diff options
| author | Tim-Philipp Müller <tim.muller@collabora.co.uk> | 2009-08-11 02:31:44 +0100 |
|---|---|---|
| committer | Tim-Philipp Müller <tim.muller@collabora.co.uk> | 2009-08-11 02:43:09 +0100 |
| commit | 4701696a92664c1b88c7368441c73893e04698a8 (patch) | |
| tree | c2bd8c69c98fdaa86a4119107396911c38a5e7f4 /docs | |
| parent | 92abe07e8011a9ac17d9fc9440fcf56106003dec (diff) | |
Move rtpmanager from -bad to -good.
Hook up build infrastructure (autotools, docs, unit test).
Diffstat (limited to 'docs')
| -rw-r--r-- | docs/plugins/Makefile.am | 5 | ||||
| -rw-r--r-- | docs/plugins/gst-plugins-good-plugins-docs.sgml | 6 | ||||
| -rw-r--r-- | docs/plugins/gst-plugins-good-plugins-sections.txt | 76 | ||||
| -rw-r--r-- | docs/plugins/inspect/plugin-gstrtpmanager.xml | 190 |
4 files changed, 277 insertions, 0 deletions
diff --git a/docs/plugins/Makefile.am b/docs/plugins/Makefile.am index dd189640..3175dd1b 100644 --- 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 \ diff --git a/docs/plugins/gst-plugins-good-plugins-docs.sgml b/docs/plugins/gst-plugins-good-plugins-docs.sgml index 75714e3e..742f0954 100644 --- 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" /> diff --git a/docs/plugins/gst-plugins-good-plugins-sections.txt b/docs/plugins/gst-plugins-good-plugins-sections.txt index 73d7aff2..7f85d55d 100644 --- a/docs/plugins/gst-plugins-good-plugins-sections.txt +++ b/docs/plugins/gst-plugins-good-plugins-sections.txt @@ -844,6 +844,82 @@ 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> GstHalAudioSink diff --git a/docs/plugins/inspect/plugin-gstrtpmanager.xml b/docs/plugins/inspect/plugin-gstrtpmanager.xml new file mode 100644 index 00000000..377f1d19 --- /dev/null +++ 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 <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>
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