch (prop_id) {
-    case PROP_INTERNAL_SSRC:
-      rtp_session_set_internal_ssrc (sess, g_value_get_uint (value));
-      break;
-    case PROP_BANDWIDTH:
-      rtp_session_set_bandwidth (sess, g_value_get_double (value));
-      break;
-    case PROP_RTCP_FRACTION:
-      rtp_session_set_rtcp_fraction (sess, g_value_get_double (value));
-      break;
-    case PROP_RTCP_MTU:
-      sess->mtu = g_value_get_uint (value);
-      break;
-    case PROP_SDES:
-      rtp_session_set_sdes_struct (sess, g_value_get_boxed (value));
-      break;
-    default:
-      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
-      break;
-  }
-}
-
-static void
-rtp_session_get_property (GObject * object, guint prop_id,
-    GValue * value, GParamSpec * pspec)
-{
-  RTPSession *sess;
-
-  sess = RTP_SESSION (object);
-
-  switch (prop_id) {
-    case PROP_INTERNAL_SSRC:
-      g_value_set_uint (value, rtp_session_get_internal_ssrc (sess));
-      break;
-    case PROP_INTERNAL_SOURCE:
-      g_value_take_object (value, rtp_session_get_internal_source (sess));
-      break;
-    case PROP_BANDWIDTH:
-      g_value_set_double (value, rtp_session_get_bandwidth (sess));
-      break;
-    case PROP_RTCP_FRACTION:
-      g_value_set_double (value, rtp_session_get_rtcp_fraction (sess));
-      break;
-    case PROP_RTCP_MTU:
-      g_value_set_uint (value, sess->mtu);
-      break;
-    case PROP_SDES:
-      g_value_take_boxed (value, rtp_session_get_sdes_struct (sess));
-      break;
-    case PROP_NUM_SOURCES:
-      g_value_set_uint (value, rtp_session_get_num_sources (sess));
-      break;
-    case PROP_NUM_ACTIVE_SOURCES:
-      g_value_set_uint (value, rtp_session_get_num_active_sources (sess));
-      break;
-    case PROP_SOURCES:
-      g_value_take_boxed (value, rtp_session_create_sources (sess));
-      break;
-    default:
-      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
-      break;
-  }
-}
-
-static void
-on_new_ssrc (RTPSession * sess, RTPSource * source)
-{
-  g_object_ref (source);
-  RTP_SESSION_UNLOCK (sess);
-  g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_NEW_SSRC], 0, source);
-  RTP_SESSION_LOCK (sess);
-  g_object_unref (source);
-}
-
-static void
-on_ssrc_collision (RTPSession * sess, RTPSource * source)
-{
-  g_object_ref (source);
-  RTP_SESSION_UNLOCK (sess);
-  g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SSRC_COLLISION], 0,
-      source);
-  RTP_SESSION_LOCK (sess);
-  g_object_unref (source);
-}
-
-static void
-on_ssrc_validated (RTPSession * sess, RTPSource * source)
-{
-  g_object_ref (source);
-  RTP_SESSION_UNLOCK (sess);
-  g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SSRC_VALIDATED], 0,
-      source);
-  RTP_SESSION_LOCK (sess);
-  g_object_unref (source);
-}
-
-static void
-on_ssrc_active (RTPSession * sess, RTPSource * source)
-{
-  g_object_ref (source);
-  RTP_SESSION_UNLOCK (sess);
-  g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SSRC_ACTIVE], 0, source);
-  RTP_SESSION_LOCK (sess);
-  g_object_unref (source);
-}
-
-static void
-on_ssrc_sdes (RTPSession * sess, RTPSource * source)
-{
-  g_object_ref (source);
-  GST_DEBUG ("SDES changed for SSRC %08x", source->ssrc);
-  RTP_SESSION_UNLOCK (sess);
-  g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SSRC_SDES], 0, source);
-  RTP_SESSION_LOCK (sess);
-  g_object_unref (source);
-}
-
-static void
-on_bye_ssrc (RTPSession * sess, RTPSource * source)
-{
-  g_object_ref (source);
-  RTP_SESSION_UNLOCK (sess);
-  g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_BYE_SSRC], 0, source);
-  RTP_SESSION_LOCK (sess);
-  g_object_unref (source);
-}
-
-static void
-on_bye_timeout (RTPSession * sess, RTPSource * source)
-{
-  g_object_ref (source);
-  RTP_SESSION_UNLOCK (sess);
-  g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_BYE_TIMEOUT], 0, source);
-  RTP_SESSION_LOCK (sess);
-  g_object_unref (source);
-}
-
-static void
-on_timeout (RTPSession * sess, RTPSource * source)
-{
-  g_object_ref (source);
-  RTP_SESSION_UNLOCK (sess);
-  g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_TIMEOUT], 0, source);
-  RTP_SESSION_LOCK (sess);
-  g_object_unref (source);
-}
-
-static void
-on_sender_timeout (RTPSession * sess, RTPSource * source)
-{
-  g_object_ref (source);
-  RTP_SESSION_UNLOCK (sess);
-  g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SENDER_TIMEOUT], 0,
-      source);
-  RTP_SESSION_LOCK (sess);
-  g_object_unref (source);
-}
-
-/**
- * rtp_session_new:
- *
- * Create a new session object.
- *
- * Returns: a new #RTPSession. g_object_unref() after usage.
- */
-RTPSession *
-rtp_session_new (void)
-{
-  RTPSession *sess;
-
-  sess = g_object_new (RTP_TYPE_SESSION, NULL);
-
-  return sess;
-}
-
-/**
- * rtp_session_set_callbacks:
- * @sess: an #RTPSession
- * @callbacks: callbacks to configure
- * @user_data: user data passed in the callbacks
- *
- * Configure a set of callbacks to be notified of actions.
- */
-void
-rtp_session_set_callbacks (RTPSession * sess, RTPSessionCallbacks * callbacks,
-    gpointer user_data)
-{
-  g_return_if_fail (RTP_IS_SESSION (sess));
-
-  if (callbacks->process_rtp) {
-    sess->callbacks.process_rtp = callbacks->process_rtp;
-    sess->process_rtp_user_data = user_data;
-  }
-  if (callbacks->send_rtp) {
-    sess->callbacks.send_rtp = callbacks->send_rtp;
-    sess->send_rtp_user_data = user_data;
-  }
-  if (callbacks->send_rtcp) {
-    sess->callbacks.send_rtcp = callbacks->send_rtcp;
-    sess->send_rtcp_user_data = user_data;
-  }
-  if (callbacks->sync_rtcp) {
-    sess->callbacks.sync_rtcp = callbacks->sync_rtcp;
-    sess->sync_rtcp_user_data = user_data;
-  }
-  if (callbacks->clock_rate) {
-    sess->callbacks.clock_rate = callbacks->clock_rate;
-    sess->clock_rate_user_data = user_data;
-  }
-  if (callbacks->reconsider) {
-    sess->callbacks.reconsider = callbacks->reconsider;
-    sess->reconsider_user_data = user_data;
-  }
-}
-
-/**
- * rtp_session_set_process_rtp_callback:
- * @sess: an #RTPSession
- * @callback: callback to set
- * @user_data: user data passed in the callback
- *
- * Configure only the process_rtp callback to be notified of the process_rtp action.
- */
-void
-rtp_session_set_process_rtp_callback (RTPSession * sess,
-    RTPSessionProcessRTP callback, gpointer user_data)
-{
-  g_return_if_fail (RTP_IS_SESSION (sess));
-
-  sess->callbacks.process_rtp = callback;
-  sess->process_rtp_user_data = user_data;
-}
-
-/**
- * rtp_session_set_send_rtp_callback:
- * @sess: an #RTPSession
- * @callback: callback to set
- * @user_data: user data passed in the callback
- *
- * Configure only the send_rtp callback to be notified of the send_rtp action.
- */
-void
-rtp_session_set_send_rtp_callback (RTPSession * sess,
-    RTPSessionSendRTP callback, gpointer user_data)
-{
-  g_return_if_fail (RTP_IS_SESSION (sess));
-
-  sess->callbacks.send_rtp = callback;
-  sess->send_rtp_user_data = user_data;
-}
-
-/**
- * rtp_session_set_send_rtcp_callback:
- * @sess: an #RTPSession
- * @callback: callback to set
- * @user_data: user data passed in the callback
- *
- * Configure only the send_rtcp callback to be notified of the send_rtcp action.
- */
-void
-rtp_session_set_send_rtcp_callback (RTPSession * sess,
-    RTPSessionSendRTCP callback, gpointer user_data)
-{
-  g_return_if_fail (RTP_IS_SESSION (sess));
-
-  sess->callbacks.send_rtcp = callback;
-  sess->send_rtcp_user_data = user_data;
-}
-
-/**
- * rtp_session_set_sync_rtcp_callback:
- * @sess: an #RTPSession
- * @callback: callback to set
- * @user_data: user data passed in the callback
- *
- * Configure only the sync_rtcp callback to be notified of the sync_rtcp action.
- */
-void
-rtp_session_set_sync_rtcp_callback (RTPSession * sess,
-    RTPSessionSyncRTCP callback, gpointer user_data)
-{
-  g_return_if_fail (RTP_IS_SESSION (sess));
-
-  sess->callbacks.sync_rtcp = callback;
-  sess->sync_rtcp_user_data = user_data;
-}
-
-/**
- * rtp_session_set_clock_rate_callback:
- * @sess: an #RTPSession
- * @callback: callback to set
- * @user_data: user data passed in the callback
- *
- * Configure only the clock_rate callback to be notified of the clock_rate action.
- */
-void
-rtp_session_set_clock_rate_callback (RTPSession * sess,
-    RTPSessionClockRate callback, gpointer user_data)
-{
-  g_return_if_fail (RTP_IS_SESSION (sess));
-
-  sess->callbacks.clock_rate = callback;
-  sess->clock_rate_user_data = user_data;
-}
-
-/**
- * rtp_session_set_reconsider_callback:
- * @sess: an #RTPSession
- * @callback: callback to set
- * @user_data: user data passed in the callback
- *
- * Configure only the reconsider callback to be notified of the reconsider action.
- */
-void
-rtp_session_set_reconsider_callback (RTPSession * sess,
-    RTPSessionReconsider callback, gpointer user_data)
-{
-  g_return_if_fail (RTP_IS_SESSION (sess));
-
-  sess->callbacks.reconsider = callback;
-  sess->reconsider_user_data = user_data;
-}
-
-/**
- * rtp_session_set_bandwidth:
- * @sess: an #RTPSession
- * @bandwidth: the bandwidth allocated
- *
- * Set the session bandwidth in bytes per second.
- */
-void
-rtp_session_set_bandwidth (RTPSession * sess, gdouble bandwidth)
-{
-  g_return_if_fail (RTP_IS_SESSION (sess));
-
-  RTP_SESSION_LOCK (sess);
-  sess->stats.bandwidth = bandwidth;
-  RTP_SESSION_UNLOCK (sess);
-}
-
-/**
- * rtp_session_get_bandwidth:
- * @sess: an #RTPSession
- *
- * Get the session bandwidth.
- *
- * Returns: the session bandwidth.
- */
-gdouble
-rtp_session_get_bandwidth (RTPSession * sess)
-{
-  gdouble result;
-
-  g_return_val_if_fail (RTP_IS_SESSION (sess), 0);
-
-  RTP_SESSION_LOCK (sess);
-  result = sess->stats.bandwidth;
-  RTP_SESSION_UNLOCK (sess);
-
-  return result;
-}
-
-/**
- * rtp_session_set_rtcp_fraction:
- * @sess: an #RTPSession
- * @bandwidth: the RTCP bandwidth
- *
- * Set the bandwidth that should be used for RTCP
- * messages.
- */
-void
-rtp_session_set_rtcp_fraction (RTPSession * sess, gdouble bandwidth)
-{
-  g_return_if_fail (RTP_IS_SESSION (sess));
-
-  RTP_SESSION_LOCK (sess);
-  sess->stats.rtcp_bandwidth = bandwidth;
-  RTP_SESSION_UNLOCK (sess);
-}
-
-/**
- * rtp_session_get_rtcp_fraction:
- * @sess: an #RTPSession
- *
- * Get the session bandwidth used for RTCP.
- *
- * Returns: The bandwidth used for RTCP messages.
- */
-gdouble
-rtp_session_get_rtcp_fraction (RTPSession * sess)
-{
-  gdouble result;
-
-  g_return_val_if_fail (RTP_IS_SESSION (sess), 0.0);
-
-  RTP_SESSION_LOCK (sess);
-  result = sess->stats.rtcp_bandwidth;
-  RTP_SESSION_UNLOCK (sess);
-
-  return result;
-}
-
-/**
- * rtp_session_set_sdes_string:
- * @sess: an #RTPSession
- * @type: the type of the SDES item
- * @item: a null-terminated string to set. 
- *
- * Store an SDES item of @type in @sess. 
- *
- * Returns: %FALSE if the data was unchanged @type is invalid.
- */
-gboolean
-rtp_session_set_sdes_string (RTPSession * sess, GstRTCPSDESType type,
-    const gchar * item)
-{
-  gboolean result;
-
-  g_return_val_if_fail (RTP_IS_SESSION (sess), FALSE);
-
-  RTP_SESSION_LOCK (sess);
-  result = rtp_source_set_sdes_string (sess->source, type, item);
-  RTP_SESSION_UNLOCK (sess);
-
-  return result;
-}
-
-/**
- * rtp_session_get_sdes_string:
- * @sess: an #RTPSession
- * @type: the type of the SDES item
- *
- * Get the SDES item of @type from @sess. 
- *
- * Returns: a null-terminated copy of the SDES item or NULL when @type was not
- * valid. g_free() after usage.
- */
-gchar *
-rtp_session_get_sdes_string (RTPSession * sess, GstRTCPSDESType type)
-{
-  gchar *result;
-
-  g_return_val_if_fail (RTP_IS_SESSION (sess), NULL);
-
-  RTP_SESSION_LOCK (sess);
-  result = rtp_source_get_sdes_string (sess->source, type);
-  RTP_SESSION_UNLOCK (sess);
-
-  return result;
-}
-
-/**
- * rtp_session_get_sdes_struct:
- * @sess: an #RTSPSession
- *
- * Get the SDES data as a #GstStructure
- *
- * Returns: a GstStructure with SDES items for @sess.
- */
-GstStructure *
-rtp_session_get_sdes_struct (RTPSession * sess)
-{
-  GstStructure *result;
-
-  g_return_val_if_fail (RTP_IS_SESSION (sess), NULL);
-
-  RTP_SESSION_LOCK (sess);
-  result = rtp_source_get_sdes_struct (sess->source);
-  RTP_SESSION_UNLOCK (sess);
-
-  return result;
-}
-
-/**
- * rtp_session_set_sdes_struct:
- * @sess: an #RTSPSession
- * @sdes: a #GstStructure
- *
- * Set the SDES data as a #GstStructure.
- */
-void
-rtp_session_set_sdes_struct (RTPSession * sess, const GstStructure * sdes)
-{
-  g_return_if_fail (RTP_IS_SESSION (sess));
-
-  RTP_SESSION_LOCK (sess);
-  rtp_source_set_sdes_struct (sess->source, sdes);
-  RTP_SESSION_UNLOCK (sess);
-}
-
-static GstFlowReturn
-source_push_rtp (RTPSource * source, gpointer data, RTPSession * session)
-{
-  GstFlowReturn result = GST_FLOW_OK;
-
-  if (source == session->source) {
-    GST_LOG ("source %08x pushed sender RTP packet", source->ssrc);
-
-    RTP_SESSION_UNLOCK (session);
-
-    if (session->callbacks.send_rtp)
-      result =
-          session->callbacks.send_rtp (session, source, data,
-          session->send_rtp_user_data);
-    else {
-      gst_mini_object_unref (GST_MINI_OBJECT_CAST (data));
-    }
-  } else {
-    GST_LOG ("source %08x pushed receiver RTP packet", source->ssrc);
-    RTP_SESSION_UNLOCK (session);
-
-    if (session->callbacks.process_rtp)
-      result =
-          session->callbacks.process_rtp (session, source,
-          GST_BUFFER_CAST (data), session->process_rtp_user_data);
-    else
-      gst_buffer_unref (GST_BUFFER_CAST (data));
-  }
-  RTP_SESSION_LOCK (session);
-
-  return result;
-}
-
-static gint
-source_clock_rate (RTPSource * source, guint8 pt, RTPSession * session)
-{
-  gint result;
-
-  RTP_SESSION_UNLOCK (session);
-
-  if (session->callbacks.clock_rate)
-    result =
-        session->callbacks.clock_rate (session, pt,
-        session->clock_rate_user_data);
-  else
-    result = -1;
-
-  RTP_SESSION_LOCK (session);
-
-  GST_DEBUG ("got clock-rate %d for pt %d", result, pt);
-
-  return result;
-}
-
-static RTPSourceCallbacks callbacks = {
-  (RTPSourcePushRTP) source_push_rtp,
-  (RTPSourceClockRate) source_clock_rate,
-};
-
-/**
- * find_add_conflicting_addresses:
- * @sess: The session to check in
- * @arrival: The arrival stats for the buffer
- *
- * Checks if an address which has a conflict is already known,
- *  otherwise remembers it to prevent loops.
- *
- * Returns: TRUE if it was a known conflict, FALSE otherwise
- */
-
-static gboolean
-find_add_conflicting_addresses (RTPSession * sess, RTPArrivalStats * arrival)
-{
-  GList *item;
-  RTPConflictingAddress *new_conflict;
-
-  for (item = g_list_first (sess->conflicting_addresses);
-      item; item = g_list_next (item)) {
-    RTPConflictingAddress *known_conflict = item->data;
-
-    if (gst_netaddress_equal (&arrival->address, &known_conflict->address)) {
-      known_conflict->time = arrival->time;
-      return TRUE;
-    }
-  }
-
-  new_conflict = g_new0 (RTPConflictingAddress, 1);
-
-  memcpy (&new_conflict->address, &arrival->address, sizeof (GstNetAddress));
-  new_conflict->time = arrival->time;
-
-  sess->conflicting_addresses = g_list_prepend (sess->conflicting_addresses,
-      new_conflict);
-
-  return FALSE;
-}
-
-static gboolean
-check_collision (RTPSession * sess, RTPSource * source,
-    RTPArrivalStats * arrival, gboolean rtp)
-{
-  /* If we have no arrival address, we can't do collision checking */
-  if (!arrival->have_address)
-    return FALSE;
-
-  if (sess->source != source) {
-    /* This is not our local source, but lets check if two remote
-     * source collide
-     */
-    if (rtp) {
-      if (source->have_rtp_from) {
-        if (gst_netaddress_equal (&source->rtp_from, &arrival->address))
-          /* Address is the same */
-          return FALSE;
-      } else {
-        /* We don't already have a from address for RTP, just set it */
-        rtp_source_set_rtp_from (source, &arrival->address);
-        return FALSE;
-      }
-    } else {
-      if (source->have_rtcp_from) {
-        if (gst_netaddress_equal (&source->rtcp_from, &arrival->address))
-          /* Address is the same */
-          return FALSE;
-      } else {
-        /* We don't already have a from address for RTCP, just set it */
-        rtp_source_set_rtcp_from (source, &arrival->address);
-        return FALSE;
-      }
-    }
-    /* We received RTP or RTCP from this source before but the network address
-     * changed. In this case, we have third-party collision or loop */
-    GST_DEBUG ("we have a third-party collision or loop");
-
-    /* FIXME: Log 3rd party collision somehow
-     * Maybe should be done in upper layer, only the SDES can tell us
-     * if its a collision or a loop
-     */
-  } else {
-    /* This is sending with our ssrc, is it an address we already know */
-
-    if (find_add_conflicting_addresses (sess, arrival)) {
-      /* Its a known conflict, its probably a loop, not a collision
-       * lets just drop the incoming packet
-       */
-      GST_DEBUG ("Our packets are being looped back to us, dropping");
-    } else {
-      /* Its a new collision, lets change our SSRC */
-
-      GST_DEBUG ("Collision for SSRC %x", rtp_source_get_ssrc (source));
-      on_ssrc_collision (sess, source);
-
-      rtp_session_schedule_bye_locked (sess, "SSRC Collision", arrival->time);
-
-      sess->change_ssrc = TRUE;
-    }
-  }
-
-  return TRUE;
-}
-
-
-/* must be called with the session lock, the returned source needs to be
- * unreffed after usage. */
-static RTPSource *
-obtain_source (RTPSession * sess, guint32 ssrc, gboolean * created,
-    RTPArrivalStats * arrival, gboolean rtp)
-{
-  RTPSource *source;
-
-  source =
-      g_hash_table_lookup (sess->ssrcs[sess->mask_idx], GINT_TO_POINTER (ssrc));
-  if (source == NULL) {
-    /* make new Source in probation and insert */
-    source = rtp_source_new (ssrc);
-
-    /* for RTP packets we need to set the source in probation. Receiving RTCP
-     * packets of an SSRC, on the other hand, is a strong indication that we
-     * are dealing with a valid source. */
-    if (rtp)
-      source->probation = RTP_DEFAULT_PROBATION;
-    else
-      source->probation = 0;
-
-    /* store from address, if any */
-    if (arrival->have_address) {
-      if (rtp)
-        rtp_source_set_rtp_from (source, &arrival->address);
-      else
-        rtp_source_set_rtcp_from (source, &arrival->address);
-    }
-
-    /* configure a callback on the source */
-    rtp_source_set_callbacks (source, &callbacks, sess);
-
-    g_hash_table_insert (sess->ssrcs[sess->mask_idx], GINT_TO_POINTER (ssrc),
-        source);
-
-    /* we have one more source now */
-    sess->total_sources++;
-    *created = TRUE;
-  } else {
-    *created = FALSE;
-    /* check for collision, this updates the address when not previously set */
-    if (check_collision (sess, source, arrival, rtp)) {
-      return NULL;
-    }
-  }
-  /* update last activity */
-  source->last_activity = arrival->time;
-  if (rtp)
-    source->last_rtp_activity = arrival->time;
-  g_object_ref (source);
-
-  return source;
-}
-
-/**
- * rtp_session_get_internal_source:
- * @sess: a #RTPSession
- *
- * Get the internal #RTPSource of @sess.
- *
- * Returns: The internal #RTPSource. g_object_unref() after usage.
- */
-RTPSource *
-rtp_session_get_internal_source (RTPSession * sess)
-{
-  RTPSource *result;
-
-  g_return_val_if_fail (RTP_IS_SESSION (sess), NULL);
-
-  result = g_object_ref (sess->source);
-
-  return result;
-}
-
-/**
- * rtp_session_set_internal_ssrc:
- * @sess: a #RTPSession
- * @ssrc: an SSRC
- *
- * Set the SSRC of @sess to @ssrc.
- */
-void
-rtp_session_set_internal_ssrc (RTPSession * sess, guint32 ssrc)
-{
-  RTP_SESSION_LOCK (sess);
-  if (ssrc != sess->source->ssrc) {
-    g_hash_table_steal (sess->ssrcs[sess->mask_idx],
-        GINT_TO_POINTER (sess->source->ssrc));
-
-    GST_DEBUG ("setting internal SSRC to %08x", ssrc);
-    /* After this call, any receiver of the old SSRC either in RTP or RTCP
-     * packets will timeout on the old SSRC, we could potentially schedule a
-     * BYE RTCP for the old SSRC... */
-    sess->source->ssrc = ssrc;
-    rtp_source_reset (sess->source);
-
-    /* rehash with the new SSRC */
-    g_hash_table_insert (sess->ssrcs[sess->mask_idx],
-        GINT_TO_POINTER (sess->source->ssrc), sess->source);
-  }
-  RTP_SESSION_UNLOCK (sess);
-
-  g_object_notify (G_OBJECT (sess), "internal-ssrc");
-}
-
-/**
- * rtp_session_get_internal_ssrc:
- * @sess: a #RTPSession
- *
- * Get the internal SSRC of @sess.
