</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-halaudiosink</FILE>
<TITLE>halaudiosink</TITLE>
GstHalAudioSink
--- /dev/null
+<plugin>
+ <name>gstrtpmanager</name>
+ <description>RTP session management plugin library</description>
+ <filename>../../gst/rtpmanager/.libs/libgstrtpmanager.so</filename>
+ <basename>libgstrtpmanager.so</basename>
+ <version>0.10.15.1</version>
+ <license>LGPL</license>
+ <source>gst-plugins-good</source>
+ <package>GStreamer Good Plug-ins git/prerelease</package>
+ <origin>Unknown package origin</origin>
+ <elements>
+ <element>
+ <name>gstrtpbin</name>
+ <longname>RTP Bin</longname>
+ <class>Filter/Network/RTP</class>
+ <description>Implement an RTP bin</description>
+ <author>Wim Taymans <wim.taymans@gmail.com></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 <philippe.kalaf@collabora.co.uk>, Wim Taymans <wim.taymans@gmail.com></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 <kai.vehmanen@nokia.com></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 <wim.taymans@gmail.com></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 <wim.taymans@gmail.com></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