From: Tim-Philipp Müller Date: Sun, 6 Sep 2015 15:33:02 +0000 (+0100) Subject: docs: remove properties and signals that no longer exist X-Git-Tag: 1.19.3~509^2~3282 X-Git-Url: http://review.tizen.org/git/?a=commitdiff_plain;h=d46c1df74542c3fc97b6a903db762c9aafed2187;p=platform%2Fupstream%2Fgstreamer.git docs: remove properties and signals that no longer exist https://bugzilla.gnome.org/show_bug.cgi?id=726443 --- diff --git a/docs/plugins/gst-plugins-good-plugins.args b/docs/plugins/gst-plugins-good-plugins.args index a9f189c..ee8fc96 100644 --- a/docs/plugins/gst-plugins-good-plugins.args +++ b/docs/plugins/gst-plugins-good-plugins.args @@ -319,16 +319,6 @@ -GstSpectrum::message -gboolean - -rw -Message -Whether to post a 'spectrum' element message on the bus for each passed interval (deprecated, use post-messages). -TRUE - - - GstSpectrum::threshold gint <= 0 @@ -379,16 +369,6 @@ -GstVideoflip::method -GstVideoflipMethod - -rw -method -method. -Rotate clockwise 90 degrees - - - GstVideoBox::alpha gdouble [0,1] @@ -529,16 +509,6 @@ -GstUDPSrc::sockfd -gint ->= G_MAXULONG -rw -Socket Handle -Socket to use for UDP reception. (-1 == allocate). --1 - - - GstUDPSrc::buffer-size gint >= 0 @@ -559,16 +529,6 @@ -GstUDPSrc::closefd -gboolean - -rw -Close sockfd -Close sockfd if passed as property on state change. -TRUE - - - GstUDPSrc::skip-first-bytes gint >= 0 @@ -579,16 +539,6 @@ -GstUDPSrc::sock -gint ->= G_MAXULONG -r -Socket Handle -Socket currently in use for UDP reception. (-1 = no socket). --1 - - - GstUDPSrc::auto-multicast gboolean @@ -649,16 +599,6 @@ -GstUDPSrc::bind-address -gchar* - -rw -Bind Address -Address to bind the socket to. This is equivalent to the multicast-group property. -"0.0.0.0" - - - GstUDPSrc::address gchar* @@ -689,16 +629,6 @@ -GstSMPTE::fps -gfloat ->= 0 -rw -FPS -Frames per second if no input files are given (deprecated). -0 - - - GstSMPTE::type GstSMPTETransitionType @@ -1069,16 +999,6 @@ -GstRTPDec::skip -gint - -rw -Skip -skip (unused). -0 - - - GstRTPDec::latency guint @@ -1209,16 +1129,6 @@ -GstEFence::fence-top -gboolean - -rw -Fence Top -Align buffers with top of fenced region. -TRUE - - - GstAlpha::alpha gdouble [0,1] @@ -1399,16 +1309,6 @@ -GstShout2send::sync -gboolean - -rw -Sync -Sync on the clock. -FALSE - - - GstShout2send::url gchar* @@ -1449,76 +1349,6 @@ -DV1394Src::channel -gint -[0,64] -rw -Channel -Channel number for listening. -63 - - - -DV1394Src::consecutive -gint ->= 1 -rw -consecutive frames -send n consecutive frames after skipping. -1 - - - -DV1394Src::drop-incomplete -gboolean - -rw -drop_incomplete -drop incomplete frames. -TRUE - - - -DV1394Src::guid -guint64 - -rw -GUID -select one of multiple DV devices by its GUID. use a hexadecimal like 0xhhhhhhhhhhhhhhhh. (0 = no guid). -0 - - - -DV1394Src::port -gint -[-1,16] -rw -Port -Port number (-1 automatic). --1 - - - -DV1394Src::skip -gint ->= 0 -rw -skip frames -skip n frames. -0 - - - -DV1394Src::use-avc -gboolean - -rw -Use AV/C -Use AV/C VTR control. -TRUE - - - GstPngEnc::compression-level guint <= 9 @@ -1599,56 +1429,6 @@ -GstSmokeEnc::keyframe -gint -[1,100000] -rw -Keyframe -Insert keyframe every N frames. -20 - - - -GstSmokeEnc::qmax -gint -[0,100] -rw -Qmax -Maximum quality. -85 - - - -GstSmokeEnc::qmin -gint -[0,100] -rw -Qmin -Minimum quality. -10 - - - -GstSmokeEnc::threshold -gint -[0,100000000] -rw -Threshold -Motion estimation threshold. -3000 - - - -GstEsdSink::host -gchar* - -rw -Host -The host running the esound daemon. -NULL - - - GstDVDec::clamp-chroma gboolean @@ -1799,16 +1579,6 @@ -GstRtpGSMParse::frequency -gint - -rw -frequency -frequency. -8000 - - - GstSpeexEnc::abr gint >= 0 @@ -2149,16 +1919,6 @@ -GstVideoMixer::background -GstVideoMixerBackground - -rw -Background -Background type. -Checker pattern - - - GstMatroskaMux::writing-app gchar* @@ -2369,96 +2129,6 @@ -GstTextOverlay::deltax -gint - -w -X position modifier -Shift X position to the left or to the right. Unit is pixels. -0 - - - -GstTextOverlay::deltay -gint - -w -Y position modifier -Shift Y position up or down. Unit is pixels. -0 - - - -GstTextOverlay::font-desc -gchararray - -w -font description -Pango font description of font to be used for rendering. See documentation of pango_font_description_from_string for syntax. -"" - - - -GstTextOverlay::halign -gchararray - -w -horizontal alignment -Horizontal alignment of the text. Can be either 'left', 'right', or 'center'. -"center" - - - -GstTextOverlay::shaded-background -gboolean - -w -shaded background -Whether to shade the background under the text area. -FALSE - - - -GstTextOverlay::text -gchararray - -w -text -Text to be display. -"" - - - -GstTextOverlay::valign -gchararray - -w -vertical alignment -Vertical alignment of the text. Can be either 'baseline', 'bottom', or 'top'. -"baseline" - - - -GstTextOverlay::xpad -gint - -w -horizontal paddding -Horizontal paddding when using left/right alignment. -25 - - - -GstTextOverlay::ypad -gint - -w -vertical padding -Vertical padding when using top/bottom alignment. -25 - - - GstCutter::leaky gboolean @@ -2519,48 +2189,18 @@ -GstRtpMP4VPay::send-config -gboolean - +GstRtpMP4VPay::config-interval +guint +<= 3600 rw -Send Config -Send the config parameters in RTP packets as well(deprecated see config-interval). -FALSE +Config Send Interval +Send Config Insertion Interval in seconds (configuration headers will be multiplexed in the data stream when detected.) (0 = disabled). +0 -GstRtpMP4VPay::buffer-list -gboolean - -rw -Buffer Array -Use Buffer Arrays. -FALSE - - - -GstRtpMP4VPay::config-interval -guint -<= 3600 -rw -Config Send Interval -Send Config Insertion Interval in seconds (configuration headers will be multiplexed in the data stream when detected.) (0 = disabled). -0 - - - -GstRTPDepay::skip -gint - -rw -skip -skip. -0 - - - -GstMultipartMux::boundary -gchar* +GstMultipartMux::boundary +gchar* rw Boundary @@ -2569,126 +2209,6 @@ -GstCairoTextOverlay::deltax -gint - -w -X position modifier -Shift X position to the left or to the right. Unit is pixels. -0 - - - -GstCairoTextOverlay::deltay -gint - -w -Y position modifier -Shift Y position up or down. Unit is pixels. -0 - - - -GstCairoTextOverlay::font-desc -gchar* - -w -font description -Pango font description of font to be used for rendering. See documentation of pango_font_description_from_string for syntax. -"" - - - -GstCairoTextOverlay::halign -gchar* - -w -horizontal alignment -Horizontal