updating docs

Original commit message from CVS:
updating docs

diff --git a/ChangeLog b/ChangeLog
index 74ef22b..5e12562 100644 (file)
--- a/ChangeLog
+++ b/ChangeLog
@@ -1,5 +1,14 @@
 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:
        * check/elements/level.c: (GST_START_TEST):
        * gst/level/Makefile.am:

diff --git a/docs/plugins/gst-plugins-good-plugins.args b/docs/plugins/gst-plugins-good-plugins.args
index 45ebd63..fba4f7a 100644 (file)
--- a/docs/plugins/gst-plugins-good-plugins.args
+++ b/docs/plugins/gst-plugins-good-plugins.args
@@ -4239,13 +4239,23 @@
 </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>
+

diff --git a/docs/plugins/inspect/plugin-alpha.xml b/docs/plugins/inspect/plugin-alpha.xml
index d6a986d..ee50166 100644 (file)
--- a/docs/plugins/inspect/plugin-alpha.xml
+++ b/docs/plugins/inspect/plugin-alpha.xml
@@ -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>

diff --git a/docs/plugins/inspect/plugin-rtp.xml b/docs/plugins/inspect/plugin-rtp.xml
index b04402f..5e04186 100644 (file)
--- a/docs/plugins/inspect/plugin-rtp.xml
+++ b/docs/plugins/inspect/plugin-rtp.xml
@@ -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>
@@ -31,6 +31,20 @@
       <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>
       <class>Codec/Parser/Network</class>
@@ -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>

diff --git a/gst/level/gstlevel.c b/gst/level/gstlevel.c
index f7d22ed..b1d6344 100644 (file)
--- a/gst/level/gstlevel.c
+++ b/gst/level/gstlevel.c
@@ -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