mirrors/gstreamer
mirrors/gstreamer
1
1
Fork 0
mirror of https://gitlab.freedesktop.org/gstreamer/gstreamer.git synced 2024-11-23 10:11:08 +00:00
Code Issues 1 Projects Releases Wiki Activity

docs: add rtpmux and rtpdtmfmux to plugin docs

Browse source 
https://bugzilla.gnome.org/show_bug.cgi?id=629117
This commit is contained in:
Tim-Philipp Müller 2012-12-16 15:25:03 +00:00
parent 3295b5d791
commit 8a3b116d1f
7 changed files with 126 additions and 1 deletions
Show stats Download patch file Download diff file Expand all files Collapse all files

2
docs/plugins/Makefile.am
View file

@ -171,6 +171,8 @@ EXTRA_HFILES = \
$(top_srcdir)/gst/rtpmanager/gstrtpptdemux.h \
$(top_srcdir)/gst/rtpmanager/gstrtpsession.h \
$(top_srcdir)/gst/rtpmanager/gstrtpssrcdemux.h \
$(top_srcdir)/gst/rtpmanager/gstrtpmux.h \
$(top_srcdir)/gst/rtpmanager/gstrtpdtmfmux.h \
$(top_srcdir)/gst/rtsp/gstrtpdec.h \
$(top_srcdir)/gst/rtsp/gstrtspsrc.h \
$(top_srcdir)/gst/shapewipe/gstshapewipe.h \

2
docs/plugins/gst-plugins-good-plugins-docs.sgml
View file

@ -134,7 +134,9 @@
<xi:include href="xml/element-rtpjpegpay.xml" />
<xi:include href="xml/element-rtspsrc.xml" />
<xi:include href="xml/element-rtpbin.xml" />
<xi:include href="xml/element-rtpdtmfmux.xml" />
<xi:include href="xml/element-rtpjitterbuffer.xml" />
<xi:include href="xml/element-rtpmux.xml" />
<xi:include href="xml/element-rtpptdemux.xml" />
<xi:include href="xml/element-rtpsession.xml" />
<xi:include href="xml/element-rtpssrcdemux.xml" />

31
docs/plugins/gst-plugins-good-plugins-sections.txt
View file

@ -2268,6 +2268,37 @@ GST_TYPE_RIPPLETV
gst_rippletv_get_type
</SECTION>
<SECTION>
<FILE>element-rtpdtmfmux</FILE>
<TITLE>rtpdtmfmux</TITLE>
GstRTPDTMFMux
<SUBSECTION Standard>
GST_IS_RTP_DTMF_MUX
GST_IS_RTP_DTMF_MUX_CLASS
GST_TYPE_RTP_DTMF_MUX
GstRTPDTMFMuxClass
GST_RTP_DTMF_MUX
GST_RTP_DTMF_MUX_CLASS
gst_rtp_dtmf_mux_get_type
gst_rtp_dtmf_mux_plugin_init
</SECTION>
<SECTION>
<FILE>element-rtpmux</FILE>
<TITLE>rtpmux</TITLE>
GstRTPMux
<SUBSECTION Standard>
GST_IS_RTP_MUX
GST_IS_RTP_MUX_CLASS
GST_RTP_MUX
GST_RTP_MUX_CLASS
GST_RTP_MUX_GET_CLASS
GST_TYPE_RTP_MUX
GstRTPMuxClass
gst_rtp_mux_get_type
gst_rtp_mux_plugin_init
</SECTION>
<SECTION>
<FILE>element-scaletempo</FILE>
<TITLE>scaletempo</TITLE>

