X-Git-Url: http://review.tizen.org/git/?a=blobdiff_plain;f=docs%2Fplugins%2Fgst-plugins-good-plugins.args;h=0e58a4bf913cef5830618ce4bf09c6d26ab7ff23;hb=1cc4f8ec24f4d02f456caf30ed3eb7f304eb192a;hp=db7849383f8e3b5ad70053bf591cfba6424dd5d0;hpb=8f9828393e96933633e392b1b79ca471d0974fa3;p=platform%2Fupstream%2Fgst-plugins-good.git
diff --git a/docs/plugins/gst-plugins-good-plugins.args b/docs/plugins/gst-plugins-good-plugins.args
index db78493..0e58a4b 100644
--- a/docs/plugins/gst-plugins-good-plugins.args
+++ b/docs/plugins/gst-plugins-good-plugins.args
@@ -199,6 +199,26 @@
+GstMultiFileSink::aggregate-gops
+gboolean
+
+rw
+Aggregate GOPs
+Whether to aggregate GOPs and process them as a whole without splitting.
+FALSE
+
+
+
+GstMultiFileSink::max-file-duration
+guint64
+
+rw
+Maximum File Duration
+Maximum file duration before starting a new file in max-size mode (in nanoseconds).
+18446744073709551615
+
+
+
GstMultiFileSrc::caps
GstCaps*
@@ -253,7 +273,7 @@
gint
>= G_MAXULONG
rw
-Start Index
+Stop Index
Stop value of index. The special value -1 means no stop.
0
@@ -281,7 +301,7 @@
GstSpectrum::bands
guint
-
+[2,1073741824]
rw
Bands
Number of frequency bands.
@@ -299,16 +319,6 @@
-GstSpectrum::message
-gboolean
-
-rw
-Message
-Whether to post a 'spectrum' element message on the bus for each passed interval (deprecated, use post-messages).
-TRUE
-
-
-
GstSpectrum::threshold
gint
<= 0
@@ -359,16 +369,6 @@
-GstVideoflip::method
-GstVideoflipMethod
-
-rw
-method
-method.
-Rotate clockwise 90 degrees
-
-
-
GstVideoBox::alpha
gdouble
[0,1]
@@ -484,7 +484,7 @@
rw
Multicast Group
-The Address of multicast group to join.
+The Address of multicast group to join. (DEPRECATED: Use address property instead).
"0.0.0.0"
@@ -509,16 +509,6 @@
-GstUDPSrc::sockfd
-gint
->= G_MAXULONG
-rw
-Socket Handle
-Socket to use for UDP reception. (-1 == allocate).
--1
-
-
-
GstUDPSrc::buffer-size
gint
>= 0
@@ -539,16 +529,6 @@
-GstUDPSrc::closefd
-gboolean
-
-rw
-Close sockfd
-Close sockfd if passed as property on state change.
-TRUE
-
-
-
GstUDPSrc::skip-first-bytes
gint
>= 0
@@ -559,16 +539,6 @@
-GstUDPSrc::sock
-gint
->= G_MAXULONG
-r
-Socket Handle
-Socket currently in use for UDP reception. (-1 = no socket).
--1
-
-
-
GstUDPSrc::auto-multicast
gboolean
@@ -584,7 +554,7 @@
rw
Multicast Interface
-The network interface on which to join the multicast group.
+The network interface on which to join the multicast group.This allows multiple interfaces seperated by comma. ("eth0,eth1").
NULL
@@ -629,6 +599,46 @@
+GstUDPSrc::address
+gchar*
+
+rw
+Address
+Address to receive packets for. This is equivalent to the multicast-group property for now.
+"0.0.0.0"
+
+
+
+GstUDPSrc::loop
+gboolean
+
+rw
+Multicast Loopback
+Used for setting the multicast loop parameter. TRUE = enable, FALSE = disable.
+TRUE
+
+
+
+GstUDPSrc::retrieve-sender-address
+gboolean
+
+rw
+Retrieve Sender Address
+Whether to retrieve the sender address and add it to buffers as meta. Disabling this might result in minor performance improvements in certain scenarios.
+TRUE
+
+
+
+GstUDPSrc::mtu
+guint
+<= G_MAXINT
+rw
+Expected Maximum Transmission Unit
+Maximum expected packet size. This directly defines the allocationsize of the receive buffer pool.
+1492
+
+
+
GstSMPTE::border
gint
>= 0
@@ -649,16 +659,6 @@
-GstSMPTE::fps
-gfloat
->= 0
-rw
-FPS
-Frames per second if no input files are given (deprecated).
-0
-
-
-
GstSMPTE::type
GstSMPTETransitionType
@@ -694,7 +694,7 @@
rw
Debug
-Dump request and response messages to stdout.
+Dump request and response messages to stdout(DEPRECATED: Printed all RTSP message to gstreamer log as 'log' level).
FALSE
@@ -879,16 +879,226 @@
-GstRTPDec::skip
+GstRTSPSrc::drop-on-latency
+gboolean
+
+rw
+Drop buffers when maximum latency is reached
+Tells the jitterbuffer to never exceed the given latency in size.
+FALSE
+
+
+
+GstRTSPSrc::probation
+guint
+
+rw
+Number of probations
+Consecutive packet sequence numbers to accept the source.
+2
+
+
+
+GstRTSPSrc::udp-reconnect
+gboolean
+
+rw
+Reconnect to the server
+Reconnect to the server if RTSP connection is closed when doing UDP.
+TRUE
+
+
+
+GstRTSPSrc::proxy-id
+gchar*
+
+rw
+proxy-id
+HTTP proxy URI user id for authentication.
+""
+
+
+
+GstRTSPSrc::proxy-pw
+gchar*
+
+rw
+proxy-pw
+HTTP proxy URI user password for authentication.
+""
+
+
+
+GstRTSPSrc::multicast-iface
+gchar*
+
+rw
+Multicast Interface
+The network interface on which to join the multicast group.
+NULL
+
+
+
+GstRTSPSrc::ntp-sync
+gboolean
+
+rw
+Sync on NTP clock
+Synchronize received streams to the NTP clock.
+FALSE
+
+
+
+GstRTSPSrc::use-pipeline-clock
+gboolean
+
+rw
+Use pipeline clock
+Use the pipeline running-time to set the NTP time in the RTCP SR messages(DEPRECATED: Use ntp-time-source property).
+FALSE
+
+
+
+GstRTSPSrc::sdes
+GstStructure*
+
+rw
+SDES
+The SDES items of this session.
+
+
+
+
+GstRTSPSrc::tls-database
+GTlsDatabase*
+
+rw
+TLS database
+TLS database with anchor certificate authorities used to validate the server certificate.
+
+
+
+
+GstRTSPSrc::tls-validation-flags
+GTlsCertificateFlags
+
+rw
+TLS validation flags
+TLS certificate validation flags used to validate the server certificate.
+G_TLS_CERTIFICATE_UNKNOWN_CA|G_TLS_CERTIFICATE_BAD_IDENTITY|G_TLS_CERTIFICATE_NOT_ACTIVATED|G_TLS_CERTIFICATE_EXPIRED|G_TLS_CERTIFICATE_REVOKED|G_TLS_CERTIFICATE_INSECURE|G_TLS_CERTIFICATE_GENERIC_ERROR
+
+
+
+GstRTSPSrc::do-retransmission
+gboolean
+
+rw
+Retransmission
+Ask the server to retransmit lost packets.
+TRUE
+
+
+
+GstRTSPSrc::tls-interaction
+GTlsInteraction*
+
+rw
+TLS interaction
+A GTlsInteraction object to promt the user for password or certificate.
+
+
+
+
+GstRTSPSrc::ntp-time-source
+GstRTSPSrcNtpTimeSource
+
+rw
+NTP Time Source
+NTP time source for RTCP packets.
+NTP time based on realtime clock
+
+
+
+GstRTSPSrc::user-agent
+gchar*
+
+rw
+User Agent
+The User-Agent string to send to the server.
+"GStreamer/1.16.1"
+
+
+
+GstRTSPSrc::max-rtcp-rtp-time-diff
gint
+>= G_MAXULONG
+rw
+Max RTCP RTP Time Diff
+Maximum amount of time in ms that the RTP time in RTCP SRs is allowed to be ahead (-1 disabled).
+1000
+
+
+
+GstRTSPSrc::rfc7273-sync
+gboolean
+
+rw
+Sync on RFC7273 clock
+Synchronize received streams to the RFC7273 clock (requires clock and offset to be provided).
+FALSE
+
+
+
+GstRTSPSrc::default-rtsp-version
+GstRTSPVersion
+
+rw
+The RTSP version to try first
+The RTSP version that should be tried first when negotiating version.
+GST_RTSP_VERSION_1_0
+
+
+
+GstRTSPSrc::max-ts-offset
+gint64
+>= 0
+rw
+Max TS Offset
+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).
+3000000000
+
+
+
+GstRTSPSrc::max-ts-offset-adjustment
+guint64
rw
-Skip
-skip (unused).
+Max Timestamp Offset Adjustment
+The maximum number of nanoseconds per frame that time stamp offsets may be adjusted (0 = no limit).
0
+GstRTSPSrc::backchannel
+GstRTSPBackchannel
+
+rw
+Backchannel type
+The type of backchannel to setup. Default is 'none'.
+No backchannel
+
+
+
+GstRTSPSrc::teardown-timeout
+guint64
+
+rw
+Teardown Timeout
+When transitioning PAUSED-READY, allow up to timeout (in nanoseconds) delay in order to send teardown (0 = disabled).
+100000000
+
+
+
GstRTPDec::latency
guint
@@ -914,7 +1124,7 @@
rw
message
-Post a level message for each passed interval.
+Post a 'level' message for each passed interval (deprecated, use the post-messages property instead).
TRUE
@@ -939,6 +1149,16 @@
+GstLevel::post-messages
+gboolean
+
+rw
+Post Messages
+Whether to post a 'level' element message on the bus for each passed interval.
+TRUE
+
+
+
GstDiceTV::square-bits
gint
[0,5]
@@ -1001,7 +1221,7 @@
GstQuarkTV::planes
gint
-[0,64]
+[1,64]
rw
Planes
Number of planes.
@@ -1009,16 +1229,6 @@
-GstEFence::fence-top
-gboolean
-
-rw
-Fence Top
-Align buffers with top of fenced region.
-TRUE
-
-
-
GstAlpha::alpha
gdouble
[0,1]
@@ -1064,7 +1274,7 @@
<= 255
rw
Target Blue
-The Blue target.
+The blue color value for custom RGB chroma keying.
0
@@ -1074,7 +1284,7 @@
<= 255
rw
Target Green
-The Green target.
+The green color value for custom RGB chroma keying.
255
@@ -1084,7 +1294,7 @@
<= 255
rw
Target Red
-The Red target.
+The red color value for custom RGB chroma keying.
0
@@ -1103,7 +1313,7 @@
guint
<= 128
rw
-Sensitivity
+White Sensitivity
Sensitivity to bright colors.
100
@@ -1144,7 +1354,7 @@
rw
ip
-ip.
+IP address or hostname.
"127.0.0.1"
@@ -1199,22 +1409,12 @@
-GstShout2send::sync
-gboolean
-
-rw
-Sync
-Sync on the clock.
-FALSE
-
-
-
GstShout2send::url
gchar*
rw
url
-url.
+the stream's homepage URL.
""
@@ -1239,82 +1439,22 @@
-GstSpeexDec::enh
-gboolean
-
-rw
-Enh
-Enable perceptual enhancement.
-TRUE
-
-
-
-DV1394Src::channel
-gint
-[0,64]
-rw
-Channel
-Channel number for listening.
-63
-
-
-
-DV1394Src::consecutive
-gint
+GstShout2send::timeout
+guint
>= 1
rw
-consecutive frames
-send n consecutive frames after skipping.
-1
-
-
-
-DV1394Src::drop-incomplete
-gboolean
-
-rw
-drop_incomplete
-drop incomplete frames.
-TRUE
-
-
-
-DV1394Src::guid
-guint64
-
-rw
-GUID
-select one of multiple DV devices by its GUID. use a hexadecimal like 0xhhhhhhhhhhhhhhhh. (0 = no guid).
-0
-
-
-
-DV1394Src::port
-gint
-[-1,16]
-rw
-Port
-Port number (-1 automatic).
--1
-
-
-
-DV1394Src::skip
-gint
->= 0
-rw
-skip frames
-skip n frames.
-0
+timeout
+Max amount of time to wait for network activity, in milliseconds.
+10000
-DV1394Src::use-avc
+GstSpeexDec::enh
gboolean
rw
-Use AV/C
-Use AV/C VTR control.
+Enh
+Enable perceptual enhancement.
TRUE
@@ -1399,53 +1539,13 @@
-GstSmokeEnc::keyframe
-gint
-[1,100000]
-rw
-Keyframe
-Insert keyframe every N frames.
-20
-
-
-
-GstSmokeEnc::qmax
-gint
-[0,100]
-rw
-Qmax
-Maximum quality.
-85
-
-
-
-GstSmokeEnc::qmin
-gint
-[0,100]
-rw
-Qmin
-Minimum quality.
-10
-
-
-
-GstSmokeEnc::threshold
-gint
-[0,100000000]
-rw
-Threshold
-Motion estimation threshold.
-3000
-
-
-
-GstEsdSink::host
-gchar*
+GstJpegEnc::snapshot
+gboolean
rw
-Host
-The host running the esound daemon.
-NULL
+Snapshot
+Send EOS after encoding a frame, useful for snapshots.
+FALSE
@@ -1599,16 +1699,6 @@
-GstRtpGSMParse::frequency
-gint
-
-rw
-frequency
-frequency.
-8000
-
-
-
GstSpeexEnc::abr
gint
>= 0
@@ -1949,16 +2039,6 @@
-GstVideoMixer::background
-GstVideoMixerBackground
-
-rw
-Background
-Background type.
-Checker pattern
-
-
-
GstMatroskaMux::writing-app
gchar*
@@ -1999,6 +2079,36 @@
+GstMatroskaMux::timecodescale
+gint64
+[1,1000000000]
+rw
+Timecode Scale
+TimecodeScale used to calculate the Raw Timecode of a Block.
+1000000
+
+
+
+GstMatroskaMux::max-cluster-duration
+gint64
+>= 0
+rw
+Maximum cluster duration
+A new cluster will be created if its duration exceeds this value. 0 means no maximum duration.
+65535000000
+
+
+
+GstMatroskaMux::min-cluster-duration
+gint64
+>= 0
+rw
+Minimum cluster duration
+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.
+500000000
+
+
+
GstTest::allowed-timestamp-deviation
gint64
>= G_MAXULONG
@@ -2169,96 +2279,6 @@
-GstTextOverlay::deltax
-gint
-
-w
-X position modifier
-Shift X position to the left or to the right. Unit is pixels.
-0
-
-
-
-GstTextOverlay::deltay
-gint
-
-w
-Y position modifier
-Shift Y position up or down. Unit is pixels.
-0
-
-
-
-GstTextOverlay::font-desc
-gchararray
-
-w
-font description
-Pango font description of font to be used for rendering. See documentation of pango_font_description_from_string for syntax.
-""
-
-
-
-GstTextOverlay::halign
-gchararray
-
-w
-horizontal alignment
-Horizontal alignment of the text. Can be either 'left', 'right', or 'center'.
-"center"
-
-
-
-GstTextOverlay::shaded-background
-gboolean
-
-w
-shaded background
-Whether to shade the background under the text area.
-FALSE
-
-
-
-GstTextOverlay::text
-gchararray
-
-w
-text
-Text to be display.
-""
-
-
-
-GstTextOverlay::valign
-gchararray
-
-w
-vertical alignment
-Vertical alignment of the text. Can be either 'baseline', 'bottom', or 'top'.
-"baseline"
-
-
-
-GstTextOverlay::xpad
-gint
-
-w
-horizontal paddding
-Horizontal paddding when using left/right alignment.
-25
-
-
-
-GstTextOverlay::ypad
-gint
-
-w
-vertical padding
-Vertical padding when using top/bottom alignment.
-25
-
-
-
GstCutter::leaky
gboolean
@@ -2312,33 +2332,13 @@
GstVideoFlip::method
GstVideoFlipMethod
-rw
+rwx
method
-method.
+method (deprecated, use video-direction instead).
Identity (no rotation)
-GstRtpMP4VPay::send-config
-gboolean
-
-rw
-Send Config
-Send the config parameters in RTP packets as well(deprecated see config-interval).
-FALSE
-
-
-
-GstRtpMP4VPay::buffer-list
-gboolean
-
-rw
-Buffer Array
-Use Buffer Arrays.
-FALSE
-
-
-
GstRtpMP4VPay::config-interval
guint
<= 3600
@@ -2349,16 +2349,6 @@
-GstRTPDepay::skip
-gint
-
-rw
-skip
-skip.
