Move rtpmanager from -bad to -good.

Hook up build infrastructure (autotools, docs, unit test).
2024-07-03 13:16:04 +00:00 · 2009-08-11 02:31:44 +01:00 · 2009-08-11 02:31:44 +01:00 · 4701696a92
parent 92abe07e80
commit 4701696a92
9 changed files with 292 additions and 0 deletions
--- a/configure.ac
+++ b/configure.ac
@ -302,6 +302,7 @@ AG_GST_CHECK_PLUGIN(multipart)
 AG_GST_CHECK_PLUGIN(qtdemux)
 AG_GST_CHECK_PLUGIN(replaygain)
 AG_GST_CHECK_PLUGIN(rtp)
+AG_GST_CHECK_PLUGIN(rtpmanager)
 AG_GST_CHECK_PLUGIN(rtsp)
 AG_GST_CHECK_PLUGIN(smpte)
 AG_GST_CHECK_PLUGIN(spectrum)
@ -1065,6 +1066,7 @@ gst/multipart/Makefile
 gst/qtdemux/Makefile
 gst/replaygain/Makefile
 gst/rtp/Makefile
+gst/rtpmanager/Makefile
 gst/rtsp/Makefile
 gst/smpte/Makefile
 gst/spectrum/Makefile
--- a/docs/plugins/Makefile.am
+++ b/docs/plugins/Makefile.am
@ -180,6 +180,11 @@ EXTRA_HFILES = \
 	$(top_srcdir)/gst/replaygain/gstrglimiter.h \
 	$(top_srcdir)/gst/replaygain/gstrgvolume.h \
 	$(top_srcdir)/gst/rtp/gstrtpjpegpay.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/rtsp/gstrtpdec.h \
 	$(top_srcdir)/gst/rtsp/gstrtspsrc.h \
 	$(top_srcdir)/gst/smpte/gstsmpte.h \
--- a/docs/plugins/gst-plugins-good-plugins-docs.sgml
+++ b/docs/plugins/gst-plugins-good-plugins-docs.sgml
@ -77,6 +77,11 @@
    <xi:include href="xml/element-gdkpixbufsink.xml" />
    <xi:include href="xml/element-goom.xml" />
    <xi:include href="xml/element-goom2k1.xml" />
+    <xi:include href="xml/element-gstrtpbin.xml" />
+    <xi:include href="xml/element-gstrtpjitterbuffer.xml" />
+    <xi:include href="xml/element-gstrtpptdemux.xml" />
+    <xi:include href="xml/element-gstrtpsession.xml" />
+    <xi:include href="xml/element-gstrtpssrcdemux.xml" />
    <xi:include href="xml/element-halaudiosink.xml" />
    <xi:include href="xml/element-halaudiosrc.xml" />
    <xi:include href="xml/element-hdv1394src.xml" />
@ -204,6 +209,7 @@
    <xi:include href="xml/plugin-quicktime.xml" />
    <xi:include href="xml/plugin-replaygain.xml" />
    <xi:include href="xml/plugin-rtp.xml" />
+    <xi:include href="xml/plugin-gstrtpmanager.xml" />
    <xi:include href="xml/plugin-rtsp.xml" />
    <xi:include href="xml/plugin-shout2send.xml" />
    <xi:include href="xml/plugin-smpte.xml" />
--- a/docs/plugins/gst-plugins-good-plugins-sections.txt
+++ b/docs/plugins/gst-plugins-good-plugins-sections.txt
@ -843,6 +843,82 @@ GST_GOOM_CLASS
 GST_IS_GOOM_CLASS
 </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>
--- a/docs/plugins/inspect/plugin-gstrtpmanager.xml
+++ b/docs/plugins/inspect/plugin-gstrtpmanager.xml
@ -0,0 +1,190 @@
+<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 &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>
--- a/gst-plugins-good.spec.in
+++ b/gst-plugins-good.spec.in
@ -100,6 +100,7 @@ rm -rf $RPM_BUILD_ROOT
 %{_libdir}/gstreamer-%{majorminor}/libgstmulaw.so
 %{_libdir}/gstreamer-%{majorminor}/libgstqtdemux.so
 %{_libdir}/gstreamer-%{majorminor}/libgstrtp.so
+%{_libdir}/gstreamer-%{majorminor}/libgstrtpmanager.so
 %{_libdir}/gstreamer-%{majorminor}/libgstrtsp.so
 %{_libdir}/gstreamer-%{majorminor}/libgstsmpte.so
 %{_libdir}/gstreamer-%{majorminor}/libgstudp.so
--- a/tests/check/Makefile.am
+++ b/tests/check/Makefile.am
@ -112,6 +112,8 @@ check_PROGRAMS = \
 	elements/rglimiter \
 	elements/rgvolume \
 	elements/rtp-payloading \
+	elements/rtpbin \
+	elements/rtpbin_buffer_list \
 	elements/spectrum \
 	elements/udpsink \
 	elements/videocrop \
@ -169,6 +171,13 @@ elements_deinterleave_LDADD = $(GST_PLUGINS_BASE_LIBS) -lgstaudio-$(GST_MAJORMIN
 elements_interleave_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(CFLAGS) $(AM_CFLAGS)
 elements_interleave_LDADD = $(GST_PLUGINS_BASE_LIBS) -lgstaudio-$(GST_MAJORMINOR) $(LDADD)

+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_souphttpsrc_CFLAGS = $(SOUP_CFLAGS) $(AM_CFLAGS)
 elements_souphttpsrc_LDADD = $(SOUP_LIBS) $(LDADD)

--- a/tests/check/elements/.gitignore
+++ b/tests/check/elements/.gitignore
@ -34,6 +34,8 @@ rganalysis
 rglimiter
 rgvolume
 rtp-payloading
+rtpbin
+rtpbin_buffer_list
 souphttpsrc
 spectrum
 sunaudio
--- a/tests/check/pipelines/.gitignore
+++ b/tests/check/pipelines/.gitignore
@ -1,4 +1,5 @@
 .dirstamp
+effectv
 flacdec
 simple-launch-lines
 wavpack