Update docs

[platform/upstream/gst-plugins-good.git] / docs / plugins / gst-plugins-good-plugins.args

diff --git a/docs/plugins/gst-plugins-good-plugins.args b/docs/plugins/gst-plugins-good-plugins.args
index 8e17826..06337f9 100644 (file)
--- a/docs/plugins/gst-plugins-good-plugins.args
+++ b/docs/plugins/gst-plugins-good-plugins.args
@@ -199,6 +199,26 @@
 </ARG>
 
 <ARG>
+<NAME>GstMultiFileSink::aggregate-gops</NAME>
+<TYPE>gboolean</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Aggregate GOPs</NICK>
+<BLURB>Whether to aggregate GOPs and process them as a whole without splitting.</BLURB>
+<DEFAULT>FALSE</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstMultiFileSink::max-file-duration</NAME>
+<TYPE>guint64</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Maximum File Duration</NICK>
+<BLURB>Maximum file duration before starting a new file in max-size mode (in nanoseconds).</BLURB>
+<DEFAULT>18446744073709551615</DEFAULT>
+</ARG>
+
+<ARG>
 <NAME>GstMultiFileSrc::caps</NAME>
 <TYPE>GstCaps*</TYPE>
 <RANGE></RANGE>
@@ -281,7 +301,7 @@
 <ARG>
 <NAME>GstSpectrum::bands</NAME>
 <TYPE>guint</TYPE>
-<RANGE></RANGE>
+<RANGE>[2,1073741824]</RANGE>
 <FLAGS>rw</FLAGS>
 <NICK>Bands</NICK>
 <BLURB>Number of frequency bands.</BLURB>
@@ -299,16 +319,6 @@
 </ARG>
 
 <ARG>
-<NAME>GstSpectrum::message</NAME>
-<TYPE>gboolean</TYPE>
-<RANGE></RANGE>
-<FLAGS>rw</FLAGS>
-<NICK>Message</NICK>
-<BLURB>Whether to post a 'spectrum' element message on the bus for each passed interval (deprecated, use post-messages).</BLURB>
-<DEFAULT>TRUE</DEFAULT>
-</ARG>
-
-<ARG>
 <NAME>GstSpectrum::threshold</NAME>
 <TYPE>gint</TYPE>
 <RANGE><= 0</RANGE>
@@ -359,16 +369,6 @@
 </ARG>
 
 <ARG>
-<NAME>GstVideoflip::method</NAME>
-<TYPE>GstVideoflipMethod</TYPE>
-<RANGE></RANGE>
-<FLAGS>rw</FLAGS>
-<NICK>method</NICK>
-<BLURB>method.</BLURB>
-<DEFAULT>Rotate clockwise 90 degrees</DEFAULT>
-</ARG>
-
-<ARG>
 <NAME>GstVideoBox::alpha</NAME>
 <TYPE>gdouble</TYPE>
 <RANGE>[0,1]</RANGE>
@@ -484,7 +484,7 @@
 <RANGE></RANGE>
 <FLAGS>rw</FLAGS>
 <NICK>Multicast Group</NICK>
-<BLURB>The Address of multicast group to join. DEPRECATED: Use address property instead.</BLURB>
+<BLURB>The Address of multicast group to join. (DEPRECATED: Use address property instead).</BLURB>
 <DEFAULT>"0.0.0.0"</DEFAULT>
 </ARG>
 
@@ -509,16 +509,6 @@
 </ARG>
 
 <ARG>
-<NAME>GstUDPSrc::sockfd</NAME>
-<TYPE>gint</TYPE>
-<RANGE>>= G_MAXULONG</RANGE>
-<FLAGS>rw</FLAGS>
-<NICK>Socket Handle</NICK>
-<BLURB>Socket to use for UDP reception. (-1 == allocate).</BLURB>
-<DEFAULT>-1</DEFAULT>
-</ARG>
-
-<ARG>
 <NAME>GstUDPSrc::buffer-size</NAME>
 <TYPE>gint</TYPE>
 <RANGE>>= 0</RANGE>
@@ -539,16 +529,6 @@
 </ARG>
 
 <ARG>
-<NAME>GstUDPSrc::closefd</NAME>
-<TYPE>gboolean</TYPE>
-<RANGE></RANGE>
-<FLAGS>rw</FLAGS>
-<NICK>Close sockfd</NICK>
-<BLURB>Close sockfd if passed as property on state change.</BLURB>
-<DEFAULT>TRUE</DEFAULT>
-</ARG>
-
-<ARG>
 <NAME>GstUDPSrc::skip-first-bytes</NAME>
 <TYPE>gint</TYPE>
 <RANGE>>= 0</RANGE>
@@ -559,16 +539,6 @@
 </ARG>
 
 <ARG>
-<NAME>GstUDPSrc::sock</NAME>
-<TYPE>gint</TYPE>
-<RANGE>>= G_MAXULONG</RANGE>
-<FLAGS>r</FLAGS>
-<NICK>Socket Handle</NICK>
-<BLURB>Socket currently in use for UDP reception. (-1 = no socket).</BLURB>
-<DEFAULT>-1</DEFAULT>
-</ARG>
-
-<ARG>
 <NAME>GstUDPSrc::auto-multicast</NAME>
 <TYPE>gboolean</TYPE>
 <RANGE></RANGE>
@@ -584,7 +554,7 @@
 <RANGE></RANGE>
 <FLAGS>rw</FLAGS>
 <NICK>Multicast Interface</NICK>
-<BLURB>The network interface on which to join the multicast group.</BLURB>
+<BLURB>The network interface on which to join the multicast group.This allows multiple interfaces seperated by comma. ("eth0,eth1").</BLURB>
 <DEFAULT>NULL</DEFAULT>
 </ARG>
 
@@ -629,23 +599,43 @@
 </ARG>
 
 <ARG>
-<NAME>GstUDPSrc::bind-address</NAME>
+<NAME>GstUDPSrc::address</NAME>
 <TYPE>gchar*</TYPE>
 <RANGE></RANGE>
 <FLAGS>rw</FLAGS>
-<NICK>Bind Address</NICK>
-<BLURB>Address to bind the socket to. This is equivalent to the multicast-group property.</BLURB>
+<NICK>Address</NICK>
+<BLURB>Address to receive packets for. This is equivalent to the multicast-group property for now.</BLURB>
 <DEFAULT>"0.0.0.0"</DEFAULT>
 </ARG>
 
 <ARG>
-<NAME>GstUDPSrc::address</NAME>
-<TYPE>gchar*</TYPE>
+<NAME>GstUDPSrc::loop</NAME>
+<TYPE>gboolean</TYPE>
 <RANGE></RANGE>
 <FLAGS>rw</FLAGS>
-<NICK>Address</NICK>
-<BLURB>Address to receive packets for. This is equivalent to the multicast-group property for now.</BLURB>
-<DEFAULT>"0.0.0.0"</DEFAULT>
+<NICK>Multicast Loopback</NICK>
+<BLURB>Used for setting the multicast loop parameter. TRUE = enable, FALSE = disable.</BLURB>
+<DEFAULT>TRUE</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstUDPSrc::retrieve-sender-address</NAME>
+<TYPE>gboolean</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Retrieve Sender Address</NICK>
+<BLURB>Whether to retrieve the sender address and add it to buffers as meta. Disabling this might result in minor performance improvements in certain scenarios.</BLURB>
+<DEFAULT>TRUE</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstUDPSrc::mtu</NAME>
+<TYPE>guint</TYPE>
+<RANGE><= G_MAXINT</RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Expected Maximum Transmission Unit</NICK>
+<BLURB>Maximum expected packet size. This directly defines the allocationsize of the receive buffer pool.</BLURB>
+<DEFAULT>1492</DEFAULT>
 </ARG>
 
 <ARG>
@@ -669,16 +659,6 @@
 </ARG>
 
 <ARG>
-<NAME>GstSMPTE::fps</NAME>
-<TYPE>gfloat</TYPE>
-<RANGE>>= 0</RANGE>
-<FLAGS>rw</FLAGS>
-<NICK>FPS</NICK>
-<BLURB>Frames per second if no input files are given (deprecated).</BLURB>
-<DEFAULT>0</DEFAULT>
-</ARG>
-
-<ARG>
 <NAME>GstSMPTE::type</NAME>
 <TYPE>GstSMPTETransitionType</TYPE>
 <RANGE></RANGE>
@@ -714,7 +694,7 @@
 <RANGE></RANGE>
 <FLAGS>rw</FLAGS>
 <NICK>Debug</NICK>
-<BLURB>Dump request and response messages to stdout.</BLURB>
+<BLURB>Dump request and response messages to stdout(DEPRECATED: Printed all RTSP message to gstreamer log as 'log' level).</BLURB>
 <DEFAULT>FALSE</DEFAULT>
 </ARG>
 
@@ -974,7 +954,7 @@
 <RANGE></RANGE>
 <FLAGS>rw</FLAGS>
 <NICK>Use pipeline clock</NICK>
-<BLURB>Use the pipeline running-time to set the NTP time in the RTCP SR messages.</BLURB>
+<BLURB>Use the pipeline running-time to set the NTP time in the RTCP SR messages(DEPRECATED: Use ntp-time-source property).</BLURB>
 <DEFAULT>FALSE</DEFAULT>
 </ARG>
 
@@ -1009,16 +989,116 @@
 </ARG>
 
 <ARG>
-<NAME>GstRTPDec::skip</NAME>
+<NAME>GstRTSPSrc::do-retransmission</NAME>
+<TYPE>gboolean</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Retransmission</NICK>
+<BLURB>Ask the server to retransmit lost packets.</BLURB>
+<DEFAULT>TRUE</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstRTSPSrc::tls-interaction</NAME>
+<TYPE>GTlsInteraction*</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>TLS interaction</NICK>
+<BLURB>A GTlsInteraction object to promt the user for password or certificate.</BLURB>
+<DEFAULT></DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstRTSPSrc::ntp-time-source</NAME>
+<TYPE>GstRTSPSrcNtpTimeSource</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>NTP Time Source</NICK>
+<BLURB>NTP time source for RTCP packets.</BLURB>
+<DEFAULT>NTP time based on realtime clock</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstRTSPSrc::user-agent</NAME>
+<TYPE>gchar*</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>User Agent</NICK>
+<BLURB>The User-Agent string to send to the server.</BLURB>
+<DEFAULT>"GStreamer/1.16.2"</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstRTSPSrc::max-rtcp-rtp-time-diff</NAME>
 <TYPE>gint</TYPE>
+<RANGE>>= G_MAXULONG</RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Max RTCP RTP Time Diff</NICK>
+<BLURB>Maximum amount of time in ms that the RTP time in RTCP SRs is allowed to be ahead (-1 disabled).</BLURB>
+<DEFAULT>1000</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstRTSPSrc::rfc7273-sync</NAME>
+<TYPE>gboolean</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Sync on RFC7273 clock</NICK>
+<BLURB>Synchronize received streams to the RFC7273 clock (requires clock and offset to be provided).</BLURB>
+<DEFAULT>FALSE</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstRTSPSrc::default-rtsp-version</NAME>
+<TYPE>GstRTSPVersion</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>The RTSP version to try first</NICK>
+<BLURB>The RTSP version that should be tried first when negotiating version.</BLURB>
+<DEFAULT>GST_RTSP_VERSION_1_0</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstRTSPSrc::max-ts-offset</NAME>
+<TYPE>gint64</TYPE>
+<RANGE>>= 0</RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Max TS Offset</NICK>
+<BLURB>The maximum absolute value of the time offset in (nanoseconds). Note, if the ntp-sync parameter is set the default value is changed to 0 (no limit).</BLURB>
+<DEFAULT>3000000000</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstRTSPSrc::max-ts-offset-adjustment</NAME>
+<TYPE>guint64</TYPE>
 <RANGE></RANGE>
 <FLAGS>rw</FLAGS>
-<NICK>Skip</NICK>
-<BLURB>skip (unused).</BLURB>
+<NICK>Max Timestamp Offset Adjustment</NICK>
+<BLURB>The maximum number of nanoseconds per frame that time stamp offsets may be adjusted (0 = no limit).</BLURB>
 <DEFAULT>0</DEFAULT>
 </ARG>
 
 <ARG>
+<NAME>GstRTSPSrc::backchannel</NAME>
+<TYPE>GstRTSPBackchannel</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Backchannel type</NICK>
+<BLURB>The type of backchannel to setup. Default is 'none'.</BLURB>
+<DEFAULT>No backchannel</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstRTSPSrc::teardown-timeout</NAME>
+<TYPE>guint64</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Teardown Timeout</NICK>
+<BLURB>When transitioning PAUSED-READY, allow up to timeout (in nanoseconds) delay in order to send teardown (0 = disabled).</BLURB>
+<DEFAULT>100000000</DEFAULT>
+</ARG>
+
+<ARG>
 <NAME>GstRTPDec::latency</NAME>
 <TYPE>guint</TYPE>
 <RANGE></RANGE>
@@ -1044,7 +1124,7 @@
 <RANGE></RANGE>
 <FLAGS>rw</FLAGS>
 <NICK>message</NICK>
-<BLURB>Post a 'level' message for each passed interval (deprecated).</BLURB>
+<BLURB>Post a 'level' message for each passed interval (deprecated, use the post-messages property instead).</BLURB>
 <DEFAULT>TRUE</DEFAULT>
 </ARG>
 
@@ -1141,7 +1221,7 @@
 <ARG>
 <NAME>GstQuarkTV::planes</NAME>
 <TYPE>gint</TYPE>
-<RANGE>[0,64]</RANGE>
+<RANGE>[1,64]</RANGE>
 <FLAGS>rw</FLAGS>
 <NICK>Planes</NICK>
 <BLURB>Number of planes.</BLURB>
@@ -1149,16 +1229,6 @@
 </ARG>
 
 <ARG>
-<NAME>GstEFence::fence-top</NAME>
-<TYPE>gboolean</TYPE>
-<RANGE></RANGE>
-<FLAGS>rw</FLAGS>
-<NICK>Fence Top</NICK>
-<BLURB>Align buffers with top of fenced region.</BLURB>
-<DEFAULT>TRUE</DEFAULT>
-</ARG>
-
-<ARG>
 <NAME>GstAlpha::alpha</NAME>
 <TYPE>gdouble</TYPE>
 <RANGE>[0,1]</RANGE>
@@ -1339,16 +1409,6 @@
 </ARG>
 
 <ARG>
-<NAME>GstShout2send::sync</NAME>
-<TYPE>gboolean</TYPE>
-<RANGE></RANGE>
-<FLAGS>rw</FLAGS>
-<NICK>Sync</NICK>
-<BLURB>Sync on the clock.</BLURB>
-<DEFAULT>FALSE</DEFAULT>
-</ARG>
-
-<ARG>
 <NAME>GstShout2send::url</NAME>
 <TYPE>gchar*</TYPE>
 <RANGE></RANGE>
@@ -1379,82 +1439,22 @@
 </ARG>
 
 <ARG>
-<NAME>GstSpeexDec::enh</NAME>
-<TYPE>gboolean</TYPE>
-<RANGE></RANGE>
-<FLAGS>rw</FLAGS>
-<NICK>Enh</NICK>
-<BLURB>Enable perceptual enhancement.</BLURB>
-<DEFAULT>TRUE</DEFAULT>
-</ARG>
-
-<ARG>
-<NAME>DV1394Src::channel</NAME>
-<TYPE>gint</TYPE>
-<RANGE>[0,64]</RANGE>
-<FLAGS>rw</FLAGS>
-<NICK>Channel</NICK>
-<BLURB>Channel number for listening.</BLURB>
-<DEFAULT>63</DEFAULT>
-</ARG>
-
-<ARG>
-<NAME>DV1394Src::consecutive</NAME>
-<TYPE>gint</TYPE>
+<NAME>GstShout2send::timeout</NAME>
+<TYPE>guint</TYPE>
 <RANGE>>= 1</RANGE>
 <FLAGS>rw</FLAGS>
-<NICK>consecutive frames</NICK>
-<BLURB>send n consecutive frames after skipping.</BLURB>
-<DEFAULT>1</DEFAULT>
-</ARG>
-
-<ARG>
-<NAME>DV1394Src::drop-incomplete</NAME>
-<TYPE>gboolean</TYPE>
-<RANGE></RANGE>
-<FLAGS>rw</FLAGS>
-<NICK>drop_incomplete</NICK>
-<BLURB>drop incomplete frames.</BLURB>
-<DEFAULT>TRUE</DEFAULT>
-</ARG>
-
-<ARG>
-<NAME>DV1394Src::guid</NAME>
-<TYPE>guint64</TYPE>
-<RANGE></RANGE>
-<FLAGS>rw</FLAGS>
-<NICK>GUID</NICK>
-<BLURB>select one of multiple DV devices by its GUID. use a hexadecimal like 0xhhhhhhhhhhhhhhhh. (0 = no guid).</BLURB>
-<DEFAULT>0</DEFAULT>
-</ARG>
-
-<ARG>
-<NAME>DV1394Src::port</NAME>
-<TYPE>gint</TYPE>
-<RANGE>[-1,16]</RANGE>
-<FLAGS>rw</FLAGS>
-<NICK>Port</NICK>
-<BLURB>Port number (-1 automatic).</BLURB>
-<DEFAULT>-1</DEFAULT>
-</ARG>
-
-<ARG>
-<NAME>DV1394Src::skip</NAME>
-<TYPE>gint</TYPE>
-<RANGE>>= 0</RANGE>
-<FLAGS>rw</FLAGS>
-<NICK>skip frames</NICK>
-<BLURB>skip n frames.</BLURB>
-<DEFAULT>0</DEFAULT>
+<NICK>timeout</NICK>
+<BLURB>Max amount of time to wait for network activity, in milliseconds.</BLURB>
+<DEFAULT>10000</DEFAULT>
 </ARG>
 
 <ARG>
-<NAME>DV1394Src::use-avc</NAME>
+<NAME>GstSpeexDec::enh</NAME>
 <TYPE>gboolean</TYPE>
 <RANGE></RANGE>
 <FLAGS>rw</FLAGS>
-<NICK>Use AV/C</NICK>
-<BLURB>Use AV/C VTR control.</BLURB>
+<NICK>Enh</NICK>
+<BLURB>Enable perceptual enhancement.</BLURB>
 <DEFAULT>TRUE</DEFAULT>
 </ARG>
 
@@ -1539,79 +1539,39 @@
 </ARG>
 
 <ARG>
-<NAME>GstSmokeEnc::keyframe</NAME>
-<TYPE>gint</TYPE>
-<RANGE>[1,100000]</RANGE>
+<NAME>GstJpegEnc::snapshot</NAME>
+<TYPE>gboolean</TYPE>
+<RANGE></RANGE>
 <FLAGS>rw</FLAGS>
-<NICK>Keyframe</NICK>
-<BLURB>Insert keyframe every N frames.</BLURB>
-<DEFAULT>20</DEFAULT>
+<NICK>Snapshot</NICK>
+<BLURB>Send EOS after encoding a frame, useful for snapshots.</BLURB>
+<DEFAULT>FALSE</DEFAULT>
 </ARG>
 
 <ARG>
-<NAME>GstSmokeEnc::qmax</NAME>
-<TYPE>gint</TYPE>
-<RANGE>[0,100]</RANGE>
+<NAME>GstDVDec::clamp-chroma</NAME>
+<TYPE>gboolean</TYPE>
+<RANGE></RANGE>
 <FLAGS>rw</FLAGS>
-<NICK>Qmax</NICK>
-<BLURB>Maximum quality.</BLURB>
-<DEFAULT>85</DEFAULT>
+<NICK>Clamp chroma</NICK>
+<BLURB>Clamp chroma.</BLURB>
+<DEFAULT>FALSE</DEFAULT>
 </ARG>
 
 <ARG>
-<NAME>GstSmokeEnc::qmin</NAME>
-<TYPE>gint</TYPE>
-<RANGE>[0,100]</RANGE>
+<NAME>GstDVDec::clamp-luma</NAME>
+<TYPE>gboolean</TYPE>
+<RANGE></RANGE>
 <FLAGS>rw</FLAGS>
-<NICK>Qmin</NICK>
-<BLURB>Minimum quality.</BLURB>
-<DEFAULT>10</DEFAULT>
+<NICK>Clamp luma</NICK>
+<BLURB>Clamp luma.</BLURB>
+<DEFAULT>FALSE</DEFAULT>
 </ARG>
 
 <ARG>
-<NAME>GstSmokeEnc::threshold</NAME>
+<NAME>GstDVDec::drop-factor</NAME>
 <TYPE>gint</TYPE>
-<RANGE>[0,100000000]</RANGE>
-<FLAGS>rw</FLAGS>
-<NICK>Threshold</NICK>
-<BLURB>Motion estimation threshold.</BLURB>
-<DEFAULT>3000</DEFAULT>
-</ARG>
-
-<ARG>
-<NAME>GstEsdSink::host</NAME>
-<TYPE>gchar*</TYPE>
-<RANGE></RANGE>
-<FLAGS>rw</FLAGS>
-<NICK>Host</NICK>
-<BLURB>The host running the esound daemon.</BLURB>
-<DEFAULT>NULL</DEFAULT>
-</ARG>
-
-<ARG>
-<NAME>GstDVDec::clamp-chroma</NAME>
-<TYPE>gboolean</TYPE>
-<RANGE></RANGE>
-<FLAGS>rw</FLAGS>
-<NICK>Clamp chroma</NICK>
-<BLURB>Clamp chroma.</BLURB>
-<DEFAULT>FALSE</DEFAULT>
-</ARG>
-
-<ARG>
-<NAME>GstDVDec::clamp-luma</NAME>
-<TYPE>gboolean</TYPE>
-<RANGE></RANGE>
-<FLAGS>rw</FLAGS>
-<NICK>Clamp luma</NICK>
-<BLURB>Clamp luma.</BLURB>
-<DEFAULT>FALSE</DEFAULT>
-</ARG>
-
-<ARG>
-<NAME>GstDVDec::drop-factor</NAME>
-<TYPE>gint</TYPE>
-<RANGE>>= 1</RANGE>
+<RANGE>>= 1</RANGE>
 <FLAGS>rw</FLAGS>
 <NICK>Drop Factor</NICK>
 <BLURB>Only decode Nth frame.</BLURB>
@@ -1739,16 +1699,6 @@
 </ARG>
 