-0
-
-
-
GstMultipartMux::boundary
gchar*
@@ -2369,186 +2359,66 @@
-GstCairoTextOverlay::deltax
-gint
+GstID3Demux::prefer-v1
+gboolean
-w
-X position modifier
-Shift X position to the left or to the right. Unit is pixels.
-0
+rwx
+Prefer version 1 tag
+Prefer tags from ID3v1 tag at end of file when both ID3v1 and ID3v2 tags are present.
+FALSE
-GstCairoTextOverlay::deltay
-gint
+GstDynUDPSink::close-socket
+gboolean
-w
-Y position modifier
-Shift Y position up or down. Unit is pixels.
-0
+rw
+Close socket
+Close socket if passed as property on state change.
+TRUE
-GstCairoTextOverlay::font-desc
-gchar*
+GstDynUDPSink::socket
+GSocket*
-w
-font description
-Pango font description of font to be used for rendering. See documentation of pango_font_description_from_string for syntax.
-""
+rw
+Socket
+Socket to use for UDP sending. (NULL == allocate).
+
-GstCairoTextOverlay::halign
+GstDynUDPSink::bind-address
gchar*
-w
-horizontal alignment
-Horizontal alignment of the text. Can be either 'left', 'right', or 'center'.
-"center"
+rw
+Bind Address
+Address to bind the socket to.
+NULL
-GstCairoTextOverlay::shaded-background
-gboolean
-
-w
-shaded background
-Whether to shade the background under the text area.
-FALSE
+GstDynUDPSink::bind-port
+gint
+[0,65535]
+rw
+Bind Port
+Port to bind the socket to.
+0
-GstCairoTextOverlay::text
-gchar*
-
-w
-text
-Text to be display.
-""
-
-
-
-GstCairoTextOverlay::valign
-gchar*
-
-w
-vertical alignment
-Vertical alignment of the text. Can be either 'baseline', 'bottom', or 'top'.
-"baseline"
-
-
-
-GstCairoTextOverlay::xpad
-gint
-
-w
-horizontal paddding
-Horizontal paddding when using left/right alignment.
-25
-
-
-
-GstCairoTextOverlay::ypad
-gint
-
-w
-vertical padding
-Vertical padding when using top/bottom alignment.
-25
-
-
-
-GstCairoTextOverlay::silent
-gboolean
-
-w
-silent
-Whether to render the text string.
-FALSE
-
-
-
-GstOssMixerElement::device-name
-gchar*
-
-r
-Device name
-Human-readable name of the sound device.
-NULL
-
-
-
-GstOssMixerElement::device
-gchar*
-
-rw
-Device
-OSS mixer device (usually /dev/mixer).
-"/dev/mixer"
-
-
-
-GstID3Demux::prefer-v1
-gboolean
-
-rwx
-Prefer version 1 tag
-Prefer tags from ID3v1 tag at end of file when both ID3v1 and ID3v2 tags are present.
-FALSE
-
-
-
-GstDynUDPSink::sockfd
-gint
-[G_MAXULONG,32767]
-rw
-socket handle
-Socket to use for UDP sending. (-1 == allocate).
--1
-
-
-
-GstDynUDPSink::closefd
-gboolean
-
-rw
-Close sockfd
-Close sockfd if passed as property on state change.
-TRUE
-
-
-
-GstDynUDPSink::close-socket
-gboolean
-
-rw
-Close socket
-Close socket if passed as property on state change.
-TRUE
-
-
-
-GstDynUDPSink::socket
+GstDynUDPSink::socket-v6
GSocket*
rw
-Socket
-Socket to use for UDP sending. (NULL == allocate).
+Socket IPv6
+Socket to use for UDPv6 sending. (NULL == allocate).
-GstCdioCddaSrc::read-speed
-gint
-[-1,100]
-rw
-Read speed
-Read from device at the specified speed (-1 = default).
--1
-
-
-
GstMultiUDPSink::bytes-served
guint64
@@ -2579,36 +2449,6 @@
-GstMultiUDPSink::closefd
-gboolean
-
-rw
-Close sockfd
-Close sockfd if passed as property on state change.
-TRUE
-
-
-
-GstMultiUDPSink::sock
-gint
->= G_MAXULONG
-r
-Socket Handle
-Socket currently in use for UDP sending. (-1 == no socket).
--1
-
-
-
-GstMultiUDPSink::sockfd
-gint
->= G_MAXULONG
-rw
-Socket Handle
-Socket to use for UDP sending. (-1 == allocate).
--1
-
-
-
GstMultiUDPSink::auto-multicast
gboolean
@@ -2709,93 +2549,63 @@
-GstCmmlDec::wait-clip-end-time
+GstMultiUDPSink::force-ipv4
gboolean
rw
-Wait clip end time
-Send a tag for a clip when the clip ends, setting its end-time. Use when you need to know both clip's start-time and end-time.
+Force IPv4
+Forcing the use of an IPv4 socket (DEPRECATED, has no effect anymore).
FALSE
-GstCmmlEnc::granule-rate-denominator
-gint64
->= 0
-rwx
-Granulerate denominator
-Granulerate denominator.
-1
-
-
-
-GstCmmlEnc::granule-rate-numerator
-gint64
->= 0
-rwx
-Granulerate numerator
-Granulerate numerator.
-1000
-
-
-
-GstCmmlEnc::granule-shift
-guchar
-<= 64
-rwx
-Granuleshift
-The number of lower bits to use for partitioning a granule position.
-32
-
-
-
-GstHalAudioSrc::udi
+GstMultiUDPSink::multicast-iface
gchar*
rw
-UDI
-Unique Device Id.
+Multicast Interface
+The network interface on which to join the multicast group.
NULL
-GstHalAudioSink::udi
+GstMultiUDPSink::bind-address
gchar*
rw
-UDI
-Unique Device Id.
+Bind Address
+Address to bind the socket to.
NULL
-GstPixbufScale::method
-GstPixbufScaleMethod
-
+GstMultiUDPSink::bind-port
+gint
+[0,65535]
rw
-method
-method.
-2
+Bind Port
+Port to bind the socket to.
+0
-GstGdkPixbuf::silent
-gboolean
+GstMultiUDPSink::socket-v6
+GSocket*
rw
-Silent
-Produce verbose output ? (deprecated).
-FALSE
+Socket Handle IPv6
+Socket to use for UDPv6 sending. (NULL == allocate).
+
-GstGConfAudioSink::profile
-GstGConfProfile
+GstMultiUDPSink::used-socket-v6
+GSocket*
-rw
-Profile
-Profile.
-Sound Events
+r
+Used Socket Handle IPv6
+Socket currently in use for UDPv6 sending. (NULL == no socket).
+
@@ -2809,16 +2619,6 @@
-GstXImageSrc::screen-num
-guint
-<= G_MAXINT
-rw
-Screen number
-X Screen Number.
-0
-
-
-
GstXImageSrc::show-pointer
gboolean
@@ -2949,16 +2749,6 @@
-GstMultipartDemux::autoscan
-gboolean
-
-rw
-autoscan
-Try to autofind the prefix (deprecated unused, see boundary).
-FALSE
-
-
-
GstMultipartDemux::boundary
gchar*
@@ -3004,7 +2794,7 @@
>= G_MAXULONG
rw
Maximum Consecutive Decoding Errors
-Error out after receiving N consecutive decoding errors (-1 = never fail, 0 = automatic, 1 = fail on first error).
+(Deprecated) Error out after receiving N consecutive decoding errors (-1 = never fail, 0 = automatic, 1 = fail on first error).
0
@@ -3109,16 +2899,6 @@
-GstDirectDrawSink::force-aspect-ratio
-gboolean
-
-rw
-Force aspect ratio
-When enabled, scaling will respect original aspect ratio.
-FALSE
-
-
-
GstWavpackEnc::bitrate
guint
<= 9600000
@@ -3221,40 +3001,40 @@
GstVideoCrop::bottom
gint
->= 0
+>= G_MAXULONG
rw
Bottom
-Pixels to crop at bottom.
+Pixels to crop at bottom (-1 to auto-crop).
0
GstVideoCrop::left
gint
->= 0
+>= G_MAXULONG
rw
Left
-Pixels to crop at left.
+Pixels to crop at left (-1 to auto-crop).
0
GstVideoCrop::right
gint
->= 0
+>= G_MAXULONG
rw
Right
-Pixels to crop at right.
+Pixels to crop at right (-1 to auto-crop).
0
GstVideoCrop::top
gint
->= 0
+>= G_MAXULONG
rw
Top
-Pixels to crop at top.
+Pixels to crop at top (-1 to auto-crop).
0
@@ -3289,26 +3069,6 @@
-GstV4l2Src::queue-size
-guint
-[1,16]
-rw
-Queue size
-Number of buffers to be enqueud in the driver in streaming mode.
-2
-
-
-
-GstV4l2Src::always-copy
-gboolean
-
-rw
-Always Copy
-If the buffer will or not be used directly from mmap.
-TRUE
-
-
-
GstV4l2Src::device-fd
gint
>= G_MAXULONG
@@ -3339,16 +3099,6 @@
-GstV4l2Src::decimate
-gint
->= 1
-rw
-Decimate
-Only use every nth frame.
-1
-
-
-
GstV4l2Src::hue
gint
@@ -3389,6 +3139,36 @@
+GstV4l2Src::extra-controls
+GstStructure*
+
+rw
+Extra Controls
+Extra v4l2 controls (CIDs) for the device.
+
+
+
+
+GstV4l2Src::force-aspect-ratio
+gboolean
+
+rw
+Force aspect ratio
+When enabled, the pixel aspect ratio will be enforced.
+TRUE
+
+
+
+GstV4l2Src::pixel-aspect-ratio
+gchar*
+
+rw
+Pixel Aspect Ratio
+Overwrite the pixel aspect ratio of the device.
+"1/1"
+
+
+
GstAudioChebBand::lower-frequency
gfloat
[0,100000]
@@ -3549,16 +3329,6 @@
-GstAudioWSincLimit::frequency
-gdouble
->= 0
-rw
-Frequency
-Cut-off Frequency (Hz).
-0
-
-
-
GstAudioWSincLimit::length
gint
[3,256000]
@@ -3600,8 +3370,8 @@
GstRndBufferSize::max
-glong
-[1,G_MAXINT]
+gint
+>= 1
rwx
maximum
maximum buffer size.
@@ -3610,8 +3380,8 @@
GstRndBufferSize::min
-glong
-[0,G_MAXINT]
+gint
+>= 0
rwx
mininum
mininum buffer size.
@@ -3620,39 +3390,59 @@
GstRndBufferSize::seed
-gulong
-<= G_MAXUINT
+guint
+
rwx
random number seed
seed for randomness (initialized when going from READY to PAUSED).
-
+0
-GstAutoAudioSink::filter-caps
-GstCaps*
+GstAutoAudioSink::ts-offset
+gint64
rw
-Filter caps
-Filter sink candidates using these caps.
-
+TS Offset
+Timestamp offset in nanoseconds.
+0
-GstAutoVideoSink::filter-caps
-GstCaps*
+GstAutoAudioSink::sync
+gboolean
rw
-Filter caps
-Filter sink candidates using these caps.
-
+Sync
+Sync on the clock.
+TRUE
-GstOsxAudioSink::device
-gint
->= 0
-rw
+GstAutoVideoSink::ts-offset
+gint64
+
+rw
+TS Offset
+Timestamp offset in nanoseconds.
+0
+
+
+
+GstAutoVideoSink::sync
+gboolean
+
+rw
+Sync
+Sync on the clock.
+TRUE
+
+
+
+GstOsxAudioSink::device
+gint
+>= 0
+rw
Device ID
Device ID of output device.
0
@@ -3699,16 +3489,6 @@
-GstGdkPixbufSink::send-messages
-gboolean
-
-rw
-Send Messages
-Whether to post messages containing pixbufs on the bus (deprecated, use post-messages).
-TRUE
-
-
-
GstGdkPixbufSink::post-messages
gboolean
@@ -3739,53 +3519,13 @@
-GstSoupHTTPSrc::iradio-genre
-gchar*
-
-r
-iradio-genre
-Genre of the stream.
-NULL
-
-
-
GstSoupHTTPSrc::iradio-mode
gboolean
rw
iradio-mode
-Enable internet radio mode (extraction of shoutcast/icecast metadata).
-FALSE
-
-
-
-GstSoupHTTPSrc::iradio-name
-gchar*
-
-r
-iradio-name
-Name of the stream.
-NULL
-
-
-
-GstSoupHTTPSrc::iradio-title
-gchar*
-
-r
-iradio-title
-Name of currently playing song.
-NULL
-
-
-
-GstSoupHTTPSrc::iradio-url
-gchar*
-
-r
-iradio-url
-Homepage URL for radio stream.
-NULL
+Enable internet radio mode (ask server to send shoutcast/icecast metadata interleaved with the actual stream data).
+TRUE
@@ -3815,7 +3555,7 @@
rw
User-Agent
Value of the User-Agent HTTP request header field.
-"GStreamer souphttpsrc "
+"GStreamer souphttpsrc 1.16.1 "
@@ -3885,66 +3625,136 @@
rw
timeout
Value in seconds to timeout a blocking I/O (0 = No timeout).
-0
+15
-GstRTPDVPay::mode
-GstDVPayMode
+GstSoupHTTPSrc::compress
+gboolean
rw
-Mode
-The payload mode of payloading.
-Video only
+Compress
+Allow compressed content encodings.
+FALSE
+
+
+
+GstSoupHTTPSrc::http-log-level
+SoupLoggerLogLevel
+
+rw
+HTTP log level
+Set log level for soup's HTTP session log.
+SOUP_LOGGER_LOG_HEADERS
+
+
+
+GstSoupHTTPSrc::keep-alive
+gboolean
+
+rw
+keep-alive
+Use HTTP persistent connections.
+TRUE
+
+
+
+GstSoupHTTPSrc::retries
+gint
+>= G_MAXULONG
+rw
+Retries
+Maximum number of retries until giving up (-1=infinite).
+3
-GstRtpH264Pay::profile-level-id
+GstSoupHTTPSrc::ssl-ca-file
gchar*
rw
-profile-level-id
-The base64 profile-level-id to set in the sink caps (deprecated).
+SSL CA File
+Location of a SSL anchor CA file to use.
NULL
-GstRtpH264Pay::sprop-parameter-sets
+GstSoupHTTPSrc::ssl-strict
+gboolean
+
+rw
+SSL Strict
+Strict SSL certificate checking.
+TRUE
+
+
+
+GstSoupHTTPSrc::ssl-use-system-ca-file
+gboolean
+
+rw
+Use System CA File
+Use system CA file.
+TRUE
+
+
+
+GstSoupHTTPSrc::tls-database
+GTlsDatabase*
+
+rw
+TLS database
+TLS database with anchor certificate authorities used to validate the server certificate.
+
+
+
+
+GstSoupHTTPSrc::method
gchar*
rw
-sprop-parameter-sets
-The base64 sprop-parameter-sets to set in out caps (set to NULL to extract from stream).
+HTTP method
+The HTTP method to use (GET, HEAD, OPTIONS, etc).
NULL
-GstRtpH264Pay::scan-mode
-GstH264PayScanMode
+GstSoupHTTPSrc::tls-interaction
+GTlsInteraction*
rw
-Scan Mode
-How to scan the input buffers for NAL units. Performance can be increased when certain assumptions are made about the input buffers.
-Buffers contain multiple complete NALUs
+TLS interaction
+A GTlsInteraction object to be used when the connection or certificate database need to interact with the user.
+
-GstRtpH264Pay::buffer-list
-gboolean
+GstRTPDVPay::mode
+GstDVPayMode
rw
-Buffer List
-Use Buffer Lists.
-FALSE
+Mode
+The payload mode of payloading.
+Video only
+
+
+
+GstRtpH264Pay::sprop-parameter-sets
+gchar*
+
+rw
+sprop-parameter-sets
+The base64 sprop-parameter-sets to set in out caps (set to NULL to extract from stream).
+NULL
GstRtpH264Pay::config-interval
-guint
-<= 3600
+gint
+[G_MAXULONG,3600]
rw
SPS PPS Send Interval
-Send SPS and PPS Insertion Interval in seconds (sprop parameter sets will be multiplexed in the data stream when detected.) (0 = disabled).
+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).
0
@@ -19889,66 +19699,6 @@
-GstVideoMixerPad::alpha
-gdouble
-[0,1]
-rw
-Alpha
-Alpha of the picture.
-1
-
-
-
-GstVideoMixerPad::xpos
-gint
-
-rw
-X Position
-X Position of the picture.
-0
-
-
-
-GstVideoMixerPad::ypos
-gint
-
-rw
-Y Position
-Y Position of the picture.
-0
-
-
-
-GstVideoMixerPad::zorder
-guint
-<= 10000
-rw
-Z-Order
-Z Order of the picture.
-0
-
-
-
-GstRtpH264Depay::byte-stream
-gboolean
-
-rw
-Byte Stream
-Generate byte stream format of NALU (deprecated; use caps).