- *
- * Returns: The SSRC of the session. 
- */
-guint32
-rtp_session_get_internal_ssrc (RTPSession * sess)
-{
-  guint32 ssrc;
-
-  RTP_SESSION_LOCK (sess);
-  ssrc = sess->source->ssrc;
-  RTP_SESSION_UNLOCK (sess);
-
-  return ssrc;
-}
-
-/**
- * rtp_session_add_source:
- * @sess: a #RTPSession
- * @src: #RTPSource to add
- *
- * Add @src to @session.
- *
- * Returns: %TRUE on success, %FALSE if a source with the same SSRC already
- * existed in the session.
- */
-gboolean
-rtp_session_add_source (RTPSession * sess, RTPSource * src)
-{
-  gboolean result = FALSE;
-  RTPSource *find;
-
-  g_return_val_if_fail (RTP_IS_SESSION (sess), FALSE);
-  g_return_val_if_fail (src != NULL, FALSE);
-
-  RTP_SESSION_LOCK (sess);
-  find =
-      g_hash_table_lookup (sess->ssrcs[sess->mask_idx],
-      GINT_TO_POINTER (src->ssrc));
-  if (find == NULL) {
-    g_hash_table_insert (sess->ssrcs[sess->mask_idx],
-        GINT_TO_POINTER (src->ssrc), src);
-    /* we have one more source now */
-    sess->total_sources++;
-    result = TRUE;
-  }
-  RTP_SESSION_UNLOCK (sess);
-
-  return result;
-}
-
-/**
- * rtp_session_get_num_sources:
- * @sess: an #RTPSession
- *
- * Get the number of sources in @sess.
- *
- * Returns: The number of sources in @sess.
- */
-guint
-rtp_session_get_num_sources (RTPSession * sess)
-{
-  guint result;
-
-  g_return_val_if_fail (RTP_IS_SESSION (sess), FALSE);
-
-  RTP_SESSION_LOCK (sess);
-  result = sess->total_sources;
-  RTP_SESSION_UNLOCK (sess);
-
-  return result;
-}
-
-/**
- * rtp_session_get_num_active_sources:
- * @sess: an #RTPSession
- *
- * Get the number of active sources in @sess. A source is considered active when
- * it has been validated and has not yet received a BYE RTCP message.
- *
- * Returns: The number of active sources in @sess.
- */
-guint
-rtp_session_get_num_active_sources (RTPSession * sess)
-{
-  guint result;
-
-  g_return_val_if_fail (RTP_IS_SESSION (sess), 0);
-
-  RTP_SESSION_LOCK (sess);
-  result = sess->stats.active_sources;
-  RTP_SESSION_UNLOCK (sess);
-
-  return result;
-}
-
-/**
- * rtp_session_get_source_by_ssrc:
- * @sess: an #RTPSession
- * @ssrc: an SSRC
- *
- * Find the source with @ssrc in @sess.
- *
- * Returns: a #RTPSource with SSRC @ssrc or NULL if the source was not found.
- * g_object_unref() after usage.
- */
-RTPSource *
-rtp_session_get_source_by_ssrc (RTPSession * sess, guint32 ssrc)
-{
-  RTPSource *result;
-
-  g_return_val_if_fail (RTP_IS_SESSION (sess), NULL);
-
-  RTP_SESSION_LOCK (sess);
-  result =
-      g_hash_table_lookup (sess->ssrcs[sess->mask_idx], GINT_TO_POINTER (ssrc));
-  if (result)
-    g_object_ref (result);
-  RTP_SESSION_UNLOCK (sess);
-
-  return result;
-}
-
-/**
- * rtp_session_get_source_by_cname:
- * @sess: a #RTPSession
- * @cname: an CNAME
- *
- * Find the source with @cname in @sess.
- *
- * Returns: a #RTPSource with CNAME @cname or NULL if the source was not found.
- * g_object_unref() after usage.
- */
-RTPSource *
-rtp_session_get_source_by_cname (RTPSession * sess, const gchar * cname)
-{
-  RTPSource *result;
-
-  g_return_val_if_fail (RTP_IS_SESSION (sess), NULL);
-  g_return_val_if_fail (cname != NULL, NULL);
-
-  RTP_SESSION_LOCK (sess);
-  result = g_hash_table_lookup (sess->cnames, cname);
-  if (result)
-    g_object_ref (result);
-  RTP_SESSION_UNLOCK (sess);
-
-  return result;
-}
-
-static guint32
-rtp_session_create_new_ssrc (RTPSession * sess)
-{
-  guint32 ssrc;
-
-  while (TRUE) {
-    ssrc = g_random_int ();
-
-    /* see if it exists in the session, we're done if it doesn't */
-    if (g_hash_table_lookup (sess->ssrcs[sess->mask_idx],
-            GINT_TO_POINTER (ssrc)) == NULL)
-      break;
-  }
-  return ssrc;
-}
-
-
-/**
- * rtp_session_create_source:
- * @sess: an #RTPSession
- *
- * Create an #RTPSource for use in @sess. This function will create a source
- * with an ssrc that is currently not used by any participants in the session.
- *
- * Returns: an #RTPSource.
- */
-RTPSource *
-rtp_session_create_source (RTPSession * sess)
-{
-  guint32 ssrc;
-  RTPSource *source;
-
-  RTP_SESSION_LOCK (sess);
-  ssrc = rtp_session_create_new_ssrc (sess);
-  source = rtp_source_new (ssrc);
-  rtp_source_set_callbacks (source, &callbacks, sess);
-  /* we need an additional ref for the source in the hashtable */
-  g_object_ref (source);
-  g_hash_table_insert (sess->ssrcs[sess->mask_idx], GINT_TO_POINTER (ssrc),
-      source);
-  /* we have one more source now */
-  sess->total_sources++;
-  RTP_SESSION_UNLOCK (sess);
-
-  return source;
-}
-
-/* update the RTPArrivalStats structure with the current time and other bits
- * about the current buffer we are handling.
- * This function is typically called when a validated packet is received.
- * This function should be called with the SESSION_LOCK
- */
-static void
-update_arrival_stats (RTPSession * sess, RTPArrivalStats * arrival,
-    gboolean rtp, GstBuffer * buffer, GstClockTime current_time,
-    GstClockTime running_time, guint64 ntpnstime)
-{
-  /* get time of arrival */
-  arrival->time = current_time;
-  arrival->running_time = running_time;
-  arrival->ntpnstime = ntpnstime;
-
-  /* get packet size including header overhead */
-  arrival->bytes = GST_BUFFER_SIZE (buffer) + sess->header_len;
-
-  if (rtp) {
-    arrival->payload_len = gst_rtp_buffer_get_payload_len (buffer);
-  } else {
-    arrival->payload_len = 0;
-  }
-
-  /* for netbuffer we can store the IP address to check for collisions */
-  arrival->have_address = GST_IS_NETBUFFER (buffer);
-  if (arrival->have_address) {
-    GstNetBuffer *netbuf = (GstNetBuffer *) buffer;
-
-    memcpy (&arrival->address, &netbuf->from, sizeof (GstNetAddress));
-  }
-}
-
-/**
- * rtp_session_process_rtp:
- * @sess: and #RTPSession
- * @buffer: an RTP buffer
- * @current_time: the current system time
- * @ntpnstime: the NTP arrival time in nanoseconds
- *
- * Process an RTP buffer in the session manager. This function takes ownership
- * of @buffer.
- *
- * Returns: a #GstFlowReturn.
- */
-GstFlowReturn
-rtp_session_process_rtp (RTPSession * sess, GstBuffer * buffer,
-    GstClockTime current_time, GstClockTime running_time, guint64 ntpnstime)
-{
-  GstFlowReturn result;
-  guint32 ssrc;
-  RTPSource *source;
-  gboolean created;
-  gboolean prevsender, prevactive;
-  RTPArrivalStats arrival;
-
-  g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
-  g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
-
-  if (!gst_rtp_buffer_validate (buffer))
-    goto invalid_packet;
-
-  RTP_SESSION_LOCK (sess);
-  /* update arrival stats */
-  update_arrival_stats (sess, &arrival, TRUE, buffer, current_time,
-      running_time, ntpnstime);
-
-  /* ignore more RTP packets when we left the session */
-  if (sess->source->received_bye)
-    goto ignore;
-
-  /* get SSRC and look up in session database */
-  ssrc = gst_rtp_buffer_get_ssrc (buffer);
-  source = obtain_source (sess, ssrc, &created, &arrival, TRUE);
-  if (!source)
-    goto collision;
-
-  prevsender = RTP_SOURCE_IS_SENDER (source);
-  prevactive = RTP_SOURCE_IS_ACTIVE (source);
-
-  /* we need to ref so that we can process the CSRCs later */
-  gst_buffer_ref (buffer);
-
-  /* let source process the packet */
-  result = rtp_source_process_rtp (source, buffer, &arrival);
-
-  /* source became active */
-  if (prevactive != RTP_SOURCE_IS_ACTIVE (source)) {
-    sess->stats.active_sources++;
-    GST_DEBUG ("source: %08x became active, %d active sources", ssrc,
-        sess->stats.active_sources);
-    on_ssrc_validated (sess, source);
-  }
-  if (prevsender != RTP_SOURCE_IS_SENDER (source)) {
-    sess->stats.sender_sources++;
-    GST_DEBUG ("source: %08x became sender, %d sender sources", ssrc,
-        sess->stats.sender_sources);
-  }
-
-  if (created)
-    on_new_ssrc (sess, source);
-
-  if (source->validated) {
-    guint8 i, count;
-    gboolean created;
-
-    /* for validated sources, we add the CSRCs as well */
-    count = gst_rtp_buffer_get_csrc_count (buffer);
-
-    for (i = 0; i < count; i++) {
-      guint32 csrc;
-      RTPSource *csrc_src;
-
-      csrc = gst_rtp_buffer_get_csrc (buffer, i);
-
-      /* get source */
-      csrc_src = obtain_source (sess, csrc, &created, &arrival, TRUE);
-      if (!csrc_src)
-        continue;
-
-      if (created) {
-        GST_DEBUG ("created new CSRC: %08x", csrc);
-        rtp_source_set_as_csrc (csrc_src);
-        if (RTP_SOURCE_IS_ACTIVE (csrc_src))
-          sess->stats.active_sources++;
-        on_new_ssrc (sess, csrc_src);
-      }
-      g_object_unref (csrc_src);
-    }
-  }
-  g_object_unref (source);
-  gst_buffer_unref (buffer);
-
-  RTP_SESSION_UNLOCK (sess);
-
-  return result;
-
-  /* ERRORS */
-invalid_packet:
-  {
-    gst_buffer_unref (buffer);
-    GST_DEBUG ("invalid RTP packet received");
-    return GST_FLOW_OK;
-  }
-ignore:
-  {
-    gst_buffer_unref (buffer);
-    RTP_SESSION_UNLOCK (sess);
-    GST_DEBUG ("ignoring RTP packet because we are leaving");
-    return GST_FLOW_OK;
-  }
-collision:
-  {
-    gst_buffer_unref (buffer);
-    RTP_SESSION_UNLOCK (sess);
-    GST_DEBUG ("ignoring packet because its collisioning");
-    return GST_FLOW_OK;
-  }
-}
-
-static void
-rtp_session_process_rb (RTPSession * sess, RTPSource * source,
-    GstRTCPPacket * packet, RTPArrivalStats * arrival)
-{
-  guint count, i;
-
-  count = gst_rtcp_packet_get_rb_count (packet);
-  for (i = 0; i < count; i++) {
-    guint32 ssrc, exthighestseq, jitter, lsr, dlsr;
-    guint8 fractionlost;
-    gint32 packetslost;
-
-    gst_rtcp_packet_get_rb (packet, i, &ssrc, &fractionlost,
-        &packetslost, &exthighestseq, &jitter, &lsr, &dlsr);
-
-    GST_DEBUG ("RB %d: SSRC %08x, jitter %" G_GUINT32_FORMAT, i, ssrc, jitter);
-
-    if (ssrc == sess->source->ssrc) {
-      /* only deal with report blocks for our session, we update the stats of
-       * the sender of the RTCP message. We could also compare our stats against
-       * the other sender to see if we are better or worse. */
-      rtp_source_process_rb (source, arrival->time, fractionlost, packetslost,
-          exthighestseq, jitter, lsr, dlsr);
-
-      on_ssrc_active (sess, source);
-    }
-  }
-}
-
-/* A Sender report contains statistics about how the sender is doing. This
- * includes timing informataion such as the relation between RTP and NTP
- * timestamps and the number of packets/bytes it sent to us.
- *
- * In this report is also included a set of report blocks related to how this
- * sender is receiving data (in case we (or somebody else) is also sending stuff
- * to it). This info includes the packet loss, jitter and seqnum. It also
- * contains information to calculate the round trip time (LSR/DLSR).
- */
-static void
-rtp_session_process_sr (RTPSession * sess, GstRTCPPacket * packet,
-    RTPArrivalStats * arrival, gboolean * do_sync)
-{
-  guint32 senderssrc, rtptime, packet_count, octet_count;
-  guint64 ntptime;
-  RTPSource *source;
-  gboolean created, prevsender;
-
-  gst_rtcp_packet_sr_get_sender_info (packet, &senderssrc, &ntptime, &rtptime,
-      &packet_count, &octet_count);
-
-  GST_DEBUG ("got SR packet: SSRC %08x, time %" GST_TIME_FORMAT,
-      senderssrc, GST_TIME_ARGS (arrival->time));
-
-  source = obtain_source (sess, senderssrc, &created, arrival, FALSE);
-  if (!source)
-    return;
-
-  /* don't try to do lip-sync for sources that sent a BYE */
-  if (rtp_source_received_bye (source))
-    *do_sync = FALSE;
-  else
-    *do_sync = TRUE;
-
-  prevsender = RTP_SOURCE_IS_SENDER (source);
-
-  /* first update the source */
-  rtp_source_process_sr (source, arrival->time, ntptime, rtptime, packet_count,
-      octet_count);
-
-  if (prevsender != RTP_SOURCE_IS_SENDER (source)) {
-    sess->stats.sender_sources++;
-    GST_DEBUG ("source: %08x became sender, %d sender sources", senderssrc,
-        sess->stats.sender_sources);
-  }
-
-  if (created)
-    on_new_ssrc (sess, source);
-
-  rtp_session_process_rb (sess, source, packet, arrival);
-  g_object_unref (source);
-}
-
-/* A receiver report contains statistics about how a receiver is doing. It
- * includes stuff like packet loss, jitter and the seqnum it received last. It
- * also contains info to calculate the round trip time.
- *
- * We are only interested in how the sender of this report is doing wrt to us.
- */
-static void
-rtp_session_process_rr (RTPSession * sess, GstRTCPPacket * packet,
-    RTPArrivalStats * arrival)
-{
-  guint32 senderssrc;
-  RTPSource *source;
-  gboolean created;
-
-  senderssrc = gst_rtcp_packet_rr_get_ssrc (packet);
-
-  GST_DEBUG ("got RR packet: SSRC %08x", senderssrc);
-
-  source = obtain_source (sess, senderssrc, &created, arrival, FALSE);
-  if (!source)
-    return;
-
-  if (created)
-    on_new_ssrc (sess, source);
-
-  rtp_session_process_rb (sess, source, packet, arrival);
-  g_object_unref (source);
-}
-
-/* Get SDES items and store them in the SSRC */
-static void
-rtp_session_process_sdes (RTPSession * sess, GstRTCPPacket * packet,
-    RTPArrivalStats * arrival)
-{
-  guint items, i, j;
-  gboolean more_items, more_entries;
-
-  items = gst_rtcp_packet_sdes_get_item_count (packet);
-  GST_DEBUG ("got SDES packet with %d items", items);
-
-  more_items = gst_rtcp_packet_sdes_first_item (packet);
-  i = 0;
-  while (more_items) {
-    guint32 ssrc;
-    gboolean changed, created;
-    RTPSource *source;
-
-    ssrc = gst_rtcp_packet_sdes_get_ssrc (packet);
-
-    GST_DEBUG ("item %d, SSRC %08x", i, ssrc);
-
-    changed = FALSE;
-
-    /* find src, no probation when dealing with RTCP */
-    source = obtain_source (sess, ssrc, &created, arrival, FALSE);
-    if (!source)
-      return;
-
-    more_entries = gst_rtcp_packet_sdes_first_entry (packet);
-    j = 0;
-    while (more_entries) {
-      GstRTCPSDESType type;
-      guint8 len;
-      guint8 *data;
-
-      gst_rtcp_packet_sdes_get_entry (packet, &type, &len, &data);
-
-      GST_DEBUG ("entry %d, type %d, len %d, data %.*s", j, type, len, len,
-          data);
-
-      changed |= rtp_source_set_sdes (source, type, data, len);
-
-      more_entries = gst_rtcp_packet_sdes_next_entry (packet);
-      j++;
-    }
-
-    source->validated = TRUE;
-
-    if (created)
-      on_new_ssrc (sess, source);
-    if (changed)
-      on_ssrc_sdes (sess, source);
-
-    g_object_unref (source);
-
-    more_items = gst_rtcp_packet_sdes_next_item (packet);
-    i++;
-  }
-}
-
-/* BYE is sent when a client leaves the session
- */
-static void
-rtp_session_process_bye (RTPSession * sess, GstRTCPPacket * packet,
-    RTPArrivalStats * arrival)
-{
-  guint count, i;
-  gchar *reason;
-  gboolean reconsider = FALSE;
-
-  reason = gst_rtcp_packet_bye_get_reason (packet);
-  GST_DEBUG ("got BYE packet (reason: %s)", GST_STR_NULL (reason));
-
-  count = gst_rtcp_packet_bye_get_ssrc_count (packet);
-  for (i = 0; i < count; i++) {
-    guint32 ssrc;
-    RTPSource *source;
-    gboolean created, prevactive, prevsender;
-    guint pmembers, members;
-
-    ssrc = gst_rtcp_packet_bye_get_nth_ssrc (packet, i);
-    GST_DEBUG ("SSRC: %08x", ssrc);
-
-    /* find src and mark bye, no probation when dealing with RTCP */
-    source = obtain_source (sess, ssrc, &created, arrival, FALSE);
-    if (!source)
-      return;
-
-    /* store time for when we need to time out this source */
-    source->bye_time = arrival->time;
-
-    prevactive = RTP_SOURCE_IS_ACTIVE (source);
-    prevsender = RTP_SOURCE_IS_SENDER (source);
-
-    /* let the source handle the rest */
-    rtp_source_process_bye (source, reason);
-
-    pmembers = sess->stats.active_sources;
-
-    if (prevactive && !RTP_SOURCE_IS_ACTIVE (source)) {
-      sess->stats.active_sources--;
-      GST_DEBUG ("source: %08x became inactive, %d active sources", ssrc,
-          sess->stats.active_sources);
-    }
-    if (prevsender && !RTP_SOURCE_IS_SENDER (source)) {
-      sess->stats.sender_sources--;
-      GST_DEBUG ("source: %08x became non sender, %d sender sources", ssrc,
-          sess->stats.sender_sources);
-    }
-    members = sess->stats.active_sources;
-
-    if (!sess->source->received_bye && members < pmembers) {
-      /* some members went away since the previous timeout estimate.
-       * Perform reverse reconsideration but only when we are not scheduling a
-       * BYE ourselves. */
-      if (arrival->time < sess->next_rtcp_check_time) {
-        GstClockTime time_remaining;
-
-        time_remaining = sess->next_rtcp_check_time - arrival->time;
-        sess->next_rtcp_check_time =
-            gst_util_uint64_scale (time_remaining, members, pmembers);
-
-        GST_DEBUG ("reverse reconsideration %" GST_TIME_FORMAT,
-            GST_TIME_ARGS (sess->next_rtcp_check_time));
-
-        sess->next_rtcp_check_time += arrival->time;
-
-        /* mark pending reconsider. We only want to signal the reconsideration
-         * once after we handled all the source in the bye packet */
-        reconsider = TRUE;
-      }
-    }
-
-    if (created)
-      on_new_ssrc (sess, source);
-
-    on_bye_ssrc (sess, source);
-
-    g_object_unref (source);
-  }
-  if (reconsider) {
-    RTP_SESSION_UNLOCK (sess);
-    /* notify app of reconsideration */
-    if (sess->callbacks.reconsider)
-      sess->callbacks.reconsider (sess, sess->reconsider_user_data);
-    RTP_SESSION_LOCK (sess);
-  }
-  g_free (reason);
-}
-
-static void
-rtp_session_process_app (RTPSession * sess, GstRTCPPacket * packet,
-    RTPArrivalStats * arrival)
-{
-  GST_DEBUG ("received APP");
-}
-
-/**
- * rtp_session_process_rtcp:
- * @sess: and #RTPSession
- * @buffer: an RTCP buffer
- * @current_time: the current system time
- *
- * Process an RTCP buffer in the session manager. This function takes ownership
- * of @buffer.
- *
- * Returns: a #GstFlowReturn.
- */
-GstFlowReturn
-rtp_session_process_rtcp (RTPSession * sess, GstBuffer * buffer,
-    GstClockTime current_time)
-{
-  GstRTCPPacket packet;
-  gboolean more, is_bye = FALSE, do_sync = FALSE;
-  RTPArrivalStats arrival;
-  GstFlowReturn result = GST_FLOW_OK;
-
-  g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
-  g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
-
-  if (!gst_rtcp_buffer_validate (buffer))
-    goto invalid_packet;
-
-  GST_DEBUG ("received RTCP packet");
-
-  RTP_SESSION_LOCK (sess);
-  /* update arrival stats */
-  update_arrival_stats (sess, &arrival, FALSE, buffer, current_time, -1, -1);
-
-  if (sess->sent_bye)
-    goto ignore;
-
-  /* make writable, we might want to change the buffer */
-  buffer = gst_buffer_make_metadata_writable (buffer);
-
-  /* start processing the compound packet */
-  more = gst_rtcp_buffer_get_first_packet (buffer, &packet);
-  while (more) {
-    GstRTCPType type;
-
-    type = gst_rtcp_packet_get_type (&packet);
-
-    /* when we are leaving the session, we should ignore all non-BYE messages */
-    if (sess->source->received_bye && type != GST_RTCP_TYPE_BYE) {
-      GST_DEBUG ("ignoring non-BYE RTCP packet because we are leaving");
-      goto next;
-    }
-
-    switch (type) {
-      case GST_RTCP_TYPE_SR:
-        rtp_session_process_sr (sess, &packet, &arrival, &do_sync);
-        break;
-      case GST_RTCP_TYPE_RR:
-        rtp_session_process_rr (sess, &packet, &arrival);
-        break;
-      case GST_RTCP_TYPE_SDES:
-        rtp_session_process_sdes (sess, &packet, &arrival);
-        break;
-      case GST_RTCP_TYPE_BYE:
-        is_bye = TRUE;
-        /* don't try to attempt lip-sync anymore for streams with a BYE */
-        do_sync = FALSE;
-        rtp_session_process_bye (sess, &packet, &arrival);
-        break;
-      case GST_RTCP_TYPE_APP:
-        rtp_session_process_app (sess, &packet, &arrival);
-        break;
-      default:
-        GST_WARNING ("got unknown RTCP packet");
-        break;
-    }
-  next:
-    more = gst_rtcp_packet_move_to_next (&packet);
-  }
-
-  /* if we are scheduling a BYE, we only want to count bye packets, else we
-   * count everything */
-  if (sess->source->received_bye) {
-    if (is_bye) {
-      sess->stats.bye_members++;
-      UPDATE_AVG (sess->stats.avg_rtcp_packet_size, arrival.bytes);
-    }
-  } else {
-    /* keep track of average packet size */
-    UPDATE_AVG (sess->stats.avg_rtcp_packet_size, arrival.bytes);
-  }
-  RTP_SESSION_UNLOCK (sess);
-
-  /* notify caller of sr packets in the callback */
-  if (do_sync && sess->callbacks.sync_rtcp)
-    result = sess->callbacks.sync_rtcp (sess, sess->source, buffer,
-        sess->sync_rtcp_user_data);
-  else
-    gst_buffer_unref (buffer);
-
-  return result;
-
-  /* ERRORS */
-invalid_packet:
-  {
-    GST_DEBUG ("invalid RTCP packet received");
-    gst_buffer_unref (buffer);
-    return GST_FLOW_OK;
-  }
-ignore:
-  {
-    gst_buffer_unref (buffer);
-    RTP_SESSION_UNLOCK (sess);
-    GST_DEBUG ("ignoring RTP packet because we left");
-    return GST_FLOW_OK;
-  }
-}
-
-/**
- * rtp_session_send_rtp:
- * @sess: an #RTPSession
- * @data: pointer to either an RTP buffer or a list of RTP buffers
- * @current_time: the current system time
- * @ntpnstime: the NTP time in nanoseconds of when this buffer was captured.