 <ARG>
-<NAME>GstRtpGSMParse::frequency</NAME>
-<TYPE>gint</TYPE>
-<RANGE></RANGE>
-<FLAGS>rw</FLAGS>
-<NICK>frequency</NICK>
-<BLURB>frequency.</BLURB>
-<DEFAULT>8000</DEFAULT>
-</ARG>
-
-<ARG>
 <NAME>GstSpeexEnc::abr</NAME>
 <TYPE>gint</TYPE>
 <RANGE>>= 0</RANGE>
@@ -2089,16 +2039,6 @@
 </ARG>
 
 <ARG>
-<NAME>GstVideoMixer::background</NAME>
-<TYPE>GstVideoMixerBackground</TYPE>
-<RANGE></RANGE>
-<FLAGS>rw</FLAGS>
-<NICK>Background</NICK>
-<BLURB>Background type.</BLURB>
-<DEFAULT>Checker pattern</DEFAULT>
-</ARG>
-
-<ARG>
 <NAME>GstMatroskaMux::writing-app</NAME>
 <TYPE>gchar*</TYPE>
 <RANGE></RANGE>
@@ -2139,6 +2079,36 @@
 </ARG>
 
 <ARG>
+<NAME>GstMatroskaMux::timecodescale</NAME>
+<TYPE>gint64</TYPE>
+<RANGE>[1,1000000000]</RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Timecode Scale</NICK>
+<BLURB>TimecodeScale used to calculate the Raw Timecode of a Block.</BLURB>
+<DEFAULT>1000000</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstMatroskaMux::max-cluster-duration</NAME>
+<TYPE>gint64</TYPE>
+<RANGE>>= 0</RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Maximum cluster duration</NICK>
+<BLURB>A new cluster will be created if its duration exceeds this value. 0 means no maximum duration.</BLURB>
+<DEFAULT>65535000000</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstMatroskaMux::min-cluster-duration</NAME>
+<TYPE>gint64</TYPE>
+<RANGE>>= 0</RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Minimum cluster duration</NICK>
+<BLURB>Desidered cluster duration as nanoseconds. A new cluster will be created irrespective of this property if a force key unit event is received. 0 means create a new cluster for each video keyframe or for each audio buffer in audio only streams.</BLURB>
+<DEFAULT>500000000</DEFAULT>
+</ARG>
+
+<ARG>
 <NAME>GstTest::allowed-timestamp-deviation</NAME>
 <TYPE>gint64</TYPE>
 <RANGE>>= G_MAXULONG</RANGE>
@@ -2309,96 +2279,6 @@
 </ARG>
 
 <ARG>
-<NAME>GstTextOverlay::deltax</NAME>
-<TYPE>gint</TYPE>
-<RANGE></RANGE>
-<FLAGS>w</FLAGS>
-<NICK>X position modifier</NICK>
-<BLURB>Shift X position to the left or to the right. Unit is pixels.</BLURB>
-<DEFAULT>0</DEFAULT>
-</ARG>
-
-<ARG>
-<NAME>GstTextOverlay::deltay</NAME>
-<TYPE>gint</TYPE>
-<RANGE></RANGE>
-<FLAGS>w</FLAGS>
-<NICK>Y position modifier</NICK>
-<BLURB>Shift Y position up or down. Unit is pixels.</BLURB>
-<DEFAULT>0</DEFAULT>
-</ARG>
-
-<ARG>
-<NAME>GstTextOverlay::font-desc</NAME>
-<TYPE>gchararray</TYPE>
-<RANGE></RANGE>
-<FLAGS>w</FLAGS>
-<NICK>font description</NICK>
-<BLURB>Pango font description of font to be used for rendering. See documentation of pango_font_description_from_string for syntax.</BLURB>
-<DEFAULT>""</DEFAULT>
-</ARG>
-
-<ARG>
-<NAME>GstTextOverlay::halign</NAME>
-<TYPE>gchararray</TYPE>
-<RANGE></RANGE>
-<FLAGS>w</FLAGS>
-<NICK>horizontal alignment</NICK>
-<BLURB>Horizontal alignment of the text. Can be either 'left', 'right', or 'center'.</BLURB>
-<DEFAULT>"center"</DEFAULT>
-</ARG>
-
-<ARG>
-<NAME>GstTextOverlay::shaded-background</NAME>
-<TYPE>gboolean</TYPE>
-<RANGE></RANGE>
-<FLAGS>w</FLAGS>
-<NICK>shaded background</NICK>
-<BLURB>Whether to shade the background under the text area.</BLURB>
-<DEFAULT>FALSE</DEFAULT>
-</ARG>
-
-<ARG>
-<NAME>GstTextOverlay::text</NAME>
-<TYPE>gchararray</TYPE>
-<RANGE></RANGE>
-<FLAGS>w</FLAGS>
-<NICK>text</NICK>
-<BLURB>Text to be display.</BLURB>
-<DEFAULT>""</DEFAULT>
-</ARG>
-
-<ARG>
-<NAME>GstTextOverlay::valign</NAME>
-<TYPE>gchararray</TYPE>
-<RANGE></RANGE>
-<FLAGS>w</FLAGS>
-<NICK>vertical alignment</NICK>
-<BLURB>Vertical alignment of the text. Can be either 'baseline', 'bottom', or 'top'.</BLURB>
-<DEFAULT>"baseline"</DEFAULT>
-</ARG>
-
-<ARG>
-<NAME>GstTextOverlay::xpad</NAME>
-<TYPE>gint</TYPE>
-<RANGE></RANGE>
-<FLAGS>w</FLAGS>
-<NICK>horizontal paddding</NICK>
-<BLURB>Horizontal paddding when using left/right alignment.</BLURB>
-<DEFAULT>25</DEFAULT>
-</ARG>
-
-<ARG>
-<NAME>GstTextOverlay::ypad</NAME>
-<TYPE>gint</TYPE>
-<RANGE></RANGE>
-<FLAGS>w</FLAGS>
-<NICK>vertical padding</NICK>
-<BLURB>Vertical padding when using top/bottom alignment.</BLURB>
-<DEFAULT>25</DEFAULT>
-</ARG>
-
-<ARG>
 <NAME>GstCutter::leaky</NAME>
 <TYPE>gboolean</TYPE>
 <RANGE></RANGE>
@@ -2454,31 +2334,11 @@
 <RANGE></RANGE>
 <FLAGS>rwx</FLAGS>
 <NICK>method</NICK>
-<BLURB>method.</BLURB>
+<BLURB>method (deprecated, use video-direction instead).</BLURB>
 <DEFAULT>Identity (no rotation)</DEFAULT>
 </ARG>
 
 <ARG>
-<NAME>GstRtpMP4VPay::send-config</NAME>
-<TYPE>gboolean</TYPE>
-<RANGE></RANGE>
-<FLAGS>rw</FLAGS>
-<NICK>Send Config</NICK>
-<BLURB>Send the config parameters in RTP packets as well(deprecated see config-interval).</BLURB>
-<DEFAULT>FALSE</DEFAULT>
-</ARG>
-
-<ARG>
-<NAME>GstRtpMP4VPay::buffer-list</NAME>
-<TYPE>gboolean</TYPE>
-<RANGE></RANGE>
-<FLAGS>rw</FLAGS>
-<NICK>Buffer Array</NICK>
-<BLURB>Use Buffer Arrays.</BLURB>
-<DEFAULT>FALSE</DEFAULT>
-</ARG>
-
-<ARG>
 <NAME>GstRtpMP4VPay::config-interval</NAME>
 <TYPE>guint</TYPE>
 <RANGE><= 3600</RANGE>
@@ -2489,16 +2349,6 @@
 </ARG>
 
 <ARG>
-<NAME>GstRTPDepay::skip</NAME>
-<TYPE>gint</TYPE>
-<RANGE></RANGE>
-<FLAGS>rw</FLAGS>
-<NICK>skip</NICK>
-<BLURB>skip.</BLURB>
-<DEFAULT>0</DEFAULT>
-</ARG>
-
-<ARG>
 <NAME>GstMultipartMux::boundary</NAME>
 <TYPE>gchar*</TYPE>
 <RANGE></RANGE>
@@ -2509,126 +2359,6 @@
 </ARG>
 
 <ARG>
-<NAME>GstCairoTextOverlay::deltax</NAME>
-<TYPE>gint</TYPE>
-<RANGE></RANGE>
-<FLAGS>w</FLAGS>
-<NICK>X position modifier</NICK>
-<BLURB>Shift X position to the left or to the right. Unit is pixels.</BLURB>
-<DEFAULT>0</DEFAULT>
-</ARG>
-
-<ARG>
-<NAME>GstCairoTextOverlay::deltay</NAME>
-<TYPE>gint</TYPE>
-<RANGE></RANGE>
-<FLAGS>w</FLAGS>
-<NICK>Y position modifier</NICK>
-<BLURB>Shift Y position up or down. Unit is pixels.</BLURB>
-<DEFAULT>0</DEFAULT>
-</ARG>
-
-<ARG>
-<NAME>GstCairoTextOverlay::font-desc</NAME>
-<TYPE>gchar*</TYPE>
-<RANGE></RANGE>
-<FLAGS>w</FLAGS>
-<NICK>font description</NICK>
-<BLURB>Pango font description of font to be used for rendering. See documentation of pango_font_description_from_string for syntax.</BLURB>
-<DEFAULT>""</DEFAULT>
-</ARG>
-
-<ARG>
-<NAME>GstCairoTextOverlay::halign</NAME>
-<TYPE>gchar*</TYPE>
-<RANGE></RANGE>
-<FLAGS>w</FLAGS>
-<NICK>horizontal alignment</NICK>
-<BLURB>Horizontal alignment of the text. Can be either 'left', 'right', or 'center'.</BLURB>
-<DEFAULT>"center"</DEFAULT>
-</ARG>
-
-<ARG>
-<NAME>GstCairoTextOverlay::shaded-background</NAME>
-<TYPE>gboolean</TYPE>
-<RANGE></RANGE>
-<FLAGS>w</FLAGS>
-<NICK>shaded background</NICK>
-<BLURB>Whether to shade the background under the text area.</BLURB>
-<DEFAULT>FALSE</DEFAULT>
-</ARG>
-
-<ARG>
-<NAME>GstCairoTextOverlay::text</NAME>
-<TYPE>gchar*</TYPE>
-<RANGE></RANGE>
-<FLAGS>w</FLAGS>
-<NICK>text</NICK>
-<BLURB>Text to be display.</BLURB>
-<DEFAULT>""</DEFAULT>
-</ARG>
-
-<ARG>
-<NAME>GstCairoTextOverlay::valign</NAME>
-<TYPE>gchar*</TYPE>
-<RANGE></RANGE>
-<FLAGS>w</FLAGS>
-<NICK>vertical alignment</NICK>
-<BLURB>Vertical alignment of the text. Can be either 'baseline', 'bottom', or 'top'.</BLURB>
-<DEFAULT>"baseline"</DEFAULT>
-</ARG>
-
-<ARG>
-<NAME>GstCairoTextOverlay::xpad</NAME>
-<TYPE>gint</TYPE>
-<RANGE></RANGE>
-<FLAGS>w</FLAGS>
-<NICK>horizontal paddding</NICK>
-<BLURB>Horizontal paddding when using left/right alignment.</BLURB>
-<DEFAULT>25</DEFAULT>
-</ARG>
-
-<ARG>
-<NAME>GstCairoTextOverlay::ypad</NAME>
-<TYPE>gint</TYPE>
-<RANGE></RANGE>
-<FLAGS>w</FLAGS>
-<NICK>vertical padding</NICK>
-<BLURB>Vertical padding when using top/bottom alignment.</BLURB>
-<DEFAULT>25</DEFAULT>
-</ARG>
-
-<ARG>
-<NAME>GstCairoTextOverlay::silent</NAME>
-<TYPE>gboolean</TYPE>
-<RANGE></RANGE>
-<FLAGS>w</FLAGS>
-<NICK>silent</NICK>
-<BLURB>Whether to render the text string.</BLURB>
-<DEFAULT>FALSE</DEFAULT>
-</ARG>
-
-<ARG>
-<NAME>GstOssMixerElement::device-name</NAME>
-<TYPE>gchar*</TYPE>
-<RANGE></RANGE>
-<FLAGS>r</FLAGS>
-<NICK>Device name</NICK>
-<BLURB>Human-readable name of the sound device.</BLURB>
-<DEFAULT>NULL</DEFAULT>
-</ARG>
-
-<ARG>
-<NAME>GstOssMixerElement::device</NAME>
-<TYPE>gchar*</TYPE>
-<RANGE></RANGE>
-<FLAGS>rw</FLAGS>
-<NICK>Device</NICK>
-<BLURB>OSS mixer device (usually /dev/mixer).</BLURB>
-<DEFAULT>"/dev/mixer"</DEFAULT>
-</ARG>
-
-<ARG>
 <NAME>GstID3Demux::prefer-v1</NAME>
 <TYPE>gboolean</TYPE>
 <RANGE></RANGE>
@@ -2639,26 +2369,6 @@
 </ARG>
 
 <ARG>
-<NAME>GstDynUDPSink::sockfd</NAME>
-<TYPE>gint</TYPE>
-<RANGE>[G_MAXULONG,32767]</RANGE>
-<FLAGS>rw</FLAGS>
-<NICK>socket handle</NICK>
-<BLURB>Socket to use for UDP sending. (-1 == allocate).</BLURB>
-<DEFAULT>-1</DEFAULT>
-</ARG>
-
-<ARG>
-<NAME>GstDynUDPSink::closefd</NAME>
-<TYPE>gboolean</TYPE>
-<RANGE></RANGE>
-<FLAGS>rw</FLAGS>
-<NICK>Close sockfd</NICK>
-<BLURB>Close sockfd if passed as property on state change.</BLURB>
-<DEFAULT>TRUE</DEFAULT>
-</ARG>
-
-<ARG>
 <NAME>GstDynUDPSink::close-socket</NAME>
 <TYPE>gboolean</TYPE>
 <RANGE></RANGE>
@@ -2709,16 +2419,6 @@
 </ARG>
 
 <ARG>
-<NAME>GstCdioCddaSrc::read-speed</NAME>
-<TYPE>gint</TYPE>
-<RANGE>[-1,100]</RANGE>
-<FLAGS>rw</FLAGS>
-<NICK>Read speed</NICK>
-<BLURB>Read from device at the specified speed (-1 = default).</BLURB>
-<DEFAULT>-1</DEFAULT>
-</ARG>
-
-<ARG>
 <NAME>GstMultiUDPSink::bytes-served</NAME>
 <TYPE>guint64</TYPE>
 <RANGE></RANGE>
@@ -2749,42 +2449,12 @@
 </ARG>
 
 <ARG>
-<NAME>GstMultiUDPSink::closefd</NAME>
+<NAME>GstMultiUDPSink::auto-multicast</NAME>
 <TYPE>gboolean</TYPE>
 <RANGE></RANGE>
 <FLAGS>rw</FLAGS>
-<NICK>Close sockfd</NICK>
-<BLURB>Close sockfd if passed as property on state change.</BLURB>
-<DEFAULT>TRUE</DEFAULT>
-</ARG>
-
-<ARG>
-<NAME>GstMultiUDPSink::sock</NAME>
-<TYPE>gint</TYPE>
-<RANGE>>= G_MAXULONG</RANGE>
-<FLAGS>r</FLAGS>
-<NICK>Socket Handle</NICK>
-<BLURB>Socket currently in use for UDP sending. (-1 == no socket).</BLURB>
-<DEFAULT>-1</DEFAULT>
-</ARG>
-
-<ARG>
-<NAME>GstMultiUDPSink::sockfd</NAME>
-<TYPE>gint</TYPE>
-<RANGE>>= G_MAXULONG</RANGE>
-<FLAGS>rw</FLAGS>
-<NICK>Socket Handle</NICK>
-<BLURB>Socket to use for UDP sending. (-1 == allocate).</BLURB>
-<DEFAULT>-1</DEFAULT>
-</ARG>
-
-<ARG>
-<NAME>GstMultiUDPSink::auto-multicast</NAME>
-<TYPE>gboolean</TYPE>
-<RANGE></RANGE>
-<FLAGS>rw</FLAGS>
-<NICK>Automatically join/leave multicast groups</NICK>
-<BLURB>Automatically join/leave the multicast groups, FALSE means user has to do it himself.</BLURB>
+<NICK>Automatically join/leave multicast groups</NICK>
+<BLURB>Automatically join/leave the multicast groups, FALSE means user has to do it himself.</BLURB>
 <DEFAULT>TRUE</DEFAULT>
 </ARG>
 
@@ -2939,96 +2609,6 @@
 </ARG>
 
 <ARG>
-<NAME>GstCmmlDec::wait-clip-end-time</NAME>
-<TYPE>gboolean</TYPE>
-<RANGE></RANGE>
-<FLAGS>rw</FLAGS>
-<NICK>Wait clip end time</NICK>
-<BLURB>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.</BLURB>
-<DEFAULT>FALSE</DEFAULT>
-</ARG>
-
-<ARG>
-<NAME>GstCmmlEnc::granule-rate-denominator</NAME>
-<TYPE>gint64</TYPE>
-<RANGE>>= 0</RANGE>
-<FLAGS>rwx</FLAGS>
-<NICK>Granulerate denominator</NICK>
-<BLURB>Granulerate denominator.</BLURB>
-<DEFAULT>1</DEFAULT>
-</ARG>
-
-<ARG>
-<NAME>GstCmmlEnc::granule-rate-numerator</NAME>
-<TYPE>gint64</TYPE>
-<RANGE>>= 0</RANGE>
-<FLAGS>rwx</FLAGS>
-<NICK>Granulerate numerator</NICK>
-<BLURB>Granulerate numerator.</BLURB>
-<DEFAULT>1000</DEFAULT>
-</ARG>
-
-<ARG>
-<NAME>GstCmmlEnc::granule-shift</NAME>
-<TYPE>guchar</TYPE>
-<RANGE><= 64</RANGE>
-<FLAGS>rwx</FLAGS>
-<NICK>Granuleshift</NICK>
-<BLURB>The number of lower bits to use for partitioning a granule position.</BLURB>
-<DEFAULT>32</DEFAULT>
-</ARG>
-
-<ARG>
-<NAME>GstHalAudioSrc::udi</NAME>
-<TYPE>gchar*</TYPE>
-<RANGE></RANGE>
-<FLAGS>rw</FLAGS>
-<NICK>UDI</NICK>
-<BLURB>Unique Device Id.</BLURB>
-<DEFAULT>NULL</DEFAULT>
-</ARG>
-
-<ARG>
-<NAME>GstHalAudioSink::udi</NAME>
-<TYPE>gchar*</TYPE>
-<RANGE></RANGE>
-<FLAGS>rw</FLAGS>
-<NICK>UDI</NICK>
-<BLURB>Unique Device Id.</BLURB>
-<DEFAULT>NULL</DEFAULT>
-</ARG>
-
-<ARG>
-<NAME>GstPixbufScale::method</NAME>
-<TYPE>GstPixbufScaleMethod</TYPE>
-<RANGE></RANGE>
-<FLAGS>rw</FLAGS>
-<NICK>method</NICK>
-<BLURB>method.</BLURB>
-<DEFAULT>2</DEFAULT>
-</ARG>
-
-<ARG>
-<NAME>GstGdkPixbuf::silent</NAME>
-<TYPE>gboolean</TYPE>
-<RANGE></RANGE>
-<FLAGS>rw</FLAGS>
-<NICK>Silent</NICK>
-<BLURB>Produce verbose output ? (deprecated).</BLURB>
-<DEFAULT>FALSE</DEFAULT>
-</ARG>
-
-<ARG>
-<NAME>GstGConfAudioSink::profile</NAME>
-<TYPE>GstGConfProfile</TYPE>
-<RANGE></RANGE>
-<FLAGS>rw</FLAGS>
-<NICK>Profile</NICK>
-<BLURB>Profile.</BLURB>
-<DEFAULT>Sound Events</DEFAULT>
-</ARG>
-
-<ARG>
 <NAME>GstXImageSrc::display-name</NAME>
 <TYPE>gchar*</TYPE>
 <RANGE></RANGE>
@@ -3039,16 +2619,6 @@
 </ARG>
 
 <ARG>
-<NAME>GstXImageSrc::screen-num</NAME>
-<TYPE>guint</TYPE>
-<RANGE><= G_MAXINT</RANGE>
-<FLAGS>rw</FLAGS>
-<NICK>Screen number</NICK>
-<BLURB>X Screen Number.</BLURB>
-<DEFAULT>0</DEFAULT>
-</ARG>
-
-<ARG>
 <NAME>GstXImageSrc::show-pointer</NAME>
 <TYPE>gboolean</TYPE>
 <RANGE></RANGE>
@@ -3179,16 +2749,6 @@
 </ARG>
 
 <ARG>
-<NAME>GstMultipartDemux::autoscan</NAME>
-<TYPE>gboolean</TYPE>
-<RANGE></RANGE>
-<FLAGS>rw</FLAGS>
-<NICK>autoscan</NICK>
-<BLURB>Try to autofind the prefix (deprecated unused, see boundary).</BLURB>
-<DEFAULT>FALSE</DEFAULT>
-</ARG>
-
-<ARG>
 <NAME>GstMultipartDemux::boundary</NAME>
 <TYPE>gchar*</TYPE>
 <RANGE></RANGE>
@@ -3339,16 +2899,6 @@
 </ARG>
 
 <ARG>
-<NAME>GstDirectDrawSink::force-aspect-ratio</NAME>
-<TYPE>gboolean</TYPE>
-<RANGE></RANGE>
-<FLAGS>rw</FLAGS>
-<NICK>Force aspect ratio</NICK>
-<BLURB>When enabled, scaling will respect original aspect ratio.</BLURB>
-<DEFAULT>FALSE</DEFAULT>
-</ARG>
-
-<ARG>
 <NAME>GstWavpackEnc::bitrate</NAME>
 <TYPE>guint</TYPE>
 <RANGE><= 9600000</RANGE>
@@ -3519,26 +3069,6 @@
 </ARG>
 