-TRUE
-
-
-
-GstRtpH264Depay::access-unit
-gboolean
-
-rw
-Access Unit
-Merge NALU into AU (picture) (deprecated; use caps).
-FALSE
-
-
-
GstAudioKaraoke::filter-band
gfloat
[0,441]
@@ -20039,16 +19789,6 @@
-GstPulseSink::client
-gchar*
-
-rw
-Client
-The PulseAudio client name to use.
-""
-
-
-
GstPulseSink::stream-properties
GstStructure*
@@ -20065,7 +19805,17 @@
rw
Client Name
The PulseAudio client name to use.
-"lt-gst-plugins-good-plugins-scan"
+"gst-plugins-good-plugins-scan"
+
+
+
+GstPulseSink::current-device
+gchar*
+
+r
+Current Device
+The current PulseAudio sink device.
+NULL
@@ -20109,16 +19859,6 @@
-GstPulseSrc::client
-gchar*
-
-rw
-Client
-The PulseAudio client_name_to_use.
-""
-
-
-
GstPulseSrc::mute
gboolean
@@ -20155,36 +19895,16 @@
rw
Client Name
The PulseAudio client_name_to_use.
-"lt-gst-plugins-good-plugins-scan"
-
-
-
-GstPulseMixer::device
-gchar*
-
-rw
-Device
-The PulseAudio sink or source to control.
-NULL
+"gst-plugins-good-plugins-scan"
-GstPulseMixer::device-name
+GstPulseSrc::current-device
gchar*
r
-Device name
-Human-readable name of the sound device.
-NULL
-
-
-
-GstPulseMixer::server
-gchar*
-
-rw
-Server
-The PulseAudio server to connect to.
+Current Device
+The current PulseAudio source device.
NULL
@@ -20389,26 +20109,6 @@
-GstAutoVideoSrc::filter-caps
-GstCaps*
-
-rw
-Filter caps
-Filter src candidates using these caps.
-
-
-
-
-GstAutoAudioSrc::filter-caps
-GstCaps*
-
-rw
-Filter caps
-Filter sink candidates using these caps.
-
-
-
-
GstRtpJPEGPay::quality
gint
[0,255]
@@ -20429,16 +20129,6 @@
-GstRtpJPEGPay::buffer-list
-gboolean
-
-rw
-Buffer List
-Use Buffer Lists.
-FALSE
-
-
-
GstAudioFIRFilter::kernel
GValueArray*
@@ -20479,17 +20169,17 @@
-GstAudioDelay::delay
+GstAudioEcho::delay
guint64
>= 1
rw
Delay
-Delay in nanoseconds.
+Delay of the echo in nanoseconds.
1
-GstAudioDelay::feedback
+GstAudioEcho::feedback
gfloat
[0,1]
rw
@@ -20499,7 +20189,7 @@
-GstAudioDelay::intensity
+GstAudioEcho::intensity
gfloat
[0,1]
rw
@@ -20509,73 +20199,33 @@
-GstAudioReverb::delay
+GstAudioEcho::max-delay
guint64
>= 1
rw
-Delay
-Delay of the echo in nanoseconds.
+Maximum Delay
+Maximum delay of the echo in nanoseconds (can't be changed in PLAYING or PAUSED state).
1
-GstAudioReverb::feedback
-gfloat
-[0,1]
+GstAudioEcho::surround-delay
+gboolean
+
rw
-Feedback
-Amount of feedback.
-0
+Enable Surround Delay
+Delay Surround Channels when TRUE instead of applying an echo effect.
+FALSE
-GstAudioReverb::intensity
-gfloat
-[0,1]
+GstAudioEcho::surround-mask
+guint64
+>= 1
rw
-Intensity
-Intensity of the echo.
-0
-
-
-
-GstAudioEcho::delay
-guint64
->= 1
-rw
-Delay
-Delay of the echo in nanoseconds.
-1
-
-
-
-GstAudioEcho::feedback
-gfloat
-[0,1]
-rw
-Feedback
-Amount of feedback.
-0
-
-
-
-GstAudioEcho::intensity
-gfloat
-[0,1]
-rw
-Intensity
-Intensity of the echo.
-0
-
-
-
-GstAudioEcho::max-delay
-guint64
->= 1
-rw
-Maximum Delay
-Maximum delay of the echo in nanoseconds (can't be changed in PLAYING or PAUSED state).
-1
+Surround Mask
+A bitmask of channels that are considered surround and delayed when surround-delay = TRUE.
+18446744073709551612
@@ -20635,7 +20285,7 @@
rw
Method
Deinterlace Method.
-Television: Full resolution
+Linear
@@ -20645,7 +20295,7 @@
rw
Mode
Deinterlace Mode.
-Auto detection
+Auto detection (best effort)
@@ -20904,7 +20554,7 @@
rw
Use pipeline clock
-Use the pipeline clock to set the NTP time in the RTCP SR messages.
+Use the pipeline running-time to set the NTP time in the RTCP SR messages (DEPRECATED: Use ntp-time-source property).
FALSE
@@ -20929,6 +20579,136 @@
+GstRtpBin::drop-on-latency
+gboolean
+
+rw
+Drop buffers when maximum latency is reached
+Tells the jitterbuffer to never exceed the given latency in size.
+FALSE
+
+
+
+GstRtpBin::do-sync-event
+gboolean
+
+rw
+Do Sync Event
+Send event downstream when a stream is synchronized to the sender.
+FALSE
+
+
+
+GstRtpBin::do-retransmission
+gboolean
+
+rw
+Do retransmission
+Enable retransmission on all streams.
+FALSE
+
+
+
+GstRtpBin::rtp-profile
+GstRTPProfile
+
+rw
+RTP Profile
+Default RTP profile of newly created sessions.
+GST_RTP_PROFILE_AVP
+
+
+
+GstRtpBin::ntp-time-source
+GstRtpNtpTimeSource
+
+rw
+NTP Time Source
+NTP time source for RTCP packets.
+NTP time based on realtime clock
+
+
+
+GstRtpBin::max-dropout-time
+guint
+
+rw
+Max dropout time
+The maximum time (milliseconds) of missing packets tolerated.
+60000
+
+
+
+GstRtpBin::max-misorder-time
+guint
+
+rw
+Max misorder time
+The maximum time (milliseconds) of misordered packets tolerated.
+2000
+
+
+
+GstRtpBin::max-rtcp-rtp-time-diff
+gint
+>= G_MAXULONG
+rw
+Max RTCP RTP Time Diff
+Maximum amount of time in ms that the RTP time in RTCP SRs is allowed to be ahead (-1 disabled).
+1000
+
+
+
+GstRtpBin::rtcp-sync-send-time
+gboolean
+
+rw
+RTCP Sync Send Time
+Use send time or capture time for RTCP sync (TRUE = send time, FALSE = capture time).
+TRUE
+
+
+
+GstRtpBin::rfc7273-sync
+gboolean
+
+rw
+Sync on RFC7273 clock
+Synchronize received streams to the RFC7273 clock (requires clock and offset to be provided).
+FALSE
+
+
+
+GstRtpBin::max-streams
+guint
+
+rw
+Max Streams
+The maximum number of streams to create for one session.
+4294967295
+
+
+
+GstRtpBin::max-ts-offset
+gint64
+>= 0
+rw
+Max TS Offset
+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).
+3000000000
+
+
+
+GstRtpBin::max-ts-offset-adjustment
+guint64
+
+rw
+Max Timestamp Offset Adjustment
+The maximum number of nanoseconds per frame that time stamp offsets may be adjusted (0 = no limit).
+0
+
+
+
GstRtpJitterBuffer::do-lost
gboolean
@@ -20989,777 +20769,987 @@
-GstRtpSession::bandwidth
-gdouble
->= 0
+GstRtpJitterBuffer::do-retransmission
+gboolean
+
rw
-Bandwidth
-The bandwidth of the session in bytes per second (0 for auto-discover).
-64000
+Do Retransmission
+Send retransmission events upstream when a packet is late.
+FALSE
-GstRtpSession::internal-session
-RTPSession*
-
-r
-Internal Session
-The internal RTPSession object.
-
+GstRtpJitterBuffer::rtx-delay
+gint
+>= G_MAXULONG
+rw
+RTX Delay
+Extra time in ms to wait before sending retransmission event (-1 automatic).
+-1
-GstRtpSession::ntp-ns-base
-guint64
-
+GstRtpJitterBuffer::rtx-delay-reorder
+gint
+>= G_MAXULONG
rw
-NTP base time
-The NTP base time corresponding to running_time 0 (deprecated).
-0
+RTX Delay Reorder
+Sending retransmission event when this much reordering (0 disable).
+3
-GstRtpSession::num-active-sources
-guint
-
-r
-Num Active Sources
-The number of active sources in the session.
-0
+GstRtpJitterBuffer::rtx-retry-period
+gint
+>= G_MAXULONG
+rw
+RTX Retry Period
+Try to get a retransmission for this many ms (-1 automatic).
+-1
-GstRtpSession::num-sources
-guint
-
-r
-Num Sources
-The number of sources in the session.
-0
+GstRtpJitterBuffer::rtx-retry-timeout
+gint
+>= G_MAXULONG
+rw
+RTX Retry Timeout
+Retry sending a transmission event after this timeout in ms (-1 automatic).
+-1
-GstRtpSession::rtcp-fraction
-gdouble
->= 0
-rw
-RTCP Fraction
-The RTCP bandwidth of the session in bytes per second (or as a real fraction of the RTP bandwidth if < 1.0).
-3200
+GstRtpJitterBuffer::stats
+GstStructure*
+
+r
+Statistics
+Various statistics.
+
-GstRtpSession::sdes
-GstStructure*
+GstRtpJitterBuffer::rtx-min-delay
+guint
rw
-SDES
-The SDES items of this session.
-
+Minimum RTX Delay
+Minimum time in ms to wait before sending retransmission event.
+0
-GstRtpSession::rtcp-rr-bandwidth
+GstRtpJitterBuffer::rtx-min-retry-timeout
gint
>= G_MAXULONG
rw
-RTCP RR bandwidth
-The RTCP bandwidth used for receivers in bytes per second (-1 = default).
+RTX Min Retry Timeout
+Minimum timeout between sending a transmission event in ms (-1 automatic).
-1
-GstRtpSession::rtcp-rs-bandwidth
+GstRtpJitterBuffer::rtx-max-retries
gint
>= G_MAXULONG
rw
-RTCP RS bandwidth
-The RTCP bandwidth used for senders in bytes per second (-1 = default).
+RTX Max Retries
+The maximum number of retries to request a retransmission. (-1 not limited).
-1
-GstRtpSession::use-pipeline-clock
+GstRtpJitterBuffer::rtx-next-seqnum
gboolean
rw
-Use pipeline clock
-Use the pipeline clock to set the NTP time in the RTCP SR messages.
-FALSE
+RTX next seqnum
+Estimate when the next packet should arrive and schedule a retransmission request for it.
+TRUE
-GstRtpSession::rtcp-min-interval
-guint64
-
+GstRtpJitterBuffer::max-dropout-time
+guint
+<= G_MAXINT
rw
-Minimum RTCP interval
-Minimum interval between Regular RTCP packet (in ns).
-5000000000
+Max dropout time
+The maximum time (milliseconds) of missing packets tolerated.
+60000
-GstV4l2Sink::device
-gchar*
+GstRtpJitterBuffer::max-misorder-time
+guint
rw
-Device
-Device location.
-"/dev/video1"
+Max misorder time
+The maximum time (milliseconds) of misordered packets tolerated.
+2000
-GstV4l2Sink::device-fd
+GstRtpJitterBuffer::max-rtcp-rtp-time-diff
gint
>= G_MAXULONG
-r
-File descriptor
-File descriptor of the device.
--1
+rw
+Max RTCP RTP Time Diff
+Maximum amount of time in ms that the RTP time in RTCP SRs is allowed to be ahead (-1 disabled).
+1000
-GstV4l2Sink::device-name
-gchar*
+GstRtpJitterBuffer::rfc7273-sync
+gboolean
-r
-Device name
-Name of the device.
-NULL
+rw
+Sync on RFC7273 clock
+Synchronize received streams to the RFC7273 clock (requires clock and offset to be provided).
+FALSE
-GstV4l2Sink::flags
-GstV4l2DeviceTypeFlags
-
-r
-Flags
-Device type flags.
-
+GstRtpJitterBuffer::rtx-deadline
+gint
+>= G_MAXULONG
+rw
+RTX Deadline (ms)
+The deadline for a valid RTX request in milliseconds. (-1 automatic).
+-1
-GstV4l2Sink::overlay-height
+GstRtpJitterBuffer::rtx-stats-timeout
guint
rw
-Overlay height
-The height of the video overlay; default is equal to negotiated image height.
-0
+RTX Statistics Timeout
+The time to wait for a retransmitted packet after it has been considered lost in order to collect statistics (ms).
+1000
-GstV4l2Sink::overlay-left
-gint
+GstRtpJitterBuffer::faststart-min-packets
+guint
rw
-Overlay left
-The leftmost (x) coordinate of the video overlay; top left corner of screen is 0,0.
+Faststart minimum packets
+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).
0
-GstV4l2Sink::overlay-top
-gint
+GstRtpJitterBuffer::max-ts-offset-adjustment
+guint64
rw
-Overlay top
-The topmost (y) coordinate of the video overlay; top left corner of screen is 0,0.
+Max Timestamp Offset Adjustment
+The maximum number of nanoseconds per frame that time stamp offsets may be adjusted (0 = no limit).
0
-GstV4l2Sink::overlay-width
+GstRtpSession::bandwidth
+gdouble
+>= 0
+rw
+Bandwidth
+The bandwidth of the session in bytes per second (0 for auto-discover).
+0
+
+
+
+GstRtpSession::internal-session
+RTPSession*
+
+r
+Internal Session
+The internal RTPSession object.
+
+
+
+
+GstRtpSession::num-active-sources
guint
-rw
-Overlay width
-The width of the video overlay; default is equal to negotiated image width.
+r
+Num Active Sources
+The number of active sources in the session.
0
-GstV4l2Sink::queue-size
+GstRtpSession::num-sources
guint
-[1,16]
+
+r
+Num Sources
+The number of sources in the session.
+0
+
+
+
+GstRtpSession::rtcp-fraction
+gdouble
+>= 0
rw
-Queue size
-Number of buffers to be enqueud in the driver in streaming mode.
-12
+RTCP Fraction
+The RTCP bandwidth of the session in bytes per second (or as a real fraction of the RTP bandwidth if < 1.0).
+0.05
-GstV4l2Sink::brightness
-gint
+GstRtpSession::sdes
+GstStructure*
rw
-Brightness
-Picture brightness, or more precisely, the black level.
-0
+SDES
+The SDES items of this session.
+
-GstV4l2Sink::contrast
+GstRtpSession::rtcp-rr-bandwidth
gint
-
+>= G_MAXULONG
rw
-Contrast
-Picture contrast or luma gain.
-0
+RTCP RR bandwidth
+The RTCP bandwidth used for receivers in bytes per second (-1 = default).
+-1
-GstV4l2Sink::hue
+GstRtpSession::rtcp-rs-bandwidth
gint
+>= G_MAXULONG
+rw
+RTCP RS bandwidth
+The RTCP bandwidth used for senders in bytes per second (-1 = default).
+-1
+
+
+
+GstRtpSession::use-pipeline-clock
+gboolean
rw
-Hue
-Hue or color balance.
-0
+Use pipeline clock
+Use the pipeline running-time to set the NTP time in the RTCP SR messages (DEPRECATED: Use ntp-time-source property).
+FALSE
-GstV4l2Sink::saturation
-gint
+GstRtpSession::rtcp-min-interval
+guint64
rw
-Saturation
-Picture color saturation or chroma gain.
-0
+Minimum RTCP interval
+Minimum interval between Regular RTCP packet (in ns).
+5000000000
-GstV4l2Sink::crop-height
+GstRtpSession::probation
guint
rw
-Crop height
-The height of the video crop; default is equal to negotiated image height.
-0
+Number of probations
+Consecutive packet sequence numbers to accept the source.
+2
-GstV4l2Sink::crop-left
-gint
+GstRtpSession::stats
+GstStructure*
-rw
-Crop left
-The leftmost (x) coordinate of the video crop; top left corner of image is 0,0.
-0
+r
+Statistics
+Various statistics.
+
-GstV4l2Sink::crop-top
-gint
+GstRtpSession::rtp-profile
+GstRTPProfile
rw
-Crop top
-The topmost (y) coordinate of the video crop; top left corner of image is 0,0.
-0
+RTP Profile
+RTP profile to use.
+GST_RTP_PROFILE_AVP
-GstV4l2Sink::crop-width
-guint
+GstRtpSession::ntp-time-source
+GstRtpNtpTimeSource
rw
-Crop width
-The width of the video crop; default is equal to negotiated image width.
