mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-11-23 18:21:04 +00:00
updating docs
Original commit message from CVS: updating docs
This commit is contained in:
Thomas Vander Stichele
parent
6115f0b56c
commit
204755d0ff
5 changed files with 57 additions and 14 deletions
|
|
@ -1,3 +1,12 @@
|
|||
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>
|
||||
|
||||
|
|
|
|||
|
|
@ -1,6 +1,6 @@
|
|||
<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 <wim@fluendo.com></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 <wim@fluendo.com></author>
|
||||
</element>
|
||||
<element>
|
||||
|
|
@ -30,6 +30,20 @@
|
|||
<description>Accepts raw RTP and RTCP packets and sends them forward</description>
|
||||
<author>Wim Taymans <wim@fluendo.com></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 <zak147@yahoo.com></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 <zak147@yahoo.com></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 <wim@fluendo.com></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>"level"</classname>:
|
||||
* after each interval of time given by the
|
||||
|
|
|
|||
Loading…
Reference in a new issue