docs/plugins/: Updates; update inspect info for rtpmanager => gstrtpmanager rename, hopefully that makes the build bo...

Original commit message from CVS:
* docs/plugins/gst-plugins-bad-plugins.args:
* docs/plugins/gst-plugins-bad-plugins.signals:
* docs/plugins/inspect/plugin-dtsdec.xml:
* docs/plugins/inspect/plugin-gstrtpmanager.xml:
* docs/plugins/inspect/plugin-musepack.xml:
* docs/plugins/inspect/plugin-rtpmanager.xml:
* docs/plugins/inspect/plugin-sdl.xml:
* docs/plugins/inspect/plugin-spcdec.xml:
* docs/plugins/inspect/plugin-swfdec.xml:
Updates; update inspect info for rtpmanager => gstrtpmanager rename,
hopefully that makes the build bots happy again.

8 files changed, 247 insertions, 36 deletions

diff --git a/docs/plugins/gst-plugins-bad-plugins.args b/docs/plugins/gst-plugins-bad-plugins.args
index 3315db1d..20c5f6ee 100644
--- a/docs/plugins/gst-plugins-bad-plugins.args
+++ b/docs/plugins/gst-plugins-bad-plugins.args
@@ -450,11 +450,11 @@
 
 <ARG>
 <NAME>GstWavpackEnc::bitrate</NAME>
-<TYPE>gdouble</TYPE>
-<RANGE>[0,9.6e+06]</RANGE>
+<TYPE>guint</TYPE>
+<RANGE><= 9600000</RANGE>
 <FLAGS>rw</FLAGS>
 <NICK>Bitrate</NICK>
-<BLURB>Try to encode with this average bitrate (bits/sec). This enables lossy encoding! A value smaller than 24000.0 disables this.</BLURB>
+<BLURB>Try to encode with this average bitrate (bits/sec). This enables lossy encoding, values smaller than 24000 disable it again.</BLURB>
 <DEFAULT>0</DEFAULT>
 </ARG>
 
@@ -464,7 +464,7 @@
 <RANGE>[0,24]</RANGE>
 <FLAGS>rw</FLAGS>
 <NICK>Bits per sample</NICK>
-<BLURB>Try to encode with this amount of bits per sample. This enables lossy encoding! A value smaller than 2.0 disables this.</BLURB>
+<BLURB>Try to encode with this amount of bits per sample. This enables lossy encoding, values smaller than 2.0 disable it again.</BLURB>
 <DEFAULT>0</DEFAULT>
 </ARG>
 
@@ -473,19 +473,19 @@
 <TYPE>GstWavpackEncCorrectionMode</TYPE>
 <RANGE></RANGE>
 <FLAGS>rw</FLAGS>
-<NICK>Correction file mode</NICK>
-<BLURB>Use this mode for correction file creation. Only works in lossy mode!.</BLURB>
+<NICK>Correction stream mode</NICK>
+<BLURB>Use this mode for the correction stream. Only works in lossy mode!.</BLURB>
 <DEFAULT>Create no correction file</DEFAULT>
 </ARG>
 
 <ARG>
 <NAME>GstWavpackEnc::extra-processing</NAME>
-<TYPE>gboolean</TYPE>
-<RANGE></RANGE>
+<TYPE>guint</TYPE>
+<RANGE><= 6</RANGE>
 <FLAGS>rw</FLAGS>
 <NICK>Extra processing</NICK>
-<BLURB>Extra encode processing.</BLURB>
-<DEFAULT>FALSE</DEFAULT>
+<BLURB>Use better but slower filters for better compression/quality.</BLURB>
+<DEFAULT>0</DEFAULT>
 </ARG>
 
 <ARG>
@@ -572,7 +572,7 @@
 <FLAGS>rw</FLAGS>
 <NICK>automatic-redirect</NICK>
 <BLURB>Enable Neon HTTP Redirects (HTTP Status Code 302).</BLURB>
-<DEFAULT>FALSE</DEFAULT>
+<DEFAULT>TRUE</DEFAULT>
 </ARG>
 
 <ARG>
@@ -1072,7 +1072,7 @@
 <FLAGS>rw</FLAGS>
 <NICK>Output format</NICK>
 <BLURB>Format of output frames.</BLURB>
-<DEFAULT>OUTPUTFORMAT_ADTS</DEFAULT>
+<DEFAULT>OUTPUTFORMAT_RAW</DEFAULT>
 </ARG>
 
