mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2025-01-27 09:38:17 +00:00
Move rtpmanager from -bad to -good.
Hook up build infrastructure (autotools, docs, unit test).
This commit is contained in:
parent
92abe07e80
commit
4701696a92
9 changed files with 292 additions and 0 deletions
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@ -302,6 +302,7 @@ AG_GST_CHECK_PLUGIN(multipart)
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AG_GST_CHECK_PLUGIN(qtdemux)
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AG_GST_CHECK_PLUGIN(replaygain)
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AG_GST_CHECK_PLUGIN(rtp)
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AG_GST_CHECK_PLUGIN(rtpmanager)
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AG_GST_CHECK_PLUGIN(rtsp)
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AG_GST_CHECK_PLUGIN(smpte)
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AG_GST_CHECK_PLUGIN(spectrum)
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@ -1065,6 +1066,7 @@ gst/multipart/Makefile
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gst/qtdemux/Makefile
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gst/replaygain/Makefile
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gst/rtp/Makefile
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gst/rtpmanager/Makefile
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gst/rtsp/Makefile
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gst/smpte/Makefile
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gst/spectrum/Makefile
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@ -180,6 +180,11 @@ EXTRA_HFILES = \
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$(top_srcdir)/gst/replaygain/gstrglimiter.h \
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$(top_srcdir)/gst/replaygain/gstrgvolume.h \
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$(top_srcdir)/gst/rtp/gstrtpjpegpay.h \
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$(top_srcdir)/gst/rtpmanager/gstrtpbin.h \
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$(top_srcdir)/gst/rtpmanager/gstrtpjitterbuffer.h \
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$(top_srcdir)/gst/rtpmanager/gstrtpptdemux.h \
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$(top_srcdir)/gst/rtpmanager/gstrtpsession.h \
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$(top_srcdir)/gst/rtpmanager/gstrtpssrcdemux.h \
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$(top_srcdir)/gst/rtsp/gstrtpdec.h \
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$(top_srcdir)/gst/rtsp/gstrtspsrc.h \
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$(top_srcdir)/gst/smpte/gstsmpte.h \
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@ -77,6 +77,11 @@
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<xi:include href="xml/element-gdkpixbufsink.xml" />
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<xi:include href="xml/element-goom.xml" />
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<xi:include href="xml/element-goom2k1.xml" />
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<xi:include href="xml/element-gstrtpbin.xml" />
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<xi:include href="xml/element-gstrtpjitterbuffer.xml" />
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<xi:include href="xml/element-gstrtpptdemux.xml" />
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<xi:include href="xml/element-gstrtpsession.xml" />
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<xi:include href="xml/element-gstrtpssrcdemux.xml" />
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<xi:include href="xml/element-halaudiosink.xml" />
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<xi:include href="xml/element-halaudiosrc.xml" />
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<xi:include href="xml/element-hdv1394src.xml" />
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@ -204,6 +209,7 @@
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<xi:include href="xml/plugin-quicktime.xml" />
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<xi:include href="xml/plugin-replaygain.xml" />
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<xi:include href="xml/plugin-rtp.xml" />
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<xi:include href="xml/plugin-gstrtpmanager.xml" />
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<xi:include href="xml/plugin-rtsp.xml" />
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<xi:include href="xml/plugin-shout2send.xml" />
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<xi:include href="xml/plugin-smpte.xml" />
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@ -843,6 +843,82 @@ GST_GOOM_CLASS
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GST_IS_GOOM_CLASS
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</SECTION>
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<SECTION>
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<FILE>element-gstrtpbin</FILE>
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<TITLE>gstrtpbin</TITLE>
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GstRtpBin
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<SUBSECTION Standard>
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GstRtpBinPrivate
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GstRtpBinClass
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GST_RTP_BIN
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GST_IS_RTP_BIN
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GST_TYPE_RTP_BIN
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gst_rtp_bin_get_type
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GST_RTP_BIN_CLASS
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GST_IS_RTP_BIN_CLASS
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</SECTION>
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<SECTION>
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<FILE>element-gstrtpjitterbuffer</FILE>
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<TITLE>gstrtpjitterbuffer</TITLE>
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GstRtpJitterBuffer
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<SUBSECTION Standard>
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GstRtpJitterBufferClass
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GstRtpJitterBufferPrivate
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GST_RTP_JITTER_BUFFER
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GST_IS_RTP_JITTER_BUFFER
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GST_TYPE_RTP_JITTER_BUFFER
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gst_rtp_jitter_buffer_get_type
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GST_RTP_JITTER_BUFFER_CLASS
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GST_IS_RTP_JITTER_BUFFER_CLASS
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</SECTION>
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<SECTION>
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<FILE>element-gstrtpptdemux</FILE>
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<TITLE>gstrtpptdemux</TITLE>
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GstRtpPtDemux
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<SUBSECTION Standard>
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GstRtpPtDemuxClass