-0
+NTP Time Source
+NTP time source for RTCP packets.
+NTP time based on realtime clock
-GstV4l2Sink::min-queued-bufs
+GstRtpSession::max-dropout-time
guint
-<= 16
+
rw
-Minimum queued bufs
-Minimum number of queued bufs; v4l2sink won't dqbuf if the driver doesn't have more than this number (which normally you shouldn't change).
-1
+Max dropout time
+The maximum time (milliseconds) of missing packets tolerated.
+60000
-GstV4l2Sink::io-mode
-GstV4l2IOMode
+GstRtpSession::max-misorder-time
+guint
rw
-IO mode
-I/O mode.
-GST_V4L2_IO_AUTO
+Max misorder time
+The maximum time (milliseconds) of misordered packets tolerated.
+2000
-GstV4l2Sink::norm
-V4L2_TV_norms
+GstRtpSession::rtcp-sync-send-time
+gboolean
rw
-TV norm
-video standard.
-none
+RTCP Sync Send Time
+Use send time or capture time for RTCP sync (TRUE = send time, FALSE = capture time).
+TRUE
-GstShapeWipe::border
-gfloat
-[0,1]
+GstRtpRtxSend::max-size-time
+guint
+
rw
-Border
-Border of the mask.
+Max Size Time
+Amount of ms to queue (0 = unlimited).
0
-GstShapeWipe::position
-gfloat
-[0,1]
+GstRtpRtxSend::max-size-packets
+guint
+<= 32767
rw
-Position
-Position of the mask.
-0
+Max Size Packets
+Amount of packets to queue (0 = unlimited).
+100
-GstFlvMux::is-live
-gboolean
+GstRtpRtxSend::num-rtx-requests
+guint
-rw
-Is Live
-The stream is live and does not need an index.
-FALSE
+r
+Num RTX Requests
+Number of retransmission events received.
+0
-GstFlvMux::streamable
-gboolean
+GstRtpRtxSend::num-rtx-packets
+guint
-rw
-streamable
-If set to true, the output should be as if it is to be streamed and hence no indexes written or duration written.
-FALSE
+r
+Num RTX Packets
+ Number of retransmission packets sent.
+0
-GstCapsSetter::caps
-GstCaps*
+GstRtpRtxSend::payload-type-map
+GstStructure*
rw
-Merge caps
-Merge these caps (thereby overwriting) in the stream.
+Payload Type Map
+Map of original payload types to their retransmission payload types.
-GstCapsSetter::join
-gboolean
+GstRtpRtxSend::ssrc-map
+GstStructure*
-rw
-Join
-Match incoming caps' mime-type to mime-type of provided caps.
-TRUE
+w
+SSRC Map
+Map of SSRCs to their retransmission SSRCs for SSRC-multiplexed mode (default = random).
+
-GstCapsSetter::replace
-gboolean
+GstRtpRtxReceive::num-rtx-requests
+guint
-rw
-Replace
-Drop fields of incoming caps.
-FALSE
+r
+Num RTX Requests
+Number of retransmission events received.
+0
-GstOss4Mixer::device
-gchar*
+GstRtpRtxReceive::num-rtx-packets
+guint
-rw
-Device
-OSS mixer device (e.g. /dev/oss/hdaudio0/mix0 or /dev/mixerN) (NULL = use first mixer device found).
-NULL
+r
+Num RTX Packets
+ Number of retransmission packets received.
+0
-GstOss4Mixer::device-name
-gchar*
+GstRtpRtxReceive::num-rtx-assoc-packets
+guint
r
-Device name
-Human-readable name of the sound device.
-NULL
+Num RTX Associated Packets
+Number of retransmission packets correctly associated with retransmission requests.
+0
-GstOss4Source::device
-gchar*
+GstRtpRtxReceive::payload-type-map
+GstStructure*
rw
-Device
-OSS4 device (e.g. /dev/oss/hdaudio0/pcm0 or /dev/dspN) (NULL = use first available device).
-NULL
+Payload Type Map
+Map of original payload types to their retransmission payload types.
+
-GstOss4Source::device-name
+GstV4l2Sink::device
gchar*
+rw
+Device
+Device location.
+"/dev/video1"
+
+
+
+GstV4l2Sink::device-fd
+gint
+>= G_MAXULONG
r
-Device name
-Human-readable name of the sound device.
-NULL
+File descriptor
+File descriptor of the device.
+-1
-GstOss4Sink::device
+GstV4l2Sink::device-name
gchar*
-rw
-Device
-OSS4 device (e.g. /dev/oss/hdaudio0/pcm0 or /dev/dspN) (NULL = use first available playback device).
+r
+Device name
+Name of the device.
NULL
-GstOss4Sink::device-name
-gchar*
+GstV4l2Sink::flags
+GstV4l2DeviceTypeFlags
r
-Device name
-Human-readable name of the sound device.
-NULL
+Flags
+Device type flags.
+
-GstOss4Sink::mute
-gboolean
+GstV4l2Sink::overlay-height
+guint
rw
-Mute
-Mute state of this stream.
-FALSE
+Overlay height
+The height of the video overlay; default is equal to negotiated image height.
+0
-GstOss4Sink::volume
-gdouble
-[0,10]
-rw
-Volume
-Linear volume of this stream, 1.0=100%.
-1
-
-
-
-GstRtpTheoraPay::config-interval
-guint
-<= 3600
+GstV4l2Sink::overlay-left
+gint
+
rw
-Config Send Interval
-Send Config Insertion Interval in seconds (configuration headers will be multiplexed in the data stream when detected.) (0 = disabled).
+Overlay left
+The leftmost (x) coordinate of the video overlay; top left corner of screen is 0,0.
0
-GstVideoMixer2::background
-GstVideoMixer2Background
+GstV4l2Sink::overlay-top
+gint
rw
-Background
-Background type.
-Checker pattern
+Overlay top
+The topmost (y) coordinate of the video overlay; top left corner of screen is 0,0.
+0
-GstRtpJ2KPay::buffer-list
-gboolean
+GstV4l2Sink::overlay-width
+guint
rw
-Buffer List
-Use Buffer Lists.
-TRUE
+Overlay width
+The width of the video overlay; default is equal to negotiated image width.
+0
-GstRtpJ2KDepay::buffer-list
-gboolean
+GstV4l2Sink::brightness
+gint
rw
-Buffer List
-Use Buffer Lists.
-TRUE
+Brightness
+Picture brightness, or more precisely, the black level.
+0
-GstJackAudioSrc::client
-JackClient*
+GstV4l2Sink::contrast
+gint
rw
-JackClient
-Handle for jack client.
-
+Contrast
+Picture contrast or luma gain.
+0
-GstJackAudioSrc::connect
-GstJackConnect
+GstV4l2Sink::hue
+gint
rw
-Connect
-Specify how the input ports will be connected.
-Automatically connect ports to physical ports
+Hue
+Hue or color balance.
+0
-GstJackAudioSrc::server
-gchar*
+GstV4l2Sink::saturation
+gint
rw
-Server
-The Jack server to connect to (NULL = default).
-NULL
+Saturation
+Picture color saturation or chroma gain.
+0
-GstJackAudioSrc::client-name
-gchar*
+GstV4l2Sink::crop-height
+guint
rw
-Client name
-The client name of the Jack instance (NULL = default).
-NULL
+Crop height
+The height of the video crop; default is equal to negotiated image height.
+0
-GstJackAudioSrc::transport
-GstJackTransport
+GstV4l2Sink::crop-left
+gint
rw
-Transport mode
-Jack transport behaviour of the client.
-
+Crop left
+The leftmost (x) coordinate of the video crop; top left corner of image is 0,0.
+0
-GstJackAudioSink::client
-JackClient*
+GstV4l2Sink::crop-top
+gint
rw
-JackClient
-Handle for jack client.
-
+Crop top
+The topmost (y) coordinate of the video crop; top left corner of image is 0,0.
+0
-GstJackAudioSink::connect
-GstJackConnect
+GstV4l2Sink::crop-width
+guint
rw
-Connect
-Specify how the output ports will be connected.
-Automatically connect ports to physical ports
+Crop width
+The width of the video crop; default is equal to negotiated image width.
+0
-GstJackAudioSink::server
-gchar*
+GstV4l2Sink::io-mode
+GstV4l2IOMode
rw
-Server
-The Jack server to connect to (NULL = default).
-NULL
+IO mode
+I/O mode.
+GST_V4L2_IO_AUTO
-GstJackAudioSink::client-name
-gchar*
+GstV4l2Sink::norm
+V4L2_TV_norms
rw
-Client name
-The client name of the Jack instance (NULL = default).
-NULL
+TV norm
+video standard.
+none
-GstJackAudioSink::transport
-GstJackTransport
+GstV4l2Sink::extra-controls
+GstStructure*
rw
-Transport mode
-Jack transport behaviour of the client.
+Extra Controls
+Extra v4l2 controls (CIDs) for the device.
-GstFlacParse::check-frame-checksums
+GstV4l2Sink::force-aspect-ratio
gboolean
rw
-Check Frame Checksums
-Check the overall checksums of every frame.
-FALSE
+Force aspect ratio
+When enabled, the pixel aspect ratio will be enforced.
+TRUE
-GstV4l2Radio::device
+GstV4l2Sink::pixel-aspect-ratio
gchar*
rw
-Radio device location
-Video4Linux2 radio device location.
-"/dev/radio0"
+Pixel Aspect Ratio
+Overwrite the pixel aspect ratio of the device.
+"1/1"
-GstV4l2Radio::frequency
-gint
-[87500000,108000000]
+GstShapeWipe::border
+gfloat
+[0,1]
rw
-Station frequency
-Station frequency in Hz.
-100000000
+Border
+Border of the mask.
+0
-GstQTMux::dts-method
-GstQTMuxDtsMethods
-
-rwx
-dts-method
-Method to determine DTS time.
-reorder
+GstShapeWipe::position
+gfloat
+[0,1]
+rw
+Position
+Position of the mask.
+0
-GstQTMux::faststart
+GstFlvMux::streamable
gboolean
rw
-Format file to faststart
-If the file should be formatted for faststart (headers first).
+streamable
+If set to true, the output should be as if it is to be streamed and hence no indexes written or duration written.
FALSE
-GstQTMux::faststart-file
+GstFlvMux::metadatacreator
gchar*
-rwx
-File to use for storing buffers
-File that will be used temporarily to store data from the stream when creating a faststart file. If null a filepath will be created automatically.
+rw
+metadatacreator
+The value of metadatacreator in the meta packet.
NULL
-GstQTMux::fragment-duration
-guint
-
-rwx
-Fragment duration
-Fragment durations in ms (produce a fragmented file if > 0).
-0
-
-
-
-GstQTMux::moov-recovery-file
+GstFlvMux::encoder
gchar*
-rwx
-File to store data for posterior moov atom recovery
-File to be used to store data for moov atom making movie file recovery possible in case of a crash during muxing. Null for disabled. (Experimental).
+rw
+encoder
+The value of encoder in the meta packet.
NULL
-GstQTMux::movie-timescale
-guint
->= 1
-rwx
-Movie timescale
-Timescale to use in the movie (units per second).
-1000
+GstCapsSetter::caps
+GstCaps*
+
+rw
+Merge caps
+Merge these caps (thereby overwriting) in the stream.
+
-GstQTMux::presentation-time
+GstCapsSetter::join
gboolean
-rwx
-Include presentation-time info
-Calculate and include presentation/composition time (in addition to decoding time).
+rw
+Join
+Match incoming caps' mime-type to mime-type of provided caps.
TRUE
-GstQTMux::streamable
+GstCapsSetter::replace
gboolean
-rwx
-Streamable
-If set to true, the output should be as if it is to be streamed and hence no indexes written or duration written.
+rw
+Replace
+Drop fields of incoming caps.
FALSE
-GstQTMux::trak-timescale
-guint
+GstOss4Source::device
+gchar*
-rwx
-Track timescale
-Timescale to use for the tracks (units per second, 0 is automatic).
-0
+rw
+Device
+OSS4 device (e.g. /dev/oss/hdaudio0/pcm0 or /dev/dspN) (NULL = use first available device).
+NULL
-GstQTMoovRecover::broken-input
+GstOss4Source::device-name
gchar*
-rw
-Path to broken input file
-Path to broken input file. (If qtmux was on faststart mode, this file is the faststart file).
+r
+Device name
+Human-readable name of the sound device.
NULL
-GstQTMoovRecover::faststart-mode
-gboolean
+GstOss4Sink::device
+gchar*
rw
-If the broken input is from faststart mode
-If the broken input is from faststart mode.
-FALSE
-
+Device
+OSS4 device (e.g. /dev/oss/hdaudio0/pcm0 or /dev/dspN) (NULL = use first available playback device).
+NULL
+
-GstQTMoovRecover::fixed-output
+GstOss4Sink::device-name
gchar*
+r
+Device name
+Human-readable name of the sound device.
+NULL
+
+
+
+GstOss4Sink::mute
+gboolean
+
rw
-Path to write the fixed file
-Path to write the fixed file to (used as output).
+Mute
+Mute state of this stream.
+FALSE
+
+
+
+GstOss4Sink::volume
+gdouble
+[0,10]
+rw
+Volume
+Linear volume of this stream, 1.0=100%.
+1
+
+
+
+GstRtpTheoraPay::config-interval
+guint
+<= 3600
+rw
+Config Send Interval
+Send Config Insertion Interval in seconds (configuration headers will be multiplexed in the data stream when detected.) (0 = disabled).
+0
+
+
+
+GstVideoMixer2::background
+GstVideoMixer2Background
+
+rw
+Background
+Background type.
+Checker pattern
+
+
+
+GstJackAudioSrc::client
+JackClient*
+
+rw
+JackClient
+Handle for jack client.
+
+
+
+
+GstJackAudioSrc::connect
+GstJackConnect
+
+rw
+Connect
+Specify how the input ports will be connected.
+Automatically connect ports to physical ports
+
+
+
+GstJackAudioSrc::server
+gchar*
+
+rw
+Server
+The Jack server to connect to (NULL = default).
NULL
-GstQTMoovRecover::recovery-input
+GstJackAudioSrc::client-name
gchar*
rw
-Path to recovery file
-Path to recovery file (used as input).
+Client name
+The client name of the Jack instance (NULL = default).
NULL
-GstMP4Mux::dts-method
+GstJackAudioSrc::transport
+GstJackTransport
+
+rw
+Transport mode
+Jack transport behaviour of the client.
+
+
+
+
+GstJackAudioSrc::port-pattern
+gchar*
+
+rw
+port pattern
+A pattern to select which ports to connect to (NULL = first physical ports).
+NULL
+
+
+
+GstJackAudioSink::client
+JackClient*
+
+rw
+JackClient
+Handle for jack client.
+
+
+
+
+GstJackAudioSink::connect
+GstJackConnect
+
+rw
+Connect
+Specify how the output ports will be connected.
+Automatically connect ports to physical ports
+
+
+
+GstJackAudioSink::server
+gchar*
+
+rw
+Server
+The Jack server to connect to (NULL = default).
+NULL
+
+
+
+GstJackAudioSink::client-name
+gchar*
+
+rw
+Client name
+The client name of the Jack instance (NULL = default).
+NULL
+
+
+
+GstJackAudioSink::transport
+GstJackTransport
+
+rw
+Transport mode
+Jack transport behaviour of the client.
+
+
+
+
+GstJackAudioSink::port-pattern
+gchar*
+
+rw
+port pattern
+A pattern to select which ports to connect to (NULL = first physical ports).
+NULL
+
+
+
+GstFlacParse::check-frame-checksums
+gboolean
+
+rw
+Check Frame Checksums
+Check the overall checksums of every frame.
+FALSE
+
+
+
+GstV4l2Radio::device
+gchar*
+
+rw
+Radio device location
+Video4Linux2 radio device location.
+"/dev/radio0"
+
+
+
+GstV4l2Radio::frequency
+gint
+[87500000,108000000]
+rw
+Station frequency
+Station frequency in Hz.
+100000000
+
+
+
+GstQTMux::dts-method
GstQTMuxDtsMethods
rwx
dts-method
-Method to determine DTS time.
+Method to determine DTS time (DEPRECATED).
reorder
-GstMP4Mux::faststart
+GstQTMux::faststart
gboolean
rw
@@ -21769,7 +21759,7 @@
-GstMP4Mux::faststart-file
+GstQTMux::faststart-file
gchar*
rwx
@@ -21779,7 +21769,7 @@
-GstMP4Mux::fragment-duration
+GstQTMux::fragment-duration
guint
rwx
@@ -21789,7 +21779,7 @@
-GstMP4Mux::moov-recovery-file
+GstQTMux::moov-recovery-file
gchar*
rwx
@@ -21799,17 +21789,17 @@
-GstMP4Mux::movie-timescale
+GstQTMux::movie-timescale
guint
->= 1
+
rwx
Movie timescale
-Timescale to use in the movie (units per second).
