Commit 204755d0 authored by Thomas Vander Stichele's avatar Thomas Vander Stichele
Browse files

updating docs

Original commit message from CVS:
updating docs
parent 6115f0b5
2005-09-23 Thomas Vander Stichele <thomas at apestaart dot org>
* docs/plugins/gst-plugins-good-plugins.args:
* docs/plugins/inspect/plugin-alpha.xml:
* docs/plugins/inspect/plugin-rtp.xml:
* gst/level/gstlevel.c: (gst_level_set_caps),
(gst_level_transform_ip):
updating docs
2005-09-23 Thomas Vander Stichele <thomas at apestaart dot org>
* Makefile.am:
......
......@@ -4238,14 +4238,24 @@
<DEFAULT>0</DEFAULT>
</ARG>
<ARG>
<NAME>GstRtpMP4VEnc::send-config</NAME>
<TYPE>gboolean</TYPE>
<RANGE></RANGE>
<FLAGS>rw</FLAGS>
<NICK>Send Config</NICK>
<BLURB>Send the config parameters in RTP packets as well.</BLURB>
<DEFAULT>FALSE</DEFAULT>
</ARG>
<ARG>
<NAME>GstLevel::interval</NAME>
<TYPE>gdouble</TYPE>
<RANGE>[0.01,100]</RANGE>
<TYPE>guint64</TYPE>
<RANGE>>= 1</RANGE>
<FLAGS>rw</FLAGS>
<NICK>Interval</NICK>
<BLURB>Interval between posts (in seconds).</BLURB>
<DEFAULT>0.1</DEFAULT>
<BLURB>Interval of time between message posts (in nanoseconds).</BLURB>
<DEFAULT>100000000</DEFAULT>
</ARG>
<ARG>
......@@ -4254,7 +4264,7 @@
<RANGE></RANGE>
<FLAGS>rw</FLAGS>
<NICK>mesage</NICK>
<BLURB>Post a level message for each interval.</BLURB>
<BLURB>Post a level message for each passed interval.</BLURB>
<DEFAULT>TRUE</DEFAULT>
</ARG>
......@@ -4270,12 +4280,12 @@
<ARG>
<NAME>GstLevel::peak-ttl</NAME>
<TYPE>gdouble</TYPE>
<RANGE>[0,100]</RANGE>
<TYPE>guint64</TYPE>
<RANGE></RANGE>
<FLAGS>rw</FLAGS>
<NICK>Peak TTL</NICK>
<BLURB>Time To Live of decay peak before it falls back.</BLURB>
<DEFAULT>0.3</DEFAULT>
<BLURB>Time To Live of decay peak before it falls back (in nanoseconds).</BLURB>
<DEFAULT>300000000</DEFAULT>
</ARG>
<ARG>
......@@ -6498,3 +6508,13 @@
<DEFAULT>2000000000</DEFAULT>
</ARG>
<ARG>
<NAME>GstRtpGSMParse::frequency</NAME>
<TYPE>gint</TYPE>
<RANGE></RANGE>
<FLAGS>rw</FLAGS>
<NICK>frequency</NICK>
<BLURB>frequency.</BLURB>
<DEFAULT>8000</DEFAULT>
</ARG>
<plugin>
<name>alpha</name>
<description>resizes a video by adding borders or cropping</description>
<description>adds an alpha channel to video</description>
<filename>../../gst/alpha/.libs/libgstalpha.so</filename>
<basename>libgstalpha.so</basename>
<version>0.9.1.1</version>
......
......@@ -13,14 +13,14 @@
<name>rtpamrdec</name>
<longname>RTP packet parser</longname>
<class>Codec/Parser/Network</class>
<description>Extracts MPEG audio from RTP packets</description>
<description>Extracts AMR audio from RTP packets (RFC 3267)</description>
<author>Wim Taymans &lt;wim@fluendo.com&gt;</author>
</element>
<element>
<name>rtpamrenc</name>
<longname>RTP packet parser</longname>
<class>Codec/Parser/Network</class>
<description>Encode AMR audio into RTP packets</description>
<description>Encode AMR audio into RTP packets (RFC 3267)</description>
<author>Wim Taymans &lt;wim@fluendo.com&gt;</author>
</element>
<element>
......@@ -30,6 +30,20 @@
<description>Accepts raw RTP and RTCP packets and sends them forward</description>
<author>Wim Taymans &lt;wim@fluendo.com&gt;</author>
</element>
<element>
<name>rtpgsmenc</name>
<longname>RTP GSM Audio Encoder</longname>
<class>Codec/Encoder/Network</class>
<description>Encodes GSM audio into a RTP packet</description>
<author>Zeeshan Ali &lt;zak147@yahoo.com&gt;</author>
</element>
<element>
<name>rtpgsmparse</name>
<longname>RTP packet parser</longname>
<class>Codec/Parser/Network</class>
<description>Extracts GSM audio from RTP packets</description>
<author>Zeeshan Ali &lt;zak147@yahoo.com&gt;</author>
</element>
<element>
<name>rtph263pdec</name>
<longname>RTP packet parser</longname>
......@@ -41,7 +55,7 @@
<name>rtph263penc</name>
<longname>RTP packet parser</longname>
<class>Codec/Parser/Network</class>
<description>Extracts H263+ video from RTP packets</description>
<description>Encodes H263+ video in RTP packets (RFC 2429)</description>
<author>Wim Taymans &lt;wim@fluendo.com&gt;</author>
</element>
<element>
......
......@@ -25,7 +25,7 @@
* <refsect2>
* <para>
* Level analyses incoming audio buffers and, if the
* <link linkend="GstLevel--message">message property</link> is #TRUE.
* <link linkend="GstLevel--message">message property</link> is #TRUE,
* generates an application message named
* <classname>&quot;level&quot;</classname>:
* after each interval of time given by the
......
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