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

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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.
This commit is contained in:
Tim-Philipp Müller 2007-05-30 09:08:24 +00:00
parent 3a496fd7eb
commit 3e6a09941d
9 changed files with 261 additions and 36 deletions
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14
ChangeLog
View file

@ -1,3 +1,17 @@
2007-05-30 Tim-Philipp Müller <tim at centricular dot net>
* 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.
2007-05-28 Wim Taymans <wim@fluendo.com> 2007-05-28 Wim Taymans <wim@fluendo.com>
* docs/plugins/gst-plugins-bad-plugins-docs.sgml: * docs/plugins/gst-plugins-bad-plugins-docs.sgml:

219
docs/plugins/gst-plugins-bad-plugins.args
View file

@ -450,11 +450,11 @@
<ARG> <ARG>
<NAME>GstWavpackEnc::bitrate</NAME> <NAME>GstWavpackEnc::bitrate</NAME>
<TYPE>gdouble</TYPE> <TYPE>guint</TYPE>
<RANGE>[0,9.6e+06]</RANGE> <RANGE><= 9600000</RANGE>
<FLAGS>rw</FLAGS> <FLAGS>rw</FLAGS>
<NICK>Bitrate</NICK> <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> <DEFAULT>0</DEFAULT>
</ARG> </ARG>
@ -464,7 +464,7 @@
<RANGE>[0,24]</RANGE> <RANGE>[0,24]</RANGE>
<FLAGS>rw</FLAGS> <FLAGS>rw</FLAGS>
<NICK>Bits per sample</NICK> <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> <DEFAULT>0</DEFAULT>
</ARG> </ARG>
@ -473,19 +473,19 @@
<TYPE>GstWavpackEncCorrectionMode</TYPE> <TYPE>GstWavpackEncCorrectionMode</TYPE>
<RANGE></RANGE> <RANGE></RANGE>
<FLAGS>rw</FLAGS> <FLAGS>rw</FLAGS>
<NICK>Correction file mode</NICK> <NICK>Correction stream mode</NICK>
<BLURB>Use this mode for correction file creation. Only works in lossy mode!.</BLURB> <BLURB>Use this mode for the correction stream. Only works in lossy mode!.</BLURB>
<DEFAULT>Create no correction file</DEFAULT> <DEFAULT>Create no correction file</DEFAULT>
</ARG> </ARG>
<ARG> <ARG>
<NAME>GstWavpackEnc::extra-processing</NAME> <NAME>GstWavpackEnc::extra-processing</NAME>
<TYPE>gboolean</TYPE> <TYPE>guint</TYPE>
<RANGE></RANGE> <RANGE><= 6</RANGE>
<FLAGS>rw</FLAGS> <FLAGS>rw</FLAGS>
<NICK>Extra processing</NICK> <NICK>Extra processing</NICK>
<BLURB>Extra encode processing.</BLURB> <BLURB>Use better but slower filters for better compression/quality.</BLURB>
<DEFAULT>FALSE</DEFAULT> <DEFAULT>0</DEFAULT>
</ARG> </ARG>
<ARG> <ARG>
@ -572,7 +572,7 @@
<FLAGS>rw</FLAGS> <FLAGS>rw</FLAGS>
<NICK>automatic-redirect</NICK> <NICK>automatic-redirect</NICK>
<BLURB>Enable Neon HTTP Redirects (HTTP Status Code 302).</BLURB> <BLURB>Enable Neon HTTP Redirects (HTTP Status Code 302).</BLURB>
<DEFAULT>FALSE</DEFAULT> <DEFAULT>TRUE</DEFAULT>
</ARG> </ARG>
<ARG> <ARG>
@ -1072,7 +1072,7 @@
<FLAGS>rw</FLAGS> <FLAGS>rw</FLAGS>
<NICK>Output format</NICK> <NICK>Output format</NICK>
<BLURB>Format of output frames.</BLURB> <BLURB>Format of output frames.</BLURB>
<DEFAULT>OUTPUTFORMAT_ADTS</DEFAULT> <DEFAULT>OUTPUTFORMAT_RAW</DEFAULT>
</ARG> </ARG>
<ARG> <ARG>
@ -1092,7 +1092,7 @@
<FLAGS>rw</FLAGS> <FLAGS>rw</FLAGS>
<NICK>Block type</NICK> <NICK>Block type</NICK>
<BLURB>Block type encorcing.</BLURB> <BLURB>Block type encorcing.</BLURB>
<DEFAULT>SHORTCTL_NOSHORT</DEFAULT> <DEFAULT>SHORTCTL_NORMAL</DEFAULT>
</ARG> </ARG>
<ARG> <ARG>
@ -1200,8 +1200,8 @@
<TYPE>gboolean</TYPE> <TYPE>gboolean</TYPE>
<RANGE></RANGE> <RANGE></RANGE>
<FLAGS>rw</FLAGS> <FLAGS>rw</FLAGS>
<NICK>Force processing</NICK> <NICK>Forced</NICK>
<BLURB>Analyze streams even when ReplayGain tags exist.</BLURB> <BLURB>Analyze even if ReplayGain tags exist.</BLURB>
<DEFAULT>TRUE</DEFAULT> <DEFAULT>TRUE</DEFAULT>
</ARG> </ARG>
@ -1211,17 +1211,17 @@
<RANGE>>= 0</RANGE> <RANGE>>= 0</RANGE>
<FLAGS>rw</FLAGS> <FLAGS>rw</FLAGS>
<NICK>Number of album tracks</NICK> <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> <DEFAULT>0</DEFAULT>
</ARG> </ARG>
<ARG> <ARG>
<NAME>GstRgAnalysis::reference-level</NAME> <NAME>GstRgAnalysis::reference-level</NAME>
<TYPE>gdouble</TYPE> <TYPE>gdouble</TYPE>
<RANGE>>= 0</RANGE> <RANGE>[0,150]</RANGE>
<FLAGS>rw</FLAGS> <FLAGS>rw</FLAGS>
<NICK>Reference level</NICK> <NICK>Reference level</NICK>
<BLURB>Reference level in dB (83.0 for original proposal).</BLURB> <BLURB>Reference level [dB].</BLURB>
<DEFAULT>89</DEFAULT> <DEFAULT>89</DEFAULT>
</ARG> </ARG>
@ -17081,7 +17081,7 @@
<RANGE></RANGE> <RANGE></RANGE>
<FLAGS>rw</FLAGS> <FLAGS>rw</FLAGS>
<NICK>Buffer latency in ms</NICK> <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> <DEFAULT>200</DEFAULT>
</ARG> </ARG>
@ -17114,3 +17114,184 @@
<BLURB>When enabled, the view is fullscreen.</BLURB> <BLURB>When enabled, the view is fullscreen.</BLURB>
<DEFAULT>FALSE</DEFAULT> <DEFAULT>FALSE</DEFAULT>
</ARG> </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>