-1000
+Timescale to use in the movie (units per second, 0 == default).
+0
-GstMP4Mux::presentation-time
+GstQTMux::presentation-time
gboolean
rwx
@@ -21819,17 +21809,17 @@
-GstMP4Mux::streamable
+GstQTMux::streamable
gboolean
rwx
Streamable
-If set to true, the output should be as if it is to be streamed and hence no indexes written or duration written.
+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).
FALSE
-GstMP4Mux::trak-timescale
+GstQTMux::trak-timescale
guint
rwx
@@ -21839,67 +21829,197 @@
-GstMJ2Mux::dts-method
-GstQTMuxDtsMethods
-
+GstQTMux::reserved-bytes-per-sec
+guint
+<= 10000
rwx
-dts-method
-Method to determine DTS time.
-reorder
+Reserved MOOV bytes per second, per track
+Multiplier for converting reserved-max-duration into bytes of header to reserve, per second, per track.
+550
-GstMJ2Mux::faststart
-gboolean
+GstQTMux::reserved-duration-remaining
+guint64
+
+r
+Report the approximate amount of remaining recording space (ns)
+Reports the approximate amount of remaining moov header space reserved using reserved-max-duration.
+0
+
+
+
+GstQTMux::reserved-max-duration
+guint64
rw
-Format file to faststart
-If the file should be formatted for faststart (headers first).
-FALSE
+Reserved maximum file duration (ns)
+When set to a value > 0, reserves space for index tables at the beginning of the file.
+18446744073709551615
-GstMJ2Mux::faststart-file
-gchar*
+GstQTMux::reserved-moov-update-period
+guint64
-rwx
-File to use for storing buffers
-File that will be used temporarily to store data from the stream when creating a faststart file. If null a filepath will be created automatically.
-NULL
+rw
+Interval at which to update index tables (ns)
+When used with reserved-max-duration, periodically updates the index tables with information muxed so far.
+18446744073709551615
-GstMJ2Mux::fragment-duration
-guint
+GstQTMux::interleave-bytes
+guint64
-rwx
-Fragment duration
-Fragment durations in ms (produce a fragmented file if > 0).
+rw
+Interleave (bytes)
+Interleave between streams in bytes.
0
-GstMJ2Mux::moov-recovery-file
-gchar*
+GstQTMux::interleave-time
+guint64
-rwx
-File to store data for posterior moov atom recovery
-File to be used to store data for moov atom making movie file recovery possible in case of a crash during muxing. Null for disabled. (Experimental).
-NULL
+rw
+Interleave (time)
+Interleave between streams in nanoseconds.
+250000000
-GstMJ2Mux::movie-timescale
+GstQTMux::max-raw-audio-drift
+guint64
+
+rw
+Max Raw Audio Drift
+Maximum allowed drift of raw audio samples vs. timestamps in nanoseconds.
+40000000
+
+
+
+GstQTMux::reserved-prefill
+gboolean
+
+rwx
+Reserved Prefill Samples Table
+Prefill samples table of reserved duration.
+FALSE
+
+
+
+GstQTMux::start-gap-threshold
+guint64
+
+rw
+Start Gap Threshold
+Threshold for creating an edit list for gaps at the start in nanoseconds.
+0
+
+
+
+GstQTMoovRecover::broken-input
+gchar*
+
+rw
+Path to broken input file
+Path to broken input file. (If qtmux was on faststart mode, this file is the faststart file).
+NULL
+
+
+
+GstQTMoovRecover::faststart-mode
+gboolean
+
+rw
+If the broken input is from faststart mode
+If the broken input is from faststart mode.
+FALSE
+
+
+
+GstQTMoovRecover::fixed-output
+gchar*
+
+rw
+Path to write the fixed file
+Path to write the fixed file to (used as output).
+NULL
+
+
+
+GstQTMoovRecover::recovery-input
+gchar*
+
+rw
+Path to recovery file
+Path to recovery file (used as input).
+NULL
+
+
+
+GstMP4Mux::dts-method
+GstQTMuxDtsMethods
+
+rwx
+dts-method
+Method to determine DTS time (DEPRECATED).
+reorder
+
+
+
+GstMP4Mux::faststart
+gboolean
+
+rw
+Format file to faststart
+If the file should be formatted for faststart (headers first).
+FALSE
+
+
+
+GstMP4Mux::faststart-file
+gchar*
+
+rwx
+File to use for storing buffers
+File that will be used temporarily to store data from the stream when creating a faststart file. If null a filepath will be created automatically.
+NULL
+
+
+
+GstMP4Mux::fragment-duration
guint
->= 1
+
+rwx
+Fragment duration
+Fragment durations in ms (produce a fragmented file if > 0).
+0
+
+
+
+GstMP4Mux::moov-recovery-file
+gchar*
+
+rwx
+File to store data for posterior moov atom recovery
+File to be used to store data for moov atom making movie file recovery possible in case of a crash during muxing. Null for disabled. (Experimental).
+NULL
+
+
+
+GstMP4Mux::movie-timescale
+guint
+
rwx
Movie timescale
-Timescale to use in the movie (units per second).
-1000
+Timescale to use in the movie (units per second, 0 == default).
+0
-GstMJ2Mux::presentation-time
+GstMP4Mux::presentation-time
gboolean
rwx
@@ -21909,752 +22029,3022 @@
-GstMJ2Mux::streamable
-gboolean
-
-rwx
-Streamable
-If set to true, the output should be as if it is to be streamed and hence no indexes written or duration written.
-FALSE
+GstMP4Mux::streamable
+gboolean
+
+rwx
+Streamable
+If set to true, the output should be as if it is to be streamed and hence no indexes written or duration written. (DEPRECATED, only valid for fragmented MP4).
+FALSE
+
+
+
+GstMP4Mux::trak-timescale
+guint
+
+rwx
+Track timescale
+Timescale to use for the tracks (units per second, 0 is automatic).
+0
+
+
+
+GstMP4Mux::reserved-bytes-per-sec
+guint
+<= 10000
+rwx
+Reserved MOOV bytes per second, per track
+Multiplier for converting reserved-max-duration into bytes of header to reserve, per second, per track.
+550
+
+
+
+GstMP4Mux::reserved-duration-remaining
+guint64
+
+r
+Report the approximate amount of remaining recording space (ns)
+Reports the approximate amount of remaining moov header space reserved using reserved-max-duration.
+0
+
+
+
+GstMP4Mux::reserved-max-duration
+guint64
+
+rw
+Reserved maximum file duration (ns)
+When set to a value > 0, reserves space for index tables at the beginning of the file.
+18446744073709551615
+
+
+
+GstMP4Mux::reserved-moov-update-period
+guint64
+
+rw
+Interval at which to update index tables (ns)
+When used with reserved-max-duration, periodically updates the index tables with information muxed so far.
+18446744073709551615
+
+
+
+GstMP4Mux::interleave-bytes
+guint64
+
+rw
+Interleave (bytes)
+Interleave between streams in bytes.
+0
+
+
+
+GstMP4Mux::interleave-time
+guint64
+
+rw
+Interleave (time)
+Interleave between streams in nanoseconds.
+250000000
+
+
+
+GstMP4Mux::max-raw-audio-drift
+guint64
+
+rw
+Max Raw Audio Drift
+Maximum allowed drift of raw audio samples vs. timestamps in nanoseconds.
+40000000
+
+
+
+GstMP4Mux::reserved-prefill
+gboolean
+
+rwx
+Reserved Prefill Samples Table
+Prefill samples table of reserved duration.
+FALSE
+
+
+
+GstMP4Mux::start-gap-threshold
+guint64
+
+rw
+Start Gap Threshold
+Threshold for creating an edit list for gaps at the start in nanoseconds.
+0
+
+
+
+GstMJ2Mux::dts-method
+GstQTMuxDtsMethods
+
+rwx
+dts-method
+Method to determine DTS time (DEPRECATED).
+reorder
+
+
+
+GstMJ2Mux::faststart
+gboolean
+
+rw
+Format file to faststart
+If the file should be formatted for faststart (headers first).
+FALSE
+
+
+
+GstMJ2Mux::faststart-file
+gchar*
+
+rwx
+File to use for storing buffers
+File that will be used temporarily to store data from the stream when creating a faststart file. If null a filepath will be created automatically.
+NULL
+
+
+
+GstMJ2Mux::fragment-duration
+guint
+
+rwx
+Fragment duration
+Fragment durations in ms (produce a fragmented file if > 0).
+0
+
+
+
+GstMJ2Mux::moov-recovery-file
+gchar*
+
+rwx
+File to store data for posterior moov atom recovery
+File to be used to store data for moov atom making movie file recovery possible in case of a crash during muxing. Null for disabled. (Experimental).
+NULL
+
+
+
+GstMJ2Mux::movie-timescale
+guint
+
+rwx
+Movie timescale
+Timescale to use in the movie (units per second, 0 == default).
+0
+
+
+
+GstMJ2Mux::presentation-time
+gboolean
+
+rwx
+Include presentation-time info
+Calculate and include presentation/composition time (in addition to decoding time).
+TRUE
+
+
+
+GstMJ2Mux::streamable
+gboolean
+
+rwx
+Streamable
+If set to true, the output should be as if it is to be streamed and hence no indexes written or duration written. (DEPRECATED, only valid for fragmented MP4).
+FALSE
+
+
+
+GstMJ2Mux::trak-timescale
+guint
+
+rwx
+Track timescale
+Timescale to use for the tracks (units per second, 0 is automatic).
+0
+
+
+
+GstMJ2Mux::reserved-bytes-per-sec
+guint
+<= 10000
+rwx
+Reserved MOOV bytes per second, per track
+Multiplier for converting reserved-max-duration into bytes of header to reserve, per second, per track.
+550
+
+
+
+GstMJ2Mux::reserved-duration-remaining
+guint64
+
+r
+Report the approximate amount of remaining recording space (ns)
+Reports the approximate amount of remaining moov header space reserved using reserved-max-duration.
+0
+
+
+
+GstMJ2Mux::reserved-max-duration
+guint64
+
+rw
+Reserved maximum file duration (ns)
+When set to a value > 0, reserves space for index tables at the beginning of the file.
+18446744073709551615
+
+
+
+GstMJ2Mux::reserved-moov-update-period
+guint64
+
+rw
+Interval at which to update index tables (ns)
+When used with reserved-max-duration, periodically updates the index tables with information muxed so far.
+18446744073709551615
+
+
+
+GstMJ2Mux::interleave-bytes
+guint64
+
+rw
+Interleave (bytes)
+Interleave between streams in bytes.
+0
+
+
+
+GstMJ2Mux::interleave-time
+guint64
+
+rw
+Interleave (time)
+Interleave between streams in nanoseconds.
+250000000
+
+
+
+GstMJ2Mux::max-raw-audio-drift
+guint64
+
+rw
+Max Raw Audio Drift
+Maximum allowed drift of raw audio samples vs. timestamps in nanoseconds.
+40000000
+
+
+
+GstMJ2Mux::reserved-prefill
+gboolean
+
+rwx
+Reserved Prefill Samples Table
+Prefill samples table of reserved duration.
+FALSE
+
+
+
+GstMJ2Mux::start-gap-threshold
+guint64
+
+rw
+Start Gap Threshold
+Threshold for creating an edit list for gaps at the start in nanoseconds.
+0
+
+
+
+GstISMLMux::dts-method
+GstQTMuxDtsMethods
+
+rwx
+dts-method
+Method to determine DTS time (DEPRECATED).
+reorder
+
+
+
+GstISMLMux::faststart
+gboolean
+
+rw
+Format file to faststart
+If the file should be formatted for faststart (headers first).
+FALSE
+
+
+
+GstISMLMux::faststart-file
+gchar*
+
+rwx
+File to use for storing buffers
+File that will be used temporarily to store data from the stream when creating a faststart file. If null a filepath will be created automatically.
+NULL
+
+
+
+GstISMLMux::fragment-duration
+guint
+
+rwx
+Fragment duration
+Fragment durations in ms (produce a fragmented file if > 0).
+2000
+
+
+
+GstISMLMux::moov-recovery-file
+gchar*
+
+rwx
+File to store data for posterior moov atom recovery
+File to be used to store data for moov atom making movie file recovery possible in case of a crash during muxing. Null for disabled. (Experimental).
+NULL
+
+
+
+GstISMLMux::movie-timescale
+guint
+
+rwx
+Movie timescale
+Timescale to use in the movie (units per second, 0 == default).
+0
+
+
+
+GstISMLMux::presentation-time
+gboolean
+
+rwx
+Include presentation-time info
+Calculate and include presentation/composition time (in addition to decoding time).
+TRUE
+
+
+
+GstISMLMux::streamable
+gboolean
+
+rwx
+Streamable
+If set to true, the output should be as if it is to be streamed and hence no indexes written or duration written.
+TRUE
+
+
+
+GstISMLMux::trak-timescale
+guint
+
+rwx
+Track timescale
+Timescale to use for the tracks (units per second, 0 is automatic).
+0
+
+
+
+GstISMLMux::reserved-bytes-per-sec
+guint
+<= 10000
+rwx
+Reserved MOOV bytes per second, per track
+Multiplier for converting reserved-max-duration into bytes of header to reserve, per second, per track.
+550
+
+
+
+GstISMLMux::reserved-duration-remaining
+guint64
+
+r
+Report the approximate amount of remaining recording space (ns)
+Reports the approximate amount of remaining moov header space reserved using reserved-max-duration.
+0
+
+
+
+GstISMLMux::reserved-max-duration
+guint64
+
+rw
+Reserved maximum file duration (ns)
+When set to a value > 0, reserves space for index tables at the beginning of the file.
+18446744073709551615
+
+
+
+GstISMLMux::reserved-moov-update-period
+guint64
+
+rw
+Interval at which to update index tables (ns)
+When used with reserved-max-duration, periodically updates the index tables with information muxed so far.
+18446744073709551615
+
+
+
+GstISMLMux::interleave-bytes
+guint64
+
+rw
+Interleave (bytes)
+Interleave between streams in bytes.
+0
+
+
+
+GstISMLMux::interleave-time
+guint64
+
+rw
+Interleave (time)
+Interleave between streams in nanoseconds.
+250000000
+
+
+
+GstISMLMux::max-raw-audio-drift
+guint64
+
+rw
+Max Raw Audio Drift
+Maximum allowed drift of raw audio samples vs. timestamps in nanoseconds.
+40000000
+
+
+
+GstISMLMux::reserved-prefill
+gboolean
+
+rwx
+Reserved Prefill Samples Table
+Prefill samples table of reserved duration.
+FALSE
+
+
+
+GstISMLMux::start-gap-threshold
+guint64
+
+rw
+Start Gap Threshold
+Threshold for creating an edit list for gaps at the start in nanoseconds.
+0
+
+
+
+Gst3GPPMux::dts-method
+GstQTMuxDtsMethods
+
+rwx
+dts-method
+Method to determine DTS time (DEPRECATED).
+reorder
+
+
+
+Gst3GPPMux::faststart
+gboolean
+
+rw
+Format file to faststart
+If the file should be formatted for faststart (headers first).
+FALSE
+
+
+
+Gst3GPPMux::faststart-file
+gchar*
+
+rwx
+File to use for storing buffers
+File that will be used temporarily to store data from the stream when creating a faststart file. If null a filepath will be created automatically.
+NULL
+
+
+
+Gst3GPPMux::fragment-duration
+guint
+
+rwx
+Fragment duration
+Fragment durations in ms (produce a fragmented file if > 0).
+0
+
+
+
+Gst3GPPMux::moov-recovery-file
+gchar*
+
+rwx
+File to store data for posterior moov atom recovery
+File to be used to store data for moov atom making movie file recovery possible in case of a crash during muxing. Null for disabled. (Experimental).
+NULL
+
+
+
+Gst3GPPMux::movie-timescale
+guint
+
+rwx
+Movie timescale
+Timescale to use in the movie (units per second, 0 == default).
+0
+
+
+
+Gst3GPPMux::presentation-time
+gboolean
+
+rwx
+Include presentation-time info
+Calculate and include presentation/composition time (in addition to decoding time).
+TRUE
+
+
+
+Gst3GPPMux::streamable
+gboolean
+
+rwx
+Streamable
+If set to true, the output should be as if it is to be streamed and hence no indexes written or duration written. (DEPRECATED, only valid for fragmented MP4).
+FALSE
+
+
+
+Gst3GPPMux::trak-timescale
+guint
+
+rwx
+Track timescale
+Timescale to use for the tracks (units per second, 0 is automatic).
+0
+
+
+
+Gst3GPPMux::reserved-bytes-per-sec
+guint
+<= 10000
+rwx
+Reserved MOOV bytes per second, per track
+Multiplier for converting reserved-max-duration into bytes of header to reserve, per second, per track.
+550
+
+
+
+Gst3GPPMux::reserved-duration-remaining
+guint64
+
+r
+Report the approximate amount of remaining recording space (ns)
+Reports the approximate amount of remaining moov header space reserved using reserved-max-duration.