 <ARG>
-<NAME>GstV4l2Src::queue-size</NAME>
-<TYPE>guint</TYPE>
-<RANGE>[1,16]</RANGE>
-<FLAGS>rw</FLAGS>
-<NICK>Queue size</NICK>
-<BLURB>Number of buffers to be enqueud in the driver in streaming mode.</BLURB>
-<DEFAULT>2</DEFAULT>
-</ARG>
-
-<ARG>
-<NAME>GstV4l2Src::always-copy</NAME>
-<TYPE>gboolean</TYPE>
-<RANGE></RANGE>
-<FLAGS>rw</FLAGS>
-<NICK>Always Copy</NICK>
-<BLURB>If the buffer will or not be used directly from mmap.</BLURB>
-<DEFAULT>TRUE</DEFAULT>
-</ARG>
-
-<ARG>
 <NAME>GstV4l2Src::device-fd</NAME>
 <TYPE>gint</TYPE>
 <RANGE>>= G_MAXULONG</RANGE>
@@ -3569,16 +3099,6 @@
 </ARG>
 
 <ARG>
-<NAME>GstV4l2Src::decimate</NAME>
-<TYPE>gint</TYPE>
-<RANGE>>= 1</RANGE>
-<FLAGS>rw</FLAGS>
-<NICK>Decimate</NICK>
-<BLURB>Only use every nth frame.</BLURB>
-<DEFAULT>1</DEFAULT>
-</ARG>
-
-<ARG>
 <NAME>GstV4l2Src::hue</NAME>
 <TYPE>gint</TYPE>
 <RANGE></RANGE>
@@ -3809,16 +3329,6 @@
 </ARG>
 
 <ARG>
-<NAME>GstAudioWSincLimit::frequency</NAME>
-<TYPE>gdouble</TYPE>
-<RANGE>>= 0</RANGE>
-<FLAGS>rw</FLAGS>
-<NICK>Frequency</NICK>
-<BLURB>Cut-off Frequency (Hz).</BLURB>
-<DEFAULT>0</DEFAULT>
-</ARG>
-
-<ARG>
 <NAME>GstAudioWSincLimit::length</NAME>
 <TYPE>gint</TYPE>
 <RANGE>[3,256000]</RANGE>
@@ -3889,16 +3399,6 @@
 </ARG>
 
 <ARG>
-<NAME>GstAutoAudioSink::filter-caps</NAME>
-<TYPE>GstCaps*</TYPE>
-<RANGE></RANGE>
-<FLAGS>rw</FLAGS>
-<NICK>Filter caps</NICK>
-<BLURB>Filter sink candidates using these caps.</BLURB>
-<DEFAULT></DEFAULT>
-</ARG>
-
-<ARG>
 <NAME>GstAutoAudioSink::ts-offset</NAME>
 <TYPE>gint64</TYPE>
 <RANGE></RANGE>
@@ -3919,16 +3419,6 @@
 </ARG>
 
 <ARG>
-<NAME>GstAutoVideoSink::filter-caps</NAME>
-<TYPE>GstCaps*</TYPE>
-<RANGE></RANGE>
-<FLAGS>rw</FLAGS>
-<NICK>Filter caps</NICK>
-<BLURB>Filter sink candidates using these caps.</BLURB>
-<DEFAULT></DEFAULT>
-</ARG>
-
-<ARG>
 <NAME>GstAutoVideoSink::ts-offset</NAME>
 <TYPE>gint64</TYPE>
 <RANGE></RANGE>
@@ -3999,16 +3489,6 @@
 </ARG>
 
 <ARG>
-<NAME>GstGdkPixbufSink::send-messages</NAME>
-<TYPE>gboolean</TYPE>
-<RANGE></RANGE>
-<FLAGS>rw</FLAGS>
-<NICK>Send Messages</NICK>
-<BLURB>Whether to post messages containing pixbufs on the bus  (deprecated, use post-messages).</BLURB>
-<DEFAULT>TRUE</DEFAULT>
-</ARG>
-
-<ARG>
 <NAME>GstGdkPixbufSink::post-messages</NAME>
 <TYPE>gboolean</TYPE>
 <RANGE></RANGE>
@@ -4039,16 +3519,6 @@
 </ARG>
 
 <ARG>
-<NAME>GstSoupHTTPSrc::iradio-genre</NAME>
-<TYPE>gchar*</TYPE>
-<RANGE></RANGE>
-<FLAGS>r</FLAGS>
-<NICK>iradio-genre</NICK>
-<BLURB>Genre of the stream.</BLURB>
-<DEFAULT>NULL</DEFAULT>
-</ARG>
-
-<ARG>
 <NAME>GstSoupHTTPSrc::iradio-mode</NAME>
 <TYPE>gboolean</TYPE>
 <RANGE></RANGE>
@@ -4059,36 +3529,6 @@
 </ARG>
 
 <ARG>
-<NAME>GstSoupHTTPSrc::iradio-name</NAME>
-<TYPE>gchar*</TYPE>
-<RANGE></RANGE>
-<FLAGS>r</FLAGS>
-<NICK>iradio-name</NICK>
-<BLURB>Name of the stream.</BLURB>
-<DEFAULT>NULL</DEFAULT>
-</ARG>
-
-<ARG>
-<NAME>GstSoupHTTPSrc::iradio-title</NAME>
-<TYPE>gchar*</TYPE>
-<RANGE></RANGE>
-<FLAGS>r</FLAGS>
-<NICK>iradio-title</NICK>
-<BLURB>Name of currently playing song.</BLURB>
-<DEFAULT>NULL</DEFAULT>
-</ARG>
-
-<ARG>
-<NAME>GstSoupHTTPSrc::iradio-url</NAME>
-<TYPE>gchar*</TYPE>
-<RANGE></RANGE>
-<FLAGS>r</FLAGS>
-<NICK>iradio-url</NICK>
-<BLURB>Homepage URL for radio stream.</BLURB>
-<DEFAULT>NULL</DEFAULT>
-</ARG>
-
-<ARG>
 <NAME>GstSoupHTTPSrc::location</NAME>
 <TYPE>gchar*</TYPE>
 <RANGE></RANGE>
@@ -4115,7 +3555,7 @@
 <FLAGS>rw</FLAGS>
 <NICK>User-Agent</NICK>
 <BLURB>Value of the User-Agent HTTP request header field.</BLURB>
-<DEFAULT>"GStreamer souphttpsrc "</DEFAULT>
+<DEFAULT>"GStreamer souphttpsrc 1.16.2 "</DEFAULT>
 </ARG>
 
 <ARG>
@@ -4215,7 +3655,7 @@
 <FLAGS>rw</FLAGS>
 <NICK>keep-alive</NICK>
 <BLURB>Use HTTP persistent connections.</BLURB>
-<DEFAULT>FALSE</DEFAULT>
+<DEFAULT>TRUE</DEFAULT>
 </ARG>
 
 <ARG>
@@ -4259,62 +3699,62 @@
 </ARG>
 
 <ARG>
-<NAME>GstRTPDVPay::mode</NAME>
-<TYPE>GstDVPayMode</TYPE>
+<NAME>GstSoupHTTPSrc::tls-database</NAME>
+<TYPE>GTlsDatabase*</TYPE>
 <RANGE></RANGE>
 <FLAGS>rw</FLAGS>
-<NICK>Mode</NICK>
-<BLURB>The payload mode of payloading.</BLURB>
-<DEFAULT>Video only</DEFAULT>
+<NICK>TLS database</NICK>
+<BLURB>TLS database with anchor certificate authorities used to validate the server certificate.</BLURB>
+<DEFAULT></DEFAULT>
 </ARG>
 
 <ARG>
-<NAME>GstRtpH264Pay::profile-level-id</NAME>
+<NAME>GstSoupHTTPSrc::method</NAME>
 <TYPE>gchar*</TYPE>
 <RANGE></RANGE>
 <FLAGS>rw</FLAGS>
-<NICK>profile-level-id</NICK>
-<BLURB>The base64 profile-level-id to set in the sink caps (deprecated).</BLURB>
+<NICK>HTTP method</NICK>
+<BLURB>The HTTP method to use (GET, HEAD, OPTIONS, etc).</BLURB>
 <DEFAULT>NULL</DEFAULT>
 </ARG>
 
 <ARG>
-<NAME>GstRtpH264Pay::sprop-parameter-sets</NAME>
-<TYPE>gchar*</TYPE>
+<NAME>GstSoupHTTPSrc::tls-interaction</NAME>
+<TYPE>GTlsInteraction*</TYPE>
 <RANGE></RANGE>
 <FLAGS>rw</FLAGS>
-<NICK>sprop-parameter-sets</NICK>
-<BLURB>The base64 sprop-parameter-sets to set in out caps (set to NULL to extract from stream).</BLURB>
-<DEFAULT>NULL</DEFAULT>
+<NICK>TLS interaction</NICK>
+<BLURB>A GTlsInteraction object to be used when the connection or certificate database need to interact with the user.</BLURB>
+<DEFAULT></DEFAULT>
 </ARG>
 
 <ARG>
-<NAME>GstRtpH264Pay::scan-mode</NAME>
-<TYPE>GstH264PayScanMode</TYPE>
+<NAME>GstRTPDVPay::mode</NAME>
+<TYPE>GstDVPayMode</TYPE>
 <RANGE></RANGE>
 <FLAGS>rw</FLAGS>
-<NICK>Scan Mode</NICK>
-<BLURB>How to scan the input buffers for NAL units. Performance can be increased when certain assumptions are made about the input buffers.</BLURB>
-<DEFAULT>Buffers contain multiple complete NALUs</DEFAULT>
+<NICK>Mode</NICK>
+<BLURB>The payload mode of payloading.</BLURB>
+<DEFAULT>Video only</DEFAULT>
 </ARG>
 
 <ARG>
-<NAME>GstRtpH264Pay::buffer-list</NAME>
-<TYPE>gboolean</TYPE>
+<NAME>GstRtpH264Pay::sprop-parameter-sets</NAME>
+<TYPE>gchar*</TYPE>
 <RANGE></RANGE>
 <FLAGS>rw</FLAGS>
-<NICK>Buffer List</NICK>
-<BLURB>Use Buffer Lists.</BLURB>
-<DEFAULT>FALSE</DEFAULT>
+<NICK>sprop-parameter-sets</NICK>
+<BLURB>The base64 sprop-parameter-sets to set in out caps (set to NULL to extract from stream).</BLURB>
+<DEFAULT>NULL</DEFAULT>
 </ARG>
 
 <ARG>
 <NAME>GstRtpH264Pay::config-interval</NAME>
-<TYPE>guint</TYPE>
-<RANGE><= 3600</RANGE>
+<TYPE>gint</TYPE>
+<RANGE>[G_MAXULONG,3600]</RANGE>
 <FLAGS>rw</FLAGS>
 <NICK>SPS PPS Send Interval</NICK>
-<BLURB>Send SPS and PPS Insertion Interval in seconds (sprop parameter sets will be multiplexed in the data stream when detected.) (0 = disabled).</BLURB>
+<BLURB>Send SPS and PPS Insertion Interval in seconds (sprop parameter sets will be multiplexed in the data stream when detected.) (0 = disabled, -1 = send with every IDR frame).</BLURB>
 <DEFAULT>0</DEFAULT>
 </ARG>
 
@@ -20259,108 +19699,48 @@
 </ARG>
 
 <ARG>
-<NAME>GstVideoMixerPad::alpha</NAME>
-<TYPE>gdouble</TYPE>
-<RANGE>[0,1]</RANGE>
+<NAME>GstAudioKaraoke::filter-band</NAME>
+<TYPE>gfloat</TYPE>
+<RANGE>[0,441]</RANGE>
 <FLAGS>rw</FLAGS>
-<NICK>Alpha</NICK>
-<BLURB>Alpha of the picture.</BLURB>
-<DEFAULT>1</DEFAULT>
+<NICK>Filter Band</NICK>
+<BLURB>The Frequency band of the filter.</BLURB>
+<DEFAULT>220</DEFAULT>
 </ARG>
 
 <ARG>
-<NAME>GstVideoMixerPad::xpos</NAME>
-<TYPE>gint</TYPE>
-<RANGE></RANGE>
+<NAME>GstAudioKaraoke::filter-width</NAME>
+<TYPE>gfloat</TYPE>
+<RANGE>[0,100]</RANGE>
 <FLAGS>rw</FLAGS>
-<NICK>X Position</NICK>
-<BLURB>X Position of the picture.</BLURB>
-<DEFAULT>0</DEFAULT>
+<NICK>Filter Width</NICK>
+<BLURB>The Frequency width of the filter.</BLURB>
+<DEFAULT>100</DEFAULT>
 </ARG>
 
 <ARG>
-<NAME>GstVideoMixerPad::ypos</NAME>
-<TYPE>gint</TYPE>
-<RANGE></RANGE>
+<NAME>GstAudioKaraoke::level</NAME>
+<TYPE>gfloat</TYPE>
+<RANGE>[0,1]</RANGE>
 <FLAGS>rw</FLAGS>
-<NICK>Y Position</NICK>
-<BLURB>Y Position of the picture.</BLURB>
-<DEFAULT>0</DEFAULT>
+<NICK>Level</NICK>
+<BLURB>Level of the effect (1.0 = full).</BLURB>
+<DEFAULT>1</DEFAULT>
 </ARG>
 
 <ARG>
-<NAME>GstVideoMixerPad::zorder</NAME>
-<TYPE>guint</TYPE>
-<RANGE><= 10000</RANGE>
+<NAME>GstAudioKaraoke::mono-level</NAME>
+<TYPE>gfloat</TYPE>
+<RANGE>[0,1]</RANGE>
 <FLAGS>rw</FLAGS>
-<NICK>Z-Order</NICK>
-<BLURB>Z Order of the picture.</BLURB>
-<DEFAULT>0</DEFAULT>
+<NICK>Mono Level</NICK>
+<BLURB>Level of the mono channel (1.0 = full).</BLURB>
+<DEFAULT>1</DEFAULT>
 </ARG>
 
 <ARG>
-<NAME>GstRtpH264Depay::byte-stream</NAME>
-<TYPE>gboolean</TYPE>
-<RANGE></RANGE>
-<FLAGS>rw</FLAGS>
-<NICK>Byte Stream</NICK>
-<BLURB>Generate byte stream format of NALU (deprecated; use caps).</BLURB>
-<DEFAULT>TRUE</DEFAULT>
-</ARG>
-
-<ARG>
-<NAME>GstRtpH264Depay::access-unit</NAME>
-<TYPE>gboolean</TYPE>
-<RANGE></RANGE>
-<FLAGS>rw</FLAGS>
-<NICK>Access Unit</NICK>
-<BLURB>Merge NALU into AU (picture) (deprecated; use caps).</BLURB>
-<DEFAULT>FALSE</DEFAULT>
-</ARG>
-
-<ARG>
-<NAME>GstAudioKaraoke::filter-band</NAME>
-<TYPE>gfloat</TYPE>
-<RANGE>[0,441]</RANGE>
-<FLAGS>rw</FLAGS>
-<NICK>Filter Band</NICK>
-<BLURB>The Frequency band of the filter.</BLURB>
-<DEFAULT>220</DEFAULT>
-</ARG>
-
-<ARG>
-<NAME>GstAudioKaraoke::filter-width</NAME>
-<TYPE>gfloat</TYPE>
-<RANGE>[0,100]</RANGE>
-<FLAGS>rw</FLAGS>
-<NICK>Filter Width</NICK>
-<BLURB>The Frequency width of the filter.</BLURB>
-<DEFAULT>100</DEFAULT>
-</ARG>
-
-<ARG>
-<NAME>GstAudioKaraoke::level</NAME>
-<TYPE>gfloat</TYPE>
-<RANGE>[0,1]</RANGE>
-<FLAGS>rw</FLAGS>
-<NICK>Level</NICK>
-<BLURB>Level of the effect (1.0 = full).</BLURB>
-<DEFAULT>1</DEFAULT>
-</ARG>
-
-<ARG>
-<NAME>GstAudioKaraoke::mono-level</NAME>
-<TYPE>gfloat</TYPE>
-<RANGE>[0,1]</RANGE>
-<FLAGS>rw</FLAGS>
-<NICK>Mono Level</NICK>
-<BLURB>Level of the mono channel (1.0 = full).</BLURB>
-<DEFAULT>1</DEFAULT>
-</ARG>
-
-<ARG>
-<NAME>GstPulseSink::device</NAME>
-<TYPE>gchar*</TYPE>
+<NAME>GstPulseSink::device</NAME>
+<TYPE>gchar*</TYPE>
 <RANGE></RANGE>
 <FLAGS>rw</FLAGS>
 <NICK>Device</NICK>
@@ -20409,16 +19789,6 @@
 </ARG>
 
 <ARG>
-<NAME>GstPulseSink::client</NAME>
-<TYPE>gchar*</TYPE>
-<RANGE></RANGE>
-<FLAGS>rw</FLAGS>
-<NICK>Client</NICK>
-<BLURB>The PulseAudio client name to use.</BLURB>
-<DEFAULT>"<unknown>"</DEFAULT>
-</ARG>
-
-<ARG>
 <NAME>GstPulseSink::stream-properties</NAME>
 <TYPE>GstStructure*</TYPE>
 <RANGE></RANGE>
@@ -20435,7 +19805,7 @@
 <FLAGS>rw</FLAGS>
 <NICK>Client Name</NICK>
 <BLURB>The PulseAudio client name to use.</BLURB>
-<DEFAULT>"lt-gst-plugins-good-plugins-scan"</DEFAULT>
+<DEFAULT>"gst-plugins-good-plugins-scan"</DEFAULT>
 </ARG>
 
 <ARG>
@@ -20489,16 +19859,6 @@
 </ARG>
 
 <ARG>
-<NAME>GstPulseSrc::client</NAME>
-<TYPE>gchar*</TYPE>
-<RANGE></RANGE>
-<FLAGS>rw</FLAGS>
-<NICK>Client</NICK>
-<BLURB>The PulseAudio client_name_to_use.</BLURB>
-<DEFAULT>"<unknown>"</DEFAULT>
-</ARG>
-
-<ARG>
 <NAME>GstPulseSrc::mute</NAME>
 <TYPE>gboolean</TYPE>
 <RANGE></RANGE>
@@ -20535,7 +19895,7 @@
 <FLAGS>rw</FLAGS>
 <NICK>Client Name</NICK>
 <BLURB>The PulseAudio client_name_to_use.</BLURB>
-<DEFAULT>"lt-gst-plugins-good-plugins-scan"</DEFAULT>
+<DEFAULT>"gst-plugins-good-plugins-scan"</DEFAULT>
 </ARG>
 
 <ARG>
@@ -20549,36 +19909,6 @@
 </ARG>
 
 <ARG>
-<NAME>GstPulseMixer::device</NAME>
-<TYPE>gchar*</TYPE>
-<RANGE></RANGE>
-<FLAGS>rw</FLAGS>
-<NICK>Device</NICK>
-<BLURB>The PulseAudio sink or source to control.</BLURB>
-<DEFAULT>NULL</DEFAULT>
-</ARG>
-
-<ARG>
-<NAME>GstPulseMixer::device-name</NAME>
-<TYPE>gchar*</TYPE>
-<RANGE></RANGE>
-<FLAGS>r</FLAGS>
-<NICK>Device name</NICK>
-<BLURB>Human-readable name of the sound device.</BLURB>
-<DEFAULT>NULL</DEFAULT>
-</ARG>
-
-<ARG>
-<NAME>GstPulseMixer::server</NAME>
-<TYPE>gchar*</TYPE>
-<RANGE></RANGE>
-<FLAGS>rw</FLAGS>
-<NICK>Server</NICK>
-<BLURB>The PulseAudio server to connect to.</BLURB>
-<DEFAULT>NULL</DEFAULT>
-</ARG>
-
-<ARG>
 <NAME>GstTagInject::tags</NAME>
 <TYPE>gchar*</TYPE>
 <RANGE></RANGE>
@@ -20779,26 +20109,6 @@
 </ARG>
 
 <ARG>
-<NAME>GstAutoVideoSrc::filter-caps</NAME>
-<TYPE>GstCaps*</TYPE>
-<RANGE></RANGE>
-<FLAGS>rw</FLAGS>
-<NICK>Filter caps</NICK>
-<BLURB>Filter src candidates using these caps.</BLURB>
-<DEFAULT></DEFAULT>
-</ARG>
-
-<ARG>
-<NAME>GstAutoAudioSrc::filter-caps</NAME>
-<TYPE>GstCaps*</TYPE>
-<RANGE></RANGE>
-<FLAGS>rw</FLAGS>
-<NICK>Filter caps</NICK>
-<BLURB>Filter sink candidates using these caps.</BLURB>
-<DEFAULT></DEFAULT>
-</ARG>
-
-<ARG>
 <NAME>GstRtpJPEGPay::quality</NAME>
 <TYPE>gint</TYPE>
 <RANGE>[0,255]</RANGE>
@@ -20819,16 +20129,6 @@
 </ARG>
 
 <ARG>
-<NAME>GstRtpJPEGPay::buffer-list</NAME>
-<TYPE>gboolean</TYPE>
-<RANGE></RANGE>
-<FLAGS>rw</FLAGS>
-<NICK>Buffer List</NICK>
-<BLURB>Use Buffer Lists.</BLURB>
-<DEFAULT>FALSE</DEFAULT>
-</ARG>
-
-<ARG>
 <NAME>GstAudioFIRFilter::kernel</NAME>
 <TYPE>GValueArray*</TYPE>
 <RANGE></RANGE>
@@ -20869,37 +20169,7 @@
 </ARG>
 
 <ARG>
-<NAME>GstAudioDelay::delay</NAME>
-<TYPE>guint64</TYPE>
-<RANGE>>= 1</RANGE>
-<FLAGS>rw</FLAGS>
-<NICK>Delay</NICK>
-<BLURB>Delay in nanoseconds.</BLURB>
-<DEFAULT>1</DEFAULT>
-</ARG>
-
-<ARG>
-<NAME>GstAudioDelay::feedback</NAME>
-<TYPE>gfloat</TYPE>
-<RANGE>[0,1]</RANGE>
-<FLAGS>rw</FLAGS>
-<NICK>Feedback</NICK>
-<BLURB>Amount of feedback.</BLURB>
-<DEFAULT>0</DEFAULT>
-</ARG>
-
-<ARG>
-<NAME>GstAudioDelay::intensity</NAME>
-<TYPE>gfloat</TYPE>
-<RANGE>[0,1]</RANGE>
-<FLAGS>rw</FLAGS>
-<NICK>Intensity</NICK>
-<BLURB>Intensity of the echo.</BLURB>
-<DEFAULT>0</DEFAULT>
-</ARG>
-
-<ARG>
-<NAME>GstAudioReverb::delay</NAME>
+<NAME>GstAudioEcho::delay</NAME>
 <TYPE>guint64</TYPE>
 <RANGE>>= 1</RANGE>
 <FLAGS>rw</FLAGS>
@@ -20909,7 +20179,7 @@
 </ARG>
 
