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Tim-Philipp Müller
gst-plugins-good
Commits
204755d0
Commit
204755d0
authored
Sep 23, 2005
by
Thomas Vander Stichele
Browse files
updating docs
Original commit message from CVS: updating docs
parent
6115f0b5
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ChangeLog
View file @
204755d0
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:
...
...
docs/plugins/gst-plugins-good-plugins.args
View file @
204755d0
...
...
@@ -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>g
double
</TYPE>
<RANGE>
[0.01,100]
</RANGE>
<TYPE>g
uint64
</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
nano
seconds).</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>g
double
</TYPE>
<RANGE>
[0,100]
</RANGE>
<TYPE>g
uint64
</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>
docs/plugins/inspect/plugin-alpha.xml
View file @
204755d0
<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>
...
...
docs/plugins/inspect/plugin-rtp.xml
View file @
204755d0
...
...
@@ -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>
E
xtract
s H263+ video
from
RTP packets
</description>
<description>
E
ncode
s H263+ video
in
RTP packets
(RFC 2429)
</description>
<author>
Wim Taymans
<
wim@fluendo.com
>
</author>
</element>
<element>
...
...
gst/level/gstlevel.c
View file @
204755d0
...
...
@@ -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
...
...
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