+0
+
+
+
+Gst3GPPMux::reserved-max-duration
+guint64
+
+rw
+Reserved maximum file duration (ns)
+When set to a value > 0, reserves space for index tables at the beginning of the file.
+18446744073709551615
+
+
+
+Gst3GPPMux::reserved-moov-update-period
+guint64
+
+rw
+Interval at which to update index tables (ns)
+When used with reserved-max-duration, periodically updates the index tables with information muxed so far.
+18446744073709551615
+
+
+
+Gst3GPPMux::interleave-bytes
+guint64
+
+rw
+Interleave (bytes)
+Interleave between streams in bytes.
+0
+
+
+
+Gst3GPPMux::interleave-time
+guint64
+
+rw
+Interleave (time)
+Interleave between streams in nanoseconds.
+250000000
+
+
+
+Gst3GPPMux::max-raw-audio-drift
+guint64
+
+rw
+Max Raw Audio Drift
+Maximum allowed drift of raw audio samples vs. timestamps in nanoseconds.
+40000000
+
+
+
+Gst3GPPMux::reserved-prefill
+gboolean
+
+rwx
+Reserved Prefill Samples Table
+Prefill samples table of reserved duration.
+FALSE
+
+
+
+Gst3GPPMux::start-gap-threshold
+guint64
+
+rw
+Start Gap Threshold
+Threshold for creating an edit list for gaps at the start in nanoseconds.
+0
+
+
+
+GstSplitFileSrc::location
+gchar*
+
+rw
+File Location
+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.
+NULL
+
+
+
+GstMatroskaDemux::max-gap-time
+guint64
+
+rw
+Maximum gap time
+The demuxer sends out segment events for skipping gaps longer than this (0 = disabled).
+2000000000
+
+
+
+GstMatroskaDemux::max-backtrack-distance
+guint
+
+rw
+Maximum backtrack distance
+Maximum backtrack distance in seconds when seeking without and index in pull mode and search for a keyframe (0 = disable backtracking).
+30
+
+
+
+GstSoupHttpClientSink::automatic-redirect
+gboolean
+
+rw
+automatic-redirect
+Automatically follow HTTP redirects (HTTP Status Code 3xx).
+TRUE
+
+
+
+GstSoupHttpClientSink::cookies
+GStrv
+
+rw
+Cookies
+HTTP request cookies.
+
+
+
+
+GstSoupHttpClientSink::location
+gchar*
+
+rw
+Location
+URI to send to.
+""
+
+
+
+GstSoupHttpClientSink::proxy
+gchar*
+
+rw
+Proxy
+HTTP proxy server URI.
+""
+
+
+
+GstSoupHttpClientSink::proxy-id
+gchar*
+
+rw
+proxy-id
+user id for proxy authentication.
+""
+
+
+
+GstSoupHttpClientSink::proxy-pw
+gchar*
+
+rw
+proxy-pw
+user password for proxy authentication.
+""
+
+
+
+GstSoupHttpClientSink::session
+SoupSession*
+
+rw
+session
+SoupSession object to use for communication.
+
+
+
+
+GstSoupHttpClientSink::user-agent
+gchar*
+
+rw
+User-Agent
+Value of the User-Agent HTTP request header field.
+"GStreamer souphttpclientsink "
+
+
+
+GstSoupHttpClientSink::user-id
+gchar*
+
+rw
+user-id
+user id for authentication.
+""
+
+
+
+GstSoupHttpClientSink::user-pw
+gchar*
+
+rw
+user-pw
+user password for authentication.
+""
+
+
+
+GstSoupHttpClientSink::http-log-level
+SoupLoggerLogLevel
+
+rw
+HTTP log level
+Set log level for soup's HTTP session log.
+SOUP_LOGGER_LOG_NONE
+
+
+
+GstSoupHttpClientSink::retries
+gint
+>= G_MAXULONG
+rw
+Retries
+Maximum number of retries, zero to disable, -1 to retry forever.
+0
+
+
+
+GstSoupHttpClientSink::retry-delay
+gint
+>= 1
+rw
+Retry Delay
+Delay in seconds between retries after a failure.
+5
+
+
+
+GstWavParse::ignore-length
+gboolean
+
+rw
+Ignore length
+Ignore length from the Wave header.
+FALSE
+
+
+
+GstGdkPixbufOverlay::location
+gchar*
+
+rw
+location
+Location of image file to overlay.
+NULL
+
+
+
+GstGdkPixbufOverlay::offset-x
+gint
+
+rw
+X Offset
+For positive value, horizontal offset of overlay image in pixels from left of video image. For negative value, horizontal offset of overlay image in pixels from right of video image.
+0
+
+
+
+GstGdkPixbufOverlay::offset-y
+gint
+
+rw
+Y Offset
+For positive value, vertical offset of overlay image in pixels from top of video image. For negative value, vertical offset of overlay image in pixels from bottom of video image.
+0
+
+
+
+GstGdkPixbufOverlay::overlay-height
+gint
+>= 0
+rw
+Overlay Height
+Height of overlay image in pixels (0 = same as overlay image).
+0
+
+
+
+GstGdkPixbufOverlay::overlay-width
+gint
+>= 0
+rw
+Overlay Width
+Width of overlay image in pixels (0 = same as overlay image).
+0
+
+
+
+GstGdkPixbufOverlay::relative-x
+gdouble
+[-1,1]
+rw
+Relative X Offset
+Horizontal offset of overlay image in fractions of video image width, from top-left corner of video image (in relative positioning).
+0
+
+
+
+GstGdkPixbufOverlay::relative-y
+gdouble
+[-1,1]
+rw
+Relative Y Offset
+Vertical offset of overlay image in fractions of video image height, from top-left corner of video image (in relative positioning).
+0
+
+
+
+GstGdkPixbufOverlay::alpha
+gdouble
+[0,1]
+rw
+Alpha
+Global alpha of overlay image.
+1
+
+
+
+GstGdkPixbufOverlay::pixbuf
+GdkPixbuf*
+
+rw
+Pixbuf
+GdkPixbuf object to render.
+
+
+
+
+GstGdkPixbufOverlay::positioning-mode
+GstGdkPixbufPositioningMode
+
+rw
+Positioning mode
+Positioning mode of offset-x and offset-y properties.
+pixels-relative-to-edges
+
+
+
+GstGdkPixbufOverlay::coef-x
+gdouble
+[-1,1]
+rw
+Relative X Offset
+Horizontal offset of overlay image in fractions of video image width, from top-left corner of video image (absolute positioning).
+0
+
+
+
+GstGdkPixbufOverlay::coef-y
+gdouble
+[-1,1]
+rw
+Relative Y Offset
+Vertical offset of overlay image in fractions of video image height, from top-left corner of video image (absolute positioning).
+0
+
+
+
+GstVP8Enc::arnr-maxframes
+gint
+[0,15]
+rw
+AltRef max frames
+AltRef maximum number of frames.
+0
+
+
+
+GstVP8Enc::arnr-strength
+gint
+[0,6]
+rw
+AltRef strength
+AltRef strength.
+3
+
+
+
+GstVP8Enc::arnr-type
+gint
+[1,3]
+rw
+AltRef type
+AltRef type.
+3
+
+
+
+GstVP8Enc::auto-alt-ref
+gboolean
+
+rw
+Auto alt reference frames
+Automatically generate AltRef frames.
+FALSE
+
+
+
+GstVP8Enc::buffer-initial-size
+gint
+>= 0
+rw
+Buffer initial size
+Initial client buffer size (ms).
+4000
+
+
+
+GstVP8Enc::buffer-optimal-size
+gint
+>= 0
+rw
+Buffer optimal size
+Optimal client buffer size (ms).
+5000
+
+
+
+GstVP8Enc::buffer-size
+gint
+>= 0
+rw
+Buffer size
+Client buffer size (ms).
+6000
+
+
+
+GstVP8Enc::cpu-used
+gint
+[-16,16]
+rw
+CPU used
+CPU used.
+0
+
+
+
+GstVP8Enc::cq-level
+gint
+[0,63]
+rw
+Constrained quality level
+Constrained quality level.
+10
+
+
+
+GstVP8Enc::deadline
+gint64
+>= 0
+rw
+Deadline
+Deadline per frame (usec, 0=disabled).
+0
+
+
+
+GstVP8Enc::dropframe-threshold
+gint
+[0,100]
+rw
+Drop Frame Threshold
+Temporal resampling threshold (buf %).
+0
+
+
+
+GstVP8Enc::end-usage
+GstVP8EncEndUsage
+
+rw
+Rate control mode
+Rate control mode.
+Variable Bit Rate (VBR) mode
+
+
+
+GstVP8Enc::error-resilient
+GstVP8EncErFlags
+
+rw
+Error resilient
+Error resilience flags.
+
+
+
+
+GstVP8Enc::lag-in-frames
+gint
+[0,25]
+rw
+Lag in frames
+Maximum number of frames to lag.
+0
+
+
+
+GstVP8Enc::max-quantizer
+gint
+[0,63]
+rw
+Maximum Quantizer
+Maximum Quantizer (worst).
+63
+
+
+
+GstVP8Enc::min-quantizer
+gint
+[0,63]
+rw
+Minimum Quantizer
+Minimum Quantizer (best).
+4
+
+
+
+GstVP8Enc::multipass-cache-file
+gchar*
+
+rw
+Multipass Cache File
+Multipass cache file. If stream caps reinited, multiple files will be created: file, file.1, file.2, ... and so on.
+"multipass.cache"
+
+
+
+GstVP8Enc::multipass-mode
+GstVP8EncMultipassMode
+
+rw
+Multipass Mode
+Multipass encode mode.
+One pass encoding (default)
+
+
+
+GstVP8Enc::noise-sensitivity
+gint
+[0,6]
+rw
+Noise sensitivity
+Noise sensisivity (frames to blur).
+0
+
+
+
+GstVP8Enc::resize-allowed
+gboolean
+
+rw
+Resize Allowed
+Allow spatial resampling.
+FALSE
+
+
+
+GstVP8Enc::resize-down-threshold
+gint
+[0,100]
+rw
+Resize Down Threshold
+Downscale threshold (buf %).
+60
+
+
+
+GstVP8Enc::resize-up-threshold
+gint
+[0,100]
+rw
+Resize Up Threshold
+Upscale threshold (buf %).
+30
+
+
+
+GstVP8Enc::sharpness
+gint
+[0,7]
+rw
+Sharpness
+Filter sharpness.
+0
+
+
+
+GstVP8Enc::static-threshold
+gint
+>= 0
+rw
+Static Threshold
+Motion detection threshold.
+0
+
+
+
+GstVP8Enc::target-bitrate
+gint
+>= 0
+rw
+Target bitrate
+Target bitrate (in bits/sec).
+256000
+
+
+
+GstVP8Enc::threads
+gint
+[0,64]
+rw
+Threads
+Number of threads to use.
+0
+
+
+
+GstVP8Enc::token-partitions
+GstVP8EncTokenPartitions
+
+rw
+Token partitions
+Number of token partitions.
+One token partition
+
+
+
+GstVP8Enc::tuning
+GstVP8EncTuning
+
+rw
+Tuning
+Tuning.
+Tune for PSNR
+
+
+
+GstVP8Enc::horizontal-scaling-mode
+GstVP8EncScalingMode
+
+rw
+Horizontal scaling mode
+Horizontal scaling mode.
+Normal
+
+
+
+GstVP8Enc::keyframe-max-dist
+gint
+>= 0
+rw
+Keyframe max distance
+Maximum distance between keyframes (number of frames).
+128
+
+
+
+GstVP8Enc::keyframe-mode
+GstVP8EncKfMode
+
+rw
+Keyframe Mode
+Keyframe placement.
+Determine optimal placement automatically
+
+
+
+GstVP8Enc::max-intra-bitrate
+gint
+>= 0
+rw
+Max Intra bitrate
+Maximum Intra frame bitrate.
+0
+
+
+
+GstVP8Enc::overshoot
+gint
+[0,1000]
+rw
+Overshoot PCT
+Datarate overshoot (max) target (%).
+100
+
+
+
+GstVP8Enc::temporal-scalability-layer-id
+GValueArray*
+
+rw
+Coding layer identification
+Sequence defining coding layer membership.
+
+
+
+
+GstVP8Enc::temporal-scalability-number-layers
+gint
+[1,5]
+rw
+Number of coding layers
+Number of coding layers to use.
+1
+
+
+
+GstVP8Enc::temporal-scalability-periodicity
+gint
+[0,16]
+rw
+Coding layer periodicity
+Length of sequence that defines layer membership periodicity.
+0
+
+
+
+GstVP8Enc::temporal-scalability-rate-decimator
+GValueArray*
+
+rw
+Coding layer rate decimator
+Rate decimation factors for each layer.
+
+
+
+
+GstVP8Enc::temporal-scalability-target-bitrate
+GValueArray*
+
+rw
+Coding layer target bitrates
+Target bitrates for coding layers (one per layer, decreasing).
+
+
+
+
+GstVP8Enc::twopass-vbr-bias
+gint
+[0,100]
+rw
+2-pass VBR bias
+CBR/VBR bias (0=CBR, 100=VBR).
+50
+
+
+
+GstVP8Enc::twopass-vbr-maxsection
+gint
+>= 0
+rw
+2-pass GOP max bitrate
+GOP maximum bitrate (% target).
+0
+
+
+
+GstVP8Enc::twopass-vbr-minsection
+gint
+>= 0
+rw
+2-pass GOP min bitrate
+GOP minimum bitrate (% target).
+0
+
+
+
+GstVP8Enc::undershoot
+gint
+[0,1000]
+rw
+Undershoot PCT
+Datarate undershoot (min) target (%).
+100
+
+
+
+GstVP8Enc::vertical-scaling-mode
+GstVP8EncScalingMode
+
+rw
+Vertical scaling mode
+Vertical scaling mode.
+Normal
+
+
+
+GstVP8Enc::timebase
+GstFraction
+
+rw
+Shortest interframe time
+Fraction of one second that is the shortest interframe time - normally left as zero which will default to the framerate.
+
+
+
+
+GstVP8Dec::deblocking-level
+guint
+<= 16
+rw
+Deblocking Level
+Deblocking level.
+4
+
+
+
+GstVP8Dec::noise-level
+guint
+<= 16
+rw
+Noise Level
+Noise level.
+0
+
+
+
+GstVP8Dec::post-processing
+gboolean
+
+rw
+Post Processing
+Enable post processing.
+FALSE
+
+
+
+GstVP8Dec::post-processing-flags
+GstVP8DecPostProcessingFlags
+
+rw
+Post Processing Flags
+Flags to control post processing.
+Deblock|Demacroblock|Multi-frame quality enhancement
+
+
+
+GstVP8Dec::threads
+guint
+[1,16]
+rw
+Max Threads
+Maximum number of decoding threads.
+1
+
+
+
+GstVideoMedian::filtersize
+GstVideoMedianSize
+
+rw
+Filtersize
+The size of the filter.
+Median of 5 neighbour pixels
+
+
+
+GstVideoMedian::lum-only
+gboolean
+
+rw
+Lum Only
+Only apply filter on luminance.
+TRUE
+
+
+
+GstScaletempo::overlap
+gdouble
+[0,1]
+rw
+Overlap Length
+Percentage of stride to overlap.
+0.2
+
+
+
+GstScaletempo::rate
+gdouble
+[-2.14748e+09,2.14748e+09]
+r
+Playback Rate
+Current playback rate.
+1
+
+
+
+GstScaletempo::search
+guint
+<= 500
+rw
+Search Length
+Length in milliseconds to search for best overlap position.
+14
+
+
+
+GstScaletempo::stride
+guint
+[1,5000]
+rw
+Stride Length
+Length in milliseconds to output each stride.
+30
+
+
+
+GstRTPMux::seqnum
+guint
+
+r
+Sequence number
+The RTP sequence number of the last processed packet.
+0
+
+
+
+GstRTPMux::seqnum-offset
+gint
+>= G_MAXULONG
+rw
+Sequence number Offset
+Offset to add to all outgoing seqnum (-1 = random).
+-1
+
+
+
+GstRTPMux::ssrc
+guint
+
+rw
+SSRC
+The SSRC of the packets (default == random).
+4294967295
+
+
+
+GstRTPMux::timestamp-offset
+gint
+>= G_MAXULONG
+rw
+Timestamp Offset
+Offset to add to all outgoing timestamps (-1 = random).
+-1
+
+
+
+GstRtpSBCPay::min-frames
+gint
+>= G_MAXULONG
+rw
+minimum frame number
+Minimum quantity of frames to send in one packet (-1 for maximum allowed by the mtu).
+0
+
+
+
+GstRTPDTMFSrc::clock-rate
+guint
+
+rw
+clockrate
+The clock-rate at which to generate the dtmf packets.
