diff options
| -rw-r--r-- | ChangeLog | 14 | ||||
| -rw-r--r-- | docs/plugins/gst-plugins-bad-plugins.args | 219 | ||||
| -rw-r--r-- | docs/plugins/gst-plugins-bad-plugins.signals | 28 | ||||
| -rw-r--r-- | docs/plugins/inspect/plugin-dtsdec.xml | 4 | ||||
| -rw-r--r-- | docs/plugins/inspect/plugin-gstrtpmanager.xml (renamed from docs/plugins/inspect/plugin-rtpmanager.xml) | 14 | ||||
| -rw-r--r-- | docs/plugins/inspect/plugin-musepack.xml | 2 | ||||
| -rw-r--r-- | docs/plugins/inspect/plugin-sdl.xml | 10 | ||||
| -rw-r--r-- | docs/plugins/inspect/plugin-spcdec.xml | 2 | ||||
| -rw-r--r-- | docs/plugins/inspect/plugin-swfdec.xml | 4 |
9 files changed, 261 insertions, 36 deletions
@@ -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> * docs/plugins/gst-plugins-bad-plugins-docs.sgml: 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 <wim@fluendo.com></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 <wim@fluendo.com></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 <philippe.kalaf@collabora.co.uk>, Wim Taymans <wim@fluendo.com></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 <kai.vehmanen@nokia.com></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 <wim@fluendo.com></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 <rbultje@ronald.bitfreak.net>Edgard Lima <edgard.lima@indt.org.br>Jan Schmidt <thaytan@mad.scientist.com></author> + <author>Ronald Bultje <rbultje@ronald.bitfreak.net> + Edgard Lima <edgard.lima@indt.org.br> + Jan Schmidt <thaytan@mad.scientist.com></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> |