28
docs/plugins/gst-plugins-bad-plugins.signals
View file

@ -30,6 +30,13 @@ GstRTPSession *gstrtpsession
guint arg1 guint arg1
</SIGNAL> </SIGNAL>
<SIGNAL>
<NAME>GstRTPSession::clear-pt-map</NAME>
<RETURNS>void</RETURNS>
<FLAGS>a</FLAGS>
GstRTPSession *gstrtpsession
</SIGNAL>
<SIGNAL> <SIGNAL>
<NAME>GstRTPPtDemux::new-payload-type</NAME> <NAME>GstRTPPtDemux::new-payload-type</NAME>
<RETURNS>void</RETURNS> <RETURNS>void</RETURNS>
@ -55,6 +62,13 @@ GstRTPPtDemux *gstrtpptdemux
guint arg1 guint arg1
</SIGNAL> </SIGNAL>
<SIGNAL>
<NAME>GstRTPPtDemux::clear-pt-map</NAME>
<RETURNS>void</RETURNS>
<FLAGS>la</FLAGS>
GstRTPPtDemux *gstrtpptdemux
</SIGNAL>
<SIGNAL> <SIGNAL>
<NAME>GstRTPJitterBuffer::request-pt-map</NAME> <NAME>GstRTPJitterBuffer::request-pt-map</NAME>
<RETURNS>GstCaps*</RETURNS> <RETURNS>GstCaps*</RETURNS>
@ -63,6 +77,13 @@ GstRTPJitterBuffer *gstrtpjitterbuffer
guint arg1 guint arg1
</SIGNAL> </SIGNAL>
<SIGNAL>
<NAME>GstRTPJitterBuffer::clear-pt-map</NAME>
<RETURNS>void</RETURNS>
<FLAGS>l</FLAGS>
GstRTPJitterBuffer *gstrtpjitterbuffer
</SIGNAL>
<SIGNAL> <SIGNAL>
<NAME>GstRTPBin::request-pt-map</NAME> <NAME>GstRTPBin::request-pt-map</NAME>
<RETURNS>GstCaps*</RETURNS> <RETURNS>GstCaps*</RETURNS>
@ -72,3 +93,10 @@ guint arg1
guint arg2 guint arg2
</SIGNAL> </SIGNAL>
<SIGNAL>
<NAME>GstRTPBin::clear-pt-map</NAME>
<RETURNS>void</RETURNS>
<FLAGS>a</FLAGS>
GstRTPBin *gstrtpbin
</SIGNAL>