- * This is the buffer timestamp converted to NTP time.
- *
- * Send the RTP buffer in the session manager. This function takes ownership of
- * @buffer.
- *
- * Returns: a #GstFlowReturn.
- */
-GstFlowReturn
-rtp_session_send_rtp (RTPSession * sess, gpointer data, gboolean is_list,
-    GstClockTime current_time, guint64 ntpnstime)
-{
-  GstFlowReturn result;
-  RTPSource *source;
-  gboolean prevsender;
-  gboolean valid_packet;
-
-  g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
-  g_return_val_if_fail (is_list || GST_IS_BUFFER (data), GST_FLOW_ERROR);
-
-  if (is_list) {
-    valid_packet = gst_rtp_buffer_list_validate (GST_BUFFER_LIST_CAST (data));
-  } else {
-    valid_packet = gst_rtp_buffer_validate (GST_BUFFER_CAST (data));
-  }
-
-  if (!valid_packet)
-    goto invalid_packet;
-
-  GST_LOG ("received RTP %s for sending", is_list ? "list" : "packet");
-
-  RTP_SESSION_LOCK (sess);
-  source = sess->source;
-
-  /* update last activity */
-  source->last_rtp_activity = current_time;
-
-  prevsender = RTP_SOURCE_IS_SENDER (source);
-
-  /* we use our own source to send */
-  result = rtp_source_send_rtp (source, data, is_list, ntpnstime);
-
-  if (RTP_SOURCE_IS_SENDER (source) && !prevsender)
-    sess->stats.sender_sources++;
-  RTP_SESSION_UNLOCK (sess);
-
-  return result;
-
-  /* ERRORS */
-invalid_packet:
-  {
-    gst_mini_object_unref (GST_MINI_OBJECT_CAST (data));
-    GST_DEBUG ("invalid RTP packet received");
-    return GST_FLOW_OK;
-  }
-}
-
-static GstClockTime
-calculate_rtcp_interval (RTPSession * sess, gboolean deterministic,
-    gboolean first)
-{
-  GstClockTime result;
-
-  if (sess->source->received_bye) {
-    result = rtp_stats_calculate_bye_interval (&sess->stats);
-  } else {
-    result = rtp_stats_calculate_rtcp_interval (&sess->stats,
-        RTP_SOURCE_IS_SENDER (sess->source), first);
-  }
-
-  GST_DEBUG ("next deterministic interval: %" GST_TIME_FORMAT ", first %d",
-      GST_TIME_ARGS (result), first);
-
-  if (!deterministic)
-    result = rtp_stats_add_rtcp_jitter (&sess->stats, result);
-
-  GST_DEBUG ("next interval: %" GST_TIME_FORMAT, GST_TIME_ARGS (result));
-
-  return result;
-}
-
-/* Stop the current @sess and schedule a BYE message for the other members.
- * One must have the session lock to call this function
- */
-static GstFlowReturn
-rtp_session_schedule_bye_locked (RTPSession * sess, const gchar * reason,
-    GstClockTime current_time)
-{
-  GstFlowReturn result = GST_FLOW_OK;
-  RTPSource *source;
-  GstClockTime interval;
-
-  g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
-
-  source = sess->source;
-
-  /* ignore more BYEs */
-  if (source->received_bye)
-    goto done;
-
-  /* we have BYE now */
-  source->received_bye = TRUE;
-  /* at least one member wants to send a BYE */
-  g_free (sess->bye_reason);
-  sess->bye_reason = g_strdup (reason);
-  sess->stats.avg_rtcp_packet_size = 100;
-  sess->stats.bye_members = 1;
-  sess->first_rtcp = TRUE;
-  sess->sent_bye = FALSE;
-
-  /* reschedule transmission */
-  sess->last_rtcp_send_time = current_time;
-  interval = calculate_rtcp_interval (sess, FALSE, TRUE);
-  sess->next_rtcp_check_time = current_time + interval;
-
-  GST_DEBUG ("Schedule BYE for %" GST_TIME_FORMAT ", %" GST_TIME_FORMAT,
-      GST_TIME_ARGS (interval), GST_TIME_ARGS (sess->next_rtcp_check_time));
-
-  RTP_SESSION_UNLOCK (sess);
-  /* notify app of reconsideration */
-  if (sess->callbacks.reconsider)
-    sess->callbacks.reconsider (sess, sess->reconsider_user_data);
-  RTP_SESSION_LOCK (sess);
-done:
-
-  return result;
-}
-
-/**
- * rtp_session_schedule_bye:
- * @sess: an #RTPSession
- * @reason: a reason or NULL
- * @current_time: the current system time
- *
- * Stop the current @sess and schedule a BYE message for the other members.
- *
- * Returns: a #GstFlowReturn.
- */
-GstFlowReturn
-rtp_session_schedule_bye (RTPSession * sess, const gchar * reason,
-    GstClockTime current_time)
-{
-  GstFlowReturn result = GST_FLOW_OK;
-
-  g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
-
-  RTP_SESSION_LOCK (sess);
-  result = rtp_session_schedule_bye_locked (sess, reason, current_time);
-  RTP_SESSION_UNLOCK (sess);
-
-  return result;
-}
-
-/**
- * rtp_session_next_timeout:
- * @sess: an #RTPSession
- * @current_time: the current system time
- *
- * Get the next time we should perform session maintenance tasks.
- *
- * Returns: a time when rtp_session_on_timeout() should be called with the
- * current system time.
- */
-GstClockTime
-rtp_session_next_timeout (RTPSession * sess, GstClockTime current_time)
-{
-  GstClockTime result;
-
-  g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
-
-  RTP_SESSION_LOCK (sess);
-
-  result = sess->next_rtcp_check_time;
-
-  GST_DEBUG ("current time: %" GST_TIME_FORMAT ", next :%" GST_TIME_FORMAT,
-      GST_TIME_ARGS (current_time), GST_TIME_ARGS (result));
-
-  if (result < current_time) {
-    GST_DEBUG ("take current time as base");
-    /* our previous check time expired, start counting from the current time
-     * again. */
-    result = current_time;
-  }
-
-  if (sess->source->received_bye) {
-    if (sess->sent_bye) {
-      GST_DEBUG ("we sent BYE already");
-      result = GST_CLOCK_TIME_NONE;
-    } else if (sess->stats.active_sources >= 50) {
-      GST_DEBUG ("reconsider BYE, more than 50 sources");
-      /* reconsider BYE if members >= 50 */
-      result += calculate_rtcp_interval (sess, FALSE, TRUE);
-    }
-  } else {
-    if (sess->first_rtcp) {
-      GST_DEBUG ("first RTCP packet");
-      /* we are called for the first time */
-      result += calculate_rtcp_interval (sess, FALSE, TRUE);
-    } else if (sess->next_rtcp_check_time < current_time) {
-      GST_DEBUG ("old check time expired, getting new timeout");
-      /* get a new timeout when we need to */
-      result += calculate_rtcp_interval (sess, FALSE, FALSE);
-    }
-  }
-  sess->next_rtcp_check_time = result;
-
-  GST_DEBUG ("next timeout: %" GST_TIME_FORMAT, GST_TIME_ARGS (result));
-  RTP_SESSION_UNLOCK (sess);
-
-  return result;
-}
-
-typedef struct
-{
-  RTPSession *sess;
-  GstBuffer *rtcp;
-  GstClockTime current_time;
-  guint64 ntpnstime;
-  GstClockTime interval;
-  GstRTCPPacket packet;
-  gboolean is_bye;
-  gboolean has_sdes;
-} ReportData;
-
-static void
-session_start_rtcp (RTPSession * sess, ReportData * data)
-{
-  GstRTCPPacket *packet = &data->packet;
-  RTPSource *own = sess->source;
-
-  data->rtcp = gst_rtcp_buffer_new (sess->mtu);
-
-  if (RTP_SOURCE_IS_SENDER (own)) {
-    guint64 ntptime;
-    guint32 rtptime;
-    guint32 packet_count, octet_count;
-
-    /* we are a sender, create SR */
-    GST_DEBUG ("create SR for SSRC %08x", own->ssrc);
-    gst_rtcp_buffer_add_packet (data->rtcp, GST_RTCP_TYPE_SR, packet);
-
-    /* get latest stats */
-    rtp_source_get_new_sr (own, data->ntpnstime, &ntptime, &rtptime,
-        &packet_count, &octet_count);
-    /* store stats */
-    rtp_source_process_sr (own, data->current_time, ntptime, rtptime,
-        packet_count, octet_count);
-
-    /* fill in sender report info */
-    gst_rtcp_packet_sr_set_sender_info (packet, own->ssrc,
-        ntptime, rtptime, packet_count, octet_count);
-  } else {
-    /* we are only receiver, create RR */
-    GST_DEBUG ("create RR for SSRC %08x", own->ssrc);
-    gst_rtcp_buffer_add_packet (data->rtcp, GST_RTCP_TYPE_RR, packet);
-    gst_rtcp_packet_rr_set_ssrc (packet, own->ssrc);
-  }
-}
-
-/* construct a Sender or Receiver Report */
-static void
-session_report_blocks (const gchar * key, RTPSource * source, ReportData * data)
-{
-  RTPSession *sess = data->sess;
-  GstRTCPPacket *packet = &data->packet;
-
-  /* create a new buffer if needed */
-  if (data->rtcp == NULL) {
-    session_start_rtcp (sess, data);
-  }
-  if (gst_rtcp_packet_get_rb_count (packet) < GST_RTCP_MAX_RB_COUNT) {
-    /* only report about other sender sources */
-    if (source != sess->source && RTP_SOURCE_IS_SENDER (source)) {
-      guint8 fractionlost;
-      gint32 packetslost;
-      guint32 exthighestseq, jitter;
-      guint32 lsr, dlsr;
-
-      /* get new stats */
-      rtp_source_get_new_rb (source, data->current_time, &fractionlost,
-          &packetslost, &exthighestseq, &jitter, &lsr, &dlsr);
-
-      /* packet is not yet filled, add report block for this source. */
-      gst_rtcp_packet_add_rb (packet, source->ssrc, fractionlost, packetslost,
-          exthighestseq, jitter, lsr, dlsr);
-    }
-  }
-}
-
-/* perform cleanup of sources that timed out */
-static gboolean
-session_cleanup (const gchar * key, RTPSource * source, ReportData * data)
-{
-  gboolean remove = FALSE;
-  gboolean byetimeout = FALSE;
-  gboolean sendertimeout = FALSE;
-  gboolean is_sender, is_active;
-  RTPSession *sess = data->sess;
-  GstClockTime interval;
-
-  is_sender = RTP_SOURCE_IS_SENDER (source);
-  is_active = RTP_SOURCE_IS_ACTIVE (source);
-
-  /* check for our own source, we don't want to delete our own source. */
-  if (!(source == sess->source)) {
-    if (source->received_bye) {
-      /* if we received a BYE from the source, remove the source after some
-       * time. */
-      if (data->current_time > source->bye_time &&
-          data->current_time - source->bye_time > sess->stats.bye_timeout) {
-        GST_DEBUG ("removing BYE source %08x", source->ssrc);
-        remove = TRUE;
-        byetimeout = TRUE;
-      }
-    }
-    /* sources that were inactive for more than 5 times the deterministic reporting
-     * interval get timed out. the min timeout is 5 seconds. */
-    if (data->current_time > source->last_activity) {
-      interval = MAX (data->interval * 5, 5 * GST_SECOND);
-      if (data->current_time - source->last_activity > interval) {
-        GST_DEBUG ("removing timeout source %08x, last %" GST_TIME_FORMAT,
-            source->ssrc, GST_TIME_ARGS (source->last_activity));
-        remove = TRUE;
-      }
-    }
-  }
-
-  /* senders that did not send for a long time become a receiver, this also
-   * holds for our own source. */
-  if (is_sender) {
-    if (data->current_time > source->last_rtp_activity) {
-      interval = MAX (data->interval * 2, 5 * GST_SECOND);
-      if (data->current_time - source->last_rtp_activity > interval) {
-        GST_DEBUG ("sender source %08x timed out and became receiver, last %"
-            GST_TIME_FORMAT, source->ssrc,
-            GST_TIME_ARGS (source->last_rtp_activity));
-        source->is_sender = FALSE;
-        sess->stats.sender_sources--;
-        sendertimeout = TRUE;
-      }
-    }
-  }
-
-  if (remove) {
-    sess->total_sources--;
-    if (is_sender)
-      sess->stats.sender_sources--;
-    if (is_active)
-      sess->stats.active_sources--;
-
-    if (byetimeout)
-      on_bye_timeout (sess, source);
-    else
-      on_timeout (sess, source);
-  } else {
-    if (sendertimeout)
-      on_sender_timeout (sess, source);
-  }
-  return remove;
-}
-
-static void
-session_sdes (RTPSession * sess, ReportData * data)
-{
-  GstRTCPPacket *packet = &data->packet;
-  guint8 *sdes_data;
-  guint sdes_len;
-
-  /* add SDES packet */
-  gst_rtcp_buffer_add_packet (data->rtcp, GST_RTCP_TYPE_SDES, packet);
-
-  gst_rtcp_packet_sdes_add_item (packet, sess->source->ssrc);
-
-  rtp_source_get_sdes (sess->source, GST_RTCP_SDES_CNAME, &sdes_data,
-      &sdes_len);
-  gst_rtcp_packet_sdes_add_entry (packet, GST_RTCP_SDES_CNAME, sdes_len,
-      sdes_data);
-
-  /* other SDES items must only be added at regular intervals and only when the
-   * user requests to since it might be a privacy problem */
-#if 0
-  gst_rtcp_packet_sdes_add_entry (&packet, GST_RTCP_SDES_NAME,
-      strlen (sess->name), (guint8 *) sess->name);
-  gst_rtcp_packet_sdes_add_entry (&packet, GST_RTCP_SDES_TOOL,
-      strlen (sess->tool), (guint8 *) sess->tool);
-#endif
-
-  data->has_sdes = TRUE;
-}
-
-/* schedule a BYE packet */
-static void
-session_bye (RTPSession * sess, ReportData * data)
-{
-  GstRTCPPacket *packet = &data->packet;
-
-  /* open packet */
-  session_start_rtcp (sess, data);
-
-  /* add SDES */
-  session_sdes (sess, data);
-
-  /* add a BYE packet */
-  gst_rtcp_buffer_add_packet (data->rtcp, GST_RTCP_TYPE_BYE, packet);
-  gst_rtcp_packet_bye_add_ssrc (packet, sess->source->ssrc);
-  if (sess->bye_reason)
-    gst_rtcp_packet_bye_set_reason (packet, sess->bye_reason);
-
-  /* we have a BYE packet now */
-  data->is_bye = TRUE;
-}
-
-static gboolean
-is_rtcp_time (RTPSession * sess, GstClockTime current_time, ReportData * data)
-{
-  GstClockTime new_send_time, elapsed;
-  gboolean result;
-
-  /* no need to check yet */
-  if (sess->next_rtcp_check_time > current_time) {
-    GST_DEBUG ("no check time yet, next %" GST_TIME_FORMAT " > now %"
-        GST_TIME_FORMAT, GST_TIME_ARGS (sess->next_rtcp_check_time),
-        GST_TIME_ARGS (current_time));
-    return FALSE;
-  }
-
-  /* get elapsed time since we last reported */
-  elapsed = current_time - sess->last_rtcp_send_time;
-
-  /* perform forward reconsideration */
-  new_send_time = rtp_stats_add_rtcp_jitter (&sess->stats, data->interval);
-
-  GST_DEBUG ("forward reconsideration %" GST_TIME_FORMAT ", elapsed %"
-      GST_TIME_FORMAT, GST_TIME_ARGS (new_send_time), GST_TIME_ARGS (elapsed));
-
-  new_send_time += sess->last_rtcp_send_time;
-
-  /* check if reconsideration */
-  if (current_time < new_send_time) {
-    GST_DEBUG ("reconsider RTCP for %" GST_TIME_FORMAT,
-        GST_TIME_ARGS (new_send_time));
-    result = FALSE;
-    /* store new check time */
-    sess->next_rtcp_check_time = new_send_time;
-  } else {
-    result = TRUE;
-    new_send_time = calculate_rtcp_interval (sess, FALSE, FALSE);
-
-    GST_DEBUG ("can send RTCP now, next interval %" GST_TIME_FORMAT,
-        GST_TIME_ARGS (new_send_time));
-    sess->next_rtcp_check_time = current_time + new_send_time;
-  }
-  return result;
-}
-
-/**
- * rtp_session_on_timeout:
- * @sess: an #RTPSession
- * @current_time: the current system time
- * @ntpnstime: the current NTP time in nanoseconds
- *
- * Perform maintenance actions after the timeout obtained with
- * rtp_session_next_timeout() expired.
- *
- * This function will perform timeouts of receivers and senders, send a BYE
- * packet or generate RTCP packets with current session stats.
- *
- * This function can call the #RTPSessionSendRTCP callback, possibly multiple
- * times, for each packet that should be processed.
- *
- * Returns: a #GstFlowReturn.
- */
-GstFlowReturn
-rtp_session_on_timeout (RTPSession * sess, GstClockTime current_time,
-    guint64 ntpnstime)
-{
-  GstFlowReturn result = GST_FLOW_OK;
-  GList *item;
-  ReportData data;
-  RTPSource *own;
-  gboolean notify = FALSE;
-
-  g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
-
-  GST_DEBUG ("reporting at %" GST_TIME_FORMAT ", NTP time %" GST_TIME_FORMAT,
-      GST_TIME_ARGS (current_time), GST_TIME_ARGS (ntpnstime));
-
-  data.sess = sess;
-  data.rtcp = NULL;
-  data.current_time = current_time;
-  data.ntpnstime = ntpnstime;
-  data.is_bye = FALSE;
-  data.has_sdes = FALSE;
-
-  own = sess->source;
-
-  RTP_SESSION_LOCK (sess);
-  /* get a new interval, we need this for various cleanups etc */
-  data.interval = calculate_rtcp_interval (sess, TRUE, sess->first_rtcp);
-
-  /* first perform cleanups */
-  g_hash_table_foreach_remove (sess->ssrcs[sess->mask_idx],
-      (GHRFunc) session_cleanup, &data);
-
-  /* see if we need to generate SR or RR packets */
-  if (is_rtcp_time (sess, current_time, &data)) {
-    if (own->received_bye) {
-      /* generate BYE instead */
-      GST_DEBUG ("generating BYE message");
-      session_bye (sess, &data);
-      sess->sent_bye = TRUE;
-    } else {
-      /* loop over all known sources and do something */
-      g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
-          (GHFunc) session_report_blocks, &data);
-    }
-  }
-
-  if (data.rtcp) {
-    guint size;
-
-    /* we keep track of the last report time in order to timeout inactive
-     * receivers or senders */
-    sess->last_rtcp_send_time = data.current_time;
-    sess->first_rtcp = FALSE;
-
-    /* add SDES for this source when not already added */
-    if (!data.has_sdes)
-      session_sdes (sess, &data);
-
-    /* update average RTCP size before sending */
-    size = GST_BUFFER_SIZE (data.rtcp) + sess->header_len;
-    UPDATE_AVG (sess->stats.avg_rtcp_packet_size, size);
-  }
-
-  /* check for outdated collisions */
-  GST_DEBUG ("checking collision list");
-  item = g_list_first (sess->conflicting_addresses);
-  while (item) {
-    RTPConflictingAddress *known_conflict = item->data;
-    GList *next_item = g_list_next (item);
-
-    if (known_conflict->time < current_time - (data.interval *
-            RTCP_INTERVAL_COLLISION_TIMEOUT)) {
-      sess->conflicting_addresses =
-          g_list_delete_link (sess->conflicting_addresses, item);
-      GST_DEBUG ("collision %p timed out", known_conflict);
-      g_free (known_conflict);
-    }
-    item = next_item;
-  }
-
-  if (sess->change_ssrc) {
-    GST_DEBUG ("need to change our SSRC (%08x)", own->ssrc);
-    g_hash_table_steal (sess->ssrcs[sess->mask_idx],
-        GINT_TO_POINTER (own->ssrc));
-
-    own->ssrc = rtp_session_create_new_ssrc (sess);
-    rtp_source_reset (own);
-
-    g_hash_table_insert (sess->ssrcs[sess->mask_idx],
-        GINT_TO_POINTER (own->ssrc), own);
-
-    g_free (sess->bye_reason);
-    sess->bye_reason = NULL;
-    sess->sent_bye = FALSE;
-    sess->change_ssrc = FALSE;
-    notify = TRUE;
-    GST_DEBUG ("changed our SSRC to %08x", own->ssrc);
-  }
-  RTP_SESSION_UNLOCK (sess);
-
-  if (notify)
-    g_object_notify (G_OBJECT (sess), "internal-ssrc");
-
-  /* push out the RTCP packet */
-  if (data.rtcp) {
-    /* close the RTCP packet */
-    gst_rtcp_buffer_end (data.rtcp);
-
-    GST_DEBUG ("sending packet");
-    if (sess->callbacks.send_rtcp)
-      result = sess->callbacks.send_rtcp (sess, own, data.rtcp,
-          sess->sent_bye, sess->send_rtcp_user_data);
-    else {
-      GST_DEBUG ("freeing packet");
-      gst_buffer_unref (data.rtcp);
-    }
-  }
-
-  return result;
-}
diff --git a/gst/rtpmanager/rtpsession.h b/gst/rtpmanager/rtpsession.h
deleted file mode 100644
index 25e228b0..00000000
--- a/gst/rtpmanager/rtpsession.h
+++ /dev/null
@@ -1,306 +0,0 @@
-/* GStreamer
- * Copyright (C) <2007> Wim Taymans <wim.taymans@gmail.com>
- *
- * This library is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Library General Public
- * License as published by the Free Software Foundation; either
- * version 2 of the License, or (at your option) any later version.
- *
- * This library is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Library General Public License for more details.
- *
- * You should have received a copy of the GNU Library General Public
- * License along with this library; if not, write to the
- * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
- * Boston, MA 02111-1307, USA.
- */
-
-#ifndef __RTP_SESSION_H__
-#define __RTP_SESSION_H__
-
-#include <gst/gst.h>
-#include <gst/netbuffer/gstnetbuffer.h>
-
-#include "rtpsource.h"
-
-typedef struct _RTPSession RTPSession;
-typedef struct _RTPSessionClass RTPSessionClass;
-
-#define RTP_TYPE_SESSION             (rtp_session_get_type())
-#define RTP_SESSION(sess)            (G_TYPE_CHECK_INSTANCE_CAST((sess),RTP_TYPE_SESSION,RTPSession))
-#define RTP_SESSION_CLASS(klass)     (G_TYPE_CHECK_CLASS_CAST((klass),RTP_TYPE_SESSION,RTPSessionClass))
-#define RTP_IS_SESSION(sess)         (G_TYPE_CHECK_INSTANCE_TYPE((sess),RTP_TYPE_SESSION))
-#define RTP_IS_SESSION_CLASS(klass)  (G_TYPE_CHECK_CLASS_TYPE((klass),RTP_TYPE_SESSION))
-#define RTP_SESSION_CAST(sess)       ((RTPSession *)(sess))
-
-#define RTP_SESSION_LOCK(sess)     (g_mutex_lock ((sess)->lock))
-#define RTP_SESSION_UNLOCK(sess)   (g_mutex_unlock ((sess)->lock))
-
-/**
- * RTPSessionProcessRTP:
- * @sess: an #RTPSession
- * @src: the #RTPSource
- * @buffer: the RTP buffer ready for processing
- * @user_data: user data specified when registering
- *
- * This callback will be called when @sess has @buffer ready for further
- * processing. Processing the buffer typically includes decoding and displaying
- * the buffer.