 <ARG>
@@ -1092,7 +1092,7 @@
 <FLAGS>rw</FLAGS>
 <NICK>Block type</NICK>
 <BLURB>Block type encorcing.</BLURB>
-<DEFAULT>SHORTCTL_NOSHORT</DEFAULT>
+<DEFAULT>SHORTCTL_NORMAL</DEFAULT>
 </ARG>
 
 <ARG>
@@ -1200,8 +1200,8 @@
 <TYPE>gboolean</TYPE>
 <RANGE></RANGE>
 <FLAGS>rw</FLAGS>
-<NICK>Force processing</NICK>
-<BLURB>Analyze streams even when ReplayGain tags exist.</BLURB>
+<NICK>Forced</NICK>
+<BLURB>Analyze even if ReplayGain tags exist.</BLURB>
 <DEFAULT>TRUE</DEFAULT>
 </ARG>
 
@@ -1211,17 +1211,17 @@
 <RANGE>>= 0</RANGE>
 <FLAGS>rw</FLAGS>
 <NICK>Number of album tracks</NICK>
-<BLURB>Number of remaining tracks in the album.</BLURB>
+<BLURB>Number of remaining album tracks.</BLURB>
 <DEFAULT>0</DEFAULT>
 </ARG>
 
 <ARG>
 <NAME>GstRgAnalysis::reference-level</NAME>
 <TYPE>gdouble</TYPE>
-<RANGE>>= 0</RANGE>
+<RANGE>[0,150]</RANGE>
 <FLAGS>rw</FLAGS>
 <NICK>Reference level</NICK>
-<BLURB>Reference level in dB (83.0 for original proposal).</BLURB>
+<BLURB>Reference level [dB].</BLURB>
 <DEFAULT>89</DEFAULT>
 </ARG>
 
@@ -17081,7 +17081,7 @@
 <RANGE></RANGE>
 <FLAGS>rw</FLAGS>
 <NICK>Buffer latency in ms</NICK>
-<BLURB>Amount of ms to buffer.</BLURB>
+<BLURB>Default amount of ms to buffer in the jitterbuffers.</BLURB>
 <DEFAULT>200</DEFAULT>
 </ARG>
 