4
docs/plugins/inspect/plugin-dtsdec.xml
View file

@ -3,10 +3,10 @@
<description>Decodes DTS audio streams</description> <description>Decodes DTS audio streams</description>
<filename>../../ext/dts/.libs/libgstdtsdec.so</filename> <filename>../../ext/dts/.libs/libgstdtsdec.so</filename>
<basename>libgstdtsdec.so</basename> <basename>libgstdtsdec.so</basename>
<version>0.10.4</version> <version>0.10.4.1</version>
<license>GPL</license> <license>GPL</license>
<source>gst-plugins-bad</source> <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> <origin>Unknown package origin</origin>
<elements> <elements>
<element> <element>

14
docs/plugins/inspect/plugin-rtpmanager.xml → docs/plugins/inspect/plugin-gstrtpmanager.xml
View file

@ -1,5 +1,5 @@
<plugin> <plugin>
<name>rtpmanager</name> <name>gstrtpmanager</name>
<description>RTP session management plugin library</description> <description>RTP session management plugin library</description>
<filename>../../gst/rtpmanager/.libs/libgstrtpmanager.so</filename> <filename>../../gst/rtpmanager/.libs/libgstrtpmanager.so</filename>
<basename>libgstrtpmanager.so</basename> <basename>libgstrtpmanager.so</basename>
@ -10,42 +10,42 @@
<origin>Unknown package origin</origin> <origin>Unknown package origin</origin>
<elements> <elements>
<element> <element>
<name>rtpbin</name> <name>gstrtpbin</name>
<longname>RTP Bin</longname> <longname>RTP Bin</longname>
<class>Filter/Network/RTP</class> <class>Filter/Network/RTP</class>
<description>Implement an RTP bin</description> <description>Implement an RTP bin</description>
<author>Wim Taymans &lt;wim@fluendo.com&gt;</author> <author>Wim Taymans &lt;wim@fluendo.com&gt;</author>
</element> </element>
<element> <element>
<name>rtpclient</name> <name>gstrtpclient</name>
<longname>RTP Client</longname> <longname>RTP Client</longname>
<class>Filter/Network/RTP</class> <class>Filter/Network/RTP</class>
<description>Implement an RTP client</description> <description>Implement an RTP client</description>
<author>Wim Taymans &lt;wim@fluendo.com&gt;</author> <author>Wim Taymans &lt;wim@fluendo.com&gt;</author>
</element> </element>
<element> <element>
<name>rtpjitterbuffer</name> <name>gstrtpjitterbuffer</name>
<longname>RTP packet jitter-buffer</longname> <longname>RTP packet jitter-buffer</longname>
<class>Filter/Network/RTP</class> <class>Filter/Network/RTP</class>
<description>A buffer that deals with network jitter and other transmission faults</description> <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> <author>Philippe Kalaf &lt;philippe.kalaf@collabora.co.uk&gt;, Wim Taymans &lt;wim@fluendo.com&gt;</author>
</element> </element>
<element> <element>
<name>rtpptdemux</name> <name>gstrtpptdemux</name>
<longname>RTP Demux</longname> <longname>RTP Demux</longname>
<class>Demux/Network/RTP</class> <class>Demux/Network/RTP</class>
<description>Parses codec streams transmitted in the same RTP session</description> <description>Parses codec streams transmitted in the same RTP session</description>
<author>Kai Vehmanen &lt;kai.vehmanen@nokia.com&gt;</author> <author>Kai Vehmanen &lt;kai.vehmanen@nokia.com&gt;</author>
</element> </element>
<element> <element>
<name>rtpsession</name> <name>gstrtpsession</name>
<longname>RTP Session</longname> <longname>RTP Session</longname>
<class>Filter/Network/RTP</class> <class>Filter/Network/RTP</class>
<description>Implement an RTP session</description> <description>Implement an RTP session</description>
<author>Wim Taymans &lt;wim@fluendo.com&gt;</author> <author>Wim Taymans &lt;wim@fluendo.com&gt;</author>
</element> </element>
<element> <element>
<name>rtpssrcdemux</name> <name>gstrtpssrcdemux</name>
<longname>RTP SSRC Demux</longname> <longname>RTP SSRC Demux</longname>
<class>Demux/Network/RTP</class> <class>Demux/Network/RTP</class>
<description>Splits RTP streams based on the SSRC</description> <description>Splits RTP streams based on the SSRC</description>