- *
- * Returns: a #GstFlowReturn.
- */
-typedef GstFlowReturn (*RTPSessionProcessRTP) (RTPSession *sess, RTPSource *src, GstBuffer *buffer, gpointer user_data);
-
-/**
- * RTPSessionSendRTP:
- * @sess: an #RTPSession
- * @src: the #RTPSource
- * @buffer: the RTP buffer ready for sending
- * @user_data: user data specified when registering
- *
- * This callback will be called when @sess has @buffer ready for sending to
- * all listening participants in this session.
- *
- * Returns: a #GstFlowReturn.
- */
-typedef GstFlowReturn (*RTPSessionSendRTP) (RTPSession *sess, RTPSource *src, gpointer data, gpointer user_data);
-
-/**
- * RTPSessionSendRTCP:
- * @sess: an #RTPSession
- * @src: the #RTPSource
- * @buffer: the RTCP buffer ready for sending
- * @eos: if an EOS event should be pushed
- * @user_data: user data specified when registering
- *
- * This callback will be called when @sess has @buffer ready for sending to
- * all listening participants in this session.
- *
- * Returns: a #GstFlowReturn.
- */
-typedef GstFlowReturn (*RTPSessionSendRTCP) (RTPSession *sess, RTPSource *src, GstBuffer *buffer, 
-    gboolean eos, gpointer user_data);
-
-/**
- * RTPSessionSyncRTCP:
- * @sess: an #RTPSession
- * @src: the #RTPSource
- * @buffer: the RTCP buffer ready for synchronisation
- * @user_data: user data specified when registering
- *
- * This callback will be called when @sess has an SR @buffer ready for doing
- * synchronisation between streams.
- *
- * Returns: a #GstFlowReturn.
- */
-typedef GstFlowReturn (*RTPSessionSyncRTCP) (RTPSession *sess, RTPSource *src, GstBuffer *buffer, gpointer user_data);
-
-/**
- * RTPSessionClockRate:
- * @sess: an #RTPSession
- * @payload: the payload
- * @user_data: user data specified when registering
- *
- * This callback will be called when @sess needs the clock-rate of @payload.
- *
- * Returns: the clock-rate of @pt.
- */
-typedef gint (*RTPSessionClockRate) (RTPSession *sess, guint8 payload, gpointer user_data);
-
-/**
- * RTPSessionReconsider:
- * @sess: an #RTPSession
- * @user_data: user data specified when registering
- *
- * This callback will be called when @sess needs to cancel the current timeout. 
- * The currently running timeout should be canceled and a new reporting interval
- * should be requested from @sess.
- */
-typedef void (*RTPSessionReconsider) (RTPSession *sess, gpointer user_data);
-
-/**
- * RTPSessionCallbacks:
- * @RTPSessionProcessRTP: callback to process RTP packets
- * @RTPSessionSendRTP: callback for sending RTP packets
- * @RTPSessionSendRTCP: callback for sending RTCP packets
- * @RTPSessionSyncRTCP: callback for handling SR packets
- * @RTPSessionReconsider: callback for reconsidering the timeout
- *
- * These callbacks can be installed on the session manager to get notification
- * when RTP and RTCP packets are ready for further processing. These callbacks
- * are not implemented with signals for performance reasons.
- */
-typedef struct {
-  RTPSessionProcessRTP  process_rtp;
-  RTPSessionSendRTP     send_rtp;
-  RTPSessionSyncRTCP    sync_rtcp;
-  RTPSessionSendRTCP    send_rtcp;
-  RTPSessionClockRate   clock_rate;
-  RTPSessionReconsider  reconsider;
-} RTPSessionCallbacks;
-
-/**
- * RTPConflictingAddress:
- * @address: #GstNetAddress which conflicted
- * @last_conflict_time: time when the last conflict was seen
- *
- * This structure is used to account for addresses that have conflicted to find
- * loops.
- */
-typedef struct {
-  GstNetAddress address;
-  GstClockTime time;
-} RTPConflictingAddress;
-
-/**
- * RTPSession:
- * @lock: lock to protect the session
- * @source: the source of this session
- * @ssrcs: Hashtable of sources indexed by SSRC
- * @cnames: Hashtable of sources indexed by CNAME
- * @num_sources: the number of sources
- * @activecount: the number of active sources
- * @callbacks: callbacks
- * @user_data: user data passed in callbacks
- * @stats: session statistics
- * @conflicting_addresses: GList of conflicting addresses
- *
- * The RTP session manager object
- */
-struct _RTPSession {
-  GObject       object;
-
-  GMutex       *lock;
-
-  guint         header_len;
-  guint         mtu;
-
-  RTPSource    *source;
-
-  /* for sender/receiver counting */
-  guint32       key;
-  guint32       mask_idx;
-  guint32       mask;
-  GHashTable   *ssrcs[32];
-  GHashTable   *cnames;
-  guint         total_sources;
-
-  GstClockTime  next_rtcp_check_time;
-  GstClockTime  last_rtcp_send_time;
-  gboolean      first_rtcp;
-
-  gchar        *bye_reason;
-  gboolean      sent_bye;
-
-  RTPSessionCallbacks   callbacks;
-  gpointer              process_rtp_user_data;
-  gpointer              send_rtp_user_data;
-  gpointer              send_rtcp_user_data;
-  gpointer              sync_rtcp_user_data;
-  gpointer              clock_rate_user_data;
-  gpointer              reconsider_user_data;
-
-  RTPSessionStats stats;
-
-  GList         *conflicting_addresses;
-  gboolean      change_ssrc;
-};
-
-/**
- * RTPSessionClass:
- * @on_new_ssrc: emited when a new source is found
- * @on_bye_ssrc: emited when a source is gone
- *
- * The session class.
- */
-struct _RTPSessionClass {
-  GObjectClass   parent_class;
-
-  /* action signals */
-  RTPSource* (*get_source_by_ssrc) (RTPSession *sess, guint32 ssrc);
-
-  /* signals */
-  void (*on_new_ssrc)       (RTPSession *sess, RTPSource *source);
-  void (*on_ssrc_collision) (RTPSession *sess, RTPSource *source);
-  void (*on_ssrc_validated) (RTPSession *sess, RTPSource *source);
-  void (*on_ssrc_active)    (RTPSession *sess, RTPSource *source);
-  void (*on_ssrc_sdes)      (RTPSession *sess, RTPSource *source);
-  void (*on_bye_ssrc)       (RTPSession *sess, RTPSource *source);
-  void (*on_bye_timeout)    (RTPSession *sess, RTPSource *source);
-  void (*on_timeout)        (RTPSession *sess, RTPSource *source);
-  void (*on_sender_timeout) (RTPSession *sess, RTPSource *source);
-};
-
-GType rtp_session_get_type (void);
-
-/* create and configure */
-RTPSession*     rtp_session_new           (void);
-void            rtp_session_set_callbacks          (RTPSession *sess,
-		                                    RTPSessionCallbacks *callbacks,
-                                                    gpointer user_data);
-void            rtp_session_set_process_rtp_callback   (RTPSession * sess,
-                                                    RTPSessionProcessRTP callback,
-                                                    gpointer user_data);
-void            rtp_session_set_send_rtp_callback  (RTPSession * sess,
-                                                    RTPSessionSendRTP callback,
-                                                    gpointer user_data);
-void            rtp_session_set_send_rtcp_callback   (RTPSession * sess,
-                                                    RTPSessionSendRTCP callback,
-                                                    gpointer user_data);
-void            rtp_session_set_sync_rtcp_callback   (RTPSession * sess,
-                                                    RTPSessionSyncRTCP callback,
-                                                    gpointer user_data);
-void            rtp_session_set_clock_rate_callback   (RTPSession * sess,
-                                                    RTPSessionClockRate callback,
-                                                    gpointer user_data);
-void            rtp_session_set_reconsider_callback (RTPSession * sess,
-                                                    RTPSessionReconsider callback,
-                                                    gpointer user_data);
-void            rtp_session_set_bandwidth          (RTPSession *sess, gdouble bandwidth);
-gdouble         rtp_session_get_bandwidth          (RTPSession *sess);
-void            rtp_session_set_rtcp_fraction      (RTPSession *sess, gdouble fraction);
-gdouble         rtp_session_get_rtcp_fraction      (RTPSession *sess);
-
-gboolean        rtp_session_set_sdes_string        (RTPSession *sess, GstRTCPSDESType type,
-                                                    const gchar *cname);
-gchar*          rtp_session_get_sdes_string        (RTPSession *sess, GstRTCPSDESType type);
-
-GstStructure *  rtp_session_get_sdes_struct        (RTPSession *sess);
-void            rtp_session_set_sdes_struct        (RTPSession *sess, const GstStructure *sdes);
-
-/* handling sources */
-RTPSource*      rtp_session_get_internal_source    (RTPSession *sess);
-
-void            rtp_session_set_internal_ssrc      (RTPSession *sess, guint32 ssrc);
-guint32         rtp_session_get_internal_ssrc      (RTPSession *sess);
-
-gboolean        rtp_session_add_source             (RTPSession *sess, RTPSource *src);
-guint           rtp_session_get_num_sources        (RTPSession *sess);
-guint           rtp_session_get_num_active_sources (RTPSession *sess);
-RTPSource*      rtp_session_get_source_by_ssrc     (RTPSession *sess, guint32 ssrc);
-RTPSource*      rtp_session_get_source_by_cname    (RTPSession *sess, const gchar *cname);
-RTPSource*      rtp_session_create_source          (RTPSession *sess);
-
-/* processing packets from receivers */
-GstFlowReturn   rtp_session_process_rtp            (RTPSession *sess, GstBuffer *buffer,
-                                                    GstClockTime current_time,
-						    GstClockTime running_time, guint64 ntpnstime);
-GstFlowReturn   rtp_session_process_rtcp           (RTPSession *sess, GstBuffer *buffer,
-                                                    GstClockTime current_time);
-
-/* processing packets for sending */
-GstFlowReturn   rtp_session_send_rtp               (RTPSession *sess, gpointer data, gboolean is_list,
-                                                    GstClockTime current_time, guint64 ntpnstime);
-
-/* stopping the session */
-GstFlowReturn   rtp_session_schedule_bye           (RTPSession *sess, const gchar *reason,
-                                                    GstClockTime current_time);
-
-/* get interval for next RTCP interval */
-GstClockTime    rtp_session_next_timeout           (RTPSession *sess, GstClockTime current_time);
-GstFlowReturn   rtp_session_on_timeout             (RTPSession *sess, GstClockTime current_time,
-                                                    guint64 ntpnstime);
-
-#endif /* __RTP_SESSION_H__ */
diff --git a/gst/rtpmanager/rtpsource.c b/gst/rtpmanager/rtpsource.c
deleted file mode 100644
index 28fa23ef..00000000
--- a/gst/rtpmanager/rtpsource.c
+++ /dev/null
@@ -1,1625 +0,0 @@
-/* GStreamer
- * Copyright (C) <2007> Wim Taymans <wim.taymans@gmail.com>
- *
- * This library is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Library General Public
- * License as published by the Free Software Foundation; either
- * version 2 of the License, or (at your option) any later version.
- *
- * This library is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Library General Public License for more details.
- *
- * You should have received a copy of the GNU Library General Public
- * License along with this library; if not, write to the
- * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
- * Boston, MA 02111-1307, USA.
- */
-#include <string.h>
-
-#include <gst/rtp/gstrtpbuffer.h>
-#include <gst/rtp/gstrtcpbuffer.h>
-
-#include "rtpsource.h"
-
-GST_DEBUG_CATEGORY_STATIC (rtp_source_debug);
-#define GST_CAT_DEFAULT rtp_source_debug
-
-#define RTP_MAX_PROBATION_LEN  32
-
-/* signals and args */
-enum
-{
-  LAST_SIGNAL
-};
-
-#define DEFAULT_SSRC                 0
-#define DEFAULT_IS_CSRC              FALSE
-#define DEFAULT_IS_VALIDATED         FALSE
-#define DEFAULT_IS_SENDER            FALSE
-#define DEFAULT_SDES                 NULL
-
-enum
-{
-  PROP_0,
-  PROP_SSRC,
-  PROP_IS_CSRC,
-  PROP_IS_VALIDATED,
-  PROP_IS_SENDER,
-  PROP_SDES,
-  PROP_STATS,
-  PROP_LAST
-};
-
-/* GObject vmethods */
-static void rtp_source_finalize (GObject * object);
-static void rtp_source_set_property (GObject * object, guint prop_id,
-    const GValue * value, GParamSpec * pspec);
-static void rtp_source_get_property (GObject * object, guint prop_id,
-    GValue * value, GParamSpec * pspec);
-
-/* static guint rtp_source_signals[LAST_SIGNAL] = { 0 }; */
-
-G_DEFINE_TYPE (RTPSource, rtp_source, G_TYPE_OBJECT);
-
-static void
-rtp_source_class_init (RTPSourceClass * klass)
-{
-  GObjectClass *gobject_class;
-
-  gobject_class = (GObjectClass *) klass;
-
-  gobject_class->finalize = rtp_source_finalize;
-
-  gobject_class->set_property = rtp_source_set_property;
-  gobject_class->get_property = rtp_source_get_property;
-
-  g_object_class_install_property (gobject_class, PROP_SSRC,
-      g_param_spec_uint ("ssrc", "SSRC",
-          "The SSRC of this source", 0, G_MAXUINT, DEFAULT_SSRC,
-          G_PARAM_READWRITE | G_PARAM_CONSTRUCT_ONLY | G_PARAM_STATIC_STRINGS));
-
-  g_object_class_install_property (gobject_class, PROP_IS_CSRC,
-      g_param_spec_boolean ("is-csrc", "Is CSRC",
-          "If this SSRC is acting as a contributing source",
-          DEFAULT_IS_CSRC, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
-
-  g_object_class_install_property (gobject_class, PROP_IS_VALIDATED,
-      g_param_spec_boolean ("is-validated", "Is Validated",
-          "If this SSRC is validated", DEFAULT_IS_VALIDATED,
-          G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
-
-  g_object_class_install_property (gobject_class, PROP_IS_SENDER,
-      g_param_spec_boolean ("is-sender", "Is Sender",
-          "If this SSRC is a sender", DEFAULT_IS_SENDER,
-          G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
-
-  /**
-   * RTPSource::sdes
-   *
-   * The current SDES items of the source. Returns a structure with the
-   * following fields:
-   *
-   *  'cname'    G_TYPE_STRING  : The canonical name 
-   *  'name'     G_TYPE_STRING  : The user name 
-   *  'email'    G_TYPE_STRING  : The user's electronic mail address
-   *  'phone'    G_TYPE_STRING  : The user's phone number
-   *  'location' G_TYPE_STRING  : The geographic user location
-   *  'tool'     G_TYPE_STRING  : The name of application or tool
-   *  'note'     G_TYPE_STRING  : A notice about the source
-   */
-  g_object_class_install_property (gobject_class, PROP_SDES,
-      g_param_spec_boxed ("sdes", "SDES",
-          "The SDES information for this source",
-          GST_TYPE_STRUCTURE, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
-
-  /**
-   * RTPSource::stats
-   *
-   * The statistics of the source. This property returns a GstStructure with
-   * name application/x-rtp-source-stats with the following fields:
-   * 
-   */
-  g_object_class_install_property (gobject_class, PROP_STATS,
-      g_param_spec_boxed ("stats", "Stats",
-          "The stats of this source", GST_TYPE_STRUCTURE,
-          G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
-
-  GST_DEBUG_CATEGORY_INIT (rtp_source_debug, "rtpsource", 0, "RTP Source");
-}
-
-/**
- * rtp_source_reset:
- * @src: an #RTPSource
- *
- * Reset the stats of @src.
- */
-void
-rtp_source_reset (RTPSource * src)
-{
-  src->received_bye = FALSE;
-
-  src->stats.cycles = -1;
-  src->stats.jitter = 0;
-  src->stats.transit = -1;
-  src->stats.curr_sr = 0;
-  src->stats.curr_rr = 0;
-}
-
-static void
-rtp_source_init (RTPSource * src)
-{
-  /* sources are initialy on probation until we receive enough valid RTP
-   * packets or a valid RTCP packet */
-  src->validated = FALSE;
-  src->internal = FALSE;
-  src->probation = RTP_DEFAULT_PROBATION;
-
-  src->payload = -1;
-  src->clock_rate = -1;
-  src->packets = g_queue_new ();
-  src->seqnum_base = -1;
-  src->last_rtptime = -1;
-
-  rtp_source_reset (src);
-}
-
-static void
-rtp_source_finalize (GObject * object)
-{
-  RTPSource *src;
-  GstBuffer *buffer;
-  gint i;
-
-  src = RTP_SOURCE_CAST (object);
-
-  while ((buffer = g_queue_pop_head (src->packets)))
-    gst_buffer_unref (buffer);
-  g_queue_free (src->packets);
-
-  for (i = 0; i < 9; i++)
-    g_free (src->sdes[i]);
-
-  g_free (src->bye_reason);
-
-  gst_caps_replace (&src->caps, NULL);
-
-  G_OBJECT_CLASS (rtp_source_parent_class)->finalize (object);
-}
-
-static GstStructure *
-rtp_source_create_stats (RTPSource * src)
-{
-  GstStructure *s;
-  gboolean is_sender = src->is_sender;
-  gboolean internal = src->internal;
-  gchar address_str[GST_NETADDRESS_MAX_LEN];
-
-  /* common data for all types of sources */
-  s = gst_structure_new ("application/x-rtp-source-stats",
-      "ssrc", G_TYPE_UINT, (guint) src->ssrc,
-      "internal", G_TYPE_BOOLEAN, internal,
-      "validated", G_TYPE_BOOLEAN, src->validated,
-      "received-bye", G_TYPE_BOOLEAN, src->received_bye,
-      "is-csrc", G_TYPE_BOOLEAN, src->is_csrc,
-      "is-sender", G_TYPE_BOOLEAN, is_sender, NULL);
-
-  /* add address and port */
-  if (src->have_rtp_from) {
-    gst_netaddress_to_string (&src->rtp_from, address_str,
-        sizeof (address_str));
-    gst_structure_set (s, "rtp-from", G_TYPE_STRING, address_str, NULL);
-  }
-  if (src->have_rtcp_from) {
-    gst_netaddress_to_string (&src->rtcp_from, address_str,
-        sizeof (address_str));
-    gst_structure_set (s, "rtcp-from", G_TYPE_STRING, address_str, NULL);
-  }
-
-  if (internal) {
-    /* our internal source */
-    if (is_sender) {
-      /* if we are sending, report about how much we sent, other sources will
-       * have a RB with info on reception. */
-      gst_structure_set (s,
-          "octets-sent", G_TYPE_UINT64, src->stats.octets_sent,
-          "packets-sent", G_TYPE_UINT64, src->stats.packets_sent,
-          "bitrate", G_TYPE_UINT64, src->bitrate, NULL);
-    } else {
-      /* if we are not sending we have nothing more to report */
-    }
-  } else {
-    gboolean have_rb;
-    guint8 fractionlost = 0;
-    gint32 packetslost = 0;
-    guint32 exthighestseq = 0;
-    guint32 jitter = 0;
-    guint32 lsr = 0;
-    guint32 dlsr = 0;
-    guint32 round_trip = 0;
-
-    /* other sources */
-    if (is_sender) {
-      gboolean have_sr;
-      GstClockTime time = 0;
-      guint64 ntptime = 0;
-      guint32 rtptime = 0;
-      guint32 packet_count = 0;
-      guint32 octet_count = 0;
-
-      /* this source is sending to us, get the last SR. */
-      have_sr = rtp_source_get_last_sr (src, &time, &ntptime, &rtptime,
-          &packet_count, &octet_count);
-      gst_structure_set (s,
-          "octets-received", G_TYPE_UINT64, src->stats.octets_received,
-          "packets-received", G_TYPE_UINT64, src->stats.packets_received,
-          "have-sr", G_TYPE_BOOLEAN, have_sr,
-          "sr-ntptime", G_TYPE_UINT64, ntptime,
-          "sr-rtptime", G_TYPE_UINT, (guint) rtptime,
-          "sr-octet-count", G_TYPE_UINT, (guint) octet_count,
-          "sr-packet-count", G_TYPE_UINT, (guint) packet_count, NULL);
-    }
-    /* we might be sending to this SSRC so we report about how it is
-     * receiving our data */
-    have_rb = rtp_source_get_last_rb (src, &fractionlost, &packetslost,
-        &exthighestseq, &jitter, &lsr, &dlsr, &round_trip);
-
-    gst_structure_set (s,
-        "have-rb", G_TYPE_BOOLEAN, have_rb,
-        "rb-fractionlost", G_TYPE_UINT, (guint) fractionlost,
-        "rb-packetslost", G_TYPE_INT, (gint) packetslost,
-        "rb-exthighestseq", G_TYPE_UINT, (guint) exthighestseq,
-        "rb-jitter", G_TYPE_UINT, (guint) jitter,
-        "rb-lsr", G_TYPE_UINT, (guint) lsr,
-        "rb-dlsr", G_TYPE_UINT, (guint) dlsr,
-        "rb-round-trip", G_TYPE_UINT, (guint) round_trip, NULL);
-  }
-
-  return s;
-}
-
-/**
- * rtp_source_get_sdes_struct:
- * @src: an #RTSPSource
- *
- * Get the SDES data as a GstStructure
- *
- * Returns: a GstStructure with SDES items for @src.
- */
-GstStructure *
-rtp_source_get_sdes_struct (RTPSource * src)
-{
-  GstStructure *s;
-  gchar *str;
-
-  s = gst_structure_new ("application/x-rtp-source-sdes",
-      "ssrc", G_TYPE_UINT, (guint) src->ssrc, NULL);
-
-  if ((str = rtp_source_get_sdes_string (src, GST_RTCP_SDES_CNAME))) {
-    gst_structure_set (s, "cname", G_TYPE_STRING, str, NULL);
-    g_free (str);
-  }
-  if ((str = rtp_source_get_sdes_string (src, GST_RTCP_SDES_NAME))) {
-    gst_structure_set (s, "name", G_TYPE_STRING, str, NULL);
-    g_free (str);
-  }
-  if ((str = rtp_source_get_sdes_string (src, GST_RTCP_SDES_EMAIL))) {
-    gst_structure_set (s, "email", G_TYPE_STRING, str, NULL);
-    g_free (str);
-  }
-  if ((str = rtp_source_get_sdes_string (src, GST_RTCP_SDES_PHONE))) {
-    gst_structure_set (s, "phone", G_TYPE_STRING, str, NULL);
-    g_free (str);
-  }
-  if ((str = rtp_source_get_sdes_string (src, GST_RTCP_SDES_LOC))) {
-    gst_structure_set (s, "location", G_TYPE_STRING, str, NULL);
-    g_free (str);
-  }
-  if ((str = rtp_source_get_sdes_string (src, GST_RTCP_SDES_TOOL))) {
-    gst_structure_set (s, "tool", G_TYPE_STRING, str, NULL);
-    g_free (str);
-  }
-  if ((str = rtp_source_get_sdes_string (src, GST_RTCP_SDES_NOTE))) {
-    gst_structure_set (s, "note", G_TYPE_STRING, str, NULL);
-    g_free (str);
-  }
-  return s;
-}
-
-/**
- * rtp_source_set_sdes_struct:
- * @src: an #RTSPSource
- * @sdes: a #GstStructure with SDES info
- *
- * Set the SDES items from @sdes.