 <ARG>
-<NAME>GstAudioReverb::feedback</NAME>
+<NAME>GstAudioEcho::feedback</NAME>
 <TYPE>gfloat</TYPE>
 <RANGE>[0,1]</RANGE>
 <FLAGS>rw</FLAGS>
@@ -20919,7 +20189,7 @@
 </ARG>
 
 <ARG>
-<NAME>GstAudioReverb::intensity</NAME>
+<NAME>GstAudioEcho::intensity</NAME>
 <TYPE>gfloat</TYPE>
 <RANGE>[0,1]</RANGE>
 <FLAGS>rw</FLAGS>
@@ -20929,43 +20199,33 @@
 </ARG>
 
 <ARG>
-<NAME>GstAudioEcho::delay</NAME>
+<NAME>GstAudioEcho::max-delay</NAME>
 <TYPE>guint64</TYPE>
 <RANGE>>= 1</RANGE>
 <FLAGS>rw</FLAGS>
-<NICK>Delay</NICK>
-<BLURB>Delay of the echo in nanoseconds.</BLURB>
+<NICK>Maximum Delay</NICK>
+<BLURB>Maximum delay of the echo in nanoseconds (can't be changed in PLAYING or PAUSED state).</BLURB>
 <DEFAULT>1</DEFAULT>
 </ARG>
 
 <ARG>
-<NAME>GstAudioEcho::feedback</NAME>
-<TYPE>gfloat</TYPE>
-<RANGE>[0,1]</RANGE>
-<FLAGS>rw</FLAGS>
-<NICK>Feedback</NICK>
-<BLURB>Amount of feedback.</BLURB>
-<DEFAULT>0</DEFAULT>
-</ARG>
-
-<ARG>
-<NAME>GstAudioEcho::intensity</NAME>
-<TYPE>gfloat</TYPE>
-<RANGE>[0,1]</RANGE>
+<NAME>GstAudioEcho::surround-delay</NAME>
+<TYPE>gboolean</TYPE>
+<RANGE></RANGE>
 <FLAGS>rw</FLAGS>
-<NICK>Intensity</NICK>
-<BLURB>Intensity of the echo.</BLURB>
-<DEFAULT>0</DEFAULT>
+<NICK>Enable Surround Delay</NICK>
+<BLURB>Delay Surround Channels when TRUE instead of applying an echo effect.</BLURB>
+<DEFAULT>FALSE</DEFAULT>
 </ARG>
 
 <ARG>
-<NAME>GstAudioEcho::max-delay</NAME>
+<NAME>GstAudioEcho::surround-mask</NAME>
 <TYPE>guint64</TYPE>
 <RANGE>>= 1</RANGE>
 <FLAGS>rw</FLAGS>
-<NICK>Maximum Delay</NICK>
-<BLURB>Maximum delay of the echo in nanoseconds (can't be changed in PLAYING or PAUSED state).</BLURB>
-<DEFAULT>1</DEFAULT>
+<NICK>Surround Mask</NICK>
+<BLURB>A bitmask of channels that are considered surround and delayed when surround-delay = TRUE.</BLURB>
+<DEFAULT>18446744073709551612</DEFAULT>
 </ARG>
 
 <ARG>
@@ -21025,7 +20285,7 @@
 <FLAGS>rw</FLAGS>
 <NICK>Method</NICK>
 <BLURB>Deinterlace Method.</BLURB>
-<DEFAULT>Television: Full resolution</DEFAULT>
+<DEFAULT>Linear</DEFAULT>
 </ARG>
 
 <ARG>
@@ -21035,7 +20295,7 @@
 <FLAGS>rw</FLAGS>
 <NICK>Mode</NICK>
 <BLURB>Deinterlace Mode.</BLURB>
-<DEFAULT>Auto detection</DEFAULT>
+<DEFAULT>Auto detection (best effort)</DEFAULT>
 </ARG>
 
 <ARG>
@@ -21294,7 +20554,7 @@
 <RANGE></RANGE>
 <FLAGS>rw</FLAGS>
 <NICK>Use pipeline clock</NICK>
-<BLURB>Use the pipeline running-time to set the NTP time in the RTCP SR messages.</BLURB>
+<BLURB>Use the pipeline running-time to set the NTP time in the RTCP SR messages (DEPRECATED: Use ntp-time-source property).</BLURB>
 <DEFAULT>FALSE</DEFAULT>
 </ARG>
 
@@ -21344,11 +20604,111 @@
 <RANGE></RANGE>
 <FLAGS>rw</FLAGS>
 <NICK>Do retransmission</NICK>
-<BLURB>Send an event downstream to request packet retransmission.</BLURB>
+<BLURB>Enable retransmission on all streams.</BLURB>
+<DEFAULT>FALSE</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstRtpBin::rtp-profile</NAME>
+<TYPE>GstRTPProfile</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>RTP Profile</NICK>
+<BLURB>Default RTP profile of newly created sessions.</BLURB>
+<DEFAULT>GST_RTP_PROFILE_AVP</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstRtpBin::ntp-time-source</NAME>
+<TYPE>GstRtpNtpTimeSource</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>NTP Time Source</NICK>
+<BLURB>NTP time source for RTCP packets.</BLURB>
+<DEFAULT>NTP time based on realtime clock</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstRtpBin::max-dropout-time</NAME>
+<TYPE>guint</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Max dropout time</NICK>
+<BLURB>The maximum time (milliseconds) of missing packets tolerated.</BLURB>
+<DEFAULT>60000</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstRtpBin::max-misorder-time</NAME>
+<TYPE>guint</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Max misorder time</NICK>
+<BLURB>The maximum time (milliseconds) of misordered packets tolerated.</BLURB>
+<DEFAULT>2000</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstRtpBin::max-rtcp-rtp-time-diff</NAME>
+<TYPE>gint</TYPE>
+<RANGE>>= G_MAXULONG</RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Max RTCP RTP Time Diff</NICK>
+<BLURB>Maximum amount of time in ms that the RTP time in RTCP SRs is allowed to be ahead (-1 disabled).</BLURB>
+<DEFAULT>1000</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstRtpBin::rtcp-sync-send-time</NAME>
+<TYPE>gboolean</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>RTCP Sync Send Time</NICK>
+<BLURB>Use send time or capture time for RTCP sync (TRUE = send time, FALSE = capture time).</BLURB>
+<DEFAULT>TRUE</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstRtpBin::rfc7273-sync</NAME>
+<TYPE>gboolean</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Sync on RFC7273 clock</NICK>
+<BLURB>Synchronize received streams to the RFC7273 clock (requires clock and offset to be provided).</BLURB>
 <DEFAULT>FALSE</DEFAULT>
 </ARG>
 
 <ARG>
+<NAME>GstRtpBin::max-streams</NAME>
+<TYPE>guint</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Max Streams</NICK>
+<BLURB>The maximum number of streams to create for one session.</BLURB>
+<DEFAULT>4294967295</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstRtpBin::max-ts-offset</NAME>
+<TYPE>gint64</TYPE>
+<RANGE>>= 0</RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Max TS Offset</NICK>
+<BLURB>The maximum absolute value of the time offset in (nanoseconds). Note, if the ntp-sync parameter is set the default value is changed to 0 (no limit).</BLURB>
+<DEFAULT>3000000000</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstRtpBin::max-ts-offset-adjustment</NAME>
+<TYPE>guint64</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Max Timestamp Offset Adjustment</NICK>
+<BLURB>The maximum number of nanoseconds per frame that time stamp offsets may be adjusted (0 = no limit).</BLURB>
+<DEFAULT>0</DEFAULT>
+</ARG>
+
+<ARG>
 <NAME>GstRtpJitterBuffer::do-lost</NAME>
 <TYPE>gboolean</TYPE>
 <RANGE></RANGE>
@@ -21434,7 +20794,7 @@
 <RANGE>>= G_MAXULONG</RANGE>
 <FLAGS>rw</FLAGS>
 <NICK>RTX Delay Reorder</NICK>
-<BLURB>Sending retransmission event when this much reordering (-1 automatic).</BLURB>
+<BLURB>Sending retransmission event when this much reordering (0 disable).</BLURB>
 <DEFAULT>3</DEFAULT>
 </ARG>
 
@@ -21469,36 +20829,146 @@
 </ARG>
 
 <ARG>
-<NAME>GstRtpSession::bandwidth</NAME>
-<TYPE>gdouble</TYPE>
-<RANGE>>= 0</RANGE>
+<NAME>GstRtpJitterBuffer::rtx-min-delay</NAME>
+<TYPE>guint</TYPE>
+<RANGE></RANGE>
 <FLAGS>rw</FLAGS>
-<NICK>Bandwidth</NICK>
-<BLURB>The bandwidth of the session in bytes per second (0 for auto-discover).</BLURB>
-<DEFAULT>64000</DEFAULT>
+<NICK>Minimum RTX Delay</NICK>
+<BLURB>Minimum time in ms to wait before sending retransmission event.</BLURB>
+<DEFAULT>0</DEFAULT>
 </ARG>
 
 <ARG>
-<NAME>GstRtpSession::internal-session</NAME>
-<TYPE>RTPSession*</TYPE>
-<RANGE></RANGE>
-<FLAGS>r</FLAGS>
-<NICK>Internal Session</NICK>
-<BLURB>The internal RTPSession object.</BLURB>
-<DEFAULT></DEFAULT>
-</ARG>
-
+<NAME>GstRtpJitterBuffer::rtx-min-retry-timeout</NAME>
+<TYPE>gint</TYPE>
+<RANGE>>= G_MAXULONG</RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>RTX Min Retry Timeout</NICK>
+<BLURB>Minimum timeout between sending a transmission event in ms (-1 automatic).</BLURB>
+<DEFAULT>-1</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstRtpJitterBuffer::rtx-max-retries</NAME>
+<TYPE>gint</TYPE>
+<RANGE>>= G_MAXULONG</RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>RTX Max Retries</NICK>
+<BLURB>The maximum number of retries to request a retransmission. (-1 not limited).</BLURB>
+<DEFAULT>-1</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstRtpJitterBuffer::rtx-next-seqnum</NAME>
+<TYPE>gboolean</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>RTX next seqnum</NICK>
+<BLURB>Estimate when the next packet should arrive and schedule a retransmission request for it.</BLURB>
+<DEFAULT>TRUE</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstRtpJitterBuffer::max-dropout-time</NAME>
+<TYPE>guint</TYPE>
+<RANGE><= G_MAXINT</RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Max dropout time</NICK>
+<BLURB>The maximum time (milliseconds) of missing packets tolerated.</BLURB>
+<DEFAULT>60000</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstRtpJitterBuffer::max-misorder-time</NAME>
+<TYPE>guint</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Max misorder time</NICK>
+<BLURB>The maximum time (milliseconds) of misordered packets tolerated.</BLURB>
+<DEFAULT>2000</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstRtpJitterBuffer::max-rtcp-rtp-time-diff</NAME>
+<TYPE>gint</TYPE>
+<RANGE>>= G_MAXULONG</RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Max RTCP RTP Time Diff</NICK>
+<BLURB>Maximum amount of time in ms that the RTP time in RTCP SRs is allowed to be ahead (-1 disabled).</BLURB>
+<DEFAULT>1000</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstRtpJitterBuffer::rfc7273-sync</NAME>
+<TYPE>gboolean</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Sync on RFC7273 clock</NICK>
+<BLURB>Synchronize received streams to the RFC7273 clock (requires clock and offset to be provided).</BLURB>
+<DEFAULT>FALSE</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstRtpJitterBuffer::rtx-deadline</NAME>
+<TYPE>gint</TYPE>
+<RANGE>>= G_MAXULONG</RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>RTX Deadline (ms)</NICK>
+<BLURB>The deadline for a valid RTX request in milliseconds. (-1 automatic).</BLURB>
+<DEFAULT>-1</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstRtpJitterBuffer::rtx-stats-timeout</NAME>
+<TYPE>guint</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>RTX Statistics Timeout</NICK>
+<BLURB>The time to wait for a retransmitted packet after it has been considered lost in order to collect statistics (ms).</BLURB>
+<DEFAULT>1000</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstRtpJitterBuffer::faststart-min-packets</NAME>
+<TYPE>guint</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Faststart minimum packets</NICK>
+<BLURB>The number of consecutive packets needed to start (set to 0 to disable faststart. The jitterbuffer will by default start after the latency has elapsed).</BLURB>
+<DEFAULT>0</DEFAULT>
+</ARG>
+
 <ARG>
-<NAME>GstRtpSession::ntp-ns-base</NAME>
+<NAME>GstRtpJitterBuffer::max-ts-offset-adjustment</NAME>
 <TYPE>guint64</TYPE>
 <RANGE></RANGE>
 <FLAGS>rw</FLAGS>
-<NICK>NTP base time</NICK>
-<BLURB>The NTP base time corresponding to running_time 0 (deprecated).</BLURB>
+<NICK>Max Timestamp Offset Adjustment</NICK>
+<BLURB>The maximum number of nanoseconds per frame that time stamp offsets may be adjusted (0 = no limit).</BLURB>
+<DEFAULT>0</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstRtpSession::bandwidth</NAME>
+<TYPE>gdouble</TYPE>
+<RANGE>>= 0</RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Bandwidth</NICK>
+<BLURB>The bandwidth of the session in bytes per second (0 for auto-discover).</BLURB>
 <DEFAULT>0</DEFAULT>
 </ARG>
 
 <ARG>
+<NAME>GstRtpSession::internal-session</NAME>
+<TYPE>RTPSession*</TYPE>
+<RANGE></RANGE>
+<FLAGS>r</FLAGS>
+<NICK>Internal Session</NICK>
+<BLURB>The internal RTPSession object.</BLURB>
+<DEFAULT></DEFAULT>
+</ARG>
+
+<ARG>
 <NAME>GstRtpSession::num-active-sources</NAME>
 <TYPE>guint</TYPE>
 <RANGE></RANGE>
@@ -21525,7 +20995,7 @@
 <FLAGS>rw</FLAGS>
 <NICK>RTCP Fraction</NICK>
 <BLURB>The RTCP bandwidth of the session in bytes per second (or as a real fraction of the RTP bandwidth if < 1.0).</BLURB>
-<DEFAULT>3200</DEFAULT>
+<DEFAULT>0.05</DEFAULT>
 </ARG>
 
 <ARG>
@@ -21564,7 +21034,7 @@
 <RANGE></RANGE>
 <FLAGS>rw</FLAGS>
 <NICK>Use pipeline clock</NICK>
-<BLURB>Use the pipeline running-time to set the NTP time in the RTCP SR messages.</BLURB>
+<BLURB>Use the pipeline running-time to set the NTP time in the RTCP SR messages (DEPRECATED: Use ntp-time-source property).</BLURB>
 <DEFAULT>FALSE</DEFAULT>
 </ARG>
 
@@ -21599,13 +21069,53 @@
 </ARG>
 
 <ARG>
-<NAME>GstRtpRtxSend::rtx-payload-type</NAME>
+<NAME>GstRtpSession::rtp-profile</NAME>
+<TYPE>GstRTPProfile</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>RTP Profile</NICK>
+<BLURB>RTP profile to use.</BLURB>
+<DEFAULT>GST_RTP_PROFILE_AVP</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstRtpSession::ntp-time-source</NAME>
+<TYPE>GstRtpNtpTimeSource</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>NTP Time Source</NICK>
+<BLURB>NTP time source for RTCP packets.</BLURB>
+<DEFAULT>NTP time based on realtime clock</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstRtpSession::max-dropout-time</NAME>
 <TYPE>guint</TYPE>
 <RANGE></RANGE>
 <FLAGS>rw</FLAGS>
-<NICK>RTX Payload Type</NICK>
-<BLURB>Payload type of the retransmission stream (fmtp in SDP).</BLURB>
-<DEFAULT>0</DEFAULT>
+<NICK>Max dropout time</NICK>
+<BLURB>The maximum time (milliseconds) of missing packets tolerated.</BLURB>
+<DEFAULT>60000</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstRtpSession::max-misorder-time</NAME>
+<TYPE>guint</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Max misorder time</NICK>
+<BLURB>The maximum time (milliseconds) of misordered packets tolerated.</BLURB>
+<DEFAULT>2000</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstRtpSession::rtcp-sync-send-time</NAME>
+<TYPE>gboolean</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>RTCP Sync Send Time</NICK>
+<BLURB>Use send time or capture time for RTCP sync (TRUE = send time, FALSE = capture time).</BLURB>
+<DEFAULT>TRUE</DEFAULT>
 </ARG>
 
 <ARG>
@@ -21669,16 +21179,6 @@
 </ARG>
 
 <ARG>
-<NAME>GstRtpRtxReceive::rtx-payload-types</NAME>
-<TYPE>string</TYPE>
-<RANGE></RANGE>
-<FLAGS>rw</FLAGS>
-<NICK>Colon separated list of payload format type</NICK>
-<BLURB>Set through SDP (fmtp), it helps to detect restransmission streams.</BLURB>
-<DEFAULT>""</DEFAULT>
-</ARG>
-
-<ARG>
 <NAME>GstRtpRtxReceive::num-rtx-requests</NAME>
 <TYPE>guint</TYPE>
 <RANGE></RANGE>
@@ -21799,16 +21299,6 @@
 </ARG>
 
 <ARG>
-<NAME>GstV4l2Sink::queue-size</NAME>
-<TYPE>guint</TYPE>
-<RANGE>[1,16]</RANGE>
-<FLAGS>rw</FLAGS>
-<NICK>Queue size</NICK>
-<BLURB>Number of buffers to be enqueud in the driver in streaming mode.</BLURB>
-<DEFAULT>12</DEFAULT>
-</ARG>
-
-<ARG>
 <NAME>GstV4l2Sink::brightness</NAME>
 <TYPE>gint</TYPE>
 <RANGE></RANGE>
@@ -21889,16 +21379,6 @@
 </ARG>
 
 <ARG>
-<NAME>GstV4l2Sink::min-queued-bufs</NAME>
-<TYPE>guint</TYPE>
-<RANGE><= 16</RANGE>
-<FLAGS>rw</FLAGS>
-<NICK>Minimum queued bufs</NICK>
-<BLURB>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).</BLURB>
-<DEFAULT>1</DEFAULT>
-</ARG>
-
-<ARG>
 <NAME>GstV4l2Sink::io-mode</NAME>
 <TYPE>GstV4l2IOMode</TYPE>
 <RANGE></RANGE>
@@ -21969,23 +21449,33 @@
 </ARG>
 
 <ARG>
-<NAME>GstFlvMux::is-live</NAME>
+<NAME>GstFlvMux::streamable</NAME>
 <TYPE>gboolean</TYPE>
 <RANGE></RANGE>
 <FLAGS>rw</FLAGS>
-<NICK>Is Live</NICK>
-<BLURB>The stream is live and does not need an index.</BLURB>
+<NICK>streamable</NICK>
+<BLURB>If set to true, the output should be as if it is to be streamed and hence no indexes written or duration written.</BLURB>
 <DEFAULT>FALSE</DEFAULT>
 </ARG>
 
 <ARG>
-<NAME>GstFlvMux::streamable</NAME>
-<TYPE>gboolean</TYPE>
+<NAME>GstFlvMux::metadatacreator</NAME>
+<TYPE>gchar*</TYPE>
 <RANGE></RANGE>
 <FLAGS>rw</FLAGS>
-<NICK>streamable</NICK>
-<BLURB>If set to true, the output should be as if it is to be streamed and hence no indexes written or duration written.</BLURB>
-<DEFAULT>FALSE</DEFAULT>
+<NICK>metadatacreator</NICK>
+<BLURB>The value of metadatacreator in the meta packet.</BLURB>
+<DEFAULT>NULL</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstFlvMux::encoder</NAME>
+<TYPE>gchar*</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>encoder</NICK>
+<BLURB>The value of encoder in the meta packet.</BLURB>
+<DEFAULT>NULL</DEFAULT>
 </ARG>
 
 <ARG>
@@ -22019,26 +21509,6 @@
 </ARG>
 
 <ARG>
-<NAME>GstOss4Mixer::device</NAME>
-<TYPE>gchar*</TYPE>
-<RANGE></RANGE>
-<FLAGS>rw</FLAGS>
-<NICK>Device</NICK>
-<BLURB>OSS mixer device (e.g. /dev/oss/hdaudio0/mix0 or /dev/mixerN) (NULL = use first mixer device found).</BLURB>
-<DEFAULT>NULL</DEFAULT>
-</ARG>
-
-<ARG>
-<NAME>GstOss4Mixer::device-name</NAME>
-<TYPE>gchar*</TYPE>
-<RANGE></RANGE>
-<FLAGS>r</FLAGS>
-<NICK>Device name</NICK>
-<BLURB>Human-readable name of the sound device.</BLURB>
-<DEFAULT>NULL</DEFAULT>
-</ARG>
-
-<ARG>
 <NAME>GstOss4Source::device</NAME>
 <TYPE>gchar*</TYPE>
 <RANGE></RANGE>
@@ -22119,26 +21589,6 @@
 </ARG>
 
 <ARG>
-<NAME>GstRtpJ2KPay::buffer-list</NAME>
-<TYPE>gboolean</TYPE>
-<RANGE></RANGE>
-<FLAGS>rw</FLAGS>
-<NICK>Buffer List</NICK>
-<BLURB>Use Buffer Lists.</BLURB>
-<DEFAULT>TRUE</DEFAULT>
-</ARG>
-
-<ARG>
-<NAME>GstRtpJ2KDepay::buffer-list</NAME>
-<TYPE>gboolean</TYPE>
-<RANGE></RANGE>
-<FLAGS>rw</FLAGS>
-<NICK>Buffer List</NICK>
-<BLURB>Use Buffer Lists.</BLURB>
-<DEFAULT>TRUE</DEFAULT>
-</ARG>
-
-<ARG>
 <NAME>GstJackAudioSrc::client</NAME>
 <TYPE>JackClient*</TYPE>
 <RANGE></RANGE>
@@ -22189,6 +21639,16 @@
 </ARG>
 