+8000
+
+
+
+GstRTPDTMFSrc::packet-redundancy
+guint
+[1,5]
+rw
+Packet Redundancy
+Number of packets to send to indicate start and stop dtmf events.
+1
+
+
+
+GstRTPDTMFSrc::pt
+guint
+<= 128
+rw
+payload type
+The payload type of the packets.
+96
+
+
+
+GstRTPDTMFSrc::seqnum
+guint
+
+r
+Sequence number
+The RTP sequence number of the last processed packet.
+0
+
+
+
+GstRTPDTMFSrc::seqnum-offset
+gint
+>= G_MAXULONG
+rw
+Sequence number Offset
+Offset to add to all outgoing seqnum (-1 = random).
+-1
+
+
+
+GstRTPDTMFSrc::ssrc
+guint
+
+rw
+SSRC
+The SSRC of the packets (-1 == random).
+4294967295
+
+
+
+GstRTPDTMFSrc::timestamp
+guint
+
+r
+Timestamp
+The RTP timestamp of the last processed packet.
+0
+
+
+
+GstRTPDTMFSrc::timestamp-offset
+gint
+>= G_MAXULONG
+rw
+Timestamp Offset
+Offset to add to all outgoing timestamps (-1 = random).
+-1
+
+
+
+GstRtpDTMFDepay::max-duration
+guint
+
+rw
+Maximum duration
+The maxumimum duration (ms) of the outgoing soundpacket. (0 = no limit).
+0
+
+
+
+GstRtpDTMFDepay::unit-time
+guint
+<= 1000
+rw
+Duration unittime
+The smallest unit (ms) the duration must be a multiple of (0 disables it).
+0
+
+
+
+GstDTMFSrc::interval
+guint
+[10,50]
+rw
+Interval between tone packets
+Interval in ms between two tone packets.
+50
+
+
+
+GstRtpVorbisPay::config-interval
+guint
+<= 3600
+rw
+Config Send Interval
+Send Config Insertion Interval in seconds (configuration headers will be multiplexed in the data stream when detected.) (0 = disabled).
+0
+
+
+
+GstRTPRtxQueue::max-size-packets
+guint
+
+rw
+Max Size Packets
+Amount of packets to queue (0 = unlimited).
+100
+
+
+
+GstRTPRtxQueue::max-size-time
+guint
+
+rw
+Max Size Times
+Amount of ms to queue (0 = unlimited).
+0
+
+
+
+GstRTPRtxQueue::fulfilled-requests
+guint
+
+r
+Fulfilled Requests
+Number of fulfilled retransmission requests.
+0
+
+
+
+GstRTPRtxQueue::requests
+guint
+
+r
+Requests
+Total number of retransmission requests.
+0
+
+
+
+GstRtpGSTPay::config-interval
+guint
+<= 3600
+rw
+Caps/Tags Send Interval
+Interval for sending caps and TAG events in seconds (0 = disabled).
+0
+
+
+
+GstRtpVP8Pay::picture-id-mode
+GstVP8RTPPayMode
+
+rw
+Picture ID Mode
+The picture ID mode for payloading.
+No Picture ID
+
+
+
+GstVP9Enc::arnr-maxframes
+gint
+[0,15]
+rw
+AltRef max frames
+AltRef maximum number of frames.
+0
+
+
+
+GstVP9Enc::arnr-strength
+gint
+[0,6]
+rw
+AltRef strength
+AltRef strength.
+3
+
+
+
+GstVP9Enc::arnr-type
+gint
+[1,3]
+rw
+AltRef type
+AltRef type.
+3
+
+
+
+GstVP9Enc::auto-alt-ref
+gboolean
+
+rw
+Auto alt reference frames
+Automatically generate AltRef frames.
+FALSE
+
+
+
+GstVP9Enc::buffer-initial-size
+gint
+>= 0
+rw
+Buffer initial size
+Initial client buffer size (ms).
+4000
+
+
+
+GstVP9Enc::buffer-optimal-size
+gint
+>= 0
+rw
+Buffer optimal size
+Optimal client buffer size (ms).
+5000
+
+
+
+GstVP9Enc::buffer-size
+gint
+>= 0
+rw
+Buffer size
+Client buffer size (ms).
+6000
+
+
+
+GstVP9Enc::cpu-used
+gint
+[-16,16]
+rw
+CPU used
+CPU used.
+0
+
+
+
+GstVP9Enc::cq-level
+gint
+[0,63]
+rw
+Constrained quality level
+Constrained quality level.
+10
+
+
+
+GstVP9Enc::deadline
+gint64
+>= 0
+rw
+Deadline
+Deadline per frame (usec, 0=disabled).
+0
+
+
+
+GstVP9Enc::dropframe-threshold
+gint
+[0,100]
+rw
+Drop Frame Threshold
+Temporal resampling threshold (buf %).
+0
+
+
+
+GstVP9Enc::end-usage
+GstVP9EncEndUsage
+
+rw
+Rate control mode
+Rate control mode.
+Variable Bit Rate (VBR) mode
+
+
+
+GstVP9Enc::error-resilient
+GstVP9EncErFlags
+
+rw
+Error resilient
+Error resilience flags.
+
+
+
+
+GstVP9Enc::horizontal-scaling-mode
+GstVP9EncScalingMode
+
+rw
+Horizontal scaling mode
+Horizontal scaling mode.
+Normal
+
+
+
+GstVP9Enc::keyframe-max-dist
+gint
+>= 0
+rw
+Keyframe max distance
+Maximum distance between keyframes (number of frames).
+128
+
+
+
+GstVP9Enc::keyframe-mode
+GstVP9EncKfMode
+
+rw
+Keyframe Mode
+Keyframe placement.
+Determine optimal placement automatically
+
+
+
+GstVP9Enc::lag-in-frames
+gint
+[0,25]
+rw
+Lag in frames
+Maximum number of frames to lag.
+0
+
+
+
+GstVP9Enc::max-intra-bitrate
+gint
+>= 0
+rw
+Max Intra bitrate
+Maximum Intra frame bitrate.
+0
+
+
+
+GstVP9Enc::max-quantizer
+gint
+[0,63]
+rw
+Maximum Quantizer
+Maximum Quantizer (worst).
+63
+
+
+
+GstVP9Enc::min-quantizer
+gint
+[0,63]
+rw
+Minimum Quantizer
+Minimum Quantizer (best).
+4
+
+
+
+GstVP9Enc::multipass-cache-file
+gchar*
+
+rw
+Multipass Cache File
+Multipass cache file. If stream caps reinited, multiple files will be created: file, file.1, file.2, ... and so on.
+"multipass.cache"
+
+
+
+GstVP9Enc::multipass-mode
+GstVP9EncMultipassMode
+
+rw
+Multipass Mode
+Multipass encode mode.
+One pass encoding (default)
+
+
+
+GstVP9Enc::noise-sensitivity
+gint
+[0,6]
+rw
+Noise sensitivity
+Noise sensisivity (frames to blur).
+0
+
+
+
+GstVP9Enc::overshoot
+gint
+[0,1000]
+rw
+Overshoot PCT
+Datarate overshoot (max) target (%).
+100
+
+
+
+GstVP9Enc::resize-allowed
+gboolean
+
+rw
+Resize Allowed
+Allow spatial resampling.
+FALSE
+
+
+
+GstVP9Enc::resize-down-threshold
+gint
+[0,100]
+rw
+Resize Down Threshold
+Downscale threshold (buf %).
+60
+
+
+
+GstVP9Enc::resize-up-threshold
+gint
+[0,100]
+rw
+Resize Up Threshold
+Upscale threshold (buf %).
+30
+
+
+
+GstVP9Enc::sharpness
+gint
+[0,7]
+rw
+Sharpness
+Filter sharpness.
+0
+
+
+
+GstVP9Enc::static-threshold
+gint
+>= 0
+rw
+Static Threshold
+Motion detection threshold.
+0
+
+
+
+GstVP9Enc::target-bitrate
+gint
+>= 0
+rw
+Target bitrate
+Target bitrate (in bits/sec).
+256000
+
+
+
+GstVP9Enc::temporal-scalability-layer-id
+GValueArray*
+
+rw
+Coding layer identification
+Sequence defining coding layer membership.
+
+
+
+
+GstVP9Enc::temporal-scalability-number-layers
+gint
+[1,5]
+rw
+Number of coding layers
+Number of coding layers to use.
+1
+
+
+
+GstVP9Enc::temporal-scalability-periodicity
+gint
+[0,16]
+rw
+Coding layer periodicity
+Length of sequence that defines layer membership periodicity.
+0
+
+
+
+GstVP9Enc::temporal-scalability-rate-decimator
+GValueArray*
+
+rw
+Coding layer rate decimator
+Rate decimation factors for each layer.
+
+
+
+
+GstVP9Enc::temporal-scalability-target-bitrate
+GValueArray*
+
+rw
+Coding layer target bitrates
+Target bitrates for coding layers (one per layer, decreasing).
+
+
+
+
+GstVP9Enc::threads
+gint
+[0,64]
+rw
+Threads
+Number of threads to use.
+0
+
+
+
+GstVP9Enc::timebase
+GstFraction
+
+rw
+Shortest interframe time
+Fraction of one second that is the shortest interframe time - normally left as zero which will default to the framerate.
+
+
+
+
+GstVP9Enc::token-partitions
+GstVP9EncTokenPartitions
+
+rw
+Token partitions
+Number of token partitions.
+One token partition
+
+
+
+GstVP9Enc::tuning
+GstVP9EncTuning
+
+rw
+Tuning
+Tuning.
+Tune for PSNR
+
+
+
+GstVP9Enc::twopass-vbr-bias
+gint
+[0,100]
+rw
+2-pass VBR bias
+CBR/VBR bias (0=CBR, 100=VBR).
+50
+
+
+
+GstVP9Enc::twopass-vbr-maxsection
+gint
+>= 0
+rw
+2-pass GOP max bitrate
+GOP maximum bitrate (% target).
+0
+
+
+
+GstVP9Enc::twopass-vbr-minsection
+gint
+>= 0
+rw
+2-pass GOP min bitrate
+GOP minimum bitrate (% target).
+0
+
+
+
+GstVP9Enc::undershoot
+gint
+[0,1000]
+rw
+Undershoot PCT
+Datarate undershoot (min) target (%).
+100
+
+
+
+GstVP9Enc::vertical-scaling-mode
+GstVP9EncScalingMode
+
+rw
+Vertical scaling mode
+Vertical scaling mode.
+Normal
+
+
+
+GstVP9Dec::deblocking-level
+guint
+<= 16
+rw
+Deblocking Level
+Deblocking level.
+4
+
+
+
+GstVP9Dec::noise-level
+guint
+<= 16
+rw
+Noise Level
+Noise level.
+0
+
+
+
+GstVP9Dec::post-processing
+gboolean
+
+rw
+Post Processing
+Enable post processing.
+FALSE
+
+
+
+GstVP9Dec::post-processing-flags
+GstVP9DecPostProcessingFlags
+
+rw
+Post Processing Flags
+Flags to control post processing.
+Deblock|Demacroblock
+
+
+
+GstVP9Dec::threads
+guint
+[1,16]
+rw
+Max Threads
+Maximum number of decoding threads.
+1
+
+
+
+GstRtpVRawPay::chunks-per-frame
+gint
+>= 1
+rw
+Chunks per Frame
+Split and send out each frame in multiple chunks to reduce overhead.
+10
+
+
+
+GstPushFileSrc::applied-rate
+gdouble
+>= G_MINDOUBLE
+rw
+Applied Rate
+Applied rate to use in TIME SEGMENT.
+1
+
+
+
+GstPushFileSrc::initial-timestamp
+guint64
+
+rw
+Initial Timestamp
+Initial Buffer Timestamp (if time-segment TRUE).
+18446744073709551615
+
+
+
+GstPushFileSrc::location
+gchar*
+
+rw
+File Location
+Location of the file to read.
+NULL
+
+
+
+GstPushFileSrc::rate
+gdouble
+>= G_MINDOUBLE
+rw
+Rate
+Rate to use in TIME SEGMENT.
+1
+
+
+
+GstPushFileSrc::start-time
+gint64
+>= 0
+rw
+Start Time
+Initial Start Time (if time-segment TRUE).
+0
-GstMJ2Mux::trak-timescale
-guint
-
-rwx
-Track timescale
-Timescale to use for the tracks (units per second, 0 is automatic).
+GstPushFileSrc::stream-time
+gint64
+>= 0
+rw
+Stream Time
+Initial Stream Time (if time-segment TRUE).
0
-GstISMLMux::dts-method
-GstQTMuxDtsMethods
+GstPushFileSrc::time-segment
+gboolean
-rwx
-dts-method
-Method to determine DTS time.
-reorder
+rw
+Time Segment
+Emit TIME SEGMENTS.
+FALSE
-GstISMLMux::faststart
-gboolean
+GstSplitMuxSrc::location
+gchar*
rw
-Format file to faststart
-If the file should be formatted for faststart (headers first).
-FALSE
+File Input Pattern
+Glob pattern for the location of the files to read.
+NULL
-GstISMLMux::faststart-file
+GstSplitMuxSink::location
gchar*
-rwx
-File to use for storing buffers
-File that will be used temporarily to store data from the stream when creating a faststart file. If null a filepath will be created automatically.
+rw
+File Output Pattern
+Format string pattern for the location of the files to write (e.g. video%05d.mp4).
NULL
-GstISMLMux::fragment-duration
-guint
+GstSplitMuxSink::max-size-bytes
+guint64
-rwx
-Fragment duration
-Fragment durations in ms (produce a fragmented file if > 0).
-2000
+rw
+Max. size bytes
+Max. amount of data per file (in bytes, 0=disable).
+0
-GstISMLMux::moov-recovery-file
-gchar*
+GstSplitMuxSink::max-size-time
+guint64
-rwx
-File to store data for posterior moov atom recovery
-File to be used to store data for moov atom making movie file recovery possible in case of a crash during muxing. Null for disabled. (Experimental).
-NULL
+rw
+Max. size (ns)
+Max. amount of time per file (in ns, 0=disable).
+0
-GstISMLMux::movie-timescale
-guint
->= 1
-rwx
-Movie timescale
-Timescale to use in the movie (units per second).
-1000
+GstSplitMuxSink::mux-overhead
+gdouble
+[0,1]
+rw
+Muxing Overhead
+Extra size overhead of muxing (0.02 = 2%).
+0.02
-GstISMLMux::presentation-time
-gboolean
+GstSplitMuxSink::muxer
+GstElement*
-rwx
-Include presentation-time info
-Calculate and include presentation/composition time (in addition to decoding time).
-TRUE
+rw
+Muxer
+The muxer element to use (NULL = default mp4mux). Valid only for async-finalize = FALSE.
+
-GstISMLMux::streamable
-gboolean
+GstSplitMuxSink::sink
+GstElement*
-rwx
-Streamable
-If set to true, the output should be as if it is to be streamed and hence no indexes written or duration written.
-FALSE
+rw
+Sink
+The sink element (or element chain) to use (NULL = default filesink). Valid only for async-finalize = FALSE.
+
-GstISMLMux::trak-timescale
+GstSplitMuxSink::max-files
guint
-rwx
-Track timescale
-Timescale to use for the tracks (units per second, 0 is automatic).
+rw
+Max files
+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.
0
-GstGPPMux::dts-method
-GstQTMuxDtsMethods
-
-rwx
-dts-method
-Method to determine DTS time.
-reorder
-
-
-
-GstGPPMux::faststart
+GstSplitMuxSink::send-keyframe-requests
gboolean
rw
-Format file to faststart
-If the file should be formatted for faststart (headers first).
+Request keyframes at max-size-time
+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.
FALSE
-GstGPPMux::faststart-file
+GstSplitMuxSink::max-size-timecode
gchar*
-rwx
-File to use for storing buffers
-File that will be used temporarily to store data from the stream when creating a faststart file. If null a filepath will be created automatically.
+rw
+Maximum timecode difference
+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.
NULL
-GstGPPMux::fragment-duration
-guint
+GstSplitMuxSink::alignment-threshold
+guint64
-rwx
-Fragment duration
-Fragment durations in ms (produce a fragmented file if > 0).
+rw
+Alignment threshold (ns)
+Allow non-reference streams to be that many ns before the reference stream.
0
-GstGPPMux::moov-recovery-file
-gchar*
-
-rwx
-File to store data for posterior moov atom recovery
-File to be used to store data for moov atom making movie file recovery possible in case of a crash during muxing. Null for disabled. (Experimental).
-NULL
-
-
-
-GstGPPMux::movie-timescale
-guint
->= 1
-rwx
-Movie timescale
-Timescale to use in the movie (units per second).
-1000
-
-
-
-GstGPPMux::presentation-time
+GstSplitMuxSink::use-robust-muxing
gboolean
-rwx
-Include presentation-time info
-Calculate and include presentation/composition time (in addition to decoding time).
-TRUE
+rw
+Support robust-muxing mode of some muxers
+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.