- */
-void
-rtp_source_set_sdes_struct (RTPSource * src, const GstStructure * sdes)
-{
-  const gchar *str;
-
-  if (!gst_structure_has_name (sdes, "application/x-rtp-source-sdes"))
-    return;
-
-  if ((str = gst_structure_get_string (sdes, "cname"))) {
-    rtp_source_set_sdes_string (src, GST_RTCP_SDES_CNAME, str);
-  }
-  if ((str = gst_structure_get_string (sdes, "name"))) {
-    rtp_source_set_sdes_string (src, GST_RTCP_SDES_NAME, str);
-  }
-  if ((str = gst_structure_get_string (sdes, "email"))) {
-    rtp_source_set_sdes_string (src, GST_RTCP_SDES_EMAIL, str);
-  }
-  if ((str = gst_structure_get_string (sdes, "phone"))) {
-    rtp_source_set_sdes_string (src, GST_RTCP_SDES_PHONE, str);
-  }
-  if ((str = gst_structure_get_string (sdes, "location"))) {
-    rtp_source_set_sdes_string (src, GST_RTCP_SDES_LOC, str);
-  }
-  if ((str = gst_structure_get_string (sdes, "tool"))) {
-    rtp_source_set_sdes_string (src, GST_RTCP_SDES_TOOL, str);
-  }
-  if ((str = gst_structure_get_string (sdes, "note"))) {
-    rtp_source_set_sdes_string (src, GST_RTCP_SDES_NOTE, str);
-  }
-}
-
-static void
-rtp_source_set_property (GObject * object, guint prop_id,
-    const GValue * value, GParamSpec * pspec)
-{
-  RTPSource *src;
-
-  src = RTP_SOURCE (object);
-
-  switch (prop_id) {
-    case PROP_SSRC:
-      src->ssrc = g_value_get_uint (value);
-      break;
-    default:
-      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
-      break;
-  }
-}
-
-static void
-rtp_source_get_property (GObject * object, guint prop_id,
-    GValue * value, GParamSpec * pspec)
-{
-  RTPSource *src;
-
-  src = RTP_SOURCE (object);
-
-  switch (prop_id) {
-    case PROP_SSRC:
-      g_value_set_uint (value, rtp_source_get_ssrc (src));
-      break;
-    case PROP_IS_CSRC:
-      g_value_set_boolean (value, rtp_source_is_as_csrc (src));
-      break;
-    case PROP_IS_VALIDATED:
-      g_value_set_boolean (value, rtp_source_is_validated (src));
-      break;
-    case PROP_IS_SENDER:
-      g_value_set_boolean (value, rtp_source_is_sender (src));
-      break;
-    case PROP_SDES:
-      g_value_take_boxed (value, rtp_source_get_sdes_struct (src));
-      break;
-    case PROP_STATS:
-      g_value_take_boxed (value, rtp_source_create_stats (src));
-      break;
-    default:
-      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
-      break;
-  }
-}
-
-/**
- * rtp_source_new:
- * @ssrc: an SSRC
- *
- * Create a #RTPSource with @ssrc.
- *
- * Returns: a new #RTPSource. Use g_object_unref() after usage.
- */
-RTPSource *
-rtp_source_new (guint32 ssrc)
-{
-  RTPSource *src;
-
-  src = g_object_new (RTP_TYPE_SOURCE, NULL);
-  src->ssrc = ssrc;
-
-  return src;
-}
-
-/**
- * rtp_source_set_callbacks:
- * @src: an #RTPSource
- * @cb: callback functions
- * @user_data: user data
- *
- * Set the callbacks for the source.
- */
-void
-rtp_source_set_callbacks (RTPSource * src, RTPSourceCallbacks * cb,
-    gpointer user_data)
-{
-  g_return_if_fail (RTP_IS_SOURCE (src));
-
-  src->callbacks.push_rtp = cb->push_rtp;
-  src->callbacks.clock_rate = cb->clock_rate;
-  src->user_data = user_data;
-}
-
-/**
- * rtp_source_get_ssrc:
- * @src: an #RTPSource
- *
- * Get the SSRC of @source.
- *
- * Returns: the SSRC of src.
- */
-guint32
-rtp_source_get_ssrc (RTPSource * src)
-{
-  guint32 result;
-
-  g_return_val_if_fail (RTP_IS_SOURCE (src), 0);
-
-  result = src->ssrc;
-
-  return result;
-}
-
-/**
- * rtp_source_set_as_csrc:
- * @src: an #RTPSource
- *
- * Configure @src as a CSRC, this will also validate @src.
- */
-void
-rtp_source_set_as_csrc (RTPSource * src)
-{
-  g_return_if_fail (RTP_IS_SOURCE (src));
-
-  src->validated = TRUE;
-  src->is_csrc = TRUE;
-}
-
-/**
- * rtp_source_is_as_csrc:
- * @src: an #RTPSource
- *
- * Check if @src is a contributing source.
- *
- * Returns: %TRUE if @src is acting as a contributing source.
- */
-gboolean
-rtp_source_is_as_csrc (RTPSource * src)
-{
-  gboolean result;
-
-  g_return_val_if_fail (RTP_IS_SOURCE (src), FALSE);
-
-  result = src->is_csrc;
-
-  return result;
-}
-
-/**
- * rtp_source_is_active:
- * @src: an #RTPSource
- *
- * Check if @src is an active source. A source is active if it has been
- * validated and has not yet received a BYE packet
- *
- * Returns: %TRUE if @src is an qactive source.
- */
-gboolean
-rtp_source_is_active (RTPSource * src)
-{
-  gboolean result;
-
-  g_return_val_if_fail (RTP_IS_SOURCE (src), FALSE);
-
-  result = RTP_SOURCE_IS_ACTIVE (src);
-
-  return result;
-}
-
-/**
- * rtp_source_is_validated:
- * @src: an #RTPSource
- *
- * Check if @src is a validated source.
- *
- * Returns: %TRUE if @src is a validated source.
- */
-gboolean
-rtp_source_is_validated (RTPSource * src)
-{
-  gboolean result;
-
-  g_return_val_if_fail (RTP_IS_SOURCE (src), FALSE);
-
-  result = src->validated;
-
-  return result;
-}
-
-/**
- * rtp_source_is_sender:
- * @src: an #RTPSource
- *
- * Check if @src is a sending source.
- *
- * Returns: %TRUE if @src is a sending source.
- */
-gboolean
-rtp_source_is_sender (RTPSource * src)
-{
-  gboolean result;
-
-  g_return_val_if_fail (RTP_IS_SOURCE (src), FALSE);
-
-  result = RTP_SOURCE_IS_SENDER (src);
-
-  return result;
-}
-
-/**
- * rtp_source_received_bye:
- * @src: an #RTPSource
- *
- * Check if @src has receoved a BYE packet.
- *
- * Returns: %TRUE if @src has received a BYE packet.
- */
-gboolean
-rtp_source_received_bye (RTPSource * src)
-{
-  gboolean result;
-
-  g_return_val_if_fail (RTP_IS_SOURCE (src), FALSE);
-
-  result = src->received_bye;
-
-  return result;
-}
-
-
-/**
- * rtp_source_get_bye_reason:
- * @src: an #RTPSource
- *
- * Get the BYE reason for @src. Check if the source receoved a BYE message first
- * with rtp_source_received_bye().
- *
- * Returns: The BYE reason or NULL when no reason was given or the source did
- * not receive a BYE message yet. g_fee() after usage.
- */
-gchar *
-rtp_source_get_bye_reason (RTPSource * src)
-{
-  gchar *result;
-
-  g_return_val_if_fail (RTP_IS_SOURCE (src), NULL);
-
-  result = g_strdup (src->bye_reason);
-
-  return result;
-}
-
-/**
- * rtp_source_update_caps:
- * @src: an #RTPSource
- * @caps: a #GstCaps
- *
- * Parse @caps and store all relevant information in @source.
- */
-void
-rtp_source_update_caps (RTPSource * src, GstCaps * caps)
-{
-  GstStructure *s;
-  guint val;
-  gint ival;
-
-  /* nothing changed, return */
-  if (caps == NULL || src->caps == caps)
-    return;
-
-  s = gst_caps_get_structure (caps, 0);
-
-  if (gst_structure_get_int (s, "payload", &ival))
-    src->payload = ival;
-  else
-    src->payload = -1;
-  GST_DEBUG ("got payload %d", src->payload);
-
-  if (gst_structure_get_int (s, "clock-rate", &ival))
-    src->clock_rate = ival;
-  else
-    src->clock_rate = -1;
-
-  GST_DEBUG ("got clock-rate %d", src->clock_rate);
-
-  if (gst_structure_get_uint (s, "seqnum-base", &val))
-    src->seqnum_base = val;
-  else
-    src->seqnum_base = -1;
-
-  GST_DEBUG ("got seqnum-base %" G_GINT32_FORMAT, src->seqnum_base);
-
-  gst_caps_replace (&src->caps, caps);
-}
-
-/**
- * rtp_source_set_sdes:
- * @src: an #RTPSource
- * @type: the type of the SDES item
- * @data: the SDES data
- * @len: the SDES length
- *
- * Store an SDES item of @type in @src. 
- *
- * Returns: %FALSE if the SDES item was unchanged or @type is unknown.
- */
-gboolean
-rtp_source_set_sdes (RTPSource * src, GstRTCPSDESType type,
-    const guint8 * data, guint len)
-{
-  guint8 *old;
-
-  g_return_val_if_fail (RTP_IS_SOURCE (src), FALSE);
-
-  if (type < 0 || type > GST_RTCP_SDES_PRIV)
-    return FALSE;
-
-  old = src->sdes[type];
-
-  /* lengths are the same, check if the data is the same */
-  if ((src->sdes_len[type] == len))
-    if (data != NULL && old != NULL && (memcmp (old, data, len) == 0))
-      return FALSE;
-
-  /* NULL data, make sure we store 0 length or if no length is given,
-   * take strlen */
-  if (data == NULL)
-    len = 0;
-
-  g_free (src->sdes[type]);
-  src->sdes[type] = g_memdup (data, len);
-  src->sdes_len[type] = len;
-
-  return TRUE;
-}
-
-/**
- * rtp_source_set_sdes_string:
- * @src: an #RTPSource
- * @type: the type of the SDES item
- * @data: the SDES data
- *
- * Store an SDES item of @type in @src. This function is similar to
- * rtp_source_set_sdes() but takes a null-terminated string for convenience.
- *
- * Returns: %FALSE if the SDES item was unchanged or @type is unknown.
- */
-gboolean
-rtp_source_set_sdes_string (RTPSource * src, GstRTCPSDESType type,
-    const gchar * data)
-{
-  guint len;
-  gboolean result;
-
-  if (data)
-    len = strlen (data);
-  else
-    len = 0;
-
-  result = rtp_source_set_sdes (src, type, (guint8 *) data, len);
-
-  return result;
-}
-
-/**
- * rtp_source_get_sdes:
- * @src: an #RTPSource
- * @type: the type of the SDES item
- * @data: location to store the SDES data or NULL
- * @len: location to store the SDES length or NULL
- *
- * Get the SDES item of @type from @src. Note that @data does not always point
- * to a null-terminated string, use rtp_source_get_sdes_string() to retrieve a
- * null-terminated string instead.
- *
- * @data remains valid until the next call to rtp_source_set_sdes().
- *
- * Returns: %TRUE if @type was valid and @data and @len contain valid
- * data. @data can be NULL when the item was unset.
- */
-gboolean
-rtp_source_get_sdes (RTPSource * src, GstRTCPSDESType type, guint8 ** data,
-    guint * len)
-{
-  g_return_val_if_fail (RTP_IS_SOURCE (src), FALSE);
-
-  if (type < 0 || type > GST_RTCP_SDES_PRIV)
-    return FALSE;
-
-  if (data)
-    *data = src->sdes[type];
-  if (len)
-    *len = src->sdes_len[type];
-
-  return TRUE;
-}
-
-/**
- * rtp_source_get_sdes_string:
- * @src: an #RTPSource
- * @type: the type of the SDES item
- *
- * Get the SDES item of @type from @src. 
- *
- * Returns: a null-terminated copy of the SDES item or NULL when @type was not
- * valid or the SDES item was unset. g_free() after usage.
- */
-gchar *
-rtp_source_get_sdes_string (RTPSource * src, GstRTCPSDESType type)
-{
-  gchar *result;
-
-  g_return_val_if_fail (RTP_IS_SOURCE (src), NULL);
-
-  if (type < 0 || type > GST_RTCP_SDES_PRIV)
-    return NULL;
-
-  result = g_strndup ((const gchar *) src->sdes[type], src->sdes_len[type]);
-
-  return result;
-}
-
-/**
- * rtp_source_set_rtp_from:
- * @src: an #RTPSource
- * @address: the RTP address to set
- *
- * Set that @src is receiving RTP packets from @address. This is used for
- * collistion checking.
- */
-void
-rtp_source_set_rtp_from (RTPSource * src, GstNetAddress * address)
-{
-  g_return_if_fail (RTP_IS_SOURCE (src));
-
-  src->have_rtp_from = TRUE;
-  memcpy (&src->rtp_from, address, sizeof (GstNetAddress));
-}
-
-/**
- * rtp_source_set_rtcp_from:
- * @src: an #RTPSource
- * @address: the RTCP address to set
- *
- * Set that @src is receiving RTCP packets from @address. This is used for
- * collistion checking.
- */
-void
-rtp_source_set_rtcp_from (RTPSource * src, GstNetAddress * address)
-{
-  g_return_if_fail (RTP_IS_SOURCE (src));
-
-  src->have_rtcp_from = TRUE;
-  memcpy (&src->rtcp_from, address, sizeof (GstNetAddress));
-}
-
-static GstFlowReturn
-push_packet (RTPSource * src, GstBuffer * buffer)
-{
-  GstFlowReturn ret = GST_FLOW_OK;
-
-  /* push queued packets first if any */
-  while (!g_queue_is_empty (src->packets)) {
-    GstBuffer *buffer = GST_BUFFER_CAST (g_queue_pop_head (src->packets));
-
-    GST_LOG ("pushing queued packet");
-    if (src->callbacks.push_rtp)
-      src->callbacks.push_rtp (src, buffer, src->user_data);
-    else
-      gst_buffer_unref (buffer);
-  }
-  GST_LOG ("pushing new packet");
-  /* push packet */
-  if (src->callbacks.push_rtp)
-    ret = src->callbacks.push_rtp (src, buffer, src->user_data);
-  else
-    gst_buffer_unref (buffer);
-
-  return ret;
-}
-
-static gint
-get_clock_rate (RTPSource * src, guint8 payload)
-{
-  if (src->payload == -1) {
-    /* first payload received, nothing was in the caps, lock on to this payload */
-    src->payload = payload;
-    GST_DEBUG ("first payload %d", payload);
-  } else if (payload != src->payload) {
-    /* we have a different payload than before, reset the clock-rate */
-    GST_DEBUG ("new payload %d", payload);
-    src->payload = payload;
-    src->clock_rate = -1;
-    src->stats.transit = -1;
-  }
-
-  if (src->clock_rate == -1) {
-    gint clock_rate = -1;
-
-    if (src->callbacks.clock_rate)
-      clock_rate = src->callbacks.clock_rate (src, payload, src->user_data);
-
-    GST_DEBUG ("got clock-rate %d", clock_rate);
-
-    src->clock_rate = clock_rate;
-  }
-  return src->clock_rate;
-}
-
-/* Jitter is the variation in the delay of received packets in a flow. It is
- * measured by comparing the interval when RTP packets were sent to the interval
- * at which they were received. For instance, if packet #1 and packet #2 leave
- * 50 milliseconds apart and arrive 60 milliseconds apart, then the jitter is 10
- * milliseconds. */
-static void
-calculate_jitter (RTPSource * src, GstBuffer * buffer,
-    RTPArrivalStats * arrival)
-{
-  guint64 ntpnstime;
-  guint32 rtparrival, transit, rtptime;
-  gint32 diff;
-  gint clock_rate;
-  guint8 pt;
-
-  /* get arrival time */
-  if ((ntpnstime = arrival->ntpnstime) == GST_CLOCK_TIME_NONE)
-    goto no_time;
-
-  pt = gst_rtp_buffer_get_payload_type (buffer);
-
-  GST_LOG ("SSRC %08x got payload %d", src->ssrc, pt);
-
-  /* get clockrate */
-  if ((clock_rate = get_clock_rate (src, pt)) == -1)
-    goto no_clock_rate;
-
-  rtptime = gst_rtp_buffer_get_timestamp (buffer);
-
-  /* convert arrival time to RTP timestamp units, truncate to 32 bits, we don't
-   * care about the absolute value, just the difference. */
-  rtparrival = gst_util_uint64_scale_int (ntpnstime, clock_rate, GST_SECOND);
-
-  /* transit time is difference with RTP timestamp */
-  transit = rtparrival - rtptime;
-
-  /* get ABS diff with previous transit time */
-  if (src->stats.transit != -1) {
-    if (transit > src->stats.transit)
-      diff = transit - src->stats.transit;
-    else
-      diff = src->stats.transit - transit;
-  } else
-    diff = 0;
-
-  src->stats.transit = transit;
-
-  /* update jitter, the value we store is scaled up so we can keep precision. */
-  src->stats.jitter += diff - ((src->stats.jitter + 8) >> 4);
-
-  src->stats.prev_rtptime = src->stats.last_rtptime;
-  src->stats.last_rtptime = rtparrival;
-
-  GST_LOG ("rtparrival %u, rtptime %u, clock-rate %d, diff %d, jitter: %f",
-      rtparrival, rtptime, clock_rate, diff, (src->stats.jitter) / 16.0);
-
-  return;
-
-  /* ERRORS */
-no_time:
-  {
-    GST_WARNING ("cannot get current time");
-    return;
-  }
-no_clock_rate:
-  {
-    GST_WARNING ("cannot get clock-rate for pt %d", pt);
-    return;
-  }
-}
-
-static void
-init_seq (RTPSource * src, guint16 seq)
-{
-  src->stats.base_seq = seq;
-  src->stats.max_seq = seq;
-  src->stats.bad_seq = RTP_SEQ_MOD + 1; /* so seq == bad_seq is false */
-  src->stats.cycles = 0;
-  src->stats.packets_received = 0;
-  src->stats.octets_received = 0;
-  src->stats.bytes_received = 0;
-  src->stats.prev_received = 0;
-  src->stats.prev_expected = 0;
-
-  GST_DEBUG ("base_seq %d", seq);
-}
-
-/**
- * rtp_source_process_rtp:
- * @src: an #RTPSource
- * @buffer: an RTP buffer
- *
- * Let @src handle the incomming RTP @buffer.
- *
- * Returns: a #GstFlowReturn.
- */
-GstFlowReturn
-rtp_source_process_rtp (RTPSource * src, GstBuffer * buffer,
-    RTPArrivalStats * arrival)
-{
-  GstFlowReturn result = GST_FLOW_OK;
-  guint16 seqnr, udelta;
-  RTPSourceStats *stats;
-
-  g_return_val_if_fail (RTP_IS_SOURCE (src), GST_FLOW_ERROR);
-  g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
-
-  stats = &src->stats;
-
-  seqnr = gst_rtp_buffer_get_seq (buffer);
-
-  rtp_source_update_caps (src, GST_BUFFER_CAPS (buffer));
-
-  if (stats->cycles == -1) {
-    GST_DEBUG ("received first buffer");
-    /* first time we heard of this source */
-    init_seq (src, seqnr);
-    src->stats.max_seq = seqnr - 1;
-    src->probation = RTP_DEFAULT_PROBATION;
-  }
-
-  udelta = seqnr - stats->max_seq;
-
-  /* if we are still on probation, check seqnum */
-  if (src->probation) {
-    guint16 expected;
-
-    expected = src->stats.max_seq + 1;
-
-    /* when in probation, we require consecutive seqnums */
-    if (seqnr == expected) {
-      /* expected packet */
-      GST_DEBUG ("probation: seqnr %d == expected %d", seqnr, expected);
-      src->probation--;
-      src->stats.max_seq = seqnr;
-      if (src->probation == 0) {
-        GST_DEBUG ("probation done!");
-        init_seq (src, seqnr);
-      } else {
-        GstBuffer *q;
-
-        GST_DEBUG ("probation %d: queue buffer", src->probation);
-        /* when still in probation, keep packets in a list. */
-        g_queue_push_tail (src->packets, buffer);
-        /* remove packets from queue if there are too many */
-        while (g_queue_get_length (src->packets) > RTP_MAX_PROBATION_LEN) {
-          q = g_queue_pop_head (src->packets);
-          gst_buffer_unref (q);
-        }
-        goto done;
-      }
-    } else {
-      GST_DEBUG ("probation: seqnr %d != expected %d", seqnr, expected);
-      src->probation = RTP_DEFAULT_PROBATION;
-      src->stats.max_seq = seqnr;
-      goto done;
-    }
-  } else if (udelta < RTP_MAX_DROPOUT) {
-    /* in order, with permissible gap */
-    if (seqnr < stats->max_seq) {
-      /* sequence number wrapped - count another 64K cycle. */
-      stats->cycles += RTP_SEQ_MOD;
-    }
-    stats->max_seq = seqnr;
-  } else if (udelta <= RTP_SEQ_MOD - RTP_MAX_MISORDER) {
-    /* the sequence number made a very large jump */
-    if (seqnr == stats->bad_seq) {
-      /* two sequential packets -- assume that the other side
-       * restarted without telling us so just re-sync
-       * (i.e., pretend this was the first packet).  */
-      init_seq (src, seqnr);
-    } else {
-      /* unacceptable jump */
-      stats->bad_seq = (seqnr + 1) & (RTP_SEQ_MOD - 1);
-      goto bad_sequence;
-    }
-  } else {
-    /* duplicate or reordered packet, will be filtered by jitterbuffer. */
-    GST_WARNING ("duplicate or reordered packet");
-  }
-
-  src->stats.octets_received += arrival->payload_len;
-  src->stats.bytes_received += arrival->bytes;
-  src->stats.packets_received++;
-  /* the source that sent the packet must be a sender */
-  src->is_sender = TRUE;
-  src->validated = TRUE;
-
-  GST_LOG ("seq %d, PC: %" G_GUINT64_FORMAT ", OC: %" G_GUINT64_FORMAT,
-      seqnr, src->stats.packets_received, src->stats.octets_received);
-
-  /* calculate jitter for the stats */
-  calculate_jitter (src, buffer, arrival);
-
-  /* we're ready to push the RTP packet now */
-  result = push_packet (src, buffer);
-
-done:
-  return result;
-
-  /* ERRORS */
-bad_sequence:
-  {
-    GST_WARNING ("unacceptable seqnum received");
-    gst_buffer_unref (buffer);
-    return GST_FLOW_OK;
-  }
-}
-
-/**
- * rtp_source_process_bye:
- * @src: an #RTPSource
- * @reason: the reason for leaving
- *
- * Notify @src that a BYE packet has been received. This will make the source
- * inactive.
- */
-void
-rtp_source_process_bye (RTPSource * src, const gchar * reason)
-{
-  g_return_if_fail (RTP_IS_SOURCE (src));
-
-  GST_DEBUG ("marking SSRC %08x as BYE, reason: %s", src->ssrc,
-      GST_STR_NULL (reason));
-
-  /* copy the reason and mark as received_bye */
-  g_free (src->bye_reason);
-  src->bye_reason = g_strdup (reason);
-  src->received_bye = TRUE;
-}
-
-static GstBufferListItem
-set_ssrc (GstBuffer ** buffer, guint group, guint idx, RTPSource * src)
-{
-  *buffer = gst_buffer_make_writable (*buffer);
-  gst_rtp_buffer_set_ssrc (*buffer, src->ssrc);
-  return GST_BUFFER_LIST_SKIP_GROUP;
-}
-
-/**
- * rtp_source_send_rtp:
- * @src: an #RTPSource
- * @data: an RTP buffer or a list of RTP buffers
- * @is_list: if @data is a buffer or list
- * @ntpnstime: the NTP time when this buffer was captured in nanoseconds. This
- * is the buffer timestamp converted to NTP time.