 <ARG>
+<NAME>GstJackAudioSrc::port-pattern</NAME>
+<TYPE>gchar*</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>port pattern</NICK>
+<BLURB>A pattern to select which ports to connect to (NULL = first physical ports).</BLURB>
+<DEFAULT>NULL</DEFAULT>
+</ARG>
+
+<ARG>
 <NAME>GstJackAudioSink::client</NAME>
 <TYPE>JackClient*</TYPE>
 <RANGE></RANGE>
@@ -22239,6 +21699,16 @@
 </ARG>
 
 <ARG>
+<NAME>GstJackAudioSink::port-pattern</NAME>
+<TYPE>gchar*</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>port pattern</NICK>
+<BLURB>A pattern to select which ports to connect to (NULL = first physical ports).</BLURB>
+<DEFAULT>NULL</DEFAULT>
+</ARG>
+
+<ARG>
 <NAME>GstFlacParse::check-frame-checksums</NAME>
 <TYPE>gboolean</TYPE>
 <RANGE></RANGE>
@@ -22274,7 +21744,7 @@
 <RANGE></RANGE>
 <FLAGS>rwx</FLAGS>
 <NICK>dts-method</NICK>
-<BLURB>(DEPRECATED) Method to determine DTS time.</BLURB>
+<BLURB>Method to determine DTS time (DEPRECATED).</BLURB>
 <DEFAULT>reorder</DEFAULT>
 </ARG>
 
@@ -22321,11 +21791,11 @@
 <ARG>
 <NAME>GstQTMux::movie-timescale</NAME>
 <TYPE>guint</TYPE>
-<RANGE>>= 1</RANGE>
+<RANGE></RANGE>
 <FLAGS>rwx</FLAGS>
 <NICK>Movie timescale</NICK>
-<BLURB>Timescale to use in the movie (units per second).</BLURB>
-<DEFAULT>1000</DEFAULT>
+<BLURB>Timescale to use in the movie (units per second, 0 == default).</BLURB>
+<DEFAULT>0</DEFAULT>
 </ARG>
 
 <ARG>
@@ -22359,147 +21829,147 @@
 </ARG>
 
 <ARG>
-<NAME>GstQTMoovRecover::broken-input</NAME>
-<TYPE>gchar*</TYPE>
+<NAME>GstQTMux::reserved-bytes-per-sec</NAME>
+<TYPE>guint</TYPE>
+<RANGE><= 10000</RANGE>
+<FLAGS>rwx</FLAGS>
+<NICK>Reserved MOOV bytes per second, per track</NICK>
+<BLURB>Multiplier for converting reserved-max-duration into bytes of header to reserve, per second, per track.</BLURB>
+<DEFAULT>550</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstQTMux::reserved-duration-remaining</NAME>
+<TYPE>guint64</TYPE>
 <RANGE></RANGE>
-<FLAGS>rw</FLAGS>
-<NICK>Path to broken input file</NICK>
-<BLURB>Path to broken input file. (If qtmux was on faststart mode, this file is the faststart file).</BLURB>
-<DEFAULT>NULL</DEFAULT>
+<FLAGS>r</FLAGS>
+<NICK>Report the approximate amount of remaining recording space (ns)</NICK>
+<BLURB>Reports the approximate amount of remaining moov header space reserved using reserved-max-duration.</BLURB>
+<DEFAULT>0</DEFAULT>
 </ARG>
 
 <ARG>
-<NAME>GstQTMoovRecover::faststart-mode</NAME>
-<TYPE>gboolean</TYPE>
+<NAME>GstQTMux::reserved-max-duration</NAME>
+<TYPE>guint64</TYPE>
 <RANGE></RANGE>
 <FLAGS>rw</FLAGS>
-<NICK>If the broken input is from faststart mode</NICK>
-<BLURB>If the broken input is from faststart mode.</BLURB>
-<DEFAULT>FALSE</DEFAULT>
+<NICK>Reserved maximum file duration (ns)</NICK>
+<BLURB>When set to a value > 0, reserves space for index tables at the beginning of the file.</BLURB>
+<DEFAULT>18446744073709551615</DEFAULT>
 </ARG>
 
 <ARG>
-<NAME>GstQTMoovRecover::fixed-output</NAME>
-<TYPE>gchar*</TYPE>
+<NAME>GstQTMux::reserved-moov-update-period</NAME>
+<TYPE>guint64</TYPE>
 <RANGE></RANGE>
 <FLAGS>rw</FLAGS>
-<NICK>Path to write the fixed file</NICK>
-<BLURB>Path to write the fixed file to (used as output).</BLURB>
-<DEFAULT>NULL</DEFAULT>
+<NICK>Interval at which to update index tables (ns)</NICK>
+<BLURB>When used with reserved-max-duration, periodically updates the index tables with information muxed so far.</BLURB>
+<DEFAULT>18446744073709551615</DEFAULT>
 </ARG>
 
 <ARG>
-<NAME>GstQTMoovRecover::recovery-input</NAME>
-<TYPE>gchar*</TYPE>
+<NAME>GstQTMux::interleave-bytes</NAME>
+<TYPE>guint64</TYPE>
 <RANGE></RANGE>
 <FLAGS>rw</FLAGS>
-<NICK>Path to recovery file</NICK>
-<BLURB>Path to recovery file (used as input).</BLURB>
-<DEFAULT>NULL</DEFAULT>
+<NICK>Interleave (bytes)</NICK>
+<BLURB>Interleave between streams in bytes.</BLURB>
+<DEFAULT>0</DEFAULT>
 </ARG>
 
 <ARG>
-<NAME>GstMP4Mux::dts-method</NAME>
-<TYPE>GstQTMuxDtsMethods</TYPE>
+<NAME>GstQTMux::interleave-time</NAME>
+<TYPE>guint64</TYPE>
 <RANGE></RANGE>
-<FLAGS>rwx</FLAGS>
-<NICK>dts-method</NICK>
-<BLURB>(DEPRECATED) Method to determine DTS time.</BLURB>
-<DEFAULT>reorder</DEFAULT>
+<FLAGS>rw</FLAGS>
+<NICK>Interleave (time)</NICK>
+<BLURB>Interleave between streams in nanoseconds.</BLURB>
+<DEFAULT>250000000</DEFAULT>
 </ARG>
 
 <ARG>
-<NAME>GstMP4Mux::faststart</NAME>
-<TYPE>gboolean</TYPE>
+<NAME>GstQTMux::max-raw-audio-drift</NAME>
+<TYPE>guint64</TYPE>
 <RANGE></RANGE>
 <FLAGS>rw</FLAGS>
-<NICK>Format file to faststart</NICK>
-<BLURB>If the file should be formatted for faststart (headers first).</BLURB>
-<DEFAULT>FALSE</DEFAULT>
+<NICK>Max Raw Audio Drift</NICK>
+<BLURB>Maximum allowed drift of raw audio samples vs. timestamps in nanoseconds.</BLURB>
+<DEFAULT>40000000</DEFAULT>
 </ARG>
 
 <ARG>
-<NAME>GstMP4Mux::faststart-file</NAME>
-<TYPE>gchar*</TYPE>
+<NAME>GstQTMux::reserved-prefill</NAME>
+<TYPE>gboolean</TYPE>
 <RANGE></RANGE>
 <FLAGS>rwx</FLAGS>
-<NICK>File to use for storing buffers</NICK>
-<BLURB>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.</BLURB>
-<DEFAULT>NULL</DEFAULT>
+<NICK>Reserved Prefill Samples Table</NICK>
+<BLURB>Prefill samples table of reserved duration.</BLURB>
+<DEFAULT>FALSE</DEFAULT>
 </ARG>
 
 <ARG>
-<NAME>GstMP4Mux::fragment-duration</NAME>
-<TYPE>guint</TYPE>
+<NAME>GstQTMux::start-gap-threshold</NAME>
+<TYPE>guint64</TYPE>
 <RANGE></RANGE>
-<FLAGS>rwx</FLAGS>
-<NICK>Fragment duration</NICK>
-<BLURB>Fragment durations in ms (produce a fragmented file if > 0).</BLURB>
+<FLAGS>rw</FLAGS>
+<NICK>Start Gap Threshold</NICK>
+<BLURB>Threshold for creating an edit list for gaps at the start in nanoseconds.</BLURB>
 <DEFAULT>0</DEFAULT>
 </ARG>
 
 <ARG>
-<NAME>GstMP4Mux::moov-recovery-file</NAME>
+<NAME>GstQTMoovRecover::broken-input</NAME>
 <TYPE>gchar*</TYPE>
 <RANGE></RANGE>
-<FLAGS>rwx</FLAGS>
-<NICK>File to store data for posterior moov atom recovery</NICK>
-<BLURB>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).</BLURB>
+<FLAGS>rw</FLAGS>
+<NICK>Path to broken input file</NICK>
+<BLURB>Path to broken input file. (If qtmux was on faststart mode, this file is the faststart file).</BLURB>
 <DEFAULT>NULL</DEFAULT>
 </ARG>
 
 <ARG>
-<NAME>GstMP4Mux::movie-timescale</NAME>
-<TYPE>guint</TYPE>
-<RANGE>>= 1</RANGE>
-<FLAGS>rwx</FLAGS>
-<NICK>Movie timescale</NICK>
-<BLURB>Timescale to use in the movie (units per second).</BLURB>
-<DEFAULT>1000</DEFAULT>
-</ARG>
-
-<ARG>
-<NAME>GstMP4Mux::presentation-time</NAME>
+<NAME>GstQTMoovRecover::faststart-mode</NAME>
 <TYPE>gboolean</TYPE>
 <RANGE></RANGE>
-<FLAGS>rwx</FLAGS>
-<NICK>Include presentation-time info</NICK>
-<BLURB>Calculate and include presentation/composition time (in addition to decoding time).</BLURB>
-<DEFAULT>TRUE</DEFAULT>
+<FLAGS>rw</FLAGS>
+<NICK>If the broken input is from faststart mode</NICK>
+<BLURB>If the broken input is from faststart mode.</BLURB>
+<DEFAULT>FALSE</DEFAULT>
 </ARG>
 
 <ARG>
-<NAME>GstMP4Mux::streamable</NAME>
-<TYPE>gboolean</TYPE>
+<NAME>GstQTMoovRecover::fixed-output</NAME>
+<TYPE>gchar*</TYPE>
 <RANGE></RANGE>
-<FLAGS>rwx</FLAGS>
-<NICK>Streamable</NICK>
-<BLURB>If set to true, the output should be as if it is to be streamed and hence no indexes written or duration written. (DEPRECATED, only valid for fragmented MP4).</BLURB>
-<DEFAULT>FALSE</DEFAULT>
+<FLAGS>rw</FLAGS>
+<NICK>Path to write the fixed file</NICK>
+<BLURB>Path to write the fixed file to (used as output).</BLURB>
+<DEFAULT>NULL</DEFAULT>
 </ARG>
 
 <ARG>
-<NAME>GstMP4Mux::trak-timescale</NAME>
-<TYPE>guint</TYPE>
+<NAME>GstQTMoovRecover::recovery-input</NAME>
+<TYPE>gchar*</TYPE>
 <RANGE></RANGE>
-<FLAGS>rwx</FLAGS>
-<NICK>Track timescale</NICK>
-<BLURB>Timescale to use for the tracks (units per second, 0 is automatic).</BLURB>
-<DEFAULT>0</DEFAULT>
+<FLAGS>rw</FLAGS>
+<NICK>Path to recovery file</NICK>
+<BLURB>Path to recovery file (used as input).</BLURB>
+<DEFAULT>NULL</DEFAULT>
 </ARG>
 
 <ARG>
-<NAME>GstMJ2Mux::dts-method</NAME>
+<NAME>GstMP4Mux::dts-method</NAME>
 <TYPE>GstQTMuxDtsMethods</TYPE>
 <RANGE></RANGE>
 <FLAGS>rwx</FLAGS>
 <NICK>dts-method</NICK>
-<BLURB>(DEPRECATED) Method to determine DTS time.</BLURB>
+<BLURB>Method to determine DTS time (DEPRECATED).</BLURB>
 <DEFAULT>reorder</DEFAULT>
 </ARG>
 
 <ARG>
-<NAME>GstMJ2Mux::faststart</NAME>
+<NAME>GstMP4Mux::faststart</NAME>
 <TYPE>gboolean</TYPE>
 <RANGE></RANGE>
 <FLAGS>rw</FLAGS>
@@ -22509,7 +21979,7 @@
 </ARG>
 
 <ARG>
-<NAME>GstMJ2Mux::faststart-file</NAME>
+<NAME>GstMP4Mux::faststart-file</NAME>
 <TYPE>gchar*</TYPE>
 <RANGE></RANGE>
 <FLAGS>rwx</FLAGS>
@@ -22519,7 +21989,7 @@
 </ARG>
 
 <ARG>
-<NAME>GstMJ2Mux::fragment-duration</NAME>
+<NAME>GstMP4Mux::fragment-duration</NAME>
 <TYPE>guint</TYPE>
 <RANGE></RANGE>
 <FLAGS>rwx</FLAGS>
@@ -22529,7 +21999,7 @@
 </ARG>
 
 <ARG>
-<NAME>GstMJ2Mux::moov-recovery-file</NAME>
+<NAME>GstMP4Mux::moov-recovery-file</NAME>
 <TYPE>gchar*</TYPE>
 <RANGE></RANGE>
 <FLAGS>rwx</FLAGS>
@@ -22539,17 +22009,17 @@
 </ARG>
 
 <ARG>
-<NAME>GstMJ2Mux::movie-timescale</NAME>
+<NAME>GstMP4Mux::movie-timescale</NAME>
 <TYPE>guint</TYPE>
-<RANGE>>= 1</RANGE>
+<RANGE></RANGE>
 <FLAGS>rwx</FLAGS>
 <NICK>Movie timescale</NICK>
-<BLURB>Timescale to use in the movie (units per second).</BLURB>
-<DEFAULT>1000</DEFAULT>
+<BLURB>Timescale to use in the movie (units per second, 0 == default).</BLURB>
+<DEFAULT>0</DEFAULT>
 </ARG>
 
 <ARG>
-<NAME>GstMJ2Mux::presentation-time</NAME>
+<NAME>GstMP4Mux::presentation-time</NAME>
 <TYPE>gboolean</TYPE>
 <RANGE></RANGE>
 <FLAGS>rwx</FLAGS>
@@ -22559,7 +22029,7 @@
 </ARG>
 
 <ARG>
-<NAME>GstMJ2Mux::streamable</NAME>
+<NAME>GstMP4Mux::streamable</NAME>
 <TYPE>gboolean</TYPE>
 <RANGE></RANGE>
 <FLAGS>rwx</FLAGS>
@@ -22569,7 +22039,7 @@
 </ARG>
 
 <ARG>
-<NAME>GstMJ2Mux::trak-timescale</NAME>
+<NAME>GstMP4Mux::trak-timescale</NAME>
 <TYPE>guint</TYPE>
 <RANGE></RANGE>
 <FLAGS>rwx</FLAGS>
@@ -22579,107 +22049,107 @@
 </ARG>
 
 <ARG>
-<NAME>GstISMLMux::dts-method</NAME>
-<TYPE>GstQTMuxDtsMethods</TYPE>
-<RANGE></RANGE>
+<NAME>GstMP4Mux::reserved-bytes-per-sec</NAME>
+<TYPE>guint</TYPE>
+<RANGE><= 10000</RANGE>
 <FLAGS>rwx</FLAGS>
-<NICK>dts-method</NICK>
-<BLURB>(DEPRECATED) Method to determine DTS time.</BLURB>
-<DEFAULT>reorder</DEFAULT>
+<NICK>Reserved MOOV bytes per second, per track</NICK>
+<BLURB>Multiplier for converting reserved-max-duration into bytes of header to reserve, per second, per track.</BLURB>
+<DEFAULT>550</DEFAULT>
 </ARG>
 
 <ARG>
-<NAME>GstISMLMux::faststart</NAME>
-<TYPE>gboolean</TYPE>
+<NAME>GstMP4Mux::reserved-duration-remaining</NAME>
+<TYPE>guint64</TYPE>
 <RANGE></RANGE>
-<FLAGS>rw</FLAGS>
-<NICK>Format file to faststart</NICK>
-<BLURB>If the file should be formatted for faststart (headers first).</BLURB>
-<DEFAULT>FALSE</DEFAULT>
+<FLAGS>r</FLAGS>
+<NICK>Report the approximate amount of remaining recording space (ns)</NICK>
+<BLURB>Reports the approximate amount of remaining moov header space reserved using reserved-max-duration.</BLURB>
+<DEFAULT>0</DEFAULT>
 </ARG>
 
 <ARG>
-<NAME>GstISMLMux::faststart-file</NAME>
-<TYPE>gchar*</TYPE>
+<NAME>GstMP4Mux::reserved-max-duration</NAME>
+<TYPE>guint64</TYPE>
 <RANGE></RANGE>
-<FLAGS>rwx</FLAGS>
-<NICK>File to use for storing buffers</NICK>
-<BLURB>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.</BLURB>
-<DEFAULT>NULL</DEFAULT>
+<FLAGS>rw</FLAGS>
+<NICK>Reserved maximum file duration (ns)</NICK>
+<BLURB>When set to a value > 0, reserves space for index tables at the beginning of the file.</BLURB>
+<DEFAULT>18446744073709551615</DEFAULT>
 </ARG>
 
 <ARG>
-<NAME>GstISMLMux::fragment-duration</NAME>
-<TYPE>guint</TYPE>
+<NAME>GstMP4Mux::reserved-moov-update-period</NAME>
+<TYPE>guint64</TYPE>
 <RANGE></RANGE>
-<FLAGS>rwx</FLAGS>
-<NICK>Fragment duration</NICK>
-<BLURB>Fragment durations in ms (produce a fragmented file if > 0).</BLURB>
-<DEFAULT>2000</DEFAULT>
+<FLAGS>rw</FLAGS>
+<NICK>Interval at which to update index tables (ns)</NICK>
+<BLURB>When used with reserved-max-duration, periodically updates the index tables with information muxed so far.</BLURB>
+<DEFAULT>18446744073709551615</DEFAULT>
 </ARG>
 
 <ARG>
-<NAME>GstISMLMux::moov-recovery-file</NAME>
-<TYPE>gchar*</TYPE>
+<NAME>GstMP4Mux::interleave-bytes</NAME>
+<TYPE>guint64</TYPE>
 <RANGE></RANGE>
-<FLAGS>rwx</FLAGS>
-<NICK>File to store data for posterior moov atom recovery</NICK>
-<BLURB>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).</BLURB>
-<DEFAULT>NULL</DEFAULT>
+<FLAGS>rw</FLAGS>
+<NICK>Interleave (bytes)</NICK>
+<BLURB>Interleave between streams in bytes.</BLURB>
+<DEFAULT>0</DEFAULT>
 </ARG>
 
 <ARG>
-<NAME>GstISMLMux::movie-timescale</NAME>
-<TYPE>guint</TYPE>
-<RANGE>>= 1</RANGE>
-<FLAGS>rwx</FLAGS>
-<NICK>Movie timescale</NICK>
-<BLURB>Timescale to use in the movie (units per second).</BLURB>
-<DEFAULT>1000</DEFAULT>
+<NAME>GstMP4Mux::interleave-time</NAME>
+<TYPE>guint64</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Interleave (time)</NICK>
+<BLURB>Interleave between streams in nanoseconds.</BLURB>
+<DEFAULT>250000000</DEFAULT>
 </ARG>
 
 <ARG>
-<NAME>GstISMLMux::presentation-time</NAME>
-<TYPE>gboolean</TYPE>
+<NAME>GstMP4Mux::max-raw-audio-drift</NAME>
+<TYPE>guint64</TYPE>
 <RANGE></RANGE>
-<FLAGS>rwx</FLAGS>
-<NICK>Include presentation-time info</NICK>
-<BLURB>Calculate and include presentation/composition time (in addition to decoding time).</BLURB>
-<DEFAULT>TRUE</DEFAULT>
+<FLAGS>rw</FLAGS>
+<NICK>Max Raw Audio Drift</NICK>
+<BLURB>Maximum allowed drift of raw audio samples vs. timestamps in nanoseconds.</BLURB>
+<DEFAULT>40000000</DEFAULT>
 </ARG>
 
 <ARG>
-<NAME>GstISMLMux::streamable</NAME>
+<NAME>GstMP4Mux::reserved-prefill</NAME>
 <TYPE>gboolean</TYPE>
 <RANGE></RANGE>
 <FLAGS>rwx</FLAGS>
-<NICK>Streamable</NICK>
-<BLURB>If set to true, the output should be as if it is to be streamed and hence no indexes written or duration written.</BLURB>
-<DEFAULT>TRUE</DEFAULT>
+<NICK>Reserved Prefill Samples Table</NICK>
+<BLURB>Prefill samples table of reserved duration.</BLURB>
+<DEFAULT>FALSE</DEFAULT>
 </ARG>
 
 <ARG>
-<NAME>GstISMLMux::trak-timescale</NAME>
-<TYPE>guint</TYPE>
+<NAME>GstMP4Mux::start-gap-threshold</NAME>
+<TYPE>guint64</TYPE>
 <RANGE></RANGE>
-<FLAGS>rwx</FLAGS>
-<NICK>Track timescale</NICK>
-<BLURB>Timescale to use for the tracks (units per second, 0 is automatic).</BLURB>
+<FLAGS>rw</FLAGS>
+<NICK>Start Gap Threshold</NICK>
+<BLURB>Threshold for creating an edit list for gaps at the start in nanoseconds.</BLURB>
 <DEFAULT>0</DEFAULT>
 </ARG>
 
 <ARG>
-<NAME>GstGPPMux::dts-method</NAME>
+<NAME>GstMJ2Mux::dts-method</NAME>
 <TYPE>GstQTMuxDtsMethods</TYPE>
 <RANGE></RANGE>
 <FLAGS>rwx</FLAGS>
 <NICK>dts-method</NICK>
-<BLURB>Method to determine DTS time.</BLURB>
+<BLURB>Method to determine DTS time (DEPRECATED).</BLURB>
 <DEFAULT>reorder</DEFAULT>
 </ARG>
 
 <ARG>
-<NAME>GstGPPMux::faststart</NAME>
+<NAME>GstMJ2Mux::faststart</NAME>
 <TYPE>gboolean</TYPE>
 <RANGE></RANGE>
 <FLAGS>rw</FLAGS>
@@ -22689,7 +22159,7 @@
 </ARG>
 
