Commit 4701696a authored by Tim-Philipp Müller's avatar Tim-Philipp Müller

Move rtpmanager from -bad to -good.

Hook up build infrastructure (autotools, docs, unit test).
parent 92abe07e
......@@ -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
......
......@@ -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 \
......
......@@ -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" />
......
......@@ -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>
......
<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>
\ No newline at end of file
......@@ -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
......
......@@ -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)
......
......@@ -34,6 +34,8 @@ rganalysis
rglimiter
rgvolume
rtp-payloading
rtpbin
rtpbin_buffer_list
souphttpsrc
spectrum
sunaudio
......
.dirstamp
effectv
flacdec
simple-launch-lines
wavpack
......
Markdown is supported
0% or
You are about to add 0 people to the discussion. Proceed with caution.
Finish editing this message first!
Please register or to comment