<RANGE></RANGE>
<FLAGS>rw</FLAGS>
<NICK>Maximum File Duration</NICK>
-<BLURB>Maximum file duration before starting a new file in max-size mode.</BLURB>
+<BLURB>Maximum file duration before starting a new file in max-size mode (in nanoseconds).</BLURB>
<DEFAULT>18446744073709551615</DEFAULT>
</ARG>
<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>
</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>
<NAME>GstSMPTE::border</NAME>
<TYPE>gint</TYPE>
<RANGE>>= 0</RANGE>
<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>
<FLAGS>rw</FLAGS>
<NICK>User Agent</NICK>
<BLURB>The User-Agent string to send to the server.</BLURB>
-<DEFAULT>"GStreamer/1.7.1"</DEFAULT>
+<DEFAULT>"GStreamer/1.16.2"</DEFAULT>
</ARG>
<ARG>
</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>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>
</ARG>
<ARG>
+<NAME>GstShout2send::timeout</NAME>
+<TYPE>guint</TYPE>
+<RANGE>>= 1</RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>timeout</NICK>
+<BLURB>Max amount of time to wait for network activity, in milliseconds.</BLURB>
+<DEFAULT>10000</DEFAULT>
+</ARG>
+
+<ARG>
<NAME>GstSpeexDec::enh</NAME>
<TYPE>gboolean</TYPE>
<RANGE></RANGE>
</ARG>
<ARG>
+<NAME>GstJpegEnc::snapshot</NAME>
+<TYPE>gboolean</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Snapshot</NICK>
+<BLURB>Send EOS after encoding a frame, useful for snapshots.</BLURB>
+<DEFAULT>FALSE</DEFAULT>
+</ARG>
+
+<ARG>
<NAME>GstDVDec::clamp-chroma</NAME>
<TYPE>gboolean</TYPE>
<RANGE></RANGE>
</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>
<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>
<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>
<FLAGS>rw</FLAGS>
<NICK>keep-alive</NICK>
<BLURB>Use HTTP persistent connections.</BLURB>
-<DEFAULT>FALSE</DEFAULT>
+<DEFAULT>TRUE</DEFAULT>
</ARG>
<ARG>
<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>
<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>
</ARG>
<ARG>
+<NAME>GstAudioEcho::surround-delay</NAME>
+<TYPE>gboolean</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<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::surround-mask</NAME>
+<TYPE>guint64</TYPE>
+<RANGE>>= 1</RANGE>
+<FLAGS>rw</FLAGS>
+<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>
<NAME>GstAspectRatioCrop::aspect-ratio</NAME>
<TYPE>GstFraction</TYPE>
<RANGE></RANGE>
<FLAGS>rw</FLAGS>
<NICK>Method</NICK>
<BLURB>Deinterlace Method.</BLURB>
-<DEFAULT>Television: Full resolution</DEFAULT>
+<DEFAULT>Linear</DEFAULT>
</ARG>
<ARG>
<FLAGS>rw</FLAGS>
<NICK>Mode</NICK>
<BLURB>Deinterlace Mode.</BLURB>
-<DEFAULT>Auto detection</DEFAULT>
+<DEFAULT>Auto detection (best effort)</DEFAULT>
</ARG>
<ARG>
</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>
<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>
<ARG>
<NAME>GstRtpJitterBuffer::max-dropout-time</NAME>
<TYPE>guint</TYPE>
-<RANGE></RANGE>
+<RANGE><= G_MAXINT</RANGE>
<FLAGS>rw</FLAGS>
<NICK>Max dropout time</NICK>
<BLURB>The maximum time (milliseconds) of missing packets tolerated.</BLURB>
</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>GstRtpJitterBuffer::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>GstRtpSession::bandwidth</NAME>
<TYPE>gdouble</TYPE>
<RANGE>>= 0</RANGE>
</ARG>
<ARG>
+<NAME>GstFlvMux::metadatacreator</NAME>
+<TYPE>gchar*</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<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>
<NAME>GstCapsSetter::caps</NAME>
<TYPE>GstCaps*</TYPE>
<RANGE></RANGE>
<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>1800</DEFAULT>
+<BLURB>Timescale to use in the movie (units per second, 0 == default).</BLURB>
+<DEFAULT>0</DEFAULT>
</ARG>
<ARG>
</ARG>
<ARG>
+<NAME>GstQTMux::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>GstQTMux::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>GstQTMux::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>GstQTMux::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>GstQTMux::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>GstQTMoovRecover::broken-input</NAME>
<TYPE>gchar*</TYPE>
<RANGE></RANGE>
<ARG>
<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>1800</DEFAULT>
+<BLURB>Timescale to use in the movie (units per second, 0 == default).</BLURB>
+<DEFAULT>0</DEFAULT>
</ARG>
<ARG>
</ARG>
<ARG>
+<NAME>GstMP4Mux::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>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>GstMP4Mux::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>GstMP4Mux::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>GstMP4Mux::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>GstMJ2Mux::dts-method</NAME>
<TYPE>GstQTMuxDtsMethods</TYPE>
<RANGE></RANGE>
<ARG>
<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>1800</DEFAULT>
+<BLURB>Timescale to use in the movie (units per second, 0 == default).</BLURB>
+<DEFAULT>0</DEFAULT>
</ARG>
<ARG>
</ARG>
<ARG>
-<NAME>GstISMLMux::dts-method</NAME>
-<TYPE>GstQTMuxDtsMethods</TYPE>
+<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>
<ARG>
<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>1800</DEFAULT>
+<BLURB>Timescale to use in the movie (units per second, 0 == default).</BLURB>
+<DEFAULT>0</DEFAULT>
</ARG>
<ARG>
</ARG>
<ARG>
+<NAME>GstISMLMux::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>GstISMLMux::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::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>GstISMLMux::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>GstISMLMux::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>Gst3GPPMux::dts-method</NAME>
<TYPE>GstQTMuxDtsMethods</TYPE>
<RANGE></RANGE>
<ARG>
<NAME>Gst3GPPMux::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>1800</DEFAULT>
+<BLURB>Timescale to use in the movie (units per second, 0 == default).</BLURB>
+<DEFAULT>0</DEFAULT>
</ARG>
<ARG>
</ARG>
<ARG>
+<NAME>Gst3GPPMux::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>Gst3GPPMux::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>Gst3GPPMux::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>Gst3GPPMux::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>Gst3GPPMux::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>GstSplitFileSrc::location</NAME>
<TYPE>gchar*</TYPE>
<RANGE></RANGE>
</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>
<NAME>GstSoupHttpClientSink::automatic-redirect</NAME>
<TYPE>gboolean</TYPE>
<RANGE></RANGE>
<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>
</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>
</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>
<RANGE></RANGE>
<FLAGS>rw</FLAGS>
<NICK>Muxer</NICK>
-<BLURB>The muxer element to use (NULL = default mp4mux).</BLURB>
+<BLURB>The muxer element to use (NULL = default mp4mux). Valid only for async-finalize = FALSE.</BLURB>
<DEFAULT></DEFAULT>
</ARG>
<RANGE></RANGE>
<FLAGS>rw</FLAGS>
<NICK>Sink</NICK>
-<BLURB>The sink element (or element chain) to use (NULL = default filesink).</BLURB>
+<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>
+
projects / platform / upstream / gst-plugins-good.git / blobdiff