 <ARG>
-<NAME>GstGPPMux::faststart-file</NAME>
+<NAME>GstMJ2Mux::faststart-file</NAME>
 <TYPE>gchar*</TYPE>
 <RANGE></RANGE>
 <FLAGS>rwx</FLAGS>
@@ -22699,7 +22169,7 @@
 </ARG>
 
 <ARG>
-<NAME>GstGPPMux::fragment-duration</NAME>
+<NAME>GstMJ2Mux::fragment-duration</NAME>
 <TYPE>guint</TYPE>
 <RANGE></RANGE>
 <FLAGS>rwx</FLAGS>
@@ -22709,7 +22179,7 @@
 </ARG>
 
 <ARG>
-<NAME>GstGPPMux::moov-recovery-file</NAME>
+<NAME>GstMJ2Mux::moov-recovery-file</NAME>
 <TYPE>gchar*</TYPE>
 <RANGE></RANGE>
 <FLAGS>rwx</FLAGS>
@@ -22719,17 +22189,17 @@
 </ARG>
 
 <ARG>
-<NAME>GstGPPMux::movie-timescale</NAME>
+<NAME>GstMJ2Mux::movie-timescale</NAME>
 <TYPE>guint</TYPE>
-<RANGE>>= 1</RANGE>
+<RANGE></RANGE>
 <FLAGS>rwx</FLAGS>
 <NICK>Movie timescale</NICK>
-<BLURB>Timescale to use in the movie (units per second).</BLURB>
-<DEFAULT>1000</DEFAULT>
+<BLURB>Timescale to use in the movie (units per second, 0 == default).</BLURB>
+<DEFAULT>0</DEFAULT>
 </ARG>
 
 <ARG>
-<NAME>GstGPPMux::presentation-time</NAME>
+<NAME>GstMJ2Mux::presentation-time</NAME>
 <TYPE>gboolean</TYPE>
 <RANGE></RANGE>
 <FLAGS>rwx</FLAGS>
@@ -22739,17 +22209,17 @@
 </ARG>
 
 <ARG>
-<NAME>GstGPPMux::streamable</NAME>
+<NAME>GstMJ2Mux::streamable</NAME>
 <TYPE>gboolean</TYPE>
 <RANGE></RANGE>
 <FLAGS>rwx</FLAGS>
 <NICK>Streamable</NICK>
-<BLURB>If set to true, the output should be as if it is to be streamed and hence no indexes written or duration written.</BLURB>
+<BLURB>If set to true, the output should be as if it is to be streamed and hence no indexes written or duration written. (DEPRECATED, only valid for fragmented MP4).</BLURB>
 <DEFAULT>FALSE</DEFAULT>
 </ARG>
 
 <ARG>
-<NAME>GstGPPMux::trak-timescale</NAME>
+<NAME>GstMJ2Mux::trak-timescale</NAME>
 <TYPE>guint</TYPE>
 <RANGE></RANGE>
 <FLAGS>rwx</FLAGS>
@@ -22759,17 +22229,107 @@
 </ARG>
 
 <ARG>
-<NAME>Gst3GPPMux::dts-method</NAME>
+<NAME>GstMJ2Mux::reserved-bytes-per-sec</NAME>
+<TYPE>guint</TYPE>
+<RANGE><= 10000</RANGE>
+<FLAGS>rwx</FLAGS>
+<NICK>Reserved MOOV bytes per second, per track</NICK>
+<BLURB>Multiplier for converting reserved-max-duration into bytes of header to reserve, per second, per track.</BLURB>
+<DEFAULT>550</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstMJ2Mux::reserved-duration-remaining</NAME>
+<TYPE>guint64</TYPE>
+<RANGE></RANGE>
+<FLAGS>r</FLAGS>
+<NICK>Report the approximate amount of remaining recording space (ns)</NICK>
+<BLURB>Reports the approximate amount of remaining moov header space reserved using reserved-max-duration.</BLURB>
+<DEFAULT>0</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstMJ2Mux::reserved-max-duration</NAME>
+<TYPE>guint64</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Reserved maximum file duration (ns)</NICK>
+<BLURB>When set to a value > 0, reserves space for index tables at the beginning of the file.</BLURB>
+<DEFAULT>18446744073709551615</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstMJ2Mux::reserved-moov-update-period</NAME>
+<TYPE>guint64</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Interval at which to update index tables (ns)</NICK>
+<BLURB>When used with reserved-max-duration, periodically updates the index tables with information muxed so far.</BLURB>
+<DEFAULT>18446744073709551615</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstMJ2Mux::interleave-bytes</NAME>
+<TYPE>guint64</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Interleave (bytes)</NICK>
+<BLURB>Interleave between streams in bytes.</BLURB>
+<DEFAULT>0</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstMJ2Mux::interleave-time</NAME>
+<TYPE>guint64</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Interleave (time)</NICK>
+<BLURB>Interleave between streams in nanoseconds.</BLURB>
+<DEFAULT>250000000</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstMJ2Mux::max-raw-audio-drift</NAME>
+<TYPE>guint64</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Max Raw Audio Drift</NICK>
+<BLURB>Maximum allowed drift of raw audio samples vs. timestamps in nanoseconds.</BLURB>
+<DEFAULT>40000000</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstMJ2Mux::reserved-prefill</NAME>
+<TYPE>gboolean</TYPE>
+<RANGE></RANGE>
+<FLAGS>rwx</FLAGS>
+<NICK>Reserved Prefill Samples Table</NICK>
+<BLURB>Prefill samples table of reserved duration.</BLURB>
+<DEFAULT>FALSE</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstMJ2Mux::start-gap-threshold</NAME>
+<TYPE>guint64</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Start Gap Threshold</NICK>
+<BLURB>Threshold for creating an edit list for gaps at the start in nanoseconds.</BLURB>
+<DEFAULT>0</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstISMLMux::dts-method</NAME>
 <TYPE>GstQTMuxDtsMethods</TYPE>
 <RANGE></RANGE>
 <FLAGS>rwx</FLAGS>
 <NICK>dts-method</NICK>
-<BLURB>(DEPRECATED) Method to determine DTS time.</BLURB>
+<BLURB>Method to determine DTS time (DEPRECATED).</BLURB>
 <DEFAULT>reorder</DEFAULT>
 </ARG>
 
 <ARG>
-<NAME>Gst3GPPMux::faststart</NAME>
+<NAME>GstISMLMux::faststart</NAME>
 <TYPE>gboolean</TYPE>
 <RANGE></RANGE>
 <FLAGS>rw</FLAGS>
@@ -22779,7 +22339,7 @@
 </ARG>
 
 <ARG>
-<NAME>Gst3GPPMux::faststart-file</NAME>
+<NAME>GstISMLMux::faststart-file</NAME>
 <TYPE>gchar*</TYPE>
 <RANGE></RANGE>
 <FLAGS>rwx</FLAGS>
@@ -22789,17 +22349,17 @@
 </ARG>
 
 <ARG>
-<NAME>Gst3GPPMux::fragment-duration</NAME>
+<NAME>GstISMLMux::fragment-duration</NAME>
 <TYPE>guint</TYPE>
 <RANGE></RANGE>
 <FLAGS>rwx</FLAGS>
 <NICK>Fragment duration</NICK>
 <BLURB>Fragment durations in ms (produce a fragmented file if > 0).</BLURB>
-<DEFAULT>0</DEFAULT>
+<DEFAULT>2000</DEFAULT>
 </ARG>
 
 <ARG>
-<NAME>Gst3GPPMux::moov-recovery-file</NAME>
+<NAME>GstISMLMux::moov-recovery-file</NAME>
 <TYPE>gchar*</TYPE>
 <RANGE></RANGE>
 <FLAGS>rwx</FLAGS>
@@ -22809,17 +22369,17 @@
 </ARG>
 
 <ARG>
-<NAME>Gst3GPPMux::movie-timescale</NAME>
+<NAME>GstISMLMux::movie-timescale</NAME>
 <TYPE>guint</TYPE>
-<RANGE>>= 1</RANGE>
+<RANGE></RANGE>
 <FLAGS>rwx</FLAGS>
 <NICK>Movie timescale</NICK>
-<BLURB>Timescale to use in the movie (units per second).</BLURB>
-<DEFAULT>1000</DEFAULT>
+<BLURB>Timescale to use in the movie (units per second, 0 == default).</BLURB>
+<DEFAULT>0</DEFAULT>
 </ARG>
 
 <ARG>
-<NAME>Gst3GPPMux::presentation-time</NAME>
+<NAME>GstISMLMux::presentation-time</NAME>
 <TYPE>gboolean</TYPE>
 <RANGE></RANGE>
 <FLAGS>rwx</FLAGS>
@@ -22829,17 +22389,17 @@
 </ARG>
 
 <ARG>
-<NAME>Gst3GPPMux::streamable</NAME>
+<NAME>GstISMLMux::streamable</NAME>
 <TYPE>gboolean</TYPE>
 <RANGE></RANGE>
 <FLAGS>rwx</FLAGS>
 <NICK>Streamable</NICK>
-<BLURB>If set to true, the output should be as if it is to be streamed and hence no indexes written or duration written. (DEPRECATED, only valid for fragmented MP4).</BLURB>
-<DEFAULT>FALSE</DEFAULT>
+<BLURB>If set to true, the output should be as if it is to be streamed and hence no indexes written or duration written.</BLURB>
+<DEFAULT>TRUE</DEFAULT>
 </ARG>
 
 <ARG>
-<NAME>Gst3GPPMux::trak-timescale</NAME>
+<NAME>GstISMLMux::trak-timescale</NAME>
 <TYPE>guint</TYPE>
 <RANGE></RANGE>
 <FLAGS>rwx</FLAGS>
@@ -22849,283 +22409,303 @@
 </ARG>
 
 <ARG>
-<NAME>GstSplitFileSrc::location</NAME>
-<TYPE>gchar*</TYPE>
-<RANGE></RANGE>
-<FLAGS>rw</FLAGS>
-<NICK>File Location</NICK>
-<BLURB>Wildcard pattern to match file names of the input files. If the location is an absolute path or contains directory components, only the base file name part will be considered for pattern matching. The results will be sorted.</BLURB>
-<DEFAULT>NULL</DEFAULT>
+<NAME>GstISMLMux::reserved-bytes-per-sec</NAME>
+<TYPE>guint</TYPE>
+<RANGE><= 10000</RANGE>
+<FLAGS>rwx</FLAGS>
+<NICK>Reserved MOOV bytes per second, per track</NICK>
+<BLURB>Multiplier for converting reserved-max-duration into bytes of header to reserve, per second, per track.</BLURB>
+<DEFAULT>550</DEFAULT>
 </ARG>
 
 <ARG>
-<NAME>GstMatroskaDemux::max-gap-time</NAME>
+<NAME>GstISMLMux::reserved-duration-remaining</NAME>
 <TYPE>guint64</TYPE>
 <RANGE></RANGE>
-<FLAGS>rw</FLAGS>
-<NICK>Maximum gap time</NICK>
-<BLURB>The demuxer sends out segment events for skipping gaps longer than this (0 = disabled).</BLURB>
-<DEFAULT>2000000000</DEFAULT>
+<FLAGS>r</FLAGS>
+<NICK>Report the approximate amount of remaining recording space (ns)</NICK>
+<BLURB>Reports the approximate amount of remaining moov header space reserved using reserved-max-duration.</BLURB>
+<DEFAULT>0</DEFAULT>
 </ARG>
 
 <ARG>
-<NAME>GstPulseAudioSink::alignment-threshold</NAME>
+<NAME>GstISMLMux::reserved-max-duration</NAME>
 <TYPE>guint64</TYPE>
-<RANGE>[1,18446744073709551614]</RANGE>
+<RANGE></RANGE>
 <FLAGS>rw</FLAGS>
-<NICK>Alignment Threshold</NICK>
-<BLURB>Timestamp alignment threshold in nanoseconds.</BLURB>
-<DEFAULT>40000000</DEFAULT>
+<NICK>Reserved maximum file duration (ns)</NICK>
+<BLURB>When set to a value > 0, reserves space for index tables at the beginning of the file.</BLURB>
+<DEFAULT>18446744073709551615</DEFAULT>
 </ARG>
 
 <ARG>
-<NAME>GstPulseAudioSink::async</NAME>
-<TYPE>gboolean</TYPE>
+<NAME>GstISMLMux::reserved-moov-update-period</NAME>
+<TYPE>guint64</TYPE>
 <RANGE></RANGE>
 <FLAGS>rw</FLAGS>
-<NICK>Async</NICK>
-<BLURB>Go asynchronously to PAUSED.</BLURB>
-<DEFAULT>TRUE</DEFAULT>
+<NICK>Interval at which to update index tables (ns)</NICK>
+<BLURB>When used with reserved-max-duration, periodically updates the index tables with information muxed so far.</BLURB>
+<DEFAULT>18446744073709551615</DEFAULT>
 </ARG>
 
 <ARG>
-<NAME>GstPulseAudioSink::blocksize</NAME>
-<TYPE>guint</TYPE>
+<NAME>GstISMLMux::interleave-bytes</NAME>
+<TYPE>guint64</TYPE>
 <RANGE></RANGE>
 <FLAGS>rw</FLAGS>
-<NICK>Block size</NICK>
-<BLURB>Size in bytes to pull per buffer (0 = default).</BLURB>
-<DEFAULT>4096</DEFAULT>
+<NICK>Interleave (bytes)</NICK>
+<BLURB>Interleave between streams in bytes.</BLURB>
+<DEFAULT>0</DEFAULT>
 </ARG>
 
 <ARG>
-<NAME>GstPulseAudioSink::buffer-time</NAME>
-<TYPE>gint64</TYPE>
-<RANGE>>= 1</RANGE>
+<NAME>GstISMLMux::interleave-time</NAME>
+<TYPE>guint64</TYPE>
+<RANGE></RANGE>
 <FLAGS>rw</FLAGS>
-<NICK>Buffer Time</NICK>
-<BLURB>Size of audio buffer in microseconds.</BLURB>
-<DEFAULT>200000</DEFAULT>
+<NICK>Interleave (time)</NICK>
+<BLURB>Interleave between streams in nanoseconds.</BLURB>
+<DEFAULT>250000000</DEFAULT>
 </ARG>
 
 <ARG>
-<NAME>GstPulseAudioSink::can-activate-pull</NAME>
-<TYPE>gboolean</TYPE>
+<NAME>GstISMLMux::max-raw-audio-drift</NAME>
+<TYPE>guint64</TYPE>
 <RANGE></RANGE>
 <FLAGS>rw</FLAGS>
-<NICK>Allow Pull Scheduling</NICK>
-<BLURB>Allow pull-based scheduling.</BLURB>
-<DEFAULT>FALSE</DEFAULT>
+<NICK>Max Raw Audio Drift</NICK>
+<BLURB>Maximum allowed drift of raw audio samples vs. timestamps in nanoseconds.</BLURB>
+<DEFAULT>40000000</DEFAULT>
 </ARG>
 
 <ARG>
-<NAME>GstPulseAudioSink::client</NAME>
-<TYPE>gchar*</TYPE>
+<NAME>GstISMLMux::reserved-prefill</NAME>
+<TYPE>gboolean</TYPE>
 <RANGE></RANGE>
-<FLAGS>rw</FLAGS>
-<NICK>Client</NICK>
-<BLURB>The PulseAudio client name to use.</BLURB>
-<DEFAULT>"<unknown>"</DEFAULT>
+<FLAGS>rwx</FLAGS>
+<NICK>Reserved Prefill Samples Table</NICK>
+<BLURB>Prefill samples table of reserved duration.</BLURB>
+<DEFAULT>FALSE</DEFAULT>
 </ARG>
 
 <ARG>
-<NAME>GstPulseAudioSink::device</NAME>
-<TYPE>gchar*</TYPE>
+<NAME>GstISMLMux::start-gap-threshold</NAME>
+<TYPE>guint64</TYPE>
 <RANGE></RANGE>
 <FLAGS>rw</FLAGS>
-<NICK>Device</NICK>
-<BLURB>The PulseAudio sink device to connect to.</BLURB>
-<DEFAULT>NULL</DEFAULT>
+<NICK>Start Gap Threshold</NICK>
+<BLURB>Threshold for creating an edit list for gaps at the start in nanoseconds.</BLURB>
+<DEFAULT>0</DEFAULT>
 </ARG>
 
 <ARG>
-<NAME>GstPulseAudioSink::device-name</NAME>
-<TYPE>gchar*</TYPE>
+<NAME>Gst3GPPMux::dts-method</NAME>
+<TYPE>GstQTMuxDtsMethods</TYPE>
 <RANGE></RANGE>
-<FLAGS>r</FLAGS>
-<NICK>Device name</NICK>
-<BLURB>Human-readable name of the sound device.</BLURB>
-<DEFAULT>NULL</DEFAULT>
+<FLAGS>rwx</FLAGS>
+<NICK>dts-method</NICK>
+<BLURB>Method to determine DTS time (DEPRECATED).</BLURB>
+<DEFAULT>reorder</DEFAULT>
 </ARG>
 
 <ARG>
-<NAME>GstPulseAudioSink::discont-wait</NAME>
-<TYPE>guint64</TYPE>
-<RANGE><= 18446744073709551614</RANGE>
+<NAME>Gst3GPPMux::faststart</NAME>
+<TYPE>gboolean</TYPE>
+<RANGE></RANGE>
 <FLAGS>rw</FLAGS>
-<NICK>Discont Wait</NICK>
-<BLURB>Window of time in nanoseconds to wait before creating a discontinuity.</BLURB>
-<DEFAULT>1000000000</DEFAULT>
+<NICK>Format file to faststart</NICK>
+<BLURB>If the file should be formatted for faststart (headers first).</BLURB>
+<DEFAULT>FALSE</DEFAULT>
 </ARG>
 
 <ARG>
-<NAME>GstPulseAudioSink::drift-tolerance</NAME>
-<TYPE>gint64</TYPE>
-<RANGE>>= 1</RANGE>
-<FLAGS>rw</FLAGS>
-<NICK>Drift Tolerance</NICK>
-<BLURB>Tolerance for clock drift in microseconds.</BLURB>
-<DEFAULT>40000</DEFAULT>
+<NAME>Gst3GPPMux::faststart-file</NAME>
+<TYPE>gchar*</TYPE>
+<RANGE></RANGE>
+<FLAGS>rwx</FLAGS>
+<NICK>File to use for storing buffers</NICK>
+<BLURB>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.</BLURB>
+<DEFAULT>NULL</DEFAULT>
 </ARG>
 
 <ARG>
-<NAME>GstPulseAudioSink::enable-last-buffer</NAME>
-<TYPE>gboolean</TYPE>
+<NAME>Gst3GPPMux::fragment-duration</NAME>
+<TYPE>guint</TYPE>
 <RANGE></RANGE>
-<FLAGS>rw</FLAGS>
-<NICK>Enable Last Buffer</NICK>
-<BLURB>Enable the last-buffer property.</BLURB>
-<DEFAULT>TRUE</DEFAULT>
+<FLAGS>rwx</FLAGS>
+<NICK>Fragment duration</NICK>
+<BLURB>Fragment durations in ms (produce a fragmented file if > 0).</BLURB>
+<DEFAULT>0</DEFAULT>
 </ARG>
 
 <ARG>
-<NAME>GstPulseAudioSink::last-buffer</NAME>
-<TYPE>GstBuffer*</TYPE>
+<NAME>Gst3GPPMux::moov-recovery-file</NAME>
+<TYPE>gchar*</TYPE>
 <RANGE></RANGE>
-<FLAGS>r</FLAGS>
-<NICK>Last Buffer</NICK>
-<BLURB>The last buffer received in the sink.</BLURB>
-<DEFAULT></DEFAULT>
+<FLAGS>rwx</FLAGS>
+<NICK>File to store data for posterior moov atom recovery</NICK>
+<BLURB>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).</BLURB>
+<DEFAULT>NULL</DEFAULT>
 </ARG>
 
 <ARG>
-<NAME>GstPulseAudioSink::latency-time</NAME>
-<TYPE>gint64</TYPE>
-<RANGE>>= 1</RANGE>
-<FLAGS>rw</FLAGS>
-<NICK>Latency Time</NICK>
-<BLURB>Audio latency in microseconds.</BLURB>
-<DEFAULT>10000</DEFAULT>
+<NAME>Gst3GPPMux::movie-timescale</NAME>
+<TYPE>guint</TYPE>
+<RANGE></RANGE>
+<FLAGS>rwx</FLAGS>
+<NICK>Movie timescale</NICK>
+<BLURB>Timescale to use in the movie (units per second, 0 == default).</BLURB>
+<DEFAULT>0</DEFAULT>
 </ARG>
 
 <ARG>
-<NAME>GstPulseAudioSink::max-lateness</NAME>
-<TYPE>gint64</TYPE>
-<RANGE>>= G_MAXULONG</RANGE>
-<FLAGS>rw</FLAGS>
-<NICK>Max Lateness</NICK>
-<BLURB>Maximum number of nanoseconds that a buffer can be late before it is dropped (-1 unlimited).</BLURB>
-<DEFAULT>-1</DEFAULT>
+<NAME>Gst3GPPMux::presentation-time</NAME>
+<TYPE>gboolean</TYPE>
+<RANGE></RANGE>
+<FLAGS>rwx</FLAGS>
+<NICK>Include presentation-time info</NICK>
+<BLURB>Calculate and include presentation/composition time (in addition to decoding time).</BLURB>
+<DEFAULT>TRUE</DEFAULT>
 </ARG>
 
 <ARG>
-<NAME>GstPulseAudioSink::mute</NAME>
+<NAME>Gst3GPPMux::streamable</NAME>
 <TYPE>gboolean</TYPE>
 <RANGE></RANGE>
-<FLAGS>rw</FLAGS>
-<NICK>Mute</NICK>
-<BLURB>Mute state of this stream.</BLURB>
+<FLAGS>rwx</FLAGS>
+<NICK>Streamable</NICK>
+<BLURB>If set to true, the output should be as if it is to be streamed and hence no indexes written or duration written. (DEPRECATED, only valid for fragmented MP4).</BLURB>
 <DEFAULT>FALSE</DEFAULT>
 </ARG>
 
 <ARG>
-<NAME>GstPulseAudioSink::preroll-queue-len</NAME>
+<NAME>Gst3GPPMux::trak-timescale</NAME>
 <TYPE>guint</TYPE>
 <RANGE></RANGE>
 <FLAGS>rwx</FLAGS>
-<NICK>Preroll queue length</NICK>
-<BLURB>Number of buffers to queue during preroll.</BLURB>
+<NICK>Track timescale</NICK>
+<BLURB>Timescale to use for the tracks (units per second, 0 is automatic).</BLURB>
 <DEFAULT>0</DEFAULT>
 </ARG>
 