+FALSE
-GstGPPMux::streamable
+GstSplitMuxSink::async-finalize
gboolean
-rwx
-Streamable
-If set to true, the output should be as if it is to be streamed and hence no indexes written or duration written.
+rw
+Finalize fragments asynchronously
+Finalize each fragment asynchronously and start a new one.
FALSE
-GstGPPMux::trak-timescale
-guint
+GstSplitMuxSink::muxer-factory
+gchar*
-rwx
-Track timescale
-Timescale to use for the tracks (units per second, 0 is automatic).
-0
+rw
+Muxer factory
+The muxer element factory to use (default = mp4mux). Valid only for async-finalize = TRUE.
+"mp4mux"
-Gst3GPPMux::dts-method
-GstQTMuxDtsMethods
+GstSplitMuxSink::muxer-properties
+GstStructure*
-rwx
-dts-method
-Method to determine DTS time.
-reorder
+rw
+Muxer properties
+The muxer element properties to use. Example: {properties,boolean-prop=true,string-prop="hi"}. Valid only for async-finalize = TRUE.
+
-Gst3GPPMux::faststart
+GstSplitMuxSink::reset-muxer
gboolean
rw
-Format file to faststart
-If the file should be formatted for faststart (headers first).
-FALSE
+Reset Muxer
+Reset the muxer after each segment. Disabling this will not work for most muxers.
+TRUE
-Gst3GPPMux::faststart-file
+GstSplitMuxSink::sink-factory
gchar*
-rwx
-File to use for storing buffers
-File that will be used temporarily to store data from the stream when creating a faststart file. If null a filepath will be created automatically.
-NULL
+rw
+Sink factory
+The sink element factory to use (default = filesink). Valid only for async-finalize = TRUE.
+"filesink"
-Gst3GPPMux::fragment-duration
-guint
+GstSplitMuxSink::sink-properties
+GstStructure*
-rwx
-Fragment duration
-Fragment durations in ms (produce a fragmented file if > 0).
-0
+rw
+Sink properties
+The sink element properties to use. Example: {properties,boolean-prop=true,string-prop="hi"}. Valid only for async-finalize = TRUE.
+
-Gst3GPPMux::moov-recovery-file
-gchar*
+GstRtpVP9Pay::picture-id-mode
+GstVP9RTPPayMode
-rwx
-File to store data for posterior moov atom recovery
-File to be used to store data for moov atom making movie file recovery possible in case of a crash during muxing. Null for disabled. (Experimental).
-NULL
+rw
+Picture ID Mode
+The picture ID mode for payloading.
+No Picture ID
-Gst3GPPMux::movie-timescale
-guint
->= 1
-rwx
-Movie timescale
-Timescale to use in the movie (units per second).
-1000
+GstRtpH265Pay::config-interval
+gint
+[G_MAXULONG,3600]
+rw
+VPS SPS PPS Send Interval
+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).
+0
-Gst3GPPMux::presentation-time
-gboolean
+GstRtpH265Pay::sprop-parameter-sets
+gchar*
-rwx
-Include presentation-time info
-Calculate and include presentation/composition time (in addition to decoding time).
-TRUE
+rw
+sprop-parameter-sets
+The base64 sprop-parameter-sets to set in out caps (set to NULL to extract from stream).
+NULL
-Gst3GPPMux::streamable
-gboolean
-
+GstLameMP3Enc::bitrate
+gint
+[8,320]
rwx
-Streamable
-If set to true, the output should be as if it is to be streamed and hence no indexes written or duration written.
-FALSE
+Bitrate (kb/s)
+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).
+128
-Gst3GPPMux::trak-timescale
-guint
+GstLameMP3Enc::cbr
+gboolean
rwx
-Track timescale
-Timescale to use for the tracks (units per second, 0 is automatic).
-0
+CBR
+Enforce constant bitrate encoding (Only valid if target is bitrate).
+FALSE
-GstSplitFileSrc::location
-gchar*
+GstLameMP3Enc::encoding-engine-quality
+GstLameMP3EncEncodingEngineQuality
-rw
-File Location
-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.
-NULL
+rwx
+Encoding Engine Quality
+Quality/speed of the encoding engine, this does not affect the bitrate!.
+Standard
-GstMatroskaDemux::max-gap-time
-guint64
+GstLameMP3Enc::mono
+gboolean
-rw
-Maximum gap time
-The demuxer sends out segment events for skipping gaps longer than this (0 = disabled).
-2000000000
+rwx
+Mono
+Enforce mono encoding.
+FALSE
-GstPulseAudioSink::alignment-threshold
-guint64
-[1,18446744073709551614]
-rw
-Alignment Threshold
-Timestamp alignment threshold in nanoseconds.
-40000000
+GstLameMP3Enc::quality
+gfloat
+[0,9.999]
+rwx
+Quality
+VBR Quality from 0 to 10, 0 being the best (Only valid if target is quality).
+4
-GstPulseAudioSink::async
-gboolean
+GstLameMP3Enc::target
+GstLameMP3EncTarget
-rw
-Async
-Go asynchronously to PAUSED.
-TRUE
+rwx
+Target
+Optimize for quality or bitrate.
+Quality
-GstPulseAudioSink::blocksize
-guint
+GstTwoLame::ath-level
+gfloat
rw
-Block size
-Size in bytes to pull per buffer (0 = default).
-4096
+ATH Level
+ATH Level in dB.
+0
-GstPulseAudioSink::buffer-time
-gint64
->= 1
+GstTwoLame::bitrate
+gint
+[8,384]
rw
-Buffer Time
-Size of audio buffer in microseconds.
-200000
+Bitrate (kb/s)
+Bitrate in kbit/sec (8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 144, 160, 192, 224, 256, 320, 384).
+192
-GstPulseAudioSink::can-activate-pull
+GstTwoLame::copyright
gboolean
rw
-Allow Pull Scheduling
-Allow pull-based scheduling.
-FALSE
-
-
-
-GstPulseAudioSink::client
-gchar*
-
-rw
-Client
-The PulseAudio client name to use.
-""
+Copyright
+Mark as copyright.
+FALSE
-GstPulseAudioSink::device
-gchar*
+GstTwoLame::emphasis
+GstTwoLameEmphasis
rw
-Device
-The PulseAudio sink device to connect to.
-NULL
+Emphasis
+Pre-emphasis to apply to the decoded audio.
+No emphasis
-GstPulseAudioSink::device-name
-gchar*
+GstTwoLame::energy-level-extension
+gboolean
-r
-Device name
-Human-readable name of the sound device.
-NULL
+rw
+Energy Level Extension
+Write peak PCM level to each frame.
+FALSE
-GstPulseAudioSink::discont-wait
-guint64
-<= 18446744073709551614
+GstTwoLame::error-protection
+gboolean
+
rw
-Discont Wait
-Window of time in nanoseconds to wait before creating a discontinuity.
-1000000000
+Error protection
+Adds checksum to every frame.
+FALSE
-GstPulseAudioSink::drift-tolerance
-gint64
->= 1
+GstTwoLame::mode
+GstTwoLameMode
+
rw
-Drift Tolerance
-Tolerance for clock drift in microseconds.
-40000
+Mode
+Encoding mode.
+Joint Stereo
-GstPulseAudioSink::enable-last-buffer
+GstTwoLame::original
gboolean
rw
-Enable Last Buffer
-Enable the last-buffer property.
+Original
+Mark as original.
TRUE
-GstPulseAudioSink::last-buffer
-GstBuffer*
+GstTwoLame::padding
+GstTwoLamePadding
-r
-Last Buffer
-The last buffer received in the sink.
-
-
-
-
-GstPulseAudioSink::latency-time
-gint64
->= 1
rw
-Latency Time
-Audio latency in microseconds.
-10000
+Padding
+Padding type.
+No Padding
-GstPulseAudioSink::max-lateness
-gint64
->= G_MAXULONG
+GstTwoLame::psymodel
+gint
+[G_MAXULONG,4]
rw
-Max Lateness
-Maximum number of nanoseconds that a buffer can be late before it is dropped (-1 unlimited).
--1
+Psychoacoustic Model
+Psychoacoustic model used to encode the audio.
+3
-GstPulseAudioSink::mute
+GstTwoLame::quick-mode
gboolean
rw
-Mute
-Mute state of this stream.
+Quick mode
+Calculate Psymodel every frames.
FALSE
-GstPulseAudioSink::preroll-queue-len
-guint
-
-rwx
-Preroll queue length
-Number of buffers to queue during preroll.
-0
+GstTwoLame::quick-mode-count
+gint
+>= 0
+rw
+Quick mode count
+Calculate Psymodel every n frames.
+10
-GstPulseAudioSink::provide-clock
+GstTwoLame::vbr
gboolean
rw
-Provide Clock
-Provide a clock to be used as the global pipeline clock.
-TRUE
+VBR
+Enable variable bitrate mode.
+FALSE
-GstPulseAudioSink::qos
-gboolean
-
+GstTwoLame::vbr-level
+gfloat
+[-10,10]
rw
-Qos
-Generate Quality-of-Service events upstream.
-FALSE
+VBR Level
+VBR Level.
+5
-GstPulseAudioSink::render-delay
-guint64
-
+GstTwoLame::vbr-max-bitrate
+gint
+[0,384]
rw
-Render Delay
-Additional render delay of the sink in nanoseconds.
+VBR max bitrate
+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).
0
-GstPulseAudioSink::server
-gchar*
+GstRtpSbcDepay::ignore-timestamps
+gboolean
rw
-Server
-The PulseAudio server to connect to.
-NULL
+Ignore Timestamps
+Various statistics.
+FALSE
-GstPulseAudioSink::slave-method
-GstBaseAudioSinkSlaveMethod
+GstQtSrc::use-default-fbo
+gboolean
rw
-Slave Method
-Algorithm to use to match the rate of the masterclock.
-GST_BASE_AUDIO_SINK_SLAVE_SKEW
+Whether to use default FBO
+When set it will not create a new FBO for the QML render thread.
+FALSE
-GstPulseAudioSink::stream-properties
-GstStructure*
+GstQtSrc::window
+gpointer
rw
-stream properties
-list of pulseaudio stream properties.
+QQuickWindow
+The QQuickWindow to place in the object hierarchy.
-GstPulseAudioSink::sync
+GstQtSink::force-aspect-ratio
gboolean
rw
-Sync
-Sync on the clock.
+Force aspect ratio
+When enabled, scaling will respect original aspect ratio.
TRUE
-GstPulseAudioSink::throttle-time
-guint64
+GstQtSink::pixel-aspect-ratio
+GstFraction
rw
-Throttle time
-The time to keep between rendered buffers (unused).
-0
+Pixel Aspect Ratio
+The pixel aspect ratio of the device.
+
-GstPulseAudioSink::ts-offset
-gint64
+GstQtSink::widget
+gpointer
rw
-TS Offset
-Timestamp offset in nanoseconds.
-0
+QQuickItem
+The QQuickItem to place in the object hierarchy.
+
-GstPulseAudioSink::volume
-gdouble
-[0,10]
+GstRtpPtDemux::ignored-payload-types
+GstValueArray
+
rw
-Volume
-Linear volume of this stream, 1.0=100%.
-1
+Ignored payload types
+Packets with these payload types will be dropped.
+
-GstSoupHttpClientSink::automatic-redirect
+GstRtpUlpFecEnc::multipacket
gboolean
-rw
-automatic-redirect
-Automatically follow HTTP redirects (HTTP Status Code 3xx).
+rwx
+Multipacket
+Apply FEC on multiple packets.
TRUE
-GstSoupHttpClientSink::cookies
-GStrv
+GstRtpUlpFecEnc::mux-seq
+gboolean
-rw
-Cookies
-HTTP request cookies.
-
+rwx
+Mux seq
+Mux seqnum for media and fec packets in same seqnum space.
+FALSE
-GstSoupHttpClientSink::location
-gchar*
-
-rw
-Location
-URI to send to.
-""
+GstRtpUlpFecEnc::percentage
+guint
+<= 100
+rwx
+Percentage
+FEC overhead percentage for the whole stream.
+0
-GstSoupHttpClientSink::proxy
-gchar*
+GstRtpUlpFecEnc::percentage-important
+guint
+<= 100
+rwx
+Percentage important
+FEC overhead percentage for important packets.
+0
+
+
+
+GstRtpUlpFecEnc::protected
+guint
-rw
-Proxy
-HTTP proxy server URI.
-""
+r
+Protected
+Count of protected packets.
+0
-GstSoupHttpClientSink::proxy-id
-gchar*
+GstRtpUlpFecEnc::pt
+guint
+<= 255
+rwx
+payload type
+The payload type of FEC packets.
+255
+
+
+
+GstRtpUlpFecEnc::ssrc
+guint
+rwx
+SSRC
+The SSRC to use on FEC'd packets.
+0
+
+
+
+GstRtpUlpFecDec::pt
+guint
+<= 127
rw
-proxy-id
-user id for proxy authentication.
-""
+pt
+FEC packets payload type.
+0
-GstSoupHttpClientSink::proxy-pw
-gchar*
+GstRtpUlpFecDec::recovered
+guint
-rw
-proxy-pw
-user password for proxy authentication.
-""
+r
+recovered
+The number of recovered packets.
+0
-GstSoupHttpClientSink::session
-SoupSession*
+GstRtpUlpFecDec::storage
+GObject*
rw
-session
-SoupSession object to use for communication.
+RTP storage
+RTP storage.
-GstSoupHttpClientSink::user-agent
-gchar*
+GstRtpUlpFecDec::unrecovered
+guint
-rw
-User-Agent
-Value of the User-Agent HTTP request header field.
-"GStreamer souphttpclientsink "
+r
+unrecovered
+The number of unrecovered packets.
+0
-GstSoupHttpClientSink::user-id
-gchar*
+GstRtpStorage::internal-storage
+GObject*
-rw
-user-id
-user id for authentication.
-""
+r
+Internal storage
+Internal RtpStorage object.
+
-GstSoupHttpClientSink::user-pw
-gchar*
+GstRtpStorage::size-time
+guint64
-rw
-user-pw
-user password for authentication.
-""
+rwx
+Storage size (in ns)
+The amount of data to keep in the storage (in ns, 0-disable).
+0
-GstWavParse::ignore-length
+GstRtpRedEnc::allow-no-red-blocks
gboolean
rw
-Ignore length
-Ignore length from the Wave header.
-FALSE
+Allow no redundant blocks
+true - can produce RED packets even without redundant blocks (distance==0) false - RED packets will be produced only if distance>0.
+TRUE
-GstGdkPixbufOverlay::location
-gchar*
+GstRtpRedEnc::distance
+guint
rw
-location
-Location of image file to overlay.
-NULL
+RED distance
+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.).
+0
-GstGdkPixbufOverlay::offset-x
+GstRtpRedEnc::pt
gint
-
+[0,127]
rw
-X Offset
-Horizontal offset of overlay image in pixels from top-left corner of video image.
+payload type
+Payload type FEC packets (-1 disable).
0
-GstGdkPixbufOverlay::offset-y
-gint
+GstRtpRedEnc::sent
+guint
-rw
-Y Offset
-Vertical offset of overlay image in pixels from top-left corner of video image.
+r
+Sent
+Count of sent packets.
0
-GstGdkPixbufOverlay::overlay-height
+GstRtpRedDec::pt
gint
->= 0
+[G_MAXULONG,127]
rw
-Overlay Height
-Height of overlay image in pixels (0 = same as overlay image).
+payload type
+Payload type FEC packets.
+-1
+
+
+
+GstRtpRedDec::received
+guint
+
+r
+Received
+Count of received packets.
0
-GstGdkPixbufOverlay::overlay-width
-gint
->= 0
+GstStereo::active
+gboolean
+
rw
-Overlay Width
-Width of overlay image in pixels (0 = same as overlay image).
-0
+active
+active.
+TRUE
-GstGdkPixbufOverlay::relative-x
-gdouble
+GstStereo::stereo
+gfloat
[0,1]
rw
-Relative X Offset
-Horizontal offset of overlay image in fractions of video image width, from top-left corner of video image.
-0
+stereo
+stereo.
+0.1
-GstGdkPixbufOverlay::relative-y
-gdouble
-[0,1]
+GstRtpFunnel::common-ts-offset
+gint
+>= G_MAXULONG
+rwx
+Common Timestamp Offset
+Use the same RTP timestamp offset for all sinkpads (-1 = disable).
+-1
+
+
+
+GstCairoOverlay::draw-on-transparent-surface
+gboolean
+
rw
-Relative Y Offset
-Vertical offset of overlay image in fractions of video image height, from top-left corner of video image.
-0
+Draw on transparent surface
+Let the draw signal work on a transparent surface and blend the results with the video at a later time.
+FALSE
+
+
+
+GstImageFreeze::num-buffers
+gint
+>= G_MAXULONG
+rw
+num-buffers
+Number of buffers to output before sending EOS (-1 = unlimited).
+-1