@@ -17114,3 +17114,184 @@
 <BLURB>When enabled, the view is fullscreen.</BLURB>
 <DEFAULT>FALSE</DEFAULT>
 </ARG>
+
+<ARG>
+<NAME>GstSFSrc::location</NAME>
+<TYPE>gchararray</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>File Location</NICK>
+<BLURB>Location of the file to read.</BLURB>
+<DEFAULT>NULL</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstSFSink::buffer-frames</NAME>
+<TYPE>gint</TYPE>
+<RANGE>>= 1</RANGE>
+<FLAGS>rwx</FLAGS>
+<NICK>Buffer frames</NICK>
+<BLURB>Number of frames per buffer, in pull mode.</BLURB>
+<DEFAULT>256</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstSFSink::location</NAME>
+<TYPE>gchararray</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>File Location</NICK>
+<BLURB>Location of the file to write.</BLURB>
+<DEFAULT>NULL</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstSFSink::major-type</NAME>
+<TYPE>GstSndfileMajorTypes</TYPE>
+<RANGE></RANGE>
+<FLAGS>rwx</FLAGS>
+<NICK>Major type</NICK>
+<BLURB>Major output type.</BLURB>
+<DEFAULT>WAV (Microsoft)</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstSFSink::minor-type</NAME>
+<TYPE>GstSndfileMinorTypes</TYPE>
+<RANGE></RANGE>
+<FLAGS>rwx</FLAGS>
+<NICK>Minor type</NICK>
+<BLURB>Minor output type.</BLURB>
+<DEFAULT>32 bit float</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstSwitch::active-pad</NAME>
+<TYPE>gchararray</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Active Pad</NICK>
+<BLURB>Name of the currently active sink pad.</BLURB>
+<DEFAULT>NULL</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstSwitch::last-timestamp</NAME>
+<TYPE>guint64</TYPE>
+<RANGE><= G_MAXUINT</RANGE>
+<FLAGS>r</FLAGS>
+<NICK>Time at the end of the last buffer</NICK>
+<BLURB>Time at the end of the last buffer.</BLURB>
+<DEFAULT>0</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstSwitch::num-sources</NAME>
+<TYPE>guint</TYPE>
+<RANGE></RANGE>
+<FLAGS>r</FLAGS>
+<NICK>number of sources</NICK>
+<BLURB>number of sources.</BLURB>
+<DEFAULT>0</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstSwitch::queue-buffers</NAME>
+<TYPE>gboolean</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Queue new segment and buffers instead of sending them</NICK>
+<BLURB>Queue new segment and buffers instead of sending them.</BLURB>
+<DEFAULT>FALSE</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstSwitch::start-value</NAME>
+<TYPE>guint64</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Start Value</NICK>
+<BLURB>Timestamp that next segment will start at (-1 to use first buffer).</BLURB>
+<DEFAULT>18446744073709551615</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstSwitch::stop-value</NAME>
+<TYPE>guint64</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Stop Value</NICK>
+<BLURB>Timestamp that current source will stop at (-1 if unknown or don't care).</BLURB>
+<DEFAULT>18446744073709551615</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstRgVolume::album-mode</NAME>
+<TYPE>gboolean</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Album mode</NICK>
+<BLURB>Prefer album over track gain.</BLURB>
+<DEFAULT>TRUE</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstRgVolume::fallback-gain</NAME>
+<TYPE>gdouble</TYPE>
+<RANGE>[-60,60]</RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Fallback gain</NICK>
+<BLURB>Gain for streams missing tags [dB].</BLURB>
+<DEFAULT>0</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstRgVolume::headroom</NAME>
+<TYPE>gdouble</TYPE>
+<RANGE>[0,60]</RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Headroom</NICK>
+<BLURB>Extra headroom [dB].</BLURB>
+<DEFAULT>0</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstRgVolume::pre-amp</NAME>
+<TYPE>gdouble</TYPE>
+<RANGE>[-60,60]</RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Pre-amp</NICK>
+<BLURB>Extra gain [dB].</BLURB>
+<DEFAULT>0</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstRgVolume::result-gain</NAME>
+<TYPE>gdouble</TYPE>
+<RANGE>[-120,120]</RANGE>
+<FLAGS>r</FLAGS>
+<NICK>Result-gain</NICK>
+<BLURB>Applied gain [dB].</BLURB>
+<DEFAULT>0</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstRgVolume::target-gain</NAME>
+<TYPE>gdouble</TYPE>
+<RANGE>[-120,120]</RANGE>
+<FLAGS>r</FLAGS>
+<NICK>Target-gain</NICK>
+<BLURB>Applicable gain [dB].</BLURB>
+<DEFAULT>0</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstRgLimiter::enabled</NAME>
+<TYPE>gboolean</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Enabled</NICK>
+<BLURB>Enable processing.</BLURB>
+<DEFAULT>TRUE</DEFAULT>
+</ARG>
+
diff --git a/docs/plugins/gst-plugins-bad-plugins.signals b/docs/plugins/gst-plugins-bad-plugins.signals
index 92c41b8b..bb75aa53 100644
--- a/docs/plugins/gst-plugins-bad-plugins.signals
+++ b/docs/plugins/gst-plugins-bad-plugins.signals
@@ -31,6 +31,13 @@ guint  arg1
 </SIGNAL>
 
 <SIGNAL>
+<NAME>GstRTPSession::clear-pt-map</NAME>
+<RETURNS>void</RETURNS>
+<FLAGS>a</FLAGS>
+GstRTPSession *gstrtpsession
+</SIGNAL>
+
+<SIGNAL>
 <NAME>GstRTPPtDemux::new-payload-type</NAME>
 <RETURNS>void</RETURNS>
 <FLAGS>l</FLAGS>
@@ -56,6 +63,13 @@ guint  arg1
 </SIGNAL>
 
 <SIGNAL>
+<NAME>GstRTPPtDemux::clear-pt-map</NAME>
+<RETURNS>void</RETURNS>
+<FLAGS>la</FLAGS>
+GstRTPPtDemux *gstrtpptdemux
+</SIGNAL>
+
+<SIGNAL>
 <NAME>GstRTPJitterBuffer::request-pt-map</NAME>
 <RETURNS>GstCaps*</RETURNS>
 <FLAGS>l</FLAGS>
@@ -64,6 +78,13 @@ guint  arg1
 </SIGNAL>
 