alignment of the text. Can be either 'left', 'right', or 'center'. -"center" - - - -GstCairoTextOverlay::shaded-background -gboolean - -w -shaded background -Whether to shade the background under the text area. -FALSE - - - -GstCairoTextOverlay::text -gchar* - -w -text -Text to be display. -"" - - - -GstCairoTextOverlay::valign -gchar* - -w -vertical alignment -Vertical alignment of the text. Can be either 'baseline', 'bottom', or 'top'. -"baseline" - - - -GstCairoTextOverlay::xpad -gint - -w -horizontal paddding -Horizontal paddding when using left/right alignment. -25 - - - -GstCairoTextOverlay::ypad -gint - -w -vertical padding -Vertical padding when using top/bottom alignment. -25 - - - -GstCairoTextOverlay::silent -gboolean - -w -silent -Whether to render the text string. -FALSE - - - -GstOssMixerElement::device-name -gchar* - -r -Device name -Human-readable name of the sound device. -NULL - - - -GstOssMixerElement::device -gchar* - -rw -Device -OSS mixer device (usually /dev/mixer). -"/dev/mixer" - - - GstID3Demux::prefer-v1 gboolean @@ -2699,26 +2219,6 @@ -GstDynUDPSink::sockfd -gint -[G_MAXULONG,32767] -rw -socket handle -Socket to use for UDP sending. (-1 == allocate). --1 - - - -GstDynUDPSink::closefd -gboolean - -rw -Close sockfd -Close sockfd if passed as property on state change. -TRUE - - - GstDynUDPSink::close-socket gboolean @@ -2769,16 +2269,6 @@ -GstCdioCddaSrc::read-speed -gint -[-1,100] -rw -Read speed -Read from device at the specified speed (-1 = default). --1 - - - GstMultiUDPSink::bytes-served guint64 @@ -2809,36 +2299,6 @@ -GstMultiUDPSink::closefd -gboolean - -rw -Close sockfd -Close sockfd if passed as property on state change. -TRUE - - - -GstMultiUDPSink::sock -gint ->= G_MAXULONG -r -Socket Handle -Socket currently in use for UDP sending. (-1 == no socket). --1 - - - -GstMultiUDPSink::sockfd -gint ->= G_MAXULONG -rw -Socket Handle -Socket to use for UDP sending. (-1 == allocate). --1 - - - GstMultiUDPSink::auto-multicast gboolean @@ -2999,96 +2459,6 @@ -GstCmmlDec::wait-clip-end-time -gboolean - -rw -Wait clip end time -Send a tag for a clip when the clip ends, setting its end-time. Use when you need to know both clip's start-time and end-time. -FALSE - - - -GstCmmlEnc::granule-rate-denominator -gint64 ->= 0 -rwx -Granulerate denominator -Granulerate denominator. -1 - - - -GstCmmlEnc::granule-rate-numerator -gint64 ->= 0 -rwx -Granulerate numerator -Granulerate numerator. -1000 - - - -GstCmmlEnc::granule-shift -guchar -<= 64 -rwx -Granuleshift -The number of lower bits to use for partitioning a granule position. -32 - - - -GstHalAudioSrc::udi -gchar* - -rw -UDI -Unique Device Id. -NULL - - - -GstHalAudioSink::udi -gchar* - -rw -UDI -Unique Device Id. -NULL - - - -GstPixbufScale::method -GstPixbufScaleMethod - -rw -method -method. -2 - - - -GstGdkPixbuf::silent -gboolean - -rw -Silent -Produce verbose output ? (deprecated). -FALSE - - - -GstGConfAudioSink::profile -GstGConfProfile - -rw -Profile -Profile. -Sound Events - - - GstXImageSrc::display-name gchar* @@ -3099,16 +2469,6 @@ -GstXImageSrc::screen-num -guint -<= G_MAXINT -rw -Screen number -X Screen Number. -0 - - - GstXImageSrc::show-pointer gboolean @@ -3239,16 +2599,6 @@ -GstMultipartDemux::autoscan -gboolean - -rw -autoscan -Try to autofind the prefix (deprecated unused, see boundary). -FALSE - - - GstMultipartDemux::boundary gchar* @@ -3399,16 +2749,6 @@ -GstDirectDrawSink::force-aspect-ratio -gboolean - -rw -Force aspect ratio -When enabled, scaling will respect original aspect ratio. -FALSE - - - GstWavpackEnc::bitrate guint <= 9600000 @@ -3579,26 +2919,6 @@ -GstV4l2Src::queue-size -guint -[1,16] -rw -Queue size -Number of buffers to be enqueud in the driver in streaming mode. -2 - - - -GstV4l2Src::always-copy -gboolean - -rw -Always Copy -If the buffer will or not be used directly from mmap. -TRUE - - - GstV4l2Src::device-fd gint >= G_MAXULONG @@ -3629,16 +2949,6 @@ -GstV4l2Src::decimate -gint ->= 1 -rw -Decimate -Only use every nth frame. -1 - - - GstV4l2Src::hue gint @@ -3869,16 +3179,6 @@ -GstAudioWSincLimit::frequency -gdouble ->= 0 -rw -Frequency -Cut-off Frequency (Hz). -0 - - - GstAudioWSincLimit::length gint [3,256000] @@ -3949,16 +3249,6 @@ -GstAutoAudioSink::filter-caps -GstCaps* - -rw -Filter caps -Filter sink candidates using these caps. - - - - GstAutoAudioSink::ts-offset gint64 @@ -3979,16 +3269,6 @@ -GstAutoVideoSink::filter-caps -GstCaps* - -rw -Filter caps -Filter sink candidates using these caps. - - - - GstAutoVideoSink::ts-offset gint64 @@ -4059,16 +3339,6 @@ -GstGdkPixbufSink::send-messages -gboolean - -rw -Send Messages -Whether to post messages containing pixbufs on the bus (deprecated, use post-messages). -TRUE - - - GstGdkPixbufSink::post-messages gboolean @@ -4099,16 +3369,6 @@ -GstSoupHTTPSrc::iradio-genre -gchar* - -r -iradio-genre -Genre of the stream. -NULL - - - GstSoupHTTPSrc::iradio-mode gboolean @@ -4119,36 +3379,6 @@ -GstSoupHTTPSrc::iradio-name -gchar* - -r -iradio-name -Name of the stream. -NULL - - - -GstSoupHTTPSrc::iradio-title -gchar* - -r -iradio-title -Name of currently playing song. -NULL - - - -GstSoupHTTPSrc::iradio-url -gchar* - -r -iradio-url -Homepage URL for radio stream. -NULL - - - GstSoupHTTPSrc::location gchar* @@ -4349,16 +3579,6 @@ -GstRtpH264Pay::profile-level-id -gchar* - -rw -profile-level-id -The base64 profile-level-id to set in the sink caps (deprecated). -NULL - - - GstRtpH264Pay::sprop-parameter-sets gchar* @@ -4369,26 +3589,6 @@ -GstRtpH264Pay::scan-mode -GstH264PayScanMode - -rw -Scan Mode -How to scan the input buffers for NAL units. Performance can be increased when certain assumptions are made about the input buffers. -Buffers contain multiple complete NALUs - - - -GstRtpH264Pay::buffer-list -gboolean - -rw -Buffer List -Use Buffer Lists. -FALSE - - - GstRtpH264Pay::config-interval guint <= 3600 @@ -20339,66 +19539,6 @@ -GstVideoMixerPad::alpha -gdouble -[0,1] -rw -Alpha -Alpha of the picture. -1 - - - -GstVideoMixerPad::xpos -gint - -rw -X Position -X Position of the picture. -0 - - - -GstVideoMixerPad::ypos -gint - -rw -Y Position -Y Position of the picture. -0 - - - -GstVideoMixerPad::zorder -guint -<= 10000 -rw -Z-Order -Z Order of the picture. -0 - - - -GstRtpH264Depay::byte-stream -gboolean - -rw -Byte Stream -Generate byte stream format of NALU (deprecated; use caps). -TRUE - - - -GstRtpH264Depay::access-unit -gboolean - -rw -Access Unit -Merge NALU into AU (picture) (deprecated; use caps). -FALSE - - - GstAudioKaraoke::filter-band gfloat [0,441] @@ -20489,16 +19629,6 @@ -GstPulseSink::client -gchar* - -rw -Client -The PulseAudio client name to use. -"" - - - GstPulseSink::stream-properties GstStructure* @@ -20569,16 +19699,6 @@ -GstPulseSrc::client -gchar* - -rw -Client -The PulseAudio client_name_to_use. -"" - - - GstPulseSrc::mute gboolean @@ -20629,36 +19749,6 @@ -GstPulseMixer::device -gchar* - -rw -Device -The PulseAudio sink or source to control. -NULL - - - -GstPulseMixer::device-name -gchar* - -r -Device name -Human-readable name of the sound device. -NULL - - - -GstPulseMixer::server -gchar* - -rw -Server -The PulseAudio server to connect to. -NULL - - - GstTagInject::tags gchar* @@ -20859,26 +19949,6 @@ -GstAutoVideoSrc::filter-caps -GstCaps* - -rw -Filter caps -Filter src candidates using these caps. - - - - -GstAutoAudioSrc::filter-caps -GstCaps* - -rw -Filter caps -Filter sink candidates using these caps. - - - - GstRtpJPEGPay::quality gint [0,255] @@ -20899,16 +19969,6 @@ -GstRtpJPEGPay::buffer-list -gboolean - -rw -Buffer List -Use Buffer Lists. -FALSE - - - GstAudioFIRFilter::kernel GValueArray* @@ -20949,66 +20009,6 @@ -GstAudioDelay::delay -guint64 ->= 1 -rw -Delay -Delay in nanoseconds. -1 - - - -GstAudioDelay::feedback -gfloat -[0,1] -rw -Feedback -Amount of feedback. -0 - - - -GstAudioDelay::intensity -gfloat -[0,1] -rw -Intensity -Intensity of the echo. -0 - - - -GstAudioReverb::delay -guint64 ->= 1 -rw -Delay -Delay of the echo in nanoseconds. -1 - - - -GstAudioReverb::feedback -gfloat -[0,1] -rw -Feedback -Amount of feedback. -0 - - - -GstAudioReverb::intensity -gfloat -[0,1] -rw -Intensity -Intensity of the echo. -0 - - - GstAudioEcho::delay guint64 >= 1 @@ -21629,16 +20629,6 @@ -GstRtpSession::ntp-ns-base -guint64 - -rw -NTP base time -The NTP base time corresponding to running_time 0 (deprecated). -0 - - - GstRtpSession::num-active-sources guint @@ -21759,16 +20749,6 @@ -GstRtpRtxSend::rtx-payload-type -guint - -rw -RTX Payload Type -Payload type of the retransmission stream (fmtp in SDP). -0 - - - GstRtpRtxSend::max-size-time guint @@ -21829,16 +20809,6 @@ -GstRtpRtxReceive::rtx-payload-types -string - -rw -Colon separated list of payload format type -Set through SDP (fmtp), it helps to detect restransmission streams. -"" - - - GstRtpRtxReceive::num-rtx-requests guint @@ -21959,16 +20929,6 @@ -GstV4l2Sink::queue-size -guint -[1,16] -rw -Queue size -Number of buffers to be enqueud in the driver in streaming mode. -12 - - - GstV4l2Sink::brightness gint @@ -22049,16 +21009,6 @@ -GstV4l2Sink::min-queued-bufs -guint -<= 16 -rw -Minimum queued bufs -Minimum number of queued bufs; v4l2sink won't dqbuf if the driver doesn't have more than this number (which normally you shouldn't change). -1 - - - GstV4l2Sink::io-mode GstV4l2IOMode @@ -22129,16 +21079,6 @@ -GstFlvMux::is-live -gboolean - -rw -Is Live -The stream is live and does not need an index. -FALSE - - - GstFlvMux::streamable gboolean @@ -22179,26 +21119,6 @@ -GstOss4Mixer::device -gchar* - -rw -Device -OSS mixer device (e.g. /dev/oss/hdaudio0/mix0 or /dev/mixerN) (NULL = use first mixer device found). -NULL - - - -GstOss4Mixer::device-name -gchar* - -r -Device