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GstRtpPtDemuxPad
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GST_RTP_PT_DEMUX
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GST_IS_RTP_PT_DEMUX
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GST_TYPE_RTP_PT_DEMUX
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gst_rtp_pt_demux_get_type
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GST_RTP_PT_DEMUX_CLASS
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GST_IS_RTP_PT_DEMUX_CLASS
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</SECTION>
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<SECTION>
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<FILE>element-gstrtpsession</FILE>
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<TITLE>gstrtpsession</TITLE>
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GstRtpSession
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<SUBSECTION Standard>
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GstRtpSessionClass
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GstRtpSessionPrivate
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GST_RTP_SESSION
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GST_IS_RTP_SESSION
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GST_TYPE_RTP_SESSION
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gst_rtp_session_get_type
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GST_RTP_SESSION_CLASS
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GST_IS_RTP_SESSION_CLASS
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GST_RTP_SESSION_CAST
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</SECTION>
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<SECTION>
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<FILE>element-gstrtpssrcdemux</FILE>
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<TITLE>gstrtpssrcdemux</TITLE>
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GstRtpSsrcDemux
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<SUBSECTION Standard>
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GstRtpSsrcDemuxClass
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GstRtpSsrcDemuxPad
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GST_RTP_SSRC_DEMUX
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GST_IS_RTP_SSRC_DEMUX
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GST_TYPE_RTP_SSRC_DEMUX
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gst_rtp_ssrc_demux_get_type
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GST_RTP_SSRC_DEMUX_CLASS
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GST_IS_RTP_SSRC_DEMUX_CLASS
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</SECTION>
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<SECTION>
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<FILE>element-halaudiosink</FILE>
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<TITLE>halaudiosink</TITLE>
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190
docs/plugins/inspect/plugin-gstrtpmanager.xml
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190
docs/plugins/inspect/plugin-gstrtpmanager.xml
Normal file
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@ -0,0 +1,190 @@
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<plugin>
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<name>gstrtpmanager</name>
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<description>RTP session management plugin library</description>
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<filename>../../gst/rtpmanager/.libs/libgstrtpmanager.so</filename>
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<basename>libgstrtpmanager.so</basename>
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<version>0.10.15.1</version>
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<license>LGPL</license>
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<source>gst-plugins-good</source>
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<package>GStreamer Good Plug-ins git/prerelease</package>
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<origin>Unknown package origin</origin>
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<elements>
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<element>
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<name>gstrtpbin</name>
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<longname>RTP Bin</longname>
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<class>Filter/Network/RTP</class>
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<description>Implement an RTP bin</description>
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<author>Wim Taymans <wim.taymans@gmail.com></author>
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<pads>
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<caps>
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<name>send_rtp_src_%d</name>
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<direction>source</direction>
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<presence>sometimes</presence>
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<details>application/x-rtp</details>
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</caps>
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<caps>
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<name>send_rtcp_src_%d</name>
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<direction>source</direction>
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<presence>request</presence>
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<details>application/x-rtcp</details>
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</caps>
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<caps>
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<name>recv_rtp_src_%d_%d_%d</name>
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<direction>source</direction>
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<presence>sometimes</presence>
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<details>application/x-rtp</details>
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</caps>
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<caps>
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<name>send_rtp_sink_%d</name>
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<direction>sink</direction>
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<presence>request</presence>
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<details>application/x-rtp</details>
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</caps>
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<caps>
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<name>recv_rtcp_sink_%d</name>
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<direction>sink</direction>
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<presence>request</presence>
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<details>application/x-rtcp</details>
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</caps>
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<caps>
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<name>recv_rtp_sink_%d</name>
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<direction>sink</direction>
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<presence>request</presence>