 <SIGNAL>
+<NAME>GstRTPJitterBuffer::clear-pt-map</NAME>
+<RETURNS>void</RETURNS>
+<FLAGS>l</FLAGS>
+GstRTPJitterBuffer *gstrtpjitterbuffer
+</SIGNAL>
+
+<SIGNAL>
 <NAME>GstRTPBin::request-pt-map</NAME>
 <RETURNS>GstCaps*</RETURNS>
 <FLAGS>l</FLAGS>
@@ -72,3 +93,10 @@ guint  arg1
 guint  arg2
 </SIGNAL>
 
+<SIGNAL>
+<NAME>GstRTPBin::clear-pt-map</NAME>
+<RETURNS>void</RETURNS>
+<FLAGS>a</FLAGS>
+GstRTPBin *gstrtpbin
+</SIGNAL>
+
diff --git a/docs/plugins/inspect/plugin-dtsdec.xml b/docs/plugins/inspect/plugin-dtsdec.xml
index 00573d4d..4471e5f2 100644
--- a/docs/plugins/inspect/plugin-dtsdec.xml
+++ b/docs/plugins/inspect/plugin-dtsdec.xml
@@ -3,10 +3,10 @@
   <description>Decodes DTS audio streams</description>
   <filename>../../ext/dts/.libs/libgstdtsdec.so</filename>
   <basename>libgstdtsdec.so</basename>
-  <version>0.10.4</version>
+  <version>0.10.4.1</version>
   <license>GPL</license>
   <source>gst-plugins-bad</source>
-  <package>GStreamer Bad Plug-ins source release</package>
+  <package>GStreamer Bad Plug-ins CVS/prerelease</package>
   <origin>Unknown package origin</origin>
   <elements>
     <element>
diff --git a/docs/plugins/inspect/plugin-rtpmanager.xml b/docs/plugins/inspect/plugin-gstrtpmanager.xml
index e522f49b..e3f0eb3f 100644
--- a/docs/plugins/inspect/plugin-rtpmanager.xml
+++ b/docs/plugins/inspect/plugin-gstrtpmanager.xml
@@ -1,5 +1,5 @@
 <plugin>
-  <name>rtpmanager</name>
+  <name>gstrtpmanager</name>
   <description>RTP session management plugin library</description>
   <filename>../../gst/rtpmanager/.libs/libgstrtpmanager.so</filename>
   <basename>libgstrtpmanager.so</basename>
@@ -10,42 +10,42 @@
   <origin>Unknown package origin</origin>
   <elements>
     <element>
-      <name>rtpbin</name>
+      <name>gstrtpbin</name>
       <longname>RTP Bin</longname>
       <class>Filter/Network/RTP</class>
       <description>Implement an RTP bin</description>
       <author>Wim Taymans &lt;wim@fluendo.com&gt;</author>
     </element>
     <element>
-      <name>rtpclient</name>
+      <name>gstrtpclient</name>
       <longname>RTP Client</longname>
       <class>Filter/Network/RTP</class>
       <description>Implement an RTP client</description>
       <author>Wim Taymans &lt;wim@fluendo.com&gt;</author>
     </element>
     <element>
-      <name>rtpjitterbuffer</name>
+      <name>gstrtpjitterbuffer</name>
       <longname>RTP packet jitter-buffer</longname>
       <class>Filter/Network/RTP</class>
       <description>A buffer that deals with network jitter and other transmission faults</description>
       <author>Philippe Kalaf &lt;philippe.kalaf@collabora.co.uk&gt;, Wim Taymans &lt;wim@fluendo.com&gt;</author>
     </element>
     <element>
-      <name>rtpptdemux</name>
+      <name>gstrtpptdemux</name>
       <longname>RTP Demux</longname>
       <class>Demux/Network/RTP</class>
       <description>Parses codec streams transmitted in the same RTP session</description>
       <author>Kai Vehmanen &lt;kai.vehmanen@nokia.com&gt;</author>
     </element>
     <element>
-      <name>rtpsession</name>
+      <name>gstrtpsession</name>
       <longname>RTP Session</longname>
       <class>Filter/Network/RTP</class>
       <description>Implement an RTP session</description>
       <author>Wim Taymans &lt;wim@fluendo.com&gt;</author>
     </element>
     <element>