2
docs/plugins/inspect/plugin-musepack.xml
View file

@ -3,7 +3,7 @@
<description>Musepack decoder</description> <description>Musepack decoder</description>
<filename>../../ext/musepack/.libs/libgstmusepack.so</filename> <filename>../../ext/musepack/.libs/libgstmusepack.so</filename>
<basename>libgstmusepack.so</basename> <basename>libgstmusepack.so</basename>
<version>0.10.3.1</version> <version>0.10.4.1</version>
<license>LGPL</license> <license>LGPL</license>
<source>gst-plugins-bad</source> <source>gst-plugins-bad</source>
<package>GStreamer Bad Plug-ins CVS/prerelease</package> <package>GStreamer Bad Plug-ins CVS/prerelease</package>

10
docs/plugins/inspect/plugin-sdl.xml
View file

@ -1,9 +1,9 @@
<plugin> <plugin>
<name>sdl</name> <name>sdl</name>
<description>SDL (Simple DirectMedia Layer) support for GStreamer</description> <description>SDL (Simple DirectMedia Layer) support for GStreamer</description>
<filename>../../ext/sdl/.libs/libgstsdlvideosink.so</filename> <filename>../../ext/sdl/.libs/libgstsdl.so</filename>
<basename>libgstsdlvideosink.so</basename> <basename>libgstsdl.so</basename>
<version>0.10.3.1</version> <version>0.10.4.1</version>
<license>LGPL</license> <license>LGPL</license>
<source>gst-plugins-bad</source> <source>gst-plugins-bad</source>
<package>GStreamer Bad Plug-ins CVS/prerelease</package> <package>GStreamer Bad Plug-ins CVS/prerelease</package>
@ -21,7 +21,9 @@
<longname>SDL video sink</longname> <longname>SDL video sink</longname>
<class>Sink/Video</class> <class>Sink/Video</class>
<description>An SDL-based videosink</description> <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> </element>
</elements> </elements>
</plugin> </plugin>

2
docs/plugins/inspect/plugin-spcdec.xml
View file

@ -3,7 +3,7 @@
<description>OpenSPC Audio Decoder</description> <description>OpenSPC Audio Decoder</description>
<filename>../../ext/spc/.libs/libgstspc.so</filename> <filename>../../ext/spc/.libs/libgstspc.so</filename>
<basename>libgstspc.so</basename> <basename>libgstspc.so</basename>
<version>0.10.3.1</version> <version>0.10.4.1</version>
<license>LGPL</license> <license>LGPL</license>
<source>gst-plugins-bad</source> <source>gst-plugins-bad</source>
<package>GStreamer Bad Plug-ins CVS/prerelease</package> <package>GStreamer Bad Plug-ins CVS/prerelease</package>

4
docs/plugins/inspect/plugin-swfdec.xml
View file

@ -3,10 +3,10 @@
<description>Uses libswfdec to decode Flash video streams</description> <description>Uses libswfdec to decode Flash video streams</description>
<filename>../../ext/swfdec/.libs/libgstswfdec.so</filename> <filename>../../ext/swfdec/.libs/libgstswfdec.so</filename>
<basename>libgstswfdec.so</basename> <basename>libgstswfdec.so</basename>
<version>0.10.4</version> <version>0.10.4.1</version>
<license>LGPL</license> <license>LGPL</license>
<source>gst-plugins-bad</source> <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> <origin>Unknown package origin</origin>
<elements> <elements>
<element> <element>