- *
- * Send @data (an RTP buffer or list of buffers) originating from @src.
- * This will make @src a sender. This function takes ownership of @data and
- * modifies the SSRC in the RTP packet to that of @src when needed.
- *
- * Returns: a #GstFlowReturn.
- */
-GstFlowReturn
-rtp_source_send_rtp (RTPSource * src, gpointer data, gboolean is_list,
-    guint64 ntpnstime)
-{
-  GstFlowReturn result;
-  guint len;
-  guint32 rtptime;
-  guint64 ext_rtptime;
-  guint64 ntp_diff, rtp_diff;
-  guint64 elapsed;
-  GstBufferList *list = NULL;
-  GstBuffer *buffer = NULL;
-  guint packets;
-  guint32 ssrc;
-
-  g_return_val_if_fail (RTP_IS_SOURCE (src), GST_FLOW_ERROR);
-  g_return_val_if_fail (is_list || GST_IS_BUFFER (data), GST_FLOW_ERROR);
-
-  if (is_list) {
-    list = GST_BUFFER_LIST_CAST (data);
-
-    /* We can grab the caps from the first group, since all
-     * groups of a buffer list have same caps. */
-    buffer = gst_buffer_list_get (list, 0, 0);
-    if (!buffer)
-      goto no_buffer;
-  } else {
-    buffer = GST_BUFFER_CAST (data);
-  }
-  rtp_source_update_caps (src, GST_BUFFER_CAPS (buffer));
-
-  /* we are a sender now */
-  src->is_sender = TRUE;
-
-  if (is_list) {
-    /* Each group makes up a network packet. */
-    packets = gst_buffer_list_n_groups (list);
-    len = gst_rtp_buffer_list_get_payload_len (list);
-  } else {
-    packets = 1;
-    len = gst_rtp_buffer_get_payload_len (buffer);
-  }
-
-  /* update stats for the SR */
-  src->stats.packets_sent += packets;
-  src->stats.octets_sent += len;
-  src->bytes_sent += len;
-
-  if (src->prev_ntpnstime) {
-    elapsed = ntpnstime - src->prev_ntpnstime;
-
-    if (elapsed > (G_GINT64_CONSTANT (1) << 31)) {
-      guint64 rate;
-
-      rate =
-          gst_util_uint64_scale (src->bytes_sent, elapsed,
-          (G_GINT64_CONSTANT (1) << 29));
-
-      GST_LOG ("Elapsed %" G_GUINT64_FORMAT ", bytes %" G_GUINT64_FORMAT
-          ", rate %" G_GUINT64_FORMAT, elapsed, src->bytes_sent, rate);
-
-      if (src->bitrate == 0)
-        src->bitrate = rate;
-      else
-        src->bitrate = ((src->bitrate * 3) + rate) / 4;
-
-      src->prev_ntpnstime = ntpnstime;
-      src->bytes_sent = 0;
-    }
-  } else {
-    GST_LOG ("Reset bitrate measurement");
-    src->prev_ntpnstime = ntpnstime;
-    src->bitrate = 0;
-  }
-
-  if (is_list) {
-    rtptime = gst_rtp_buffer_list_get_timestamp (list);
-  } else {
-    rtptime = gst_rtp_buffer_get_timestamp (buffer);
-  }
-  ext_rtptime = src->last_rtptime;
-  ext_rtptime = gst_rtp_buffer_ext_timestamp (&ext_rtptime, rtptime);
-
-  GST_LOG ("SSRC %08x, RTP %" G_GUINT64_FORMAT ", NTP %" GST_TIME_FORMAT,
-      src->ssrc, ext_rtptime, GST_TIME_ARGS (ntpnstime));
-
-  if (ext_rtptime > src->last_rtptime) {
-    rtp_diff = ext_rtptime - src->last_rtptime;
-    ntp_diff = ntpnstime - src->last_ntpnstime;
-
-    /* calc the diff so we can detect drift at the sender. This can also be used
-     * to guestimate the clock rate if the NTP time is locked to the RTP
-     * timestamps (as is the case when the capture device is providing the clock). */
-    GST_LOG ("SSRC %08x, diff RTP %" G_GUINT64_FORMAT ", diff NTP %"
-        GST_TIME_FORMAT, src->ssrc, rtp_diff, GST_TIME_ARGS (ntp_diff));
-  }
-
-  /* we keep track of the last received RTP timestamp and the corresponding
-   * NTP timestamp so that we can use this info when constructing SR reports */
-  src->last_rtptime = ext_rtptime;
-  src->last_ntpnstime = ntpnstime;
-
-  /* push packet */
-  if (!src->callbacks.push_rtp)
-    goto no_callback;
-
-  if (is_list) {
-    ssrc = gst_rtp_buffer_list_get_ssrc (list);
-  } else {
-    ssrc = gst_rtp_buffer_get_ssrc (buffer);
-  }
-
-  if (ssrc != src->ssrc) {
-    /* the SSRC of the packet is not correct, make a writable buffer and
-     * update the SSRC. This could involve a complete copy of the packet when
-     * it is not writable. Usually the payloader will use caps negotiation to
-     * get the correct SSRC from the session manager before pushing anything. */
-
-    /* FIXME, we don't want to warn yet because we can't inform any payloader
-     * of the changes SSRC yet because we don't implement pad-alloc. */
-    GST_LOG ("updating SSRC from %08x to %08x, fix the payloader", ssrc,
-        src->ssrc);
-
-    if (is_list) {
-      list = gst_buffer_list_make_writable (list);
-      gst_buffer_list_foreach (list, (GstBufferListFunc) set_ssrc, src);
-    } else {
-      set_ssrc (&buffer, 0, 0, src);
-    }
-  }
-  GST_LOG ("pushing RTP %s %" G_GUINT64_FORMAT, is_list ? "list" : "packet",
-      src->stats.packets_sent);
-
-  result = src->callbacks.push_rtp (src, data, src->user_data);
-
-  return result;
-
-  /* ERRORS */
-no_buffer:
-  {
-    GST_WARNING ("no buffers in buffer list");
-    gst_mini_object_unref (GST_MINI_OBJECT_CAST (data));
-    return GST_FLOW_OK;
-  }
-no_callback:
-  {
-    GST_WARNING ("no callback installed, dropping packet");
-    gst_mini_object_unref (GST_MINI_OBJECT_CAST (data));
-    return GST_FLOW_OK;
-  }
-}
-
-/**
- * rtp_source_process_sr:
- * @src: an #RTPSource
- * @time: time of packet arrival
- * @ntptime: the NTP time in 32.32 fixed point
- * @rtptime: the RTP time
- * @packet_count: the packet count
- * @octet_count: the octect count
- *
- * Update the sender report in @src.
- */
-void
-rtp_source_process_sr (RTPSource * src, GstClockTime time, guint64 ntptime,
-    guint32 rtptime, guint32 packet_count, guint32 octet_count)
-{
-  RTPSenderReport *curr;
-  gint curridx;
-
-  g_return_if_fail (RTP_IS_SOURCE (src));
-
-  GST_DEBUG ("got SR packet: SSRC %08x, NTP %08x:%08x, RTP %" G_GUINT32_FORMAT
-      ", PC %" G_GUINT32_FORMAT ", OC %" G_GUINT32_FORMAT, src->ssrc,
-      (guint32) (ntptime >> 32), (guint32) (ntptime & 0xffffffff), rtptime,
-      packet_count, octet_count);
-
-  curridx = src->stats.curr_sr ^ 1;
-  curr = &src->stats.sr[curridx];
-
-  /* this is a sender now */
-  src->is_sender = TRUE;
-
-  /* update current */
-  curr->is_valid = TRUE;
-  curr->ntptime = ntptime;
-  curr->rtptime = rtptime;
-  curr->packet_count = packet_count;
-  curr->octet_count = octet_count;
-  curr->time = time;
-
-  /* make current */
-  src->stats.curr_sr = curridx;
-}
-
-/**
- * rtp_source_process_rb:
- * @src: an #RTPSource
- * @time: the current time in nanoseconds since 1970
- * @fractionlost: fraction lost since last SR/RR
- * @packetslost: the cumululative number of packets lost
- * @exthighestseq: the extended last sequence number received
- * @jitter: the interarrival jitter
- * @lsr: the last SR packet from this source
- * @dlsr: the delay since last SR packet
- *
- * Update the report block in @src.
- */
-void
-rtp_source_process_rb (RTPSource * src, GstClockTime time, guint8 fractionlost,
-    gint32 packetslost, guint32 exthighestseq, guint32 jitter, guint32 lsr,
-    guint32 dlsr)
-{
-  RTPReceiverReport *curr;
-  gint curridx;
-  guint32 ntp, A;
-
-  g_return_if_fail (RTP_IS_SOURCE (src));
-
-  GST_DEBUG ("got RB packet: SSRC %08x, FL %2x, PL %d, HS %" G_GUINT32_FORMAT
-      ", jitter %" G_GUINT32_FORMAT ", LSR %04x:%04x, DLSR %04x:%04x",
-      src->ssrc, fractionlost, packetslost, exthighestseq, jitter, lsr >> 16,
-      lsr & 0xffff, dlsr >> 16, dlsr & 0xffff);
-
-  curridx = src->stats.curr_rr ^ 1;
-  curr = &src->stats.rr[curridx];
-
-  /* update current */
-  curr->is_valid = TRUE;
-  curr->fractionlost = fractionlost;
-  curr->packetslost = packetslost;
-  curr->exthighestseq = exthighestseq;
-  curr->jitter = jitter;
-  curr->lsr = lsr;
-  curr->dlsr = dlsr;
-
-  /* calculate round trip, round the time up */
-  ntp = ((gst_rtcp_unix_to_ntp (time) + 0xffff) >> 16) & 0xffffffff;
-  A = dlsr + lsr;
-  if (A > 0 && ntp > A)
-    A = ntp - A;
-  else
-    A = 0;
-  curr->round_trip = A;
-
-  GST_DEBUG ("NTP %04x:%04x, round trip %04x:%04x", ntp >> 16, ntp & 0xffff,
-      A >> 16, A & 0xffff);
-
-  /* make current */
-  src->stats.curr_rr = curridx;
-}
-
-/**
- * rtp_source_get_new_sr:
- * @src: an #RTPSource
- * @ntpnstime: the current time in nanoseconds since 1970
- * @ntptime: the NTP time in 32.32 fixed point
- * @rtptime: the RTP time corresponding to @ntptime
- * @packet_count: the packet count
- * @octet_count: the octect count
- *
- * Get new values to put into a new SR report from this source.
- *
- * Returns: %TRUE on success.
- */
-gboolean
-rtp_source_get_new_sr (RTPSource * src, guint64 ntpnstime,
-    guint64 * ntptime, guint32 * rtptime, guint32 * packet_count,
-    guint32 * octet_count)
-{
-  guint64 t_rtp;
-  guint64 t_current_ntp;
-  GstClockTimeDiff diff;
-
-  g_return_val_if_fail (RTP_IS_SOURCE (src), FALSE);
-
-  /* use the sync params to interpolate the date->time member to rtptime. We
-   * use the last sent timestamp and rtptime as reference points. We assume
-   * that the slope of the rtptime vs timestamp curve is 1, which is certainly
-   * sufficient for the frequency at which we report SR and the rate we send
-   * out RTP packets. */
-  t_rtp = src->last_rtptime;
-
-  GST_DEBUG ("last_ntpnstime %" GST_TIME_FORMAT ", last_rtptime %"
-      G_GUINT64_FORMAT, GST_TIME_ARGS (src->last_ntpnstime), t_rtp);
-
-  if (src->clock_rate != -1) {
-    /* get the diff with the SR time */
-    diff = GST_CLOCK_DIFF (src->last_ntpnstime, ntpnstime);
-
-    /* now translate the diff to RTP time, handle positive and negative cases.
-     * If there is no diff, we already set rtptime correctly above. */
-    if (diff > 0) {
-      GST_DEBUG ("ntpnstime %" GST_TIME_FORMAT ", diff %" GST_TIME_FORMAT,
-          GST_TIME_ARGS (ntpnstime), GST_TIME_ARGS (diff));
-      t_rtp += gst_util_uint64_scale_int (diff, src->clock_rate, GST_SECOND);
-    } else {
-      diff = -diff;
-      GST_DEBUG ("ntpnstime %" GST_TIME_FORMAT ", diff -%" GST_TIME_FORMAT,
-          GST_TIME_ARGS (ntpnstime), GST_TIME_ARGS (diff));
-      t_rtp -= gst_util_uint64_scale_int (diff, src->clock_rate, GST_SECOND);
-    }
-  } else {
-    GST_WARNING ("no clock-rate, cannot interpolate rtp time");
-  }
-
-  /* convert the NTP time in nanoseconds to 32.32 fixed point */
-  t_current_ntp = gst_util_uint64_scale (ntpnstime, (1LL << 32), GST_SECOND);
-
-  GST_DEBUG ("NTP %08x:%08x, RTP %" G_GUINT32_FORMAT,
-      (guint32) (t_current_ntp >> 32), (guint32) (t_current_ntp & 0xffffffff),
-      (guint32) t_rtp);
-
-  if (ntptime)
-    *ntptime = t_current_ntp;
-  if (rtptime)
-    *rtptime = t_rtp;
-  if (packet_count)
-    *packet_count = src->stats.packets_sent;
-  if (octet_count)
-    *octet_count = src->stats.octets_sent;
-
-  return TRUE;
-}
-
-/**
- * rtp_source_get_new_rb:
- * @src: an #RTPSource
- * @time: the current time of the system clock
- * @fractionlost: fraction lost since last SR/RR
- * @packetslost: the cumululative number of packets lost
- * @exthighestseq: the extended last sequence number received
- * @jitter: the interarrival jitter
- * @lsr: the last SR packet from this source
- * @dlsr: the delay since last SR packet
- *
- * Get new values to put into a new report block from this source.
- *
- * Returns: %TRUE on success.
- */
-gboolean
-rtp_source_get_new_rb (RTPSource * src, GstClockTime time,
-    guint8 * fractionlost, gint32 * packetslost, guint32 * exthighestseq,
-    guint32 * jitter, guint32 * lsr, guint32 * dlsr)
-{
-  RTPSourceStats *stats;
-  guint64 extended_max, expected;
-  guint64 expected_interval, received_interval, ntptime;
-  gint64 lost, lost_interval;
-  guint32 fraction, LSR, DLSR;
-  GstClockTime sr_time;
-
-  stats = &src->stats;
-
-  extended_max = stats->cycles + stats->max_seq;
-  expected = extended_max - stats->base_seq + 1;
-
-  GST_DEBUG ("ext_max %" G_GUINT64_FORMAT ", expected %" G_GUINT64_FORMAT
-      ", received %" G_GUINT64_FORMAT ", base_seq %" G_GUINT32_FORMAT,
-      extended_max, expected, stats->packets_received, stats->base_seq);
-
-  lost = expected - stats->packets_received;
-  lost = CLAMP (lost, -0x800000, 0x7fffff);
-
-  expected_interval = expected - stats->prev_expected;
-  stats->prev_expected = expected;
-  received_interval = stats->packets_received - stats->prev_received;
-  stats->prev_received = stats->packets_received;
-
-  lost_interval = expected_interval - received_interval;
-
-  if (expected_interval == 0 || lost_interval <= 0)
-    fraction = 0;
-  else
-    fraction = (lost_interval << 8) / expected_interval;
-
-  GST_DEBUG ("add RR for SSRC %08x", src->ssrc);
-  /* we scaled the jitter up for additional precision */
-  GST_DEBUG ("fraction %" G_GUINT32_FORMAT ", lost %" G_GINT64_FORMAT
-      ", extseq %" G_GUINT64_FORMAT ", jitter %d", fraction, lost,
-      extended_max, stats->jitter >> 4);
-
-  if (rtp_source_get_last_sr (src, &sr_time, &ntptime, NULL, NULL, NULL)) {
-    GstClockTime diff;
-
-    /* LSR is middle 32 bits of the last ntptime */
-    LSR = (ntptime >> 16) & 0xffffffff;
-    diff = time - sr_time;
-    GST_DEBUG ("last SR time diff %" GST_TIME_FORMAT, GST_TIME_ARGS (diff));
-    /* DLSR, delay since last SR is expressed in 1/65536 second units */
-    DLSR = gst_util_uint64_scale_int (diff, 65536, GST_SECOND);
-  } else {
-    /* No valid SR received, LSR/DLSR are set to 0 then */
-    GST_DEBUG ("no valid SR received");
-    LSR = 0;
-    DLSR = 0;
-  }
-  GST_DEBUG ("LSR %04x:%04x, DLSR %04x:%04x", LSR >> 16, LSR & 0xffff,
-      DLSR >> 16, DLSR & 0xffff);
-
-  if (fractionlost)
-    *fractionlost = fraction;
-  if (packetslost)
-    *packetslost = lost;
-  if (exthighestseq)
-    *exthighestseq = extended_max;
-  if (jitter)
-    *jitter = stats->jitter >> 4;
-  if (lsr)
-    *lsr = LSR;
-  if (dlsr)
-    *dlsr = DLSR;
-
-  return TRUE;
-}
-
-/**
- * rtp_source_get_last_sr:
- * @src: an #RTPSource
- * @time: time of packet arrival
- * @ntptime: the NTP time in 32.32 fixed point
- * @rtptime: the RTP time
- * @packet_count: the packet count
- * @octet_count: the octect count
- *
- * Get the values of the last sender report as set with rtp_source_process_sr().
- *
- * Returns: %TRUE if there was a valid SR report.
- */
-gboolean
-rtp_source_get_last_sr (RTPSource * src, GstClockTime * time, guint64 * ntptime,
-    guint32 * rtptime, guint32 * packet_count, guint32 * octet_count)
-{
-  RTPSenderReport *curr;
-
-  g_return_val_if_fail (RTP_IS_SOURCE (src), FALSE);
-
-  curr = &src->stats.sr[src->stats.curr_sr];
-  if (!curr->is_valid)
-    return FALSE;
-
-  if (ntptime)
-    *ntptime = curr->ntptime;
-  if (rtptime)
-    *rtptime = curr->rtptime;
-  if (packet_count)
-    *packet_count = curr->packet_count;
-  if (octet_count)
-    *octet_count = curr->octet_count;
-  if (time)
-    *time = curr->time;
-
-  return TRUE;
-}
-
-/**
- * rtp_source_get_last_rb:
- * @src: an #RTPSource
- * @fractionlost: fraction lost since last SR/RR
- * @packetslost: the cumululative number of packets lost
- * @exthighestseq: the extended last sequence number received
- * @jitter: the interarrival jitter
- * @lsr: the last SR packet from this source
- * @dlsr: the delay since last SR packet
- * @round_trip: the round trip time
- *
- * Get the values of the last RB report set with rtp_source_process_rb().
- *
- * Returns: %TRUE if there was a valid SB report.
- */
-gboolean
-rtp_source_get_last_rb (RTPSource * src, guint8 * fractionlost,
-    gint32 * packetslost, guint32 * exthighestseq, guint32 * jitter,
-    guint32 * lsr, guint32 * dlsr, guint32 * round_trip)
-{
-  RTPReceiverReport *curr;
-
-  g_return_val_if_fail (RTP_IS_SOURCE (src), FALSE);
-
-  curr = &src->stats.rr[src->stats.curr_rr];
-  if (!curr->is_valid)
-    return FALSE;
-
-  if (fractionlost)
-    *fractionlost = curr->fractionlost;
-  if (packetslost)
-    *packetslost = curr->packetslost;
-  if (exthighestseq)
-    *exthighestseq = curr->exthighestseq;
-  if (jitter)
-    *jitter = curr->jitter;
-  if (lsr)
-    *lsr = curr->lsr;
-  if (dlsr)
-    *dlsr = curr->dlsr;
-  if (round_trip)
-    *round_trip = curr->round_trip;
-
-  return TRUE;
-}
diff --git a/gst/rtpmanager/rtpsource.h b/gst/rtpmanager/rtpsource.h
deleted file mode 100644
index 8355bc0c..00000000
--- a/gst/rtpmanager/rtpsource.h
+++ /dev/null
@@ -1,226 +0,0 @@
-/* GStreamer
- * Copyright (C) <2007> Wim Taymans <wim.taymans@gmail.com>
- *
- * This library is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Library General Public
- * License as published by the Free Software Foundation; either
- * version 2 of the License, or (at your option) any later version.
- *
- * This library is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Library General Public License for more details.
- *
- * You should have received a copy of the GNU Library General Public
- * License along with this library; if not, write to the
- * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
- * Boston, MA 02111-1307, USA.
- */
-
-#ifndef __RTP_SOURCE_H__
-#define __RTP_SOURCE_H__
-
-#include <gst/gst.h>
-#include <gst/rtp/gstrtcpbuffer.h>
-#include <gst/netbuffer/gstnetbuffer.h>
-
-#include "rtpstats.h"
-
-/* the default number of consecutive RTP packets we need to receive before the
- * source is considered valid */
-#define RTP_NO_PROBATION        0
-#define RTP_DEFAULT_PROBATION   2
-
-#define RTP_SEQ_MOD          (1 << 16)
-
-typedef struct _RTPSource RTPSource;
-typedef struct _RTPSourceClass RTPSourceClass;
-
-#define RTP_TYPE_SOURCE             (rtp_source_get_type())
-#define RTP_SOURCE(src)             (G_TYPE_CHECK_INSTANCE_CAST((src),RTP_TYPE_SOURCE,RTPSource))
-#define RTP_SOURCE_CLASS(klass)     (G_TYPE_CHECK_CLASS_CAST((klass),RTP_TYPE_SOURCE,RTPSourceClass))
-#define RTP_IS_SOURCE(src)          (G_TYPE_CHECK_INSTANCE_TYPE((src),RTP_TYPE_SOURCE))
-#define RTP_IS_SOURCE_CLASS(klass)  (G_TYPE_CHECK_CLASS_TYPE((klass),RTP_TYPE_SOURCE))
-#define RTP_SOURCE_CAST(src)        ((RTPSource *)(src))
-
-/**
- * RTP_SOURCE_IS_ACTIVE:
- * @src: an #RTPSource
- *
- * Check if @src is active. A source is active when it has been validated
- * and has not yet received a BYE packet.
- */
-#define RTP_SOURCE_IS_ACTIVE(src)  (src->validated && !src->received_bye)
-
-/**
- * RTP_SOURCE_IS_SENDER:
- * @src: an #RTPSource
- *
- * Check if @src is a sender.
- */
-#define RTP_SOURCE_IS_SENDER(src)  (src->is_sender)
-
-/**
- * RTPSourcePushRTP:
- * @src: an #RTPSource
- * @buffer: the RTP buffer ready for processing
- * @user_data: user data specified when registering
- *
- * This callback will be called when @src has @buffer ready for further
- * processing.
- *
- * Returns: a #GstFlowReturn.
- */
-typedef GstFlowReturn (*RTPSourcePushRTP) (RTPSource *src, GstBuffer *buffer, 
-	gpointer user_data);
-
-/**
- * RTPSourceClockRate:
- * @src: an #RTPSource
- * @payload: a payload type
- * @user_data: user data specified when registering
- *
- * This callback will be called when @src needs the clock-rate of the
- * @payload.
- *
- * Returns: a clock-rate for @payload.
- */
-typedef gint (*RTPSourceClockRate) (RTPSource *src, guint8 payload, gpointer user_data);
-
-/**
- * RTPSourceCallbacks:
- * @push_rtp: a packet becomes available for handling
- * @clock_rate: a clock-rate is requested
- * @get_time: the current clock time is requested
- *
- * Callbacks performed by #RTPSource when actions need to be performed.