40
docs/plugins/gst-plugins-good-plugins.args
View file

@ -23448,3 +23448,43 @@
<DEFAULT>30</DEFAULT>
</ARG>
<ARG>
<NAME>GstRTPMux::seqnum</NAME>
<TYPE>guint</TYPE>
<RANGE></RANGE>
<FLAGS>r</FLAGS>
<NICK>Sequence number</NICK>
<BLURB>The RTP sequence number of the last processed packet.</BLURB>
<DEFAULT>0</DEFAULT>
</ARG>
<ARG>
<NAME>GstRTPMux::seqnum-offset</NAME>
<TYPE>gint</TYPE>
<RANGE>>= G_MAXULONG</RANGE>
<FLAGS>rw</FLAGS>
<NICK>Sequence number Offset</NICK>
<BLURB>Offset to add to all outgoing seqnum (-1 = random).</BLURB>
<DEFAULT>-1</DEFAULT>
</ARG>
<ARG>
<NAME>GstRTPMux::ssrc</NAME>
<TYPE>guint</TYPE>
<RANGE></RANGE>
<FLAGS>rw</FLAGS>
<NICK>SSRC</NICK>
<BLURB>The SSRC of the packets (-1 == random).</BLURB>
<DEFAULT>4294967295</DEFAULT>
</ARG>
<ARG>
<NAME>GstRTPMux::timestamp-offset</NAME>
<TYPE>gint</TYPE>
<RANGE>>= G_MAXULONG</RANGE>
<FLAGS>rw</FLAGS>
<NICK>Timestamp Offset</NICK>
<BLURB>Offset to add to all outgoing timestamps (-1 = random).</BLURB>
<DEFAULT>-1</DEFAULT>
</ARG>

2
docs/plugins/gst-plugins-good-plugins.hierarchy
View file

@ -243,6 +243,8 @@ GObject
GstRtpVRawPay
GstRtpVorbisPay
GstRTPDec
GstRTPMux
GstRTPDTMFMux
GstRndBufferSize
GstRtpJitterBuffer
GstRtpPtDemux

48
docs/plugins/inspect/plugin-rtpmanager.xml
View file

@ -54,6 +54,33 @@
</caps>
</pads>
</element>
<element>
<name>rtpdtmfmux</name>
<longname>RTP muxer</longname>
<class>Codec/Muxer</class>
<description>mixes RTP DTMF streams into other RTP streams</description>
<author>Zeeshan Ali &lt;first.last@nokia.com&gt;</author>
<pads>
<caps>
<name>priority_sink_%u</name>
<direction>sink</direction>
<presence>request</presence>
<details>application/x-rtp</details>
</caps>
<caps>
<name>sink_%u</name>
<direction>sink</direction>
<presence>request</presence>
<details>application/x-rtp</details>
</caps>
<caps>
<name>src</name>
<direction>source</direction>
<presence>always</presence>
<details>application/x-rtp</details>
</caps>
</pads>
</element>
<element>
<name>rtpjitterbuffer</name>
<longname>RTP packet jitter-buffer</longname>
@ -81,6 +108,27 @@
</caps>
</pads>
</element>
<element>
<name>rtpmux</name>
<longname>RTP muxer</longname>
<class>Codec/Muxer</class>
<description>multiplex N rtp streams into one</description>
<author>Zeeshan Ali &lt;first.last@nokia.com&gt;</author>
<pads>
<caps>
<name>sink_%u</name>
<direction>sink</direction>
<presence>request</presence>
<details>application/x-rtp</details>
</caps>
<caps>
<name>src</name>
<direction>source</direction>
<presence>always</presence>
<details>application/x-rtp</details>
</caps>
</pads>
</element>
<element>
<name>rtpptdemux</name>
<longname>RTP Demux</longname>

2
gst/rtpmanager/gstrtpdtmfmux.c
View file

@ -30,7 +30,7 @@
* @see_also: rtpdtmfsrc, dtmfsrc, rtpmux
*
* The RTP "DTMF" Muxer muxes multiple RTP streams into a valid RTP
* stream. It does exactly what it's parent (#rtpmux) does, except
* stream. It does exactly what its parent (#rtpmux) does, except
* that it prevent buffers coming over a regular sink_%%u pad from going through
* for the duration of buffers that came in a priority_sink_%%u pad.
*