diff --git a/ChangeLog b/ChangeLog index bbdfd843d6..d5c3a4ae35 100644 --- a/ChangeLog +++ b/ChangeLog @@ -1,3 +1,17 @@ +2007-05-30 Tim-Philipp Müller + + * 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 * 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 3315db1de6..20c5f6ee2e 100644 --- a/docs/plugins/gst-plugins-bad-plugins.args +++ b/docs/plugins/gst-plugins-bad-plugins.args @@ -450,11 +450,11 @@ GstWavpackEnc::bitrate -gdouble -[0,9.6e+06] +guint +<= 9600000 rw Bitrate -Try to encode with this average bitrate (bits/sec). This enables lossy encoding! A value smaller than 24000.0 disables this. +Try to encode with this average bitrate (bits/sec). This enables lossy encoding, values smaller than 24000 disable it again. 0 @@ -464,7 +464,7 @@ [0,24] rw Bits per sample -Try to encode with this amount of bits per sample. This enables lossy encoding! A value smaller than 2.0 disables this. +Try to encode with this amount of bits per sample. This enables lossy encoding, values smaller than 2.0 disable it again. 0 @@ -473,19 +473,19 @@ GstWavpackEncCorrectionMode rw -Correction file mode -Use this mode for correction file creation. Only works in lossy mode!. +Correction stream mode +Use this mode for the correction stream. Only works in lossy mode!. Create no correction file GstWavpackEnc::extra-processing -gboolean - +guint +<= 6 rw Extra processing -Extra encode processing. -FALSE +Use better but slower filters for better compression/quality. +0 @@ -572,7 +572,7 @@ rw automatic-redirect Enable Neon HTTP Redirects (HTTP Status Code 302). -FALSE +TRUE @@ -1072,7 +1072,7 @@ rw Output format Format of output frames. -OUTPUTFORMAT_ADTS +OUTPUTFORMAT_RAW @@ -1092,7 +1092,7 @@ rw Block type Block type encorcing. -SHORTCTL_NOSHORT +SHORTCTL_NORMAL @@ -1200,8 +1200,8 @@ gboolean rw -Force processing -Analyze streams even when ReplayGain tags exist. +Forced +Analyze even if ReplayGain tags exist. TRUE @@ -1211,17 +1211,17 @@ >= 0 rw Number of album tracks -Number of remaining tracks in the album. +Number of remaining album tracks. 0 GstRgAnalysis::reference-level gdouble ->= 0 +[0,150] rw Reference level -Reference level in dB (83.0 for original proposal). +Reference level [dB]. 89 @@ -17081,7 +17081,7 @@ rw Buffer latency in ms -Amount of ms to buffer. +Default amount of ms to buffer in the jitterbuffers. 200 @@ -17114,3 +17114,184 @@ When enabled, the view is fullscreen. FALSE + + +GstSFSrc::location +gchararray + +rw +File Location +Location of the file to read. +NULL + + + +GstSFSink::buffer-frames +gint +>= 1 +rwx +Buffer frames +Number of frames per buffer, in pull mode. +256 + + + +GstSFSink::location +gchararray + +rw +File Location +Location of the file to write. +NULL + + + +GstSFSink::major-type +GstSndfileMajorTypes + +rwx +Major type +Major output type. +WAV (Microsoft) + + + +GstSFSink::minor-type +GstSndfileMinorTypes + +rwx +Minor type +Minor output type. +32 bit float + + + +GstSwitch::active-pad +gchararray + +rw +Active Pad +Name of the currently active sink pad. +NULL + + + +GstSwitch::last-timestamp +guint64 +<= G_MAXUINT +r +Time at the end of the last buffer +Time at the end of the last buffer. +0 + + + +GstSwitch::num-sources +guint + +r +number of sources +number of sources. +0 + + + +GstSwitch::queue-buffers +gboolean + +rw +Queue new segment and buffers instead of sending them +Queue new segment and buffers instead of sending them. +FALSE + + + +GstSwitch::start-value +guint64 + +rw +Start Value +Timestamp that next segment will start at (-1 to use first buffer). +18446744073709551615 + + + +GstSwitch::stop-value +guint64 + +rw +Stop Value +Timestamp that current source will stop at (-1 if unknown or don't care). +18446744073709551615 + + + +GstRgVolume::album-mode +gboolean + +rw +Album mode +Prefer album over track gain. +TRUE + + + +GstRgVolume::fallback-gain +gdouble +[-60,60] +rw +Fallback gain +Gain for streams missing tags [dB]. +0 + + + +GstRgVolume::headroom +gdouble +[0,60] +rw +Headroom +Extra headroom [dB]. +0 + + + +GstRgVolume::pre-amp +gdouble +[-60,60] +rw +Pre-amp +Extra gain [dB]. +0 + + + +GstRgVolume::result-gain +gdouble +[-120,120] +r +Result-gain +Applied gain [dB]. +0 + + + +GstRgVolume::target-gain +gdouble +[-120,120] +r +Target-gain +Applicable gain [dB]. +0 + + + +GstRgLimiter::enabled +gboolean + +rw +Enabled +Enable processing. +TRUE + + diff --git a/docs/plugins/gst-plugins-bad-plugins.signals b/docs/plugins/gst-plugins-bad-plugins.signals index 92c41b8b0f..bb75aa5383 100644 --- a/docs/plugins/gst-plugins-bad-plugins.signals +++ b/docs/plugins/gst-plugins-bad-plugins.signals @@ -30,6 +30,13 @@ GstRTPSession *gstrtpsession guint arg1 + +GstRTPSession::clear-pt-map +void +a +GstRTPSession *gstrtpsession + + GstRTPPtDemux::new-payload-type void @@ -55,6 +62,13 @@ GstRTPPtDemux *gstrtpptdemux guint arg1 + +GstRTPPtDemux::clear-pt-map +void +la +GstRTPPtDemux *gstrtpptdemux + + GstRTPJitterBuffer::request-pt-map GstCaps* @@ -63,6 +77,13 @@ GstRTPJitterBuffer *gstrtpjitterbuffer guint arg1 + +GstRTPJitterBuffer::clear-pt-map +void +l +GstRTPJitterBuffer *gstrtpjitterbuffer + + GstRTPBin::request-pt-map GstCaps* @@ -72,3 +93,10 @@ guint arg1 guint arg2 + +GstRTPBin::clear-pt-map +void +a +GstRTPBin *gstrtpbin + + diff --git a/docs/plugins/inspect/plugin-dtsdec.xml b/docs/plugins/inspect/plugin-dtsdec.xml index 00573d4d49..4471e5f2ce 100644 --- a/docs/plugins/inspect/plugin-dtsdec.xml +++ b/docs/plugins/inspect/plugin-dtsdec.xml @@ -3,10 +3,10 @@ Decodes DTS audio streams ../../ext/dts/.libs/libgstdtsdec.so libgstdtsdec.so - 0.10.4 + 0.10.4.1 GPL gst-plugins-bad - GStreamer Bad Plug-ins source release + GStreamer Bad Plug-ins CVS/prerelease Unknown package origin diff --git a/docs/plugins/inspect/plugin-rtpmanager.xml b/docs/plugins/inspect/plugin-gstrtpmanager.xml similarity index 89% rename from docs/plugins/inspect/plugin-rtpmanager.xml rename to docs/plugins/inspect/plugin-gstrtpmanager.xml index e522f49b4b..e3f0eb3ffe 100644 --- a/docs/plugins/inspect/plugin-rtpmanager.xml +++ b/docs/plugins/inspect/plugin-gstrtpmanager.xml @@ -1,5 +1,5 @@ - rtpmanager + gstrtpmanager RTP session management plugin library ../../gst/rtpmanager/.libs/libgstrtpmanager.so libgstrtpmanager.so @@ -10,42 +10,42 @@ Unknown package origin - rtpbin + gstrtpbin RTP Bin Filter/Network/RTP Implement an RTP bin Wim Taymans <wim@fluendo.com> - rtpclient + gstrtpclient RTP Client Filter/Network/RTP Implement an RTP client Wim Taymans <wim@fluendo.com> - rtpjitterbuffer + gstrtpjitterbuffer RTP packet jitter-buffer Filter/Network/RTP A buffer that deals with network jitter and other transmission faults Philippe Kalaf <philippe.kalaf@collabora.co.uk>, Wim Taymans <wim@fluendo.com> - rtpptdemux + gstrtpptdemux RTP Demux Demux/Network/RTP Parses codec streams transmitted in the same RTP session Kai Vehmanen <kai.vehmanen@nokia.com> - rtpsession + gstrtpsession RTP Session Filter/Network/RTP Implement an RTP session Wim Taymans <wim@fluendo.com> - rtpssrcdemux + gstrtpssrcdemux RTP SSRC Demux Demux/Network/RTP Splits RTP streams based on the SSRC diff --git a/docs/plugins/inspect/plugin-musepack.xml b/docs/plugins/inspect/plugin-musepack.xml index 093cf759f1..69ea1ceaad 100644 --- a/docs/plugins/inspect/plugin-musepack.xml +++ b/docs/plugins/inspect/plugin-musepack.xml @@ -3,7 +3,7 @@ Musepack decoder ../../ext/musepack/.libs/libgstmusepack.so libgstmusepack.so - 0.10.3.1 + 0.10.4.1 LGPL gst-plugins-bad GStreamer Bad Plug-ins CVS/prerelease diff --git a/docs/plugins/inspect/plugin-sdl.xml b/docs/plugins/inspect/plugin-sdl.xml index d7c1339045..b0e74ad73c 100644 --- a/docs/plugins/inspect/plugin-sdl.xml +++ b/docs/plugins/inspect/plugin-sdl.xml @@ -1,9 +1,9 @@ sdl SDL (Simple DirectMedia Layer) support for GStreamer - ../../ext/sdl/.libs/libgstsdlvideosink.so - libgstsdlvideosink.so - 0.10.3.1 + ../../ext/sdl/.libs/libgstsdl.so + libgstsdl.so + 0.10.4.1 LGPL gst-plugins-bad GStreamer Bad Plug-ins CVS/prerelease @@ -21,7 +21,9 @@ SDL video sink Sink/Video An SDL-based videosink - Ronald Bultje <rbultje@ronald.bitfreak.net>Edgard Lima <edgard.lima@indt.org.br>Jan Schmidt <thaytan@mad.scientist.com> + Ronald Bultje <rbultje@ronald.bitfreak.net> + Edgard Lima <edgard.lima@indt.org.br> + Jan Schmidt <thaytan@mad.scientist.com> \ No newline at end of file diff --git a/docs/plugins/inspect/plugin-spcdec.xml b/docs/plugins/inspect/plugin-spcdec.xml index 0da64a7045..d7b8257eaf 100644 --- a/docs/plugins/inspect/plugin-spcdec.xml +++ b/docs/plugins/inspect/plugin-spcdec.xml @@ -3,7 +3,7 @@ OpenSPC Audio Decoder ../../ext/spc/.libs/libgstspc.so libgstspc.so - 0.10.3.1 + 0.10.4.1 LGPL gst-plugins-bad GStreamer Bad Plug-ins CVS/prerelease diff --git a/docs/plugins/inspect/plugin-swfdec.xml b/docs/plugins/inspect/plugin-swfdec.xml index 8d517e5ec8..938cb0f85f 100644 --- a/docs/plugins/inspect/plugin-swfdec.xml +++ b/docs/plugins/inspect/plugin-swfdec.xml @@ -3,10 +3,10 @@ Uses libswfdec to decode Flash video streams ../../ext/swfdec/.libs/libgstswfdec.so libgstswfdec.so - 0.10.4 + 0.10.4.1 LGPL gst-plugins-bad - GStreamer Bad Plug-ins source release + GStreamer Bad Plug-ins CVS/prerelease Unknown package origin