name -Human-readable name of the sound device. -NULL - - - GstOss4Source::device gchar* @@ -22279,26 +21199,6 @@ -GstRtpJ2KPay::buffer-list -gboolean - -rw -Buffer List -Use Buffer Lists. -TRUE - - - -GstRtpJ2KDepay::buffer-list -gboolean - -rw -Buffer List -Use Buffer Lists. -TRUE - - - GstJackAudioSrc::client JackClient* @@ -23009,96 +21909,6 @@ -GstGPPMux::dts-method -GstQTMuxDtsMethods - -rwx -dts-method -Method to determine DTS time. -reorder - - - -GstGPPMux::faststart -gboolean - -rw -Format file to faststart -If the file should be formatted for faststart (headers first). -FALSE - - - -GstGPPMux::faststart-file -gchar* - -rwx -File to use for storing buffers -File that will be used temporarily to store data from the stream when creating a faststart file. If null a filepath will be created automatically. -NULL - - - -GstGPPMux::fragment-duration -guint - -rwx -Fragment duration -Fragment durations in ms (produce a fragmented file if > 0). -0 - - - -GstGPPMux::moov-recovery-file -gchar* - -rwx -File to store data for posterior moov atom recovery -File to be used to store data for moov atom making movie file recovery possible in case of a crash during muxing. Null for disabled. (Experimental). -NULL - - - -GstGPPMux::movie-timescale -guint ->= 1 -rwx -Movie timescale -Timescale to use in the movie (units per second). -1000 - - - -GstGPPMux::presentation-time -gboolean - -rwx -Include presentation-time info -Calculate and include presentation/composition time (in addition to decoding time). -TRUE - - - -GstGPPMux::streamable -gboolean - -rwx -Streamable -If set to true, the output should be as if it is to be streamed and hence no indexes written or duration written. -FALSE - - - -GstGPPMux::trak-timescale -guint - -rwx -Track timescale -Timescale to use for the tracks (units per second, 0 is automatic). -0 - - - Gst3GPPMux::dts-method GstQTMuxDtsMethods @@ -23249,266 +22059,6 @@ -GstPulseAudioSink::alignment-threshold -guint64 -[1,18446744073709551614] -rw -Alignment Threshold -Timestamp alignment threshold in nanoseconds. -40000000 - - - -GstPulseAudioSink::async -gboolean - -rw -Async -Go asynchronously to PAUSED. -TRUE - - - -GstPulseAudioSink::blocksize -guint - -rw -Block size -Size in bytes to pull per buffer (0 = default). -4096 - - - -GstPulseAudioSink::buffer-time -gint64 ->= 1 -rw -Buffer Time -Size of audio buffer in microseconds. -200000 - - - -GstPulseAudioSink::can-activate-pull -gboolean - -rw -Allow Pull Scheduling -Allow pull-based scheduling. -FALSE - - - -GstPulseAudioSink::client -gchar* - -rw -Client -The PulseAudio client name to use. -"" - - - -GstPulseAudioSink::device -gchar* - -rw -Device -The PulseAudio sink device to connect to. -NULL - - - -GstPulseAudioSink::device-name -gchar* - -r -Device name -Human-readable name of the sound device. -NULL - - - -GstPulseAudioSink::discont-wait -guint64 -<= 18446744073709551614 -rw -Discont Wait -Window of time in nanoseconds to wait before creating a discontinuity. -1000000000 - - - -GstPulseAudioSink::drift-tolerance -gint64 ->= 1 -rw -Drift Tolerance -Tolerance for clock drift in microseconds. -40000 - - - -GstPulseAudioSink::enable-last-buffer -gboolean - -rw -Enable Last Buffer -Enable the