 <ARG>
-<NAME>GstPulseAudioSink::provide-clock</NAME>
-<TYPE>gboolean</TYPE>
+<NAME>Gst3GPPMux::reserved-bytes-per-sec</NAME>
+<TYPE>guint</TYPE>
+<RANGE><= 10000</RANGE>
+<FLAGS>rwx</FLAGS>
+<NICK>Reserved MOOV bytes per second, per track</NICK>
+<BLURB>Multiplier for converting reserved-max-duration into bytes of header to reserve, per second, per track.</BLURB>
+<DEFAULT>550</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>Gst3GPPMux::reserved-duration-remaining</NAME>
+<TYPE>guint64</TYPE>
 <RANGE></RANGE>
-<FLAGS>rw</FLAGS>
-<NICK>Provide Clock</NICK>
-<BLURB>Provide a clock to be used as the global pipeline clock.</BLURB>
-<DEFAULT>TRUE</DEFAULT>
+<FLAGS>r</FLAGS>
+<NICK>Report the approximate amount of remaining recording space (ns)</NICK>
+<BLURB>Reports the approximate amount of remaining moov header space reserved using reserved-max-duration.</BLURB>
+<DEFAULT>0</DEFAULT>
 </ARG>
 
 <ARG>
-<NAME>GstPulseAudioSink::qos</NAME>
-<TYPE>gboolean</TYPE>
+<NAME>Gst3GPPMux::reserved-max-duration</NAME>
+<TYPE>guint64</TYPE>
 <RANGE></RANGE>
 <FLAGS>rw</FLAGS>
-<NICK>Qos</NICK>
-<BLURB>Generate Quality-of-Service events upstream.</BLURB>
-<DEFAULT>FALSE</DEFAULT>
+<NICK>Reserved maximum file duration (ns)</NICK>
+<BLURB>When set to a value > 0, reserves space for index tables at the beginning of the file.</BLURB>
+<DEFAULT>18446744073709551615</DEFAULT>
 </ARG>
 
 <ARG>
-<NAME>GstPulseAudioSink::render-delay</NAME>
+<NAME>Gst3GPPMux::reserved-moov-update-period</NAME>
 <TYPE>guint64</TYPE>
 <RANGE></RANGE>
 <FLAGS>rw</FLAGS>
-<NICK>Render Delay</NICK>
-<BLURB>Additional render delay of the sink in nanoseconds.</BLURB>
-<DEFAULT>0</DEFAULT>
+<NICK>Interval at which to update index tables (ns)</NICK>
+<BLURB>When used with reserved-max-duration, periodically updates the index tables with information muxed so far.</BLURB>
+<DEFAULT>18446744073709551615</DEFAULT>
 </ARG>
 
 <ARG>
-<NAME>GstPulseAudioSink::server</NAME>
-<TYPE>gchar*</TYPE>
+<NAME>Gst3GPPMux::interleave-bytes</NAME>
+<TYPE>guint64</TYPE>
 <RANGE></RANGE>
 <FLAGS>rw</FLAGS>
-<NICK>Server</NICK>
-<BLURB>The PulseAudio server to connect to.</BLURB>
-<DEFAULT>NULL</DEFAULT>
+<NICK>Interleave (bytes)</NICK>
+<BLURB>Interleave between streams in bytes.</BLURB>
+<DEFAULT>0</DEFAULT>
 </ARG>
 
 <ARG>
-<NAME>GstPulseAudioSink::slave-method</NAME>
-<TYPE>GstBaseAudioSinkSlaveMethod</TYPE>
+<NAME>Gst3GPPMux::interleave-time</NAME>
+<TYPE>guint64</TYPE>
 <RANGE></RANGE>
 <FLAGS>rw</FLAGS>
-<NICK>Slave Method</NICK>
-<BLURB>Algorithm to use to match the rate of the masterclock.</BLURB>
-<DEFAULT>GST_BASE_AUDIO_SINK_SLAVE_SKEW</DEFAULT>
+<NICK>Interleave (time)</NICK>
+<BLURB>Interleave between streams in nanoseconds.</BLURB>
+<DEFAULT>250000000</DEFAULT>
 </ARG>
 
 <ARG>
-<NAME>GstPulseAudioSink::stream-properties</NAME>
-<TYPE>GstStructure*</TYPE>
+<NAME>Gst3GPPMux::max-raw-audio-drift</NAME>
+<TYPE>guint64</TYPE>
 <RANGE></RANGE>
 <FLAGS>rw</FLAGS>
-<NICK>stream properties</NICK>
-<BLURB>list of pulseaudio stream properties.</BLURB>
-<DEFAULT></DEFAULT>
+<NICK>Max Raw Audio Drift</NICK>
+<BLURB>Maximum allowed drift of raw audio samples vs. timestamps in nanoseconds.</BLURB>
+<DEFAULT>40000000</DEFAULT>
 </ARG>
 
 <ARG>
-<NAME>GstPulseAudioSink::sync</NAME>
+<NAME>Gst3GPPMux::reserved-prefill</NAME>
 <TYPE>gboolean</TYPE>
 <RANGE></RANGE>
-<FLAGS>rw</FLAGS>
-<NICK>Sync</NICK>
-<BLURB>Sync on the clock.</BLURB>
-<DEFAULT>TRUE</DEFAULT>
+<FLAGS>rwx</FLAGS>
+<NICK>Reserved Prefill Samples Table</NICK>
+<BLURB>Prefill samples table of reserved duration.</BLURB>
+<DEFAULT>FALSE</DEFAULT>
 </ARG>
 
 <ARG>
-<NAME>GstPulseAudioSink::throttle-time</NAME>
+<NAME>Gst3GPPMux::start-gap-threshold</NAME>
 <TYPE>guint64</TYPE>
 <RANGE></RANGE>
 <FLAGS>rw</FLAGS>
-<NICK>Throttle time</NICK>
-<BLURB>The time to keep between rendered buffers (unused).</BLURB>
+<NICK>Start Gap Threshold</NICK>
+<BLURB>Threshold for creating an edit list for gaps at the start in nanoseconds.</BLURB>
 <DEFAULT>0</DEFAULT>
 </ARG>
 
 <ARG>
-<NAME>GstPulseAudioSink::ts-offset</NAME>
-<TYPE>gint64</TYPE>
+<NAME>GstSplitFileSrc::location</NAME>
+<TYPE>gchar*</TYPE>
 <RANGE></RANGE>
 <FLAGS>rw</FLAGS>
-<NICK>TS Offset</NICK>
-<BLURB>Timestamp offset in nanoseconds.</BLURB>
-<DEFAULT>0</DEFAULT>
+<NICK>File Location</NICK>
+<BLURB>Wildcard pattern to match file names of the input files. If the location is an absolute path or contains directory components, only the base file name part will be considered for pattern matching. The results will be sorted.</BLURB>
+<DEFAULT>NULL</DEFAULT>
 </ARG>
 
 <ARG>
-<NAME>GstPulseAudioSink::volume</NAME>
-<TYPE>gdouble</TYPE>
-<RANGE>[0,10]</RANGE>
+<NAME>GstMatroskaDemux::max-gap-time</NAME>
+<TYPE>guint64</TYPE>
+<RANGE></RANGE>
 <FLAGS>rw</FLAGS>
-<NICK>Volume</NICK>
-<BLURB>Linear volume of this stream, 1.0=100%.</BLURB>
-<DEFAULT>1</DEFAULT>
+<NICK>Maximum gap time</NICK>
+<BLURB>The demuxer sends out segment events for skipping gaps longer than this (0 = disabled).</BLURB>
+<DEFAULT>2000000000</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstMatroskaDemux::max-backtrack-distance</NAME>
+<TYPE>guint</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Maximum backtrack distance</NICK>
+<BLURB>Maximum backtrack distance in seconds when seeking without and index in pull mode and search for a keyframe (0 = disable backtracking).</BLURB>
+<DEFAULT>30</DEFAULT>
 </ARG>
 
 <ARG>
@@ -23239,6 +22819,26 @@
 </ARG>
 
 <ARG>
+<NAME>GstSoupHttpClientSink::retries</NAME>
+<TYPE>gint</TYPE>
+<RANGE>>= G_MAXULONG</RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Retries</NICK>
+<BLURB>Maximum number of retries, zero to disable, -1 to retry forever.</BLURB>
+<DEFAULT>0</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstSoupHttpClientSink::retry-delay</NAME>
+<TYPE>gint</TYPE>
+<RANGE>>= 1</RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Retry Delay</NICK>
+<BLURB>Delay in seconds between retries after a failure.</BLURB>
+<DEFAULT>5</DEFAULT>
+</ARG>
+
+<ARG>
 <NAME>GstWavParse::ignore-length</NAME>
 <TYPE>gboolean</TYPE>
 <RANGE></RANGE>
@@ -23301,20 +22901,20 @@
 <ARG>
 <NAME>GstGdkPixbufOverlay::relative-x</NAME>
 <TYPE>gdouble</TYPE>
-<RANGE>[0,1]</RANGE>
+<RANGE>[-1,1]</RANGE>
 <FLAGS>rw</FLAGS>
 <NICK>Relative X Offset</NICK>
-<BLURB>Horizontal offset of overlay image in fractions of video image width, from top-left corner of video image.</BLURB>
+<BLURB>Horizontal offset of overlay image in fractions of video image width, from top-left corner of video image (in relative positioning).</BLURB>
 <DEFAULT>0</DEFAULT>
 </ARG>
 
 <ARG>
 <NAME>GstGdkPixbufOverlay::relative-y</NAME>
 <TYPE>gdouble</TYPE>
-<RANGE>[0,1]</RANGE>
+<RANGE>[-1,1]</RANGE>
 <FLAGS>rw</FLAGS>
 <NICK>Relative Y Offset</NICK>
-<BLURB>Vertical offset of overlay image in fractions of video image height, from top-left corner of video image.</BLURB>
+<BLURB>Vertical offset of overlay image in fractions of video image height, from top-left corner of video image (in relative positioning).</BLURB>
 <DEFAULT>0</DEFAULT>
 </ARG>
 
@@ -23329,6 +22929,46 @@
 </ARG>
 
 <ARG>
+<NAME>GstGdkPixbufOverlay::pixbuf</NAME>
+<TYPE>GdkPixbuf*</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Pixbuf</NICK>
+<BLURB>GdkPixbuf object to render.</BLURB>
+<DEFAULT></DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstGdkPixbufOverlay::positioning-mode</NAME>
+<TYPE>GstGdkPixbufPositioningMode</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Positioning mode</NICK>
+<BLURB>Positioning mode of offset-x and offset-y properties.</BLURB>
+<DEFAULT>pixels-relative-to-edges</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstGdkPixbufOverlay::coef-x</NAME>
+<TYPE>gdouble</TYPE>
+<RANGE>[-1,1]</RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Relative X Offset</NICK>
+<BLURB>Horizontal offset of overlay image in fractions of video image width, from top-left corner of video image (absolute positioning).</BLURB>
+<DEFAULT>0</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstGdkPixbufOverlay::coef-y</NAME>
+<TYPE>gdouble</TYPE>
+<RANGE>[-1,1]</RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Relative Y Offset</NICK>
+<BLURB>Vertical offset of overlay image in fractions of video image height, from top-left corner of video image (absolute positioning).</BLURB>
+<DEFAULT>0</DEFAULT>
+</ARG>
+
+<ARG>
 <NAME>GstVP8Enc::arnr-maxframes</NAME>
 <TYPE>gint</TYPE>
 <RANGE>[0,15]</RANGE>
@@ -23459,73 +23099,33 @@
 </ARG>
 
 <ARG>
-<NAME>GstVP8Enc::h-scaling-mode</NAME>
-<TYPE>GstVP8EncScalingMode</TYPE>
-<RANGE></RANGE>
+<NAME>GstVP8Enc::lag-in-frames</NAME>
+<TYPE>gint</TYPE>
+<RANGE>[0,25]</RANGE>
 <FLAGS>rw</FLAGS>
-<NICK>Horizontal scaling mode</NICK>
-<BLURB>Horizontal scaling mode.</BLURB>
-<DEFAULT>Normal</DEFAULT>
+<NICK>Lag in frames</NICK>
+<BLURB>Maximum number of frames to lag.</BLURB>
+<DEFAULT>0</DEFAULT>
 </ARG>
 
 <ARG>
-<NAME>GstVP8Enc::kf-max-dist</NAME>
+<NAME>GstVP8Enc::max-quantizer</NAME>
 <TYPE>gint</TYPE>
-<RANGE>>= 0</RANGE>
+<RANGE>[0,63]</RANGE>
 <FLAGS>rw</FLAGS>
-<NICK>Keyframe max distance</NICK>
-<BLURB>Maximum distance between keyframes (number of frames).</BLURB>
-<DEFAULT>128</DEFAULT>
+<NICK>Maximum Quantizer</NICK>
+<BLURB>Maximum Quantizer (worst).</BLURB>
+<DEFAULT>63</DEFAULT>
 </ARG>
 
 <ARG>
-<NAME>GstVP8Enc::kf-mode</NAME>
-<TYPE>GstVP8EncKfMode</TYPE>
-<RANGE></RANGE>
+<NAME>GstVP8Enc::min-quantizer</NAME>
+<TYPE>gint</TYPE>
+<RANGE>[0,63]</RANGE>
 <FLAGS>rw</FLAGS>
-<NICK>Keyframe Mode</NICK>
-<BLURB>Keyframe placement.</BLURB>
-<DEFAULT>Determine optimal placement automatically</DEFAULT>
-</ARG>
-
-<ARG>
-<NAME>GstVP8Enc::lag-in-frames</NAME>
-<TYPE>gint</TYPE>
-<RANGE>[0,25]</RANGE>
-<FLAGS>rw</FLAGS>
-<NICK>Lag in frames</NICK>
-<BLURB>Maximum number of frames to lag.</BLURB>
-<DEFAULT>0</DEFAULT>
-</ARG>
-
-<ARG>
-<NAME>GstVP8Enc::max-intra-bitrate-pct</NAME>
-<TYPE>gint</TYPE>
-<RANGE>>= 0</RANGE>
-<FLAGS>rw</FLAGS>
-<NICK>Max Intra bitrate</NICK>
-<BLURB>Maximum Intra frame bitrate.</BLURB>
-<DEFAULT>0</DEFAULT>
-</ARG>
-
-<ARG>
-<NAME>GstVP8Enc::max-quantizer</NAME>
-<TYPE>gint</TYPE>
-<RANGE>[0,63]</RANGE>
-<FLAGS>rw</FLAGS>
-<NICK>Maximum Quantizer</NICK>
-<BLURB>Maximum Quantizer (worst).</BLURB>
-<DEFAULT>63</DEFAULT>
-</ARG>
-
-<ARG>
-<NAME>GstVP8Enc::min-quantizer</NAME>
-<TYPE>gint</TYPE>
-<RANGE>[0,63]</RANGE>
-<FLAGS>rw</FLAGS>
-<NICK>Minimum Quantizer</NICK>
-<BLURB>Minimum Quantizer (best).</BLURB>
-<DEFAULT>4</DEFAULT>
+<NICK>Minimum Quantizer</NICK>
+<BLURB>Minimum Quantizer (best).</BLURB>
+<DEFAULT>4</DEFAULT>
 </ARG>
 
 <ARG>
@@ -23534,7 +23134,7 @@
 <RANGE></RANGE>
 <FLAGS>rw</FLAGS>
 <NICK>Multipass Cache File</NICK>
-<BLURB>Multipass cache file.</BLURB>
+<BLURB>Multipass cache file. If stream caps reinited, multiple files will be created: file, file.1, file.2, ... and so on.</BLURB>
 <DEFAULT>"multipass.cache"</DEFAULT>
 </ARG>
 
@@ -23559,16 +23159,6 @@
 </ARG>
 
 <ARG>
-<NAME>GstVP8Enc::overshoot-pct</NAME>
-<TYPE>gint</TYPE>
-<RANGE>[0,1000]</RANGE>
-<FLAGS>rw</FLAGS>
-<NICK>Overshoot PCT</NICK>
-<BLURB>Datarate overshoot (max) target (%).</BLURB>
-<DEFAULT>100</DEFAULT>
-</ARG>
-
-<ARG>
 <NAME>GstVP8Enc::resize-allowed</NAME>
 <TYPE>gboolean</TYPE>
 <RANGE></RANGE>
@@ -23649,56 +23239,6 @@
 </ARG>
 
 <ARG>
-<NAME>GstVP8Enc::ts-layer-id</NAME>
-<TYPE>GValueArray*</TYPE>
-<RANGE></RANGE>
-<FLAGS>rw</FLAGS>
-<NICK>Coding layer identification</NICK>
-<BLURB>Sequence defining coding layer membership.</BLURB>
-<DEFAULT></DEFAULT>
-</ARG>
-
-<ARG>
-<NAME>GstVP8Enc::ts-number-layers</NAME>
-<TYPE>gint</TYPE>
-<RANGE>[1,5]</RANGE>
-<FLAGS>rw</FLAGS>
-<NICK>Number of coding layers</NICK>
-<BLURB>Number of coding layers to use.</BLURB>
-<DEFAULT>1</DEFAULT>
-</ARG>
-
-<ARG>
-<NAME>GstVP8Enc::ts-periodicity</NAME>
-<TYPE>gint</TYPE>
-<RANGE>[0,16]</RANGE>
-<FLAGS>rw</FLAGS>
-<NICK>Layer periodicity</NICK>
-<BLURB>Length of sequence that defines layer membership periodicity.</BLURB>
-<DEFAULT>0</DEFAULT>
-</ARG>
-
-<ARG>
-<NAME>GstVP8Enc::ts-rate-decimator</NAME>
-<TYPE>GValueArray*</TYPE>
-<RANGE></RANGE>
-<FLAGS>rw</FLAGS>
-<NICK>Coding layer rate decimator</NICK>
-<BLURB>Rate decimation factors for each layer.</BLURB>
-<DEFAULT></DEFAULT>
-</ARG>
-
-<ARG>
-<NAME>GstVP8Enc::ts-target-bitrate</NAME>
-<TYPE>GValueArray*</TYPE>
-<RANGE></RANGE>
-<FLAGS>rw</FLAGS>
-<NICK>Coding layer target bitrates</NICK>
-<BLURB>Target bitrates for coding layers (one per layer, decreasing).</BLURB>
-<DEFAULT></DEFAULT>
-</ARG>
-
-<ARG>
 <NAME>GstVP8Enc::tuning</NAME>
 <TYPE>GstVP8EncTuning</TYPE>
 <RANGE></RANGE>
@@ -23709,56 +23249,6 @@
 </ARG>
 
 <ARG>
-<NAME>GstVP8Enc::twopass-vbr-bias-pct</NAME>
-<TYPE>gint</TYPE>
-<RANGE>[0,100]</RANGE>
-<FLAGS>rw</FLAGS>
-<NICK>2-pass VBR bias</NICK>
-<BLURB>CBR/VBR bias (0=CBR, 100=VBR).</BLURB>
-<DEFAULT>50</DEFAULT>
-</ARG>
-
-<ARG>
-<NAME>GstVP8Enc::twopass-vbr-maxsection-pct</NAME>
-<TYPE>gint</TYPE>
-<RANGE>>= 0</RANGE>
-<FLAGS>rw</FLAGS>
-<NICK>2-pass GOP max bitrate</NICK>
-<BLURB>GOP maximum bitrate (% target).</BLURB>
-<DEFAULT>0</DEFAULT>
-</ARG>
-
-<ARG>
-<NAME>GstVP8Enc::twopass-vbr-minsection-pct</NAME>
-<TYPE>gint</TYPE>
-<RANGE>>= 0</RANGE>
-<FLAGS>rw</FLAGS>
-<NICK>2-pass GOP min bitrate</NICK>
-<BLURB>GOP minimum bitrate (% target).</BLURB>
-<DEFAULT>0</DEFAULT>
-</ARG>
-
-<ARG>
-<NAME>GstVP8Enc::undershoot-pct</NAME>
-<TYPE>gint</TYPE>
-<RANGE>[0,1000]</RANGE>
-<FLAGS>rw</FLAGS>
-<NICK>Undershoot PCT</NICK>
-<BLURB>Datarate undershoot (min) target (%).</BLURB>
-<DEFAULT>100</DEFAULT>
-</ARG>
-
-<ARG>
-<NAME>GstVP8Enc::v-scaling-mode</NAME>
-<TYPE>GstVP8EncScalingMode</TYPE>
-<RANGE></RANGE>
-<FLAGS>rw</FLAGS>
-<NICK>Vertical scaling mode</NICK>
-<BLURB>Vertical scaling mode.</BLURB>
-<DEFAULT>Normal</DEFAULT>
-</ARG>
-
-<ARG>
 <NAME>GstVP8Enc::horizontal-scaling-mode</NAME>
 <TYPE>GstVP8EncScalingMode</TYPE>
 <RANGE></RANGE>
@@ -24054,7 +23544,7 @@
 <RANGE></RANGE>
 <FLAGS>rw</FLAGS>
 <NICK>SSRC</NICK>
-<BLURB>The SSRC of the packets (-1 == random).</BLURB>
+<BLURB>The SSRC of the packets (default == random).</BLURB>
 <DEFAULT>4294967295</DEFAULT>
 </ARG>
 
@@ -24219,6 +23709,26 @@
 </ARG>
 
 <ARG>
+<NAME>GstRTPRtxQueue::fulfilled-requests</NAME>
+<TYPE>guint</TYPE>
+<RANGE></RANGE>
+<FLAGS>r</FLAGS>
+<NICK>Fulfilled Requests</NICK>
+<BLURB>Number of fulfilled retransmission requests.</BLURB>
+<DEFAULT>0</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstRTPRtxQueue::requests</NAME>
+<TYPE>guint</TYPE>
+<RANGE></RANGE>
+<FLAGS>r</FLAGS>
+<NICK>Requests</NICK>
+<BLURB>Total number of retransmission requests.</BLURB>
+<DEFAULT>0</DEFAULT>
+</ARG>
+
+<ARG>
 <NAME>GstRtpGSTPay::config-interval</NAME>
 <TYPE>guint</TYPE>
 <RANGE><= 3600</RANGE>
@@ -24444,7 +23954,7 @@
 <RANGE></RANGE>
 <FLAGS>rw</FLAGS>
 <NICK>Multipass Cache File</NICK>
-<BLURB>Multipass cache file.</BLURB>
+<BLURB>Multipass cache file. If stream caps reinited, multiple files will be created: file, file.1, file.2, ... and so on.</BLURB>
 <DEFAULT>"multipass.cache"</DEFAULT>
 </ARG>
 