- */
-typedef struct {
-  RTPSourcePushRTP     push_rtp;
-  RTPSourceClockRate   clock_rate;
-} RTPSourceCallbacks;
-
-/**
- * RTPSource:
- *
- * A source in the #RTPSession
- */
-struct _RTPSource {
-  GObject       object;
-
-  /*< private >*/
-  guint32       ssrc;
-
-  gint          probation;
-  gboolean      validated;
-  gboolean      internal;
-  gboolean      is_csrc;
-  gboolean      is_sender;
-
-  guint8       *sdes[9];
-  guint         sdes_len[9];
-
-  gboolean      received_bye;
-  gchar        *bye_reason;
-
-  gboolean      have_rtp_from;
-  GstNetAddress rtp_from;
-  gboolean      have_rtcp_from;
-  GstNetAddress rtcp_from;
-
-  gint          payload;
-  GstCaps      *caps;
-  gint          clock_rate;
-  gint32        seqnum_base;
-
-  GstClockTime  bye_time;
-  GstClockTime  last_activity;
-  GstClockTime  last_rtp_activity;
-
-  GstClockTime  last_rtptime;
-  GstClockTime  last_ntpnstime;
-
-  /* for bitrate estimation */
-  guint64       bitrate;
-  GstClockTime  prev_ntpnstime;
-  guint64       bytes_sent;
-
-  GQueue       *packets;
-
-  RTPSourceCallbacks callbacks;
-  gpointer           user_data;
-
-  RTPSourceStats stats;
-};
-
-struct _RTPSourceClass {
-  GObjectClass   parent_class;
-};
-
-GType rtp_source_get_type (void);
-
-/* managing lifetime of sources */
-RTPSource*      rtp_source_new                 (guint32 ssrc);
-void            rtp_source_set_callbacks       (RTPSource *src, RTPSourceCallbacks *cb, gpointer data);
-
-/* properties */
-guint32         rtp_source_get_ssrc            (RTPSource *src);
-
-void            rtp_source_set_as_csrc         (RTPSource *src);
-gboolean        rtp_source_is_as_csrc          (RTPSource *src);
-
-gboolean        rtp_source_is_active           (RTPSource *src);
-gboolean        rtp_source_is_validated        (RTPSource *src);
-gboolean        rtp_source_is_sender           (RTPSource *src);
-
-gboolean        rtp_source_received_bye        (RTPSource *src);
-gchar *         rtp_source_get_bye_reason      (RTPSource *src);
-
-void            rtp_source_update_caps         (RTPSource *src, GstCaps *caps);
-
-/* SDES info */
-gboolean        rtp_source_set_sdes            (RTPSource *src, GstRTCPSDESType type, 
-                                                const guint8 *data, guint len);
-gboolean        rtp_source_set_sdes_string     (RTPSource *src, GstRTCPSDESType type,
-                                                const gchar *data);
-gboolean        rtp_source_get_sdes            (RTPSource *src, GstRTCPSDESType type,
-                                                guint8 **data, guint *len);
-gchar*          rtp_source_get_sdes_string     (RTPSource *src, GstRTCPSDESType type);
-
-GstStructure *  rtp_source_get_sdes_struct     (RTPSource * src);
-void            rtp_source_set_sdes_struct     (RTPSource * src, const GstStructure *sdes);
-
-/* handling network address */
-void            rtp_source_set_rtp_from        (RTPSource *src, GstNetAddress *address);
-void            rtp_source_set_rtcp_from       (RTPSource *src, GstNetAddress *address);
-
-/* handling RTP */
-GstFlowReturn   rtp_source_process_rtp         (RTPSource *src, GstBuffer *buffer, RTPArrivalStats *arrival);
-
-GstFlowReturn   rtp_source_send_rtp            (RTPSource *src, gpointer data, gboolean is_list, guint64 ntpnstime);
-
-/* RTCP messages */
-void            rtp_source_process_bye         (RTPSource *src, const gchar *reason);
-void            rtp_source_process_sr          (RTPSource *src, GstClockTime time, guint64 ntptime,
-                                                guint32 rtptime, guint32 packet_count, guint32 octet_count);
-void            rtp_source_process_rb          (RTPSource *src, GstClockTime time, guint8 fractionlost,
-                                                gint32 packetslost, guint32 exthighestseq, guint32 jitter,
-                                                guint32 lsr, guint32 dlsr);
-
-gboolean        rtp_source_get_new_sr          (RTPSource *src, guint64 ntpnstime, guint64 *ntptime,
-		                                guint32 *rtptime, guint32 *packet_count,
-						guint32 *octet_count);
-gboolean        rtp_source_get_new_rb          (RTPSource *src, GstClockTime time, guint8 *fractionlost,
-                                                gint32 *packetslost, guint32 *exthighestseq, guint32 *jitter,
-                                                guint32 *lsr, guint32 *dlsr);
-
-gboolean        rtp_source_get_last_sr         (RTPSource *src, GstClockTime *time, guint64 *ntptime,
-                                                guint32 *rtptime, guint32 *packet_count,
-						guint32 *octet_count);
-gboolean        rtp_source_get_last_rb         (RTPSource *src, guint8 *fractionlost, gint32 *packetslost,
-                                                guint32 *exthighestseq, guint32 *jitter,
-                                                guint32 *lsr, guint32 *dlsr, guint32 *round_trip);
-
-void            rtp_source_reset               (RTPSource * src);
-
-#endif /* __RTP_SOURCE_H__ */
diff --git a/gst/rtpmanager/rtpstats.c b/gst/rtpmanager/rtpstats.c
deleted file mode 100644
index 640c3194..00000000
--- a/gst/rtpmanager/rtpstats.c
+++ /dev/null
@@ -1,176 +0,0 @@
-/* GStreamer
- * Copyright (C) <2007> Wim Taymans <wim.taymans@gmail.com>
- *
- * This library is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Library General Public
- * License as published by the Free Software Foundation; either
- * version 2 of the License, or (at your option) any later version.
- *
- * This library is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Library General Public License for more details.
- *
- * You should have received a copy of the GNU Library General Public
- * License along with this library; if not, write to the
- * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
- * Boston, MA 02111-1307, USA.
- */
-
-#include "rtpstats.h"
-
-/**
- * rtp_stats_init_defaults:
- * @stats: an #RTPSessionStats struct
- *
- * Initialize @stats with its default values.
- */
-void
-rtp_stats_init_defaults (RTPSessionStats * stats)
-{
-  stats->bandwidth = RTP_STATS_BANDWIDTH;
-  stats->sender_fraction = RTP_STATS_SENDER_FRACTION;
-  stats->receiver_fraction = RTP_STATS_RECEIVER_FRACTION;
-  stats->rtcp_bandwidth = RTP_STATS_RTCP_BANDWIDTH;
-  stats->min_interval = RTP_STATS_MIN_INTERVAL;
-  stats->bye_timeout = RTP_STATS_BYE_TIMEOUT;
-}
-
-/**
- * rtp_stats_calculate_rtcp_interval:
- * @stats: an #RTPSessionStats struct
- * @sender: if we are a sender
- * @first: if this is the first time
- * 
- * Calculate the RTCP interval. The result of this function is the amount of
- * time to wait (in nanoseconds) before sending a new RTCP message.
- *
- * Returns: the RTCP interval.
- */
-GstClockTime
-rtp_stats_calculate_rtcp_interval (RTPSessionStats * stats, gboolean we_send,
-    gboolean first)
-{
-  gdouble members, senders, n;
-  gdouble avg_rtcp_size, rtcp_bw;
-  gdouble interval;
-  gdouble rtcp_min_time;
-
-  /* Very first call at application start-up uses half the min
-   * delay for quicker notification while still allowing some time
-   * before reporting for randomization and to learn about other
-   * sources so the report interval will converge to the correct
-   * interval more quickly.
-   */
-  rtcp_min_time = stats->min_interval;
-  if (first)
-    rtcp_min_time /= 2.0;
-
-  /* Dedicate a fraction of the RTCP bandwidth to senders unless
-   * the number of senders is large enough that their share is
-   * more than that fraction.
-   */
-  n = members = stats->active_sources;
-  senders = (gdouble) stats->sender_sources;
-  rtcp_bw = stats->rtcp_bandwidth;
-
-  if (senders <= members * RTP_STATS_SENDER_FRACTION) {
-    if (we_send) {
-      rtcp_bw *= RTP_STATS_SENDER_FRACTION;
-      n = senders;
-    } else {
-      rtcp_bw *= RTP_STATS_RECEIVER_FRACTION;
-      n -= senders;
-    }
-  }
-
-  avg_rtcp_size = stats->avg_rtcp_packet_size / 16.0;
-  /*
-   * The effective number of sites times the average packet size is
-   * the total number of octets sent when each site sends a report.
-   * Dividing this by the effective bandwidth gives the time
-   * interval over which those packets must be sent in order to
-   * meet the bandwidth target, with a minimum enforced.  In that
-   * time interval we send one report so this time is also our
-   * average time between reports.
-   */
-  interval = avg_rtcp_size * n / rtcp_bw;
-  if (interval < rtcp_min_time)
-    interval = rtcp_min_time;
-
-  return interval * GST_SECOND;
-}
-
-/**
- * rtp_stats_add_rtcp_jitter:
- * @stats: an #RTPSessionStats struct
- * @interval: an RTCP interval
- * 
- * Apply a random jitter to the @interval. @interval is typically obtained with
- * rtp_stats_calculate_rtcp_interval().
- *
- * Returns: the new RTCP interval.
- */
-GstClockTime
-rtp_stats_add_rtcp_jitter (RTPSessionStats * stats, GstClockTime interval)
-{
-  gdouble temp;
-
-  /* see RFC 3550 p 30 
-   * To compensate for "unconditional reconsideration" converging to a
-   * value below the intended average.
-   */
-#define COMPENSATION  (2.71828 - 1.5);
-
-  temp = (interval * g_random_double_range (0.5, 1.5)) / COMPENSATION;
-
-  return (GstClockTime) temp;
-}
-
-
-/**
- * rtp_stats_calculate_bye_interval:
- * @stats: an #RTPSessionStats struct
- * 
- * Calculate the BYE interval. The result of this function is the amount of
- * time to wait (in nanoseconds) before sending a BYE message.
- *
- * Returns: the BYE interval.
- */
-GstClockTime
-rtp_stats_calculate_bye_interval (RTPSessionStats * stats)
-{
-  gdouble members;
-  gdouble avg_rtcp_size, rtcp_bw;
-  gdouble interval;
-  gdouble rtcp_min_time;
-
-  /* no interval when we have less than 50 members */
-  if (stats->active_sources < 50)
-    return 0;
-  
-  rtcp_min_time = (stats->min_interval) / 2.0;
-
-  /* Dedicate a fraction of the RTCP bandwidth to senders unless
-   * the number of senders is large enough that their share is
-   * more than that fraction.
-   */
-  members = stats->bye_members;
-  rtcp_bw = stats->rtcp_bandwidth * RTP_STATS_RECEIVER_FRACTION;
-
-  avg_rtcp_size = stats->avg_rtcp_packet_size / 16.0;
-  /*
-   * The effective number of sites times the average packet size is
-   * the total number of octets sent when each site sends a report.
-   * Dividing this by the effective bandwidth gives the time
-   * interval over which those packets must be sent in order to
-   * meet the bandwidth target, with a minimum enforced.  In that
-   * time interval we send one report so this time is also our
-   * average time between reports.
-   */
-  interval = avg_rtcp_size * members / rtcp_bw;
-  if (interval < rtcp_min_time)
-    interval = rtcp_min_time;
-
-  return interval * GST_SECOND;
-}
diff --git a/gst/rtpmanager/rtpstats.h b/gst/rtpmanager/rtpstats.h
deleted file mode 100644
index e5824315..00000000
--- a/gst/rtpmanager/rtpstats.h
+++ /dev/null
@@ -1,195 +0,0 @@
-/* GStreamer
- * Copyright (C) <2007> Wim Taymans <wim.taymans@gmail.com>
- *
- * This library is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Library General Public
- * License as published by the Free Software Foundation; either
- * version 2 of the License, or (at your option) any later version.
- *
- * This library is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Library General Public License for more details.
- *
- * You should have received a copy of the GNU Library General Public
- * License along with this library; if not, write to the
- * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
- * Boston, MA 02111-1307, USA.
- */
-
-#ifndef __RTP_STATS_H__
-#define __RTP_STATS_H__
-
-#include <gst/gst.h>
-#include <gst/netbuffer/gstnetbuffer.h>
-
-/**
- * RTPSenderReport:
- *
- * A sender report structure.
- */
-typedef struct {
-  gboolean is_valid;
-  guint64 ntptime;
-  guint32 rtptime;
-  guint32 packet_count;
-  guint32 octet_count;
-  GstClockTime time;
-} RTPSenderReport;
-
-/**
- * RTPReceiverReport:
- *
- * A receiver report structure.
- */
-typedef struct {
-  gboolean is_valid;
-  guint32 ssrc; /* who the report is from */
-  guint8  fractionlost;
-  guint32 packetslost;
-  guint32 exthighestseq;
-  guint32 jitter;
-  guint32 lsr;
-  guint32 dlsr;
-  guint32 round_trip;
-} RTPReceiverReport;
-
-/**
- * RTPArrivalStats:
- * @time: arrival time of a packet according to the system clock
- * @running_time: arrival time of a packet as buffer running_time
- * @ntpnstime: arrival time of a packet as NTP time in nanoseconds
- * @have_address: if the @address field contains a valid address
- * @address: address of the sender of the packet
- * @bytes: bytes of the packet including lowlevel overhead
- * @payload_len: bytes of the RTP payload
- *
- * Structure holding information about the arrival stats of a packet.
- */
-typedef struct {
-  GstClockTime  time;
-  GstClockTime  running_time;
-  guint64       ntpnstime;
-  gboolean      have_address;
-  GstNetAddress address;
-  guint         bytes;
-  guint         payload_len;
-} RTPArrivalStats;
-
-/**
- * RTPSourceStats:
- * @packetsreceived: number of received packets in total
- * @prevpacketsreceived: number of packets received in previous reporting
- *                       interval
- * @octetsreceived: number of payload bytes received
- * @bytesreceived: number of total bytes received including headers and lower
- *                 protocol level overhead
- * @max_seqnr: highest sequence number received
- * @transit: previous transit time used for calculating @jitter
- * @jitter: current jitter
- * @prev_rtptime: previous time when an RTP packet was received
- * @prev_rtcptime: previous time when an RTCP packet was received
- * @last_rtptime: time when last RTP packet received
- * @last_rtcptime: time when last RTCP packet received
- * @curr_rr: index of current @rr block
- * @rr: previous and current receiver report block
- * @curr_sr: index of current @sr block
- * @sr: previous and current sender report block
- *
- * Stats about a source.
- */
-typedef struct {
-  guint64      packets_received;
-  guint64      octets_received;
-  guint64      bytes_received;
-
-  guint32      prev_expected;
-  guint32      prev_received;
-
-  guint16      max_seq;
-  guint64      cycles;
-  guint32      base_seq;
-  guint32      bad_seq;
-  guint32      transit;
-  guint32      jitter;
-
-  guint64      packets_sent;
-  guint64      octets_sent;
-
-  /* when we received stuff */
-  GstClockTime prev_rtptime;
-  GstClockTime prev_rtcptime;
-  GstClockTime last_rtptime;
-  GstClockTime last_rtcptime;
-
-  /* sender and receiver reports */
-  gint              curr_rr;
-  RTPReceiverReport rr[2];
-  gint              curr_sr;
-  RTPSenderReport   sr[2];
-} RTPSourceStats;
-
-#define RTP_STATS_BANDWIDTH           64000.0
-#define RTP_STATS_RTCP_BANDWIDTH      3000.0
-/*
- * Minimum average time between RTCP packets from this site (in
- * seconds).  This time prevents the reports from `clumping' when
- * sessions are small and the law of large numbers isn't helping
- * to smooth out the traffic.  It also keeps the report interval
- * from becoming ridiculously small during transient outages like
- * a network partition.
- */
-#define RTP_STATS_MIN_INTERVAL      5.0
-/*
- * Fraction of the RTCP bandwidth to be shared among active
- * senders.  (This fraction was chosen so that in a typical
- * session with one or two active senders, the computed report
- * time would be roughly equal to the minimum report time so that
- * we don't unnecessarily slow down receiver reports.) The
- * receiver fraction must be 1 - the sender fraction.
- */
-#define RTP_STATS_SENDER_FRACTION       (0.25)
-#define RTP_STATS_RECEIVER_FRACTION     (1.0 - RTP_STATS_SENDER_FRACTION)
-
-/*
- * When receiving a BYE from a source, remove the source from the database
- * after this timeout.
- */
-#define RTP_STATS_BYE_TIMEOUT           (2 * GST_SECOND)
-
-/*
- * The maximum number of missing packets we tollerate. These are packets with a
- * sequence number bigger than the last seen packet.
- */
-#define RTP_MAX_DROPOUT      3000
-/*
- * The maximum number of misordered packets we tollerate. These are packets with
- * a sequence number smaller than the last seen packet.
- */
-#define RTP_MAX_MISORDER     100
-
-/**
- * RTPSessionStats:
- *
- * Stats kept for a session and used to produce RTCP packet timeouts.
- */
-typedef struct {
-  gdouble       bandwidth;
-  gdouble       sender_fraction;
-  gdouble       receiver_fraction;
-  gdouble       rtcp_bandwidth;
-  gdouble       min_interval;
-  GstClockTime  bye_timeout;
-  guint         sender_sources;
-  guint         active_sources;
-  guint         avg_rtcp_packet_size;
-  guint         bye_members;
-} RTPSessionStats;
-
-void           rtp_stats_init_defaults               (RTPSessionStats *stats);
-
-GstClockTime   rtp_stats_calculate_rtcp_interval    (RTPSessionStats *stats, gboolean sender, gboolean first);
-GstClockTime   rtp_stats_add_rtcp_jitter            (RTPSessionStats *stats, GstClockTime interval);
-GstClockTime   rtp_stats_calculate_bye_interval     (RTPSessionStats *stats);
-
-#endif /* __RTP_STATS_H__ */
diff --git a/tests/check/Makefile.am b/tests/check/Makefile.am
index 21e291a1..3a2d92ff 100644
--- a/tests/check/Makefile.am
+++ b/tests/check/Makefile.am
@@ -99,8 +99,6 @@ check_PROGRAMS = \
 	elements/asfmux \
 	elements/legacyresample \
 	elements/qtmux \
-	elements/rtpbin \
-	elements/rtpbin_buffer_list \
 	elements/selector \
 	elements/shapewipe \
 	elements/mxfdemux \
@@ -125,13 +123,6 @@ elements_camerabin_LDADD = \
          -lgstinterfaces-@GST_MAJORMINOR@
 elements_camerabin_SOURCES = elements/camerabin.c
 
-elements_rtpbin_buffer_list_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(GST_CFLAGS) \
-	$(ERROR_CFLAGS) $(GST_CHECK_CFLAGS)
-elements_rtpbin_buffer_list_LDADD = $(GST_PLUGINS_BASE_LIBS) \
-             -lgstnetbuffer-@GST_MAJORMINOR@ -lgstrtp-@GST_MAJORMINOR@ \
-             $(GST_BASE_LIBS) $(GST_LIBS_LIBS) $(GST_CHECK_LIBS)
-elements_rtpbin_buffer_list_SOURCES = elements/rtpbin_buffer_list.c
-
 elements_timidity_CFLAGS = $(GST_BASE_CFLAGS) $(AM_CFLAGS)
 elements_timidity_LDADD = $(GST_BASE_LIBS) $(LDADD)
 
diff --git a/tests/check/elements/.gitignore b/tests/check/elements/.gitignore
index e889dfb3..30c67fe8 100644
--- a/tests/check/elements/.gitignore
+++ b/tests/check/elements/.gitignore
@@ -20,8 +20,6 @@ souphttpsrc
 rganalysis
 rglimiter
 rgvolume
-rtpbin
-rtpbin_buffer_list
 selector
 shapewipe
 spectrum
diff --git a/tests/check/elements/rtpbin.c b/tests/check/elements/rtpbin.c
deleted file mode 100644
index 8764da54..00000000
--- a/tests/check/elements/rtpbin.c
+++ /dev/null
@@ -1,421 +0,0 @@
-/* GStreamer
- *
- * unit test for gstrtpbin
- *
- * Copyright (C) <2009> Wim Taymans <wim.taymans@gmail.com>
- *
- * This library is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Library General Public
- * License as published by the Free Software Foundation; either
- * version 2 of the License, or (at your option) any later version.
- *
- * This library is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Library General Public License for more details.
- *
- * You should have received a copy of the GNU Library General Public
- * License along with this library; if not, write to the
- * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
- * Boston, MA 02111-1307, USA.