-      <name>rtpssrcdemux</name>
+      <name>gstrtpssrcdemux</name>
       <longname>RTP SSRC Demux</longname>
       <class>Demux/Network/RTP</class>
       <description>Splits RTP streams based on the SSRC</description>
diff --git a/docs/plugins/inspect/plugin-musepack.xml b/docs/plugins/inspect/plugin-musepack.xml
index 093cf759..69ea1cea 100644
--- a/docs/plugins/inspect/plugin-musepack.xml
+++ b/docs/plugins/inspect/plugin-musepack.xml
@@ -3,7 +3,7 @@
   <description>Musepack decoder</description>
   <filename>../../ext/musepack/.libs/libgstmusepack.so</filename>
   <basename>libgstmusepack.so</basename>
-  <version>0.10.3.1</version>
+  <version>0.10.4.1</version>
   <license>LGPL</license>
   <source>gst-plugins-bad</source>
   <package>GStreamer Bad Plug-ins CVS/prerelease</package>
diff --git a/docs/plugins/inspect/plugin-sdl.xml b/docs/plugins/inspect/plugin-sdl.xml
index d7c13390..b0e74ad7 100644
--- a/docs/plugins/inspect/plugin-sdl.xml
+++ b/docs/plugins/inspect/plugin-sdl.xml
@@ -1,9 +1,9 @@
 <plugin>
   <name>sdl</name>
   <description>SDL (Simple DirectMedia Layer) support for GStreamer</description>
-  <filename>../../ext/sdl/.libs/libgstsdlvideosink.so</filename>
-  <basename>libgstsdlvideosink.so</basename>
-  <version>0.10.3.1</version>
+  <filename>../../ext/sdl/.libs/libgstsdl.so</filename>
+  <basename>libgstsdl.so</basename>
+  <version>0.10.4.1</version>
   <license>LGPL</license>
   <source>gst-plugins-bad</source>
   <package>GStreamer Bad Plug-ins CVS/prerelease</package>
@@ -21,7 +21,9 @@
       <longname>SDL video sink</longname>
       <class>Sink/Video</class>
       <description>An SDL-based videosink</description>
-      <author>Ronald Bultje &lt;rbultje@ronald.bitfreak.net&gt;Edgard Lima &lt;edgard.lima@indt.org.br&gt;Jan Schmidt &lt;thaytan@mad.scientist.com&gt;</author>
+      <author>Ronald Bultje &lt;rbultje@ronald.bitfreak.net&gt;
+    Edgard Lima &lt;edgard.lima@indt.org.br&gt;
+    Jan Schmidt &lt;thaytan@mad.scientist.com&gt;</author>
     </element>
   </elements>
 </plugin>
\ No newline at end of file
diff --git a/docs/plugins/inspect/plugin-spcdec.xml b/docs/plugins/inspect/plugin-spcdec.xml
index 0da64a70..d7b8257e 100644
--- a/docs/plugins/inspect/plugin-spcdec.xml
+++ b/docs/plugins/inspect/plugin-spcdec.xml
@@ -3,7 +3,7 @@
   <description>OpenSPC Audio Decoder</description>
   <filename>../../ext/spc/.libs/libgstspc.so</filename>
   <basename>libgstspc.so</basename>
-  <version>0.10.3.1</version>
+  <version>0.10.4.1</version>
   <license>LGPL</license>
   <source>gst-plugins-bad</source>
   <package>GStreamer Bad Plug-ins CVS/prerelease</package>
diff --git a/docs/plugins/inspect/plugin-swfdec.xml b/docs/plugins/inspect/plugin-swfdec.xml
index 8d517e5e..938cb0f8 100644
--- a/docs/plugins/inspect/plugin-swfdec.xml
+++ b/docs/plugins/inspect/plugin-swfdec.xml
@@ -3,10 +3,10 @@
   <description>Uses libswfdec to decode Flash video streams</description>
   <filename>../../ext/swfdec/.libs/libgstswfdec.so</filename>
   <basename>libgstswfdec.so</basename>
-  <version>0.10.4</version>
+  <version>0.10.4.1</version>
   <license>LGPL</license>
   <source>gst-plugins-bad</source>
-  <package>GStreamer Bad Plug-ins source release</package>
+  <package>GStreamer Bad Plug-ins CVS/prerelease</package>
   <origin>Unknown package origin</origin>
   <elements>
     <element>