@@ -24728,3 +24238,813 @@
 <DEFAULT>1</DEFAULT>
 </ARG>
 
+<ARG>
+<NAME>GstRtpVRawPay::chunks-per-frame</NAME>
+<TYPE>gint</TYPE>
+<RANGE>>= 1</RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Chunks per Frame</NICK>
+<BLURB>Split and send out each frame in multiple chunks to reduce overhead.</BLURB>
+<DEFAULT>10</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstPushFileSrc::applied-rate</NAME>
+<TYPE>gdouble</TYPE>
+<RANGE>>= G_MINDOUBLE</RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Applied Rate</NICK>
+<BLURB>Applied rate to use in TIME SEGMENT.</BLURB>
+<DEFAULT>1</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstPushFileSrc::initial-timestamp</NAME>
+<TYPE>guint64</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Initial Timestamp</NICK>
+<BLURB>Initial Buffer Timestamp (if time-segment TRUE).</BLURB>
+<DEFAULT>18446744073709551615</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstPushFileSrc::location</NAME>
+<TYPE>gchar*</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>File Location</NICK>
+<BLURB>Location of the file to read.</BLURB>
+<DEFAULT>NULL</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstPushFileSrc::rate</NAME>
+<TYPE>gdouble</TYPE>
+<RANGE>>= G_MINDOUBLE</RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Rate</NICK>
+<BLURB>Rate to use in TIME SEGMENT.</BLURB>
+<DEFAULT>1</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstPushFileSrc::start-time</NAME>
+<TYPE>gint64</TYPE>
+<RANGE>>= 0</RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Start Time</NICK>
+<BLURB>Initial Start Time (if time-segment TRUE).</BLURB>
+<DEFAULT>0</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstPushFileSrc::stream-time</NAME>
+<TYPE>gint64</TYPE>
+<RANGE>>= 0</RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Stream Time</NICK>
+<BLURB>Initial Stream Time (if time-segment TRUE).</BLURB>
+<DEFAULT>0</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstPushFileSrc::time-segment</NAME>
+<TYPE>gboolean</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Time Segment</NICK>
+<BLURB>Emit TIME SEGMENTS.</BLURB>
+<DEFAULT>FALSE</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstSplitMuxSrc::location</NAME>
+<TYPE>gchar*</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>File Input Pattern</NICK>
+<BLURB>Glob pattern for the location of the files to read.</BLURB>
+<DEFAULT>NULL</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstSplitMuxSink::location</NAME>
+<TYPE>gchar*</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>File Output Pattern</NICK>
+<BLURB>Format string pattern for the location of the files to write (e.g. video%05d.mp4).</BLURB>
+<DEFAULT>NULL</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstSplitMuxSink::max-size-bytes</NAME>
+<TYPE>guint64</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Max. size bytes</NICK>
+<BLURB>Max. amount of data per file (in bytes, 0=disable).</BLURB>
+<DEFAULT>0</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstSplitMuxSink::max-size-time</NAME>
+<TYPE>guint64</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Max. size (ns)</NICK>
+<BLURB>Max. amount of time per file (in ns, 0=disable).</BLURB>
+<DEFAULT>0</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstSplitMuxSink::mux-overhead</NAME>
+<TYPE>gdouble</TYPE>
+<RANGE>[0,1]</RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Muxing Overhead</NICK>
+<BLURB>Extra size overhead of muxing (0.02 = 2%).</BLURB>
+<DEFAULT>0.02</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstSplitMuxSink::muxer</NAME>
+<TYPE>GstElement*</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Muxer</NICK>
+<BLURB>The muxer element to use (NULL = default mp4mux). Valid only for async-finalize = FALSE.</BLURB>
+<DEFAULT></DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstSplitMuxSink::sink</NAME>
+<TYPE>GstElement*</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Sink</NICK>
+<BLURB>The sink element (or element chain) to use (NULL = default filesink). Valid only for async-finalize = FALSE.</BLURB>
+<DEFAULT></DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstSplitMuxSink::max-files</NAME>
+<TYPE>guint</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Max files</NICK>
+<BLURB>Maximum number of files to keep on disk. Once the maximum is reached,old files start to be deleted to make room for new ones.</BLURB>
+<DEFAULT>0</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstSplitMuxSink::send-keyframe-requests</NAME>
+<TYPE>gboolean</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Request keyframes at max-size-time</NICK>
+<BLURB>Request a keyframe every max-size-time ns to try splitting at that point. Needs max-size-bytes to be 0 in order to be effective.</BLURB>
+<DEFAULT>FALSE</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstSplitMuxSink::max-size-timecode</NAME>
+<TYPE>gchar*</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Maximum timecode difference</NICK>
+<BLURB>Maximum difference in timecode between first and last frame. Separator is assumed to be ":" everywhere (e.g. 01:00:00:00). Will only be effective if a timecode track is present.</BLURB>
+<DEFAULT>NULL</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstSplitMuxSink::alignment-threshold</NAME>
+<TYPE>guint64</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Alignment threshold (ns)</NICK>
+<BLURB>Allow non-reference streams to be that many ns before the reference stream.</BLURB>
+<DEFAULT>0</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstSplitMuxSink::use-robust-muxing</NAME>
+<TYPE>gboolean</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Support robust-muxing mode of some muxers</NICK>
+<BLURB>Check if muxers support robust muxing via the reserved-max-duration and reserved-duration-remaining properties and use them if so. (Only present on qtmux and mp4mux for now). splitmuxsink may then also  create new fragments if the reserved header space is about to overflow. Note that for mp4mux and qtmux, reserved-moov-update-period must be set manually by the app to a non-zero value for robust muxing to have an effect.</BLURB>
+<DEFAULT>FALSE</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstSplitMuxSink::async-finalize</NAME>
+<TYPE>gboolean</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Finalize fragments asynchronously</NICK>
+<BLURB>Finalize each fragment asynchronously and start a new one.</BLURB>
+<DEFAULT>FALSE</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstSplitMuxSink::muxer-factory</NAME>
+<TYPE>gchar*</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Muxer factory</NICK>
+<BLURB>The muxer element factory to use (default = mp4mux). Valid only for async-finalize = TRUE.</BLURB>
+<DEFAULT>"mp4mux"</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstSplitMuxSink::muxer-properties</NAME>
+<TYPE>GstStructure*</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Muxer properties</NICK>
+<BLURB>The muxer element properties to use. Example: {properties,boolean-prop=true,string-prop="hi"}. Valid only for async-finalize = TRUE.</BLURB>
+<DEFAULT></DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstSplitMuxSink::reset-muxer</NAME>
+<TYPE>gboolean</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Reset Muxer</NICK>
+<BLURB>Reset the muxer after each segment. Disabling this will not work for most muxers.</BLURB>
+<DEFAULT>TRUE</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstSplitMuxSink::sink-factory</NAME>
+<TYPE>gchar*</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Sink factory</NICK>
+<BLURB>The sink element factory to use (default = filesink). Valid only for async-finalize = TRUE.</BLURB>
+<DEFAULT>"filesink"</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstSplitMuxSink::sink-properties</NAME>
+<TYPE>GstStructure*</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Sink properties</NICK>
+<BLURB>The sink element properties to use. Example: {properties,boolean-prop=true,string-prop="hi"}. Valid only for async-finalize = TRUE.</BLURB>
+<DEFAULT></DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstRtpVP9Pay::picture-id-mode</NAME>
+<TYPE>GstVP9RTPPayMode</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Picture ID Mode</NICK>
+<BLURB>The picture ID mode for payloading.</BLURB>
+<DEFAULT>No Picture ID</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstRtpH265Pay::config-interval</NAME>
+<TYPE>gint</TYPE>
+<RANGE>[G_MAXULONG,3600]</RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>VPS SPS PPS Send Interval</NICK>
+<BLURB>Send VPS, SPS and PPS Insertion Interval in seconds (sprop parameter sets will be multiplexed in the data stream when detected.) (0 = disabled, -1 = send with every IDR frame).</BLURB>
+<DEFAULT>0</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstRtpH265Pay::sprop-parameter-sets</NAME>
+<TYPE>gchar*</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>sprop-parameter-sets</NICK>
+<BLURB>The base64 sprop-parameter-sets to set in out caps (set to NULL to extract from stream).</BLURB>
+<DEFAULT>NULL</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstLameMP3Enc::bitrate</NAME>
+<TYPE>gint</TYPE>
+<RANGE>[8,320]</RANGE>
+<FLAGS>rwx</FLAGS>
+<NICK>Bitrate (kb/s)</NICK>
+<BLURB>Bitrate in kbit/sec (Only valid if target is bitrate, for CBR one of 8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256 or 320).</BLURB>
+<DEFAULT>128</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstLameMP3Enc::cbr</NAME>
+<TYPE>gboolean</TYPE>
+<RANGE></RANGE>
+<FLAGS>rwx</FLAGS>
+<NICK>CBR</NICK>
+<BLURB>Enforce constant bitrate encoding (Only valid if target is bitrate).</BLURB>
+<DEFAULT>FALSE</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstLameMP3Enc::encoding-engine-quality</NAME>
+<TYPE>GstLameMP3EncEncodingEngineQuality</TYPE>
+<RANGE></RANGE>
+<FLAGS>rwx</FLAGS>
+<NICK>Encoding Engine Quality</NICK>
+<BLURB>Quality/speed of the encoding engine, this does not affect the bitrate!.</BLURB>
+<DEFAULT>Standard</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstLameMP3Enc::mono</NAME>
+<TYPE>gboolean</TYPE>
+<RANGE></RANGE>
+<FLAGS>rwx</FLAGS>
+<NICK>Mono</NICK>
+<BLURB>Enforce mono encoding.</BLURB>
+<DEFAULT>FALSE</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstLameMP3Enc::quality</NAME>
+<TYPE>gfloat</TYPE>
+<RANGE>[0,9.999]</RANGE>
+<FLAGS>rwx</FLAGS>
+<NICK>Quality</NICK>
+<BLURB>VBR Quality from 0 to 10, 0 being the best (Only valid if target is quality).</BLURB>
+<DEFAULT>4</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstLameMP3Enc::target</NAME>
+<TYPE>GstLameMP3EncTarget</TYPE>
+<RANGE></RANGE>
+<FLAGS>rwx</FLAGS>
+<NICK>Target</NICK>
+<BLURB>Optimize for quality or bitrate.</BLURB>
+<DEFAULT>Quality</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstTwoLame::ath-level</NAME>
+<TYPE>gfloat</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>ATH Level</NICK>
+<BLURB>ATH Level in dB.</BLURB>
+<DEFAULT>0</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstTwoLame::bitrate</NAME>
+<TYPE>gint</TYPE>
+<RANGE>[8,384]</RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Bitrate (kb/s)</NICK>
+<BLURB>Bitrate in kbit/sec (8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 144, 160, 192, 224, 256, 320, 384).</BLURB>
+<DEFAULT>192</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstTwoLame::copyright</NAME>
+<TYPE>gboolean</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Copyright</NICK>
+<BLURB>Mark as copyright.</BLURB>
+<DEFAULT>FALSE</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstTwoLame::emphasis</NAME>
+<TYPE>GstTwoLameEmphasis</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Emphasis</NICK>
+<BLURB>Pre-emphasis to apply to the decoded audio.</BLURB>
+<DEFAULT>No emphasis</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstTwoLame::energy-level-extension</NAME>
+<TYPE>gboolean</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Energy Level Extension</NICK>
+<BLURB>Write peak PCM level to each frame.</BLURB>
+<DEFAULT>FALSE</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstTwoLame::error-protection</NAME>
+<TYPE>gboolean</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Error protection</NICK>
+<BLURB>Adds checksum to every frame.</BLURB>
+<DEFAULT>FALSE</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstTwoLame::mode</NAME>
+<TYPE>GstTwoLameMode</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Mode</NICK>
+<BLURB>Encoding mode.</BLURB>
+<DEFAULT>Joint Stereo</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstTwoLame::original</NAME>
+<TYPE>gboolean</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Original</NICK>
+<BLURB>Mark as original.</BLURB>
+<DEFAULT>TRUE</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstTwoLame::padding</NAME>
+<TYPE>GstTwoLamePadding</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Padding</NICK>
+<BLURB>Padding type.</BLURB>
+<DEFAULT>No Padding</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstTwoLame::psymodel</NAME>
+<TYPE>gint</TYPE>
+<RANGE>[G_MAXULONG,4]</RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Psychoacoustic Model</NICK>
+<BLURB>Psychoacoustic model used to encode the audio.</BLURB>
+<DEFAULT>3</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstTwoLame::quick-mode</NAME>
+<TYPE>gboolean</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Quick mode</NICK>
+<BLURB>Calculate Psymodel every frames.</BLURB>
+<DEFAULT>FALSE</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstTwoLame::quick-mode-count</NAME>
+<TYPE>gint</TYPE>
+<RANGE>>= 0</RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Quick mode count</NICK>
+<BLURB>Calculate Psymodel every n frames.</BLURB>
+<DEFAULT>10</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstTwoLame::vbr</NAME>
+<TYPE>gboolean</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>VBR</NICK>
+<BLURB>Enable variable bitrate mode.</BLURB>
+<DEFAULT>FALSE</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstTwoLame::vbr-level</NAME>
+<TYPE>gfloat</TYPE>
+<RANGE>[-10,10]</RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>VBR Level</NICK>
+<BLURB>VBR Level.</BLURB>
+<DEFAULT>5</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstTwoLame::vbr-max-bitrate</NAME>
+<TYPE>gint</TYPE>
+<RANGE>[0,384]</RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>VBR max bitrate</NICK>
+<BLURB>Specify maximum VBR bitrate (0=off, 8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 144, 160, 192, 224, 256, 320, 384).</BLURB>
+<DEFAULT>0</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstRtpSbcDepay::ignore-timestamps</NAME>
+<TYPE>gboolean</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Ignore Timestamps</NICK>
+<BLURB>Various statistics.</BLURB>
+<DEFAULT>FALSE</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstQtSrc::use-default-fbo</NAME>
+<TYPE>gboolean</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Whether to use default FBO</NICK>
+<BLURB>When set it will not create a new FBO for the QML render thread.</BLURB>
+<DEFAULT>FALSE</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstQtSrc::window</NAME>
+<TYPE>gpointer</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>QQuickWindow</NICK>
+<BLURB>The QQuickWindow to place in the object hierarchy.</BLURB>
+<DEFAULT></DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstQtSink::force-aspect-ratio</NAME>
+<TYPE>gboolean</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Force aspect ratio</NICK>
+<BLURB>When enabled, scaling will respect original aspect ratio.</BLURB>
+<DEFAULT>TRUE</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstQtSink::pixel-aspect-ratio</NAME>
+<TYPE>GstFraction</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Pixel Aspect Ratio</NICK>
+<BLURB>The pixel aspect ratio of the device.</BLURB>
+<DEFAULT></DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstQtSink::widget</NAME>
+<TYPE>gpointer</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>QQuickItem</NICK>
+<BLURB>The QQuickItem to place in the object hierarchy.</BLURB>
+<DEFAULT></DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstRtpPtDemux::ignored-payload-types</NAME>
+<TYPE>GstValueArray</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Ignored payload types</NICK>
+<BLURB>Packets with these payload types will be dropped.</BLURB>
+<DEFAULT></DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstRtpUlpFecEnc::multipacket</NAME>
+<TYPE>gboolean</TYPE>
+<RANGE></RANGE>
+<FLAGS>rwx</FLAGS>
+<NICK>Multipacket</NICK>
+<BLURB>Apply FEC on multiple packets.</BLURB>
+<DEFAULT>TRUE</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstRtpUlpFecEnc::mux-seq</NAME>
+<TYPE>gboolean</TYPE>
+<RANGE></RANGE>
+<FLAGS>rwx</FLAGS>
+<NICK>Mux seq</NICK>
+<BLURB>Mux seqnum for media and fec packets in same seqnum space.</BLURB>
+<DEFAULT>FALSE</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstRtpUlpFecEnc::percentage</NAME>
+<TYPE>guint</TYPE>
+<RANGE><= 100</RANGE>
+<FLAGS>rwx</FLAGS>
+<NICK>Percentage</NICK>
+<BLURB>FEC overhead percentage for the whole stream.</BLURB>
+<DEFAULT>0</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstRtpUlpFecEnc::percentage-important</NAME>
+<TYPE>guint</TYPE>
+<RANGE><= 100</RANGE>
+<FLAGS>rwx</FLAGS>
+<NICK>Percentage important</NICK>
+<BLURB>FEC overhead percentage for important packets.</BLURB>
+<DEFAULT>0</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstRtpUlpFecEnc::protected</NAME>
+<TYPE>guint</TYPE>
+<RANGE></RANGE>
+<FLAGS>r</FLAGS>
+<NICK>Protected</NICK>
+<BLURB>Count of protected packets.</BLURB>
+<DEFAULT>0</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstRtpUlpFecEnc::pt</NAME>
+<TYPE>guint</TYPE>
+<RANGE><= 255</RANGE>
+<FLAGS>rwx</FLAGS>
+<NICK>payload type</NICK>
+<BLURB>The payload type of FEC packets.</BLURB>
+<DEFAULT>255</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstRtpUlpFecEnc::ssrc</NAME>
+<TYPE>guint</TYPE>
+<RANGE></RANGE>
+<FLAGS>rwx</FLAGS>
+<NICK>SSRC</NICK>
+<BLURB>The SSRC to use on FEC'd packets.</BLURB>
+<DEFAULT>0</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstRtpUlpFecDec::pt</NAME>
+<TYPE>guint</TYPE>
+<RANGE><= 127</RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>pt</NICK>
+<BLURB>FEC packets payload type.</BLURB>
+<DEFAULT>0</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstRtpUlpFecDec::recovered</NAME>
+<TYPE>guint</TYPE>
+<RANGE></RANGE>
+<FLAGS>r</FLAGS>
+<NICK>recovered</NICK>
+<BLURB>The number of recovered packets.</BLURB>
+<DEFAULT>0</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstRtpUlpFecDec::storage</NAME>
+<TYPE>GObject*</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>RTP storage</NICK>
+<BLURB>RTP storage.</BLURB>
+<DEFAULT></DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstRtpUlpFecDec::unrecovered</NAME>
+<TYPE>guint</TYPE>
+<RANGE></RANGE>
+<FLAGS>r</FLAGS>
+<NICK>unrecovered</NICK>
+<BLURB>The number of unrecovered packets.</BLURB>
+<DEFAULT>0</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstRtpStorage::internal-storage</NAME>
+<TYPE>GObject*</TYPE>
+<RANGE></RANGE>
+<FLAGS>r</FLAGS>
+<NICK>Internal storage</NICK>
+<BLURB>Internal RtpStorage object.</BLURB>
+<DEFAULT></DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstRtpStorage::size-time</NAME>
+<TYPE>guint64</TYPE>
+<RANGE></RANGE>
+<FLAGS>rwx</FLAGS>
+<NICK>Storage size (in ns)</NICK>
+<BLURB>The amount of data to keep in the storage (in ns, 0-disable).</BLURB>
+<DEFAULT>0</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstRtpRedEnc::allow-no-red-blocks</NAME>
+<TYPE>gboolean</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Allow no redundant blocks</NICK>
+<BLURB>true - can produce RED packets even without redundant blocks (distance==0) false - RED packets will be produced only if distance>0.</BLURB>
+<DEFAULT>TRUE</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstRtpRedEnc::distance</NAME>
+<TYPE>guint</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>RED distance</NICK>
+<BLURB>Tells which media packet to use as a redundant block (0 - no redundant blocks, 1 to use previous packet, 2 to use the packet before previous, etc.).</BLURB>
+<DEFAULT>0</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstRtpRedEnc::pt</NAME>
+<TYPE>gint</TYPE>
+<RANGE>[0,127]</RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>payload type</NICK>
+<BLURB>Payload type FEC packets (-1 disable).</BLURB>
+<DEFAULT>0</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstRtpRedEnc::sent</NAME>
+<TYPE>guint</TYPE>
+<RANGE></RANGE>
+<FLAGS>r</FLAGS>
+<NICK>Sent</NICK>
+<BLURB>Count of sent packets.</BLURB>
+<DEFAULT>0</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstRtpRedDec::pt</NAME>
+<TYPE>gint</TYPE>
+<RANGE>[G_MAXULONG,127]</RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>payload type</NICK>
+<BLURB>Payload type FEC packets.</BLURB>
+<DEFAULT>-1</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstRtpRedDec::received</NAME>
+<TYPE>guint</TYPE>
+<RANGE></RANGE>
+<FLAGS>r</FLAGS>
+<NICK>Received</NICK>
+<BLURB>Count of received packets.</BLURB>
+<DEFAULT>0</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstStereo::active</NAME>
+<TYPE>gboolean</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>active</NICK>
+<BLURB>active.</BLURB>
+<DEFAULT>TRUE</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstStereo::stereo</NAME>
+<TYPE>gfloat</TYPE>
+<RANGE>[0,1]</RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>stereo</NICK>
+<BLURB>stereo.</BLURB>
+<DEFAULT>0.1</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstRtpFunnel::common-ts-offset</NAME>
+<TYPE>gint</TYPE>
+<RANGE>>= G_MAXULONG</RANGE>
+<FLAGS>rwx</FLAGS>
+<NICK>Common Timestamp Offset</NICK>
+<BLURB>Use the same RTP timestamp offset for all sinkpads (-1 = disable).</BLURB>
+<DEFAULT>-1</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstCairoOverlay::draw-on-transparent-surface</NAME>
+<TYPE>gboolean</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Draw on transparent surface</NICK>
+<BLURB>Let the draw signal work on a transparent surface and blend the results with the video at a later time.</BLURB>
+<DEFAULT>FALSE</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstImageFreeze::num-buffers</NAME>
+<TYPE>gint</TYPE>
+<RANGE>>= G_MAXULONG</RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>num-buffers</NICK>
+<BLURB>Number of buffers to output before sending EOS (-1 = unlimited).</BLURB>
+<DEFAULT>-1</DEFAULT>
+</ARG>
+