- */
-
-#include <gst/check/gstcheck.h>
-
-GST_START_TEST (test_cleanup_send)
-{
-  GstElement *rtpbin;
-  GstPad *rtp_sink, *rtp_src, *rtcp_src;
-  GObject *session;
-  gint count = 2;
-
-  rtpbin = gst_element_factory_make ("gstrtpbin", "rtpbin");
-
-  while (count--) {
-    /* request session 0 */
-    rtp_sink = gst_element_get_request_pad (rtpbin, "send_rtp_sink_0");
-    fail_unless (rtp_sink != NULL);
-    ASSERT_OBJECT_REFCOUNT (rtp_sink, "rtp_sink", 2);
-
-    /* request again */
-    rtp_sink = gst_element_get_request_pad (rtpbin, "send_rtp_sink_0");
-    fail_unless (rtp_sink != NULL);
-    ASSERT_OBJECT_REFCOUNT (rtp_sink, "rtp_sink", 3);
-    gst_object_unref (rtp_sink);
-
-    /* this static pad should be created automatically now */
-    rtp_src = gst_element_get_static_pad (rtpbin, "send_rtp_src_0");
-    fail_unless (rtp_src != NULL);
-    ASSERT_OBJECT_REFCOUNT (rtp_src, "rtp_src", 2);
-
-    /* we should be able to get an internal session 0 now */
-    g_signal_emit_by_name (rtpbin, "get-internal-session", 0, &session);
-    fail_unless (session != NULL);
-    g_object_unref (session);
-
-    /* get the send RTCP pad too */
-    rtcp_src = gst_element_get_request_pad (rtpbin, "send_rtcp_src_0");
-    fail_unless (rtcp_src != NULL);
-    ASSERT_OBJECT_REFCOUNT (rtcp_src, "rtcp_src", 2);
-
-    /* second time */
-    rtcp_src = gst_element_get_request_pad (rtpbin, "send_rtcp_src_0");
-    fail_unless (rtcp_src != NULL);
-    ASSERT_OBJECT_REFCOUNT (rtcp_src, "rtcp_src", 3);
-    gst_object_unref (rtcp_src);
-
-    gst_element_release_request_pad (rtpbin, rtp_sink);
-    /* we should only have our refs to the pads now */
-    ASSERT_OBJECT_REFCOUNT (rtp_sink, "rtp_sink", 1);
-    ASSERT_OBJECT_REFCOUNT (rtp_src, "rtp_src", 1);
-    ASSERT_OBJECT_REFCOUNT (rtcp_src, "rtp_src", 2);
-
-    /* the other pad should be gone now */
-    fail_unless (gst_element_get_static_pad (rtpbin, "send_rtp_src_0") == NULL);
-
-    /* internal session should still be there */
-    g_signal_emit_by_name (rtpbin, "get-internal-session", 0, &session);
-    fail_unless (session != NULL);
-    g_object_unref (session);
-
-    /* release the RTCP pad */
-    gst_element_release_request_pad (rtpbin, rtcp_src);
-    /* we should only have our refs to the pads now */
-    ASSERT_OBJECT_REFCOUNT (rtp_sink, "rtp_sink", 1);
-    ASSERT_OBJECT_REFCOUNT (rtp_src, "rtp_src", 1);
-    ASSERT_OBJECT_REFCOUNT (rtcp_src, "rtp_src", 1);
-
-    /* the session should be gone now */
-    g_signal_emit_by_name (rtpbin, "get-internal-session", 0, &session);
-    fail_unless (session == NULL);
-
-    /* unref the request pad and the static pad */
-    gst_object_unref (rtp_sink);
-    gst_object_unref (rtp_src);
-    gst_object_unref (rtcp_src);
-  }
-
-  gst_object_unref (rtpbin);
-}
-
-GST_END_TEST;
-
-typedef struct
-{
-  guint16 seqnum;
-  gboolean pad_added;
-  GstPad *pad;
-  GMutex *lock;
-  GCond *cond;
-  GstPad *sinkpad;
-  GList *pads;
-} CleanupData;
-
-static void
-init_data (CleanupData * data)
-{
-  data->seqnum = 10;
-  data->pad_added = FALSE;
-  data->lock = g_mutex_new ();
-  data->cond = g_cond_new ();
-  data->pads = NULL;
-}
-
-static void
-clean_data (CleanupData * data)
-{
-  g_list_foreach (data->pads, (GFunc) gst_object_unref, NULL);
-  g_list_free (data->pads);
-  g_mutex_free (data->lock);
-  g_cond_free (data->cond);
-}
-
-static guint8 rtp_packet[] = { 0x80, 0x60, 0x94, 0xbc, 0x8f, 0x37, 0x4e, 0xb8,
-  0x44, 0xa8, 0xf3, 0x7c, 0x06, 0x6a, 0x0c, 0xce,
-  0x13, 0x25, 0x19, 0x69, 0x1f, 0x93, 0x25, 0x9d,
-  0x2b, 0x82, 0x31, 0x3b, 0x36, 0xc1, 0x3c, 0x13
-};
-
-static GstBuffer *
-make_rtp_packet (CleanupData * data)
-{
-  static GstCaps *caps = NULL;
-  GstBuffer *result;
-  guint8 *datap;
-
-  if (caps == NULL) {
-    caps = gst_caps_from_string ("application/x-rtp,"
-        "media=(string)audio, clock-rate=(int)44100, "
-        "encoding-name=(string)L16, encoding-params=(string)1, channels=(int)1");
-    data->seqnum = 0;
-  }
-
-  result = gst_buffer_new_and_alloc (sizeof (rtp_packet));
-  datap = GST_BUFFER_DATA (result);
-  memcpy (datap, rtp_packet, sizeof (rtp_packet));
-
-  datap[2] = (data->seqnum >> 8) & 0xff;
-  datap[3] = data->seqnum & 0xff;
-
-  data->seqnum++;
-
-  gst_buffer_set_caps (result, caps);
-
-  return result;
-}
-
-static GstFlowReturn
-dummy_chain (GstPad * pad, GstBuffer * buffer)
-{
-  gst_buffer_unref (buffer);
-
-  return GST_FLOW_OK;
-}
-
-static GstStaticPadTemplate sink_factory = GST_STATIC_PAD_TEMPLATE ("sink",
-    GST_PAD_SINK,
-    GST_PAD_ALWAYS,
-    GST_STATIC_CAPS ("application/x-rtp"));
-
-
-static GstPad *
-make_sinkpad (CleanupData * data)
-{
-  GstPad *pad;
-
-  pad = gst_pad_new_from_static_template (&sink_factory, "sink");
-
-  gst_pad_set_chain_function (pad, dummy_chain);
-  gst_pad_set_active (pad, TRUE);
-
-  data->pads = g_list_prepend (data->pads, pad);
-
-  return pad;
-}
-
-static void
-pad_added_cb (GstElement * rtpbin, GstPad * pad, CleanupData * data)
-{
-  GstPad *sinkpad;
-
-  GST_DEBUG ("pad added %s:%s\n", GST_DEBUG_PAD_NAME (pad));
-
-  if (GST_PAD_IS_SINK (pad))
-    return;
-
-  fail_unless (data->pad_added == FALSE);
-
-  sinkpad = make_sinkpad (data);
-  fail_unless (gst_pad_link (pad, sinkpad) == GST_PAD_LINK_OK);
-
-  g_mutex_lock (data->lock);
-  data->pad_added = TRUE;
-  data->pad = pad;
-  g_cond_signal (data->cond);
-  g_mutex_unlock (data->lock);
-}
-
-static void
-pad_removed_cb (GstElement * rtpbin, GstPad * pad, CleanupData * data)
-{
-  GST_DEBUG ("pad removed %s:%s\n", GST_DEBUG_PAD_NAME (pad));
-
-  if (data->pad != pad)
-    return;
-
-  fail_unless (data->pad_added == TRUE);
-
-  g_mutex_lock (data->lock);
-  data->pad_added = FALSE;
-  g_cond_signal (data->cond);
-  g_mutex_unlock (data->lock);
-}
-
-GST_START_TEST (test_cleanup_recv)
-{
-  GstElement *rtpbin;
-  GstPad *rtp_sink;
-  CleanupData data;
-  GstStateChangeReturn ret;
-  GstFlowReturn res;
-  GstBuffer *buffer;
-  gint count = 2;
-
-  init_data (&data);
-
-  rtpbin = gst_element_factory_make ("gstrtpbin", "rtpbin");
-
-  g_signal_connect (rtpbin, "pad-added", (GCallback) pad_added_cb, &data);
-  g_signal_connect (rtpbin, "pad-removed", (GCallback) pad_removed_cb, &data);
-
-  ret = gst_element_set_state (rtpbin, GST_STATE_PLAYING);
-  fail_unless (ret == GST_STATE_CHANGE_SUCCESS);
-
-  while (count--) {
-    /* request session 0 */
-    rtp_sink = gst_element_get_request_pad (rtpbin, "recv_rtp_sink_0");
-    fail_unless (rtp_sink != NULL);
-    ASSERT_OBJECT_REFCOUNT (rtp_sink, "rtp_sink", 2);
-
-    /* no sourcepads are created yet */
-    fail_unless (rtpbin->numsinkpads == 1);
-    fail_unless (rtpbin->numsrcpads == 0);
-
-    buffer = make_rtp_packet (&data);
-    res = gst_pad_chain (rtp_sink, buffer);
-    GST_DEBUG ("res %d, %s\n", res, gst_flow_get_name (res));
-    fail_unless (res == GST_FLOW_OK);
-
-    buffer = make_rtp_packet (&data);
-    res = gst_pad_chain (rtp_sink, buffer);
-    GST_DEBUG ("res %d, %s\n", res, gst_flow_get_name (res));
-    fail_unless (res == GST_FLOW_OK);
-
-    /* we wait for the new pad to appear now */
-    g_mutex_lock (data.lock);
-    while (!data.pad_added)
-      g_cond_wait (data.cond, data.lock);
-    g_mutex_unlock (data.lock);
-
-    /* sourcepad created now */
-    fail_unless (rtpbin->numsinkpads == 1);
-    fail_unless (rtpbin->numsrcpads == 1);
-
-    /* remove the session */
-    gst_element_release_request_pad (rtpbin, rtp_sink);
-    gst_object_unref (rtp_sink);
-
-    /* pad should be gone now */
-    g_mutex_lock (data.lock);
-    while (data.pad_added)
-      g_cond_wait (data.cond, data.lock);
-    g_mutex_unlock (data.lock);
-
-    /* nothing left anymore now */
-    fail_unless (rtpbin->numsinkpads == 0);
-    fail_unless (rtpbin->numsrcpads == 0);
-  }
-
-  ret = gst_element_set_state (rtpbin, GST_STATE_NULL);
-  fail_unless (ret == GST_STATE_CHANGE_SUCCESS);
-
-  gst_object_unref (rtpbin);
-
-  clean_data (&data);
-}
-
-GST_END_TEST;
-
-GST_START_TEST (test_cleanup_recv2)
-{
-  GstElement *rtpbin;
-  GstPad *rtp_sink;
-  CleanupData data;
-  GstStateChangeReturn ret;
-  GstFlowReturn res;
-  GstBuffer *buffer;
-  gint count = 2;
-
-  init_data (&data);
-
-  rtpbin = gst_element_factory_make ("gstrtpbin", "rtpbin");
-
-  g_signal_connect (rtpbin, "pad-added", (GCallback) pad_added_cb, &data);
-  g_signal_connect (rtpbin, "pad-removed", (GCallback) pad_removed_cb, &data);
-
-  ret = gst_element_set_state (rtpbin, GST_STATE_PLAYING);
-  fail_unless (ret == GST_STATE_CHANGE_SUCCESS);
-
-  /* request session 0 */
-  rtp_sink = gst_element_get_request_pad (rtpbin, "recv_rtp_sink_0");
-  fail_unless (rtp_sink != NULL);
-  ASSERT_OBJECT_REFCOUNT (rtp_sink, "rtp_sink", 2);
-
-  while (count--) {
-    /* no sourcepads are created yet */
-    fail_unless (rtpbin->numsinkpads == 1);
-    fail_unless (rtpbin->numsrcpads == 0);
-
-    buffer = make_rtp_packet (&data);
-    res = gst_pad_chain (rtp_sink, buffer);
-    GST_DEBUG ("res %d, %s\n", res, gst_flow_get_name (res));
-    fail_unless (res == GST_FLOW_OK);
-
-    buffer = make_rtp_packet (&data);
-    res = gst_pad_chain (rtp_sink, buffer);
-    GST_DEBUG ("res %d, %s\n", res, gst_flow_get_name (res));
-    fail_unless (res == GST_FLOW_OK);
-
-    /* we wait for the new pad to appear now */
-    g_mutex_lock (data.lock);
-    while (!data.pad_added)
-      g_cond_wait (data.cond, data.lock);
-    g_mutex_unlock (data.lock);
-
-    /* sourcepad created now */
-    fail_unless (rtpbin->numsinkpads == 1);
-    fail_unless (rtpbin->numsrcpads == 1);
-
-    /* change state */
-    ret = gst_element_set_state (rtpbin, GST_STATE_NULL);
-    fail_unless (ret == GST_STATE_CHANGE_SUCCESS);
-
-    /* pad should be gone now */
-    g_mutex_lock (data.lock);
-    while (data.pad_added)
-      g_cond_wait (data.cond, data.lock);
-    g_mutex_unlock (data.lock);
-
-    /* back to playing for the next round */
-    ret = gst_element_set_state (rtpbin, GST_STATE_PLAYING);
-    fail_unless (ret == GST_STATE_CHANGE_SUCCESS);
-  }
-
-  /* remove the session */
-  gst_element_release_request_pad (rtpbin, rtp_sink);
-  gst_object_unref (rtp_sink);
-
-  /* nothing left anymore now */
-  fail_unless (rtpbin->numsinkpads == 0);
-  fail_unless (rtpbin->numsrcpads == 0);
-
-  ret = gst_element_set_state (rtpbin, GST_STATE_NULL);
-  fail_unless (ret == GST_STATE_CHANGE_SUCCESS);
-
-  gst_object_unref (rtpbin);
-
-  clean_data (&data);
-}
-
-GST_END_TEST;
-
-Suite *
-gstrtpbin_suite (void)
-{
-  Suite *s = suite_create ("gstrtpbin");
-  TCase *tc_chain = tcase_create ("general");
-
-  suite_add_tcase (s, tc_chain);
-  tcase_add_test (tc_chain, test_cleanup_send);
-  tcase_add_test (tc_chain, test_cleanup_recv);
-  tcase_add_test (tc_chain, test_cleanup_recv2);
-
-  return s;
-}
-
-int
-main (int argc, char **argv)
-{
-  int nf;
-
-  Suite *s = gstrtpbin_suite ();
-  SRunner *sr = srunner_create (s);
-
-  gst_check_init (&argc, &argv);
-
-  srunner_run_all (sr, CK_NORMAL);
-  nf = srunner_ntests_failed (sr);
-  srunner_free (sr);
-
-  return nf;
-}
diff --git a/tests/check/elements/rtpbin_buffer_list.c b/tests/check/elements/rtpbin_buffer_list.c
deleted file mode 100644
index af4003dd..00000000
--- a/tests/check/elements/rtpbin_buffer_list.c
+++ /dev/null
@@ -1,331 +0,0 @@
-/* GStreamer
- *
- * Unit test for gstrtpbin sending rtp packets using GstBufferList.
- * Copyright (C) 2009 Branko Subasic <branko dot subasic at axis dot com>
- *
- * This library is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Library General Public
- * License as published by the Free Software Foundation; either
- * version 2 of the License, or (at your option) any later version.
- *
- * This library is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Library General Public License for more details.
- *
- * You should have received a copy of the GNU Library General Public
- * License along with this library; if not, write to the
- * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
- * Boston, MA 02111-1307, USA.
- */
-
-#include <gst/check/gstcheck.h>
-
-#include <gst/rtp/gstrtpbuffer.h>
-
-
-
-/* This test makes sure that RTP packets sent as buffer lists are sent through
- * the rtpbin as they are supposed to, and not corrupted in any way.
- */
-
-
-#define TEST_CAPS \
-  "application/x-rtp, "                \
-  "media=(string)video, "              \
-  "clock-rate=(int)90000, "            \
-  "encoding-name=(string)H264, "       \
-  "profile-level-id=(string)4d4015, "  \
-  "payload=(int)96, "                  \
-  "ssrc=(guint)2633237432, "           \
-  "clock-base=(guint)1868267015, "     \
-  "seqnum-base=(guint)54229"
-
-
-/* RTP headers and the first 2 bytes of the payload (FU indicator and FU header)
- */
-static const guint8 rtp_header[2][14] = {
-  {0x80, 0x60, 0xbb, 0xb7, 0x5c, 0xe9, 0x09,
-      0x0d, 0xf5, 0x9c, 0x43, 0x55, 0x1c, 0x86},
-  {0x80, 0x60, 0xbb, 0xb8, 0x5c, 0xe9, 0x09,
-      0x0d, 0xf5, 0x9c, 0x43, 0x55, 0x1c, 0x46}
-};
-
-static const guint rtp_header_len[] = {
-  sizeof rtp_header[0],
-  sizeof rtp_header[1]
-};
-
-static GstBuffer *header_buffer[2] = { NULL, NULL };
-
-
-/* Some payload.
- */
-static char *payload =
-    "0123456789ABSDEF0123456789ABSDEF0123456789ABSDEF0123456789ABSDEF0123456789ABSDEF"
-    "0123456789ABSDEF0123456789ABSDEF0123456789ABSDEF0123456789ABSDEF0123456789ABSDEF"
-    "0123456789ABSDEF0123456789ABSDEF0123456789ABSDEF0123456789ABSDEF0123456789ABSDEF"
-    "0123456789ABSDEF0123456789ABSDEF0123456789ABSDEF0123456789ABSDEF0123456789ABSDEF"
-    "0123456789ABSDEF0123456789ABSDEF0123456789ABSDEF0123456789ABSDEF0123456789ABSDEF"
-    "0123456789ABSDEF0123456789ABSDEF0123456789ABSDEF0123456789ABSDEF0123456789ABSDEF"
-    "0123456789ABSDEF0123456";
-
-static const guint payload_offset[] = {
-  0, 498
-};
-
-static const guint payload_len[] = {
-  498, 5
-};
-
-
-static GstBuffer *original_buffer = NULL;
-
-static GstStaticPadTemplate sinktemplate = GST_STATIC_PAD_TEMPLATE ("sink",
-    GST_PAD_SINK,
-    GST_PAD_ALWAYS,
-    GST_STATIC_CAPS ("application/x-rtp"));
-
-static GstStaticPadTemplate srctemplate = GST_STATIC_PAD_TEMPLATE ("src",
-    GST_PAD_SRC,
-    GST_PAD_ALWAYS,
-    GST_STATIC_CAPS ("application/x-rtp"));
-
-
-static GstBuffer *
-_create_original_buffer (void)
-{
-  GstCaps *caps;
-
-  if (original_buffer != NULL)
-    return original_buffer;
-
-  original_buffer = gst_buffer_new ();
-  fail_unless (original_buffer != NULL);
-
-  gst_buffer_set_data (original_buffer, (guint8 *) payload, strlen (payload));
-  GST_BUFFER_TIMESTAMP (original_buffer) =
-      gst_clock_get_internal_time (gst_system_clock_obtain ());
-
-  caps = gst_caps_from_string (TEST_CAPS);
-  fail_unless (caps != NULL);
-  gst_buffer_set_caps (original_buffer, caps);
-  gst_caps_unref (caps);
-
-  return original_buffer;
-}
-
-static GstBufferList *
-_create_buffer_list (void)
-{
-  GstBufferList *list;
-  GstBufferListIterator *it;
-  GstBuffer *orig_buffer;
-  GstBuffer *buffer;
-
-  orig_buffer = _create_original_buffer ();
-  fail_if (orig_buffer == NULL);
-
-  list = gst_buffer_list_new ();
-  fail_if (list == NULL);
-
-  it = gst_buffer_list_iterate (list);
-  fail_if (it == NULL);
-
-  /*** First group, i.e. first packet. **/
-  gst_buffer_list_iterator_add_group (it);
-
-  /* Create buffer with RTP header and add it to the 1st group */
-  buffer = gst_buffer_new ();
-  GST_BUFFER_MALLOCDATA (buffer) = g_memdup (&rtp_header[0], rtp_header_len[0]);
-  GST_BUFFER_DATA (buffer) = GST_BUFFER_MALLOCDATA (buffer);
-  GST_BUFFER_SIZE (buffer) = rtp_header_len[0];
-  gst_buffer_copy_metadata (buffer, orig_buffer, GST_BUFFER_COPY_ALL);
-  header_buffer[0] = buffer;
-  gst_buffer_list_iterator_add (it, buffer);
-
-  /* Create the payload buffer and add it to the 1st group
-   */
-  buffer =
-      gst_buffer_create_sub (orig_buffer, payload_offset[0], payload_len[0]);
-  fail_if (buffer == NULL);
-  gst_buffer_list_iterator_add (it, buffer);
-
-
-  /***  Second group, i.e. second packet. ***/
-
-  /* Create a new group to hold the rtp header and the payload */
-  gst_buffer_list_iterator_add_group (it);
-
-  /* Create buffer with RTP header and add it to the 2nd group */
-  buffer = gst_buffer_new ();
-  GST_BUFFER_MALLOCDATA (buffer) = g_memdup (&rtp_header[1], rtp_header_len[1]);
-  GST_BUFFER_DATA (buffer) = GST_BUFFER_MALLOCDATA (buffer);
-  GST_BUFFER_SIZE (buffer) = rtp_header_len[1];
-  gst_buffer_copy_metadata (buffer, orig_buffer, GST_BUFFER_COPY_ALL);
-  header_buffer[1] = buffer;
-
-  /* Add the rtp header to the buffer list */
-  gst_buffer_list_iterator_add (it, buffer);
-
-  /* Create the payload buffer and add it to the 2d group
-   */
-  buffer =
-      gst_buffer_create_sub (orig_buffer, payload_offset[1], payload_len[1]);
-  fail_if (buffer == NULL);
-  gst_buffer_list_iterator_add (it, buffer);
-
-  gst_buffer_list_iterator_free (it);
-
-  return list;
-}
-
-
-static void
-_check_header (GstBuffer * buffer, guint index)
-{
-  guint8 *data;
-
-  fail_if (buffer == NULL);
-  fail_unless (index < 2);
-
-  fail_unless (GST_BUFFER_SIZE (buffer) == rtp_header_len[index]);
-
-  /* Can't do a memcmp() on the whole header, cause the SSRC (bytes 8-11) will
-   * most likely be changed in gstrtpbin.
-   */
-  fail_unless ((data = GST_BUFFER_DATA (buffer)) != NULL);
-  fail_unless_equals_uint64 (*(guint64 *) data, *(guint64 *) rtp_header[index]);
-  fail_unless (*(guint16 *) (data + 12) ==
-      *(guint16 *) (rtp_header[index] + 12));
-}
-
-
-static void
-_check_payload (GstBuffer * buffer, guint index)
-{
-  fail_if (buffer == NULL);
-  fail_unless (index < 2);
-
-  fail_unless (GST_BUFFER_SIZE (buffer) == payload_len[index]);
-  fail_if (GST_BUFFER_DATA (buffer) !=
-      (gpointer) (payload + payload_offset[index]));
-  fail_if (memcmp (GST_BUFFER_DATA (buffer), payload + payload_offset[index],
-          payload_len[index]));
-}
-
-
-static void
-_check_group (GstBufferListIterator * it, guint index, GstCaps * caps)
-{
-  GstBuffer *buffer;
-
-  fail_unless (it != NULL);
-  fail_unless (gst_buffer_list_iterator_n_buffers (it) == 2);
-  fail_unless (caps != NULL);
-
-  fail_unless ((buffer = gst_buffer_list_iterator_next (it)) != NULL);
-
-  fail_unless (GST_BUFFER_TIMESTAMP (buffer) ==
-      GST_BUFFER_TIMESTAMP (original_buffer));
-
-  fail_unless (gst_caps_is_equal (GST_BUFFER_CAPS (original_buffer),
-          GST_BUFFER_CAPS (buffer)));
-
-  _check_header (buffer, index);
-
-  fail_unless ((buffer = gst_buffer_list_iterator_next (it)) != NULL);
-  _check_payload (buffer, index);
-}
-
-
-static GstFlowReturn
-_sink_chain_list (GstPad * pad, GstBufferList * list)
-{
-  GstCaps *caps;
-  GstBufferListIterator *it;
-
-  caps = gst_caps_from_string (TEST_CAPS);
-  fail_unless (caps != NULL);
-
-  fail_unless (GST_IS_BUFFER_LIST (list));
-  fail_unless (gst_buffer_list_n_groups (list) == 2);
-
-  it = gst_buffer_list_iterate (list);
-  fail_if (it == NULL);
-
-  fail_unless (gst_buffer_list_iterator_next_group (it));
-  _check_group (it, 0, caps);
-
-  fail_unless (gst_buffer_list_iterator_next_group (it));
-  _check_group (it, 1, caps);
-
-  gst_caps_unref (caps);
-  gst_buffer_list_iterator_free (it);
-
-  gst_buffer_list_unref (list);
-
-  return GST_FLOW_OK;
-}
-
-
-static void
-_set_chain_functions (GstPad * pad)
-{
-  gst_pad_set_chain_list_function (pad, _sink_chain_list);
-}
-
-
-GST_START_TEST (test_bufferlist)
-{
-  GstElement *rtpbin;
-  GstPad *sinkpad;
-  GstPad *srcpad;
-  GstBufferList *list;
-
-  list = _create_buffer_list ();
-  fail_unless (list != NULL);
-
-  rtpbin = gst_check_setup_element ("gstrtpbin");
-
-  srcpad =
-      gst_check_setup_src_pad_by_name (rtpbin, &srctemplate, "send_rtp_sink_0");
-  fail_if (srcpad == NULL);
-  sinkpad =
-      gst_check_setup_sink_pad_by_name (rtpbin, &sinktemplate,
-      "send_rtp_src_0");
-  fail_if (sinkpad == NULL);
-
-  _set_chain_functions (sinkpad);
-
-  gst_pad_set_active (sinkpad, TRUE);
-  gst_element_set_state (rtpbin, GST_STATE_PLAYING);
-  fail_unless (gst_pad_push_list (srcpad, list) == GST_FLOW_OK);
-  gst_pad_set_active (sinkpad, FALSE);
-
-  gst_check_teardown_pad_by_name (rtpbin, "send_rtp_src_0");
-  gst_check_teardown_pad_by_name (rtpbin, "send_rtp_sink_0");
-  gst_check_teardown_element (rtpbin);
-}
-
-GST_END_TEST;
-
-
-
-static Suite *
-bufferlist_suite (void)
-{
-  Suite *s = suite_create ("BufferList");
-
-  TCase *tc_chain = tcase_create ("general");
-
-  /* time out after 30s. */
-  tcase_set_timeout (tc_chain, 10);
-
-  suite_add_tcase (s, tc_chain);
-  tcase_add_test (tc_chain, test_bufferlist);
-
-  return s;
-}
-
-GST_CHECK_MAIN (bufferlist);