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<details>application/x-rtp</details>
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</caps>
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</pads>
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</element>
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<element>
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<name>gstrtpjitterbuffer</name>
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<longname>RTP packet jitter-buffer</longname>
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<class>Filter/Network/RTP</class>
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<description>A buffer that deals with network jitter and other transmission faults</description>
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<author>Philippe Kalaf <philippe.kalaf@collabora.co.uk>, Wim Taymans <wim.taymans@gmail.com></author>
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<pads>
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<caps>
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<name>sink_rtcp</name>
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<direction>sink</direction>
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<presence>request</presence>
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<details>application/x-rtcp</details>
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</caps>
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<caps>
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<name>sink</name>
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<direction>sink</direction>
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<presence>always</presence>
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<details>application/x-rtp, clock-rate=(int)[ 1, 2147483647 ]</details>
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</caps>
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<caps>
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<name>src</name>
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<direction>source</direction>
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<presence>always</presence>
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<details>application/x-rtp</details>
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</caps>
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</pads>
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</element>
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<element>
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<name>gstrtpptdemux</name>
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<longname>RTP Demux</longname>
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<class>Demux/Network/RTP</class>
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<description>Parses codec streams transmitted in the same RTP session</description>
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<author>Kai Vehmanen <kai.vehmanen@nokia.com></author>
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<pads>
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<caps>
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<name>src_%d</name>
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<direction>source</direction>
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<presence>sometimes</presence>
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<details>application/x-rtp, payload=(int)[ 0, 255 ]</details>
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</caps>
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<caps>
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<name>sink</name>
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<direction>sink</direction>
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<presence>always</presence>
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<details>application/x-rtp</details>
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</caps>
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</pads>
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</element>
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<element>
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<name>gstrtpsession</name>
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<longname>RTP Session</longname>
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<class>Filter/Network/RTP</class>
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<description>Implement an RTP session</description>
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<author>Wim Taymans <wim.taymans@gmail.com></author>
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<pads>
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<caps>
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<name>send_rtcp_src</name>
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<direction>source</direction>
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<presence>request</presence>
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<details>application/x-rtcp</details>
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</caps>
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<caps>
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<name>send_rtp_src</name>
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<direction>source</direction>
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<presence>sometimes</presence>
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<details>application/x-rtp</details>
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</caps>
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<caps>
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<name>sync_src</name>
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<direction>source</direction>
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<presence>sometimes</presence>
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<details>application/x-rtcp</details>
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</caps>
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<caps>
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<name>recv_rtp_src</name>
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<direction>source</direction>
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<presence>sometimes</presence>
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<details>application/x-rtp</details>
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</caps>
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<caps>
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<name>send_rtp_sink</name>
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<direction>sink</direction>
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<presence>request</presence>
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<details>application/x-rtp</details>
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</caps>
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<caps>
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<name>recv_rtcp_sink</name>
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<direction>sink</direction>
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<presence>request</presence>
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<details>application/x-rtcp</details>
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</caps>