last-buffer property. -TRUE - - - -GstPulseAudioSink::last-buffer -GstBuffer* - -r -Last Buffer -The last buffer received in the sink. - - - - -GstPulseAudioSink::latency-time -gint64 ->= 1 -rw -Latency Time -Audio latency in microseconds. -10000 - - - -GstPulseAudioSink::max-lateness -gint64 ->= G_MAXULONG -rw -Max Lateness -Maximum number of nanoseconds that a buffer can be late before it is dropped (-1 unlimited). --1 - - - -GstPulseAudioSink::mute -gboolean - -rw -Mute -Mute state of this stream. -FALSE - - - -GstPulseAudioSink::preroll-queue-len -guint - -rwx -Preroll queue length -Number of buffers to queue during preroll. -0 - - - -GstPulseAudioSink::provide-clock -gboolean - -rw -Provide Clock -Provide a clock to be used as the global pipeline clock. -TRUE - - - -GstPulseAudioSink::qos -gboolean - -rw -Qos -Generate Quality-of-Service events upstream. -FALSE - - - -GstPulseAudioSink::render-delay -guint64 - -rw -Render Delay -Additional render delay of the sink in nanoseconds. -0 - - - -GstPulseAudioSink::server -gchar* - -rw -Server -The PulseAudio server to connect to. -NULL - - - -GstPulseAudioSink::slave-method -GstBaseAudioSinkSlaveMethod - -rw -Slave Method -Algorithm to use to match the rate of the masterclock. -GST_BASE_AUDIO_SINK_SLAVE_SKEW - - - -GstPulseAudioSink::stream-properties -GstStructure* - -rw -stream properties -list of pulseaudio stream properties. - - - - -GstPulseAudioSink::sync -gboolean - -rw -Sync -Sync on the clock. -TRUE - - - -GstPulseAudioSink::throttle-time -guint64 - -rw -Throttle time -The time to keep between rendered buffers (unused). -0 - - - -GstPulseAudioSink::ts-offset -gint64 - -rw -TS Offset -Timestamp offset in nanoseconds. -0 - - - -GstPulseAudioSink::volume -gdouble -[0,10] -rw -Volume -Linear volume of this stream, 1.0=100%. -1 - - - GstSoupHttpClientSink::automatic-redirect gboolean @@ -23859,36 +22409,6 @@ -GstVP8Enc::h-scaling-mode -GstVP8EncScalingMode - -rw -Horizontal scaling mode -Horizontal scaling mode. -Normal - - - -GstVP8Enc::kf-max-dist -gint ->= 0 -rw -Keyframe max distance -Maximum distance between keyframes (number of frames). -128 - - - -GstVP8Enc::kf-mode -GstVP8EncKfMode - -rw -Keyframe Mode -Keyframe placement. -Determine optimal placement automatically - - - GstVP8Enc::lag-in-frames gint [0,25] @@ -23899,16 +22419,6 @@ -GstVP8Enc::max-intra-bitrate-pct -gint ->= 0 -rw -Max Intra bitrate -Maximum Intra frame bitrate. -0 - - - GstVP8Enc::max-quantizer gint [0,63] @@ -23959,16 +22469,6 @@ -GstVP8Enc::overshoot-pct -gint -[0,1000] -rw -Overshoot PCT -Datarate overshoot (max) target (%). -100 - - - GstVP8Enc::resize-allowed gboolean @@ -24049,56 +22549,6 @@ -GstVP8Enc::ts-layer-id -GValueArray* - -rw -Coding layer identification -Sequence defining coding layer membership. - - - - -GstVP8Enc::ts-number-layers -gint -[1,5] -rw -Number of coding layers -Number of coding layers to use. -1 - - - -GstVP8Enc::ts-periodicity -gint -[0,16] -rw -Layer periodicity -Length of sequence that defines layer membership periodicity. -0 - - - -GstVP8Enc::ts-rate-decimator -GValueArray* - -rw -Coding layer rate decimator -Rate decimation factors for each layer. - - - - -GstVP8Enc::ts-target-bitrate -GValueArray* - -rw -Coding layer target bitrates -Target bitrates for coding layers (one per layer, decreasing). - - - - GstVP8Enc::tuning GstVP8EncTuning @@ -24109,56 +22559,6 @@ -GstVP8Enc::twopass-vbr-bias-pct -gint -[0,100] -rw -2-pass VBR bias -CBR/VBR bias (0=CBR, 100=VBR). -50 - - - -GstVP8Enc::twopass-vbr-maxsection-pct -gint ->= 0 -rw -2-pass GOP max bitrate -GOP maximum bitrate (% target). -0 - - - -GstVP8Enc::twopass-vbr-minsection-pct -gint ->= 0 -rw -2-pass GOP min bitrate -GOP minimum bitrate (% target). -0 - - - -GstVP8Enc::undershoot-pct -gint -[0,1000] -rw -Undershoot PCT -Datarate undershoot (min) target (%). -100 - - - -GstVP8Enc::v-scaling-mode -GstVP8EncScalingMode - -rw -Vertical scaling mode -Vertical scaling mode. -Normal - - - GstVP8Enc::horizontal-scaling-mode GstVP8EncScalingMode diff --git a/docs/plugins/gst-plugins-good-plugins.signals b/docs/plugins/gst-plugins-good-plugins.signals index 25d0c51..0496a50 100644 --- a/docs/plugins/gst-plugins-good-plugins.signals +++ b/docs/plugins/gst-plugins-good-plugins.signals @@ -1,15 +1,3 @@ - -GstQTDemux::got-redirect -void -GstQTDemux *gstqtdemux -gchar *arg1 - - - -GstGSMEnc::frame-encoded -void -GstGSMEnc *gstgsmenc - GstMultiUDPSink::add @@ -73,24 +61,6 @@ gint arg2 -GstFdSrc::timeout -void -GstFdSrc *gstfdsrc - - - -GstDiceTV::reset -void -GstDiceTV *gstdicetv - - - -GstVertigoTV::reset-parms -void -GstVertigoTV *gstvertigotv - - - GstShout2send::connection-problem void c @@ -99,163 +69,6 @@ gint arg1 -DV1394Src::frame-dropped -void -DV1394Src *dv1394src - - - -GstJpegEnc::frame-encoded -void -l -GstJpegEnc *gstjpegenc - - - -GstAASink::frame-displayed -void -l -GstAASink *gstaasink - - - -GstAASink::have-size -void -l -GstAASink *gstaasink -guint arg1 -guint arg2 - - - -GstMultiFdSink::add -void -GstMultiFdSink *gstmultifdsink -gint arg1 - - - -GstMultiFdSink::clear -void -GstMultiFdSink *gstmultifdsink - - - -GstMultiFdSink::client-added -void -GstMultiFdSink *gstmultifdsink -gint arg1 - - - -GstMultiFdSink::client-removed -void -GstMultiFdSink *gstmultifdsink -gint arg1 -GstClientStatus arg2 - - - -GstMultiFdSink::get-stats -GValueArray* -GstMultiFdSink *gstmultifdsink -gint arg1 - - - -GstMultiFdSink::remove -void -GstMultiFdSink *gstmultifdsink -gint arg1 - - - -GstDecodeBin::new-decoded-pad -void -GstDecodeBin *gstdecodebin -GstPad *arg1 -gboolean arg2 - - - -GstDecodeBin::removed-decoded-pad -void -GstDecodeBin *gstdecodebin -GstPad *arg1 - - - -GstDecodeBin::unknown-type -void -GstDecodeBin *gstdecodebin -GstPad *arg1 -GstCaps *arg2 - - - -GstFakeSrc::handoff -void -GstFakeSrc *gstfakesrc -GstBuffer arg1 -GstPad *arg2 - - - -GstFakeSink::handoff -void -GstFakeSink *gstfakesink -GstBuffer arg1 -GstPad *arg2 - - - -GstIdentity::handoff -void -GstIdentity *gstidentity -GstBuffer arg1 - - - -GstTypeFindElement::have-type -void -GstTypeFindElement *gsttypefindelement -guint arg1 -GstCaps *arg2 - - - -GstQueue::overrun -void -GstQueue *gstqueue - - - -GstQueue::running -void -GstQueue *gstqueue - - - -GstQueue::underrun -void -GstQueue *gstqueue - - - -GstBin::element-added -void -GstBin *gstbin -GstElement *arg1 - - - -GstBin::element-removed -void -GstBin *gstbin -GstElement *arg1 - - - GstDV1394Src::frame-dropped void l