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<caps>
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<name>recv_rtp_sink</name>
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<direction>sink</direction>
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<presence>request</presence>
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<details>application/x-rtp</details>
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</caps>
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</pads>
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</element>
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<element>
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<name>gstrtpssrcdemux</name>
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<longname>RTP SSRC Demux</longname>
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<class>Demux/Network/RTP</class>
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<description>Splits RTP streams based on the SSRC</description>
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<author>Wim Taymans <wim.taymans@gmail.com></author>
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<pads>
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<caps>
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<name>rtcp_src_%d</name>
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<direction>source</direction>
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<presence>sometimes</presence>
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<details>application/x-rtcp</details>
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</caps>
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<caps>
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<name>src_%d</name>
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<direction>source</direction>
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<presence>sometimes</presence>
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<details>application/x-rtp</details>
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</caps>
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<caps>
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<name>rtcp_sink</name>
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<direction>sink</direction>
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<presence>always</presence>
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<details>application/x-rtcp</details>
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</caps>
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<caps>
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<name>sink</name>
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<direction>sink</direction>
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<presence>always</presence>
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<details>application/x-rtp</details>
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</caps>
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</pads>
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</element>
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</elements>
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</plugin>
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@ -100,6 +100,7 @@ rm -rf $RPM_BUILD_ROOT
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%{_libdir}/gstreamer-%{majorminor}/libgstmulaw.so
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%{_libdir}/gstreamer-%{majorminor}/libgstqtdemux.so
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%{_libdir}/gstreamer-%{majorminor}/libgstrtp.so
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%{_libdir}/gstreamer-%{majorminor}/libgstrtpmanager.so
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%{_libdir}/gstreamer-%{majorminor}/libgstrtsp.so
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%{_libdir}/gstreamer-%{majorminor}/libgstsmpte.so
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%{_libdir}/gstreamer-%{majorminor}/libgstudp.so
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@ -112,6 +112,8 @@ check_PROGRAMS = \
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elements/rglimiter \
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elements/rgvolume \
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elements/rtp-payloading \
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elements/rtpbin \
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elements/rtpbin_buffer_list \
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elements/spectrum \
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elements/udpsink \
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elements/videocrop \
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@ -169,6 +171,13 @@ elements_deinterleave_LDADD = $(GST_PLUGINS_BASE_LIBS) -lgstaudio-$(GST_MAJORMIN
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elements_interleave_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(CFLAGS) $(AM_CFLAGS)
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elements_interleave_LDADD = $(GST_PLUGINS_BASE_LIBS) -lgstaudio-$(GST_MAJORMINOR) $(LDADD)
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elements_rtpbin_buffer_list_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(GST_CFLAGS) \
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$(ERROR_CFLAGS) $(GST_CHECK_CFLAGS)
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elements_rtpbin_buffer_list_LDADD = $(GST_PLUGINS_BASE_LIBS) \
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-lgstnetbuffer-@GST_MAJORMINOR@ -lgstrtp-@GST_MAJORMINOR@ \
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$(GST_BASE_LIBS) $(GST_LIBS_LIBS) $(GST_CHECK_LIBS)
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elements_rtpbin_buffer_list_SOURCES = elements/rtpbin_buffer_list.c
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elements_souphttpsrc_CFLAGS = $(SOUP_CFLAGS) $(AM_CFLAGS)
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elements_souphttpsrc_LDADD = $(SOUP_LIBS) $(LDADD)
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2
tests/check/elements/.gitignore
vendored
2
tests/check/elements/.gitignore
vendored
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@ -34,6 +34,8 @@ rganalysis
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rglimiter
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rgvolume
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rtp-payloading
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rtpbin
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rtpbin_buffer_list
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souphttpsrc
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spectrum
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sunaudio
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1
tests/check/pipelines/.gitignore
vendored
1
tests/check/pipelines/.gitignore
vendored
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@ -1,4 +1,5 @@
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.dirstamp
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effectv
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flacdec
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simple-launch-lines
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wavpack
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