Commit 1d4404d8 authored by Sebastian Dröge's avatar Sebastian Dröge 🍵

Release 1.3.1

parent 0ead08b2
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This is GStreamer Good Plugins 1.2.0
This is GStreamer Good Plugins 1.3.1
Changes since 1.0:
Changes since 1.2:
New API:
• GstContext negotiation / sharing / announcing for sharing a
generic context between elements, e.g. a display handle
• GL texture upload conversion meta for allowing different
buffer types to be converted to an OpenGL texture
• GstCapsFeatures as extension to GstCaps for allowing the
negotiation of specific memory or meta requirements between
elements
• GstMemory flags for contiguous and non-mappable memory
• The stream-start event has optional flags now, e.g. for signalling
sparse streams
• The stream-start even has an optional group-id field now to signal
all streams that should be played together
• Allocators library in gst-plugins-base, currently only with generic
dmabuf memory support
• insertbin library for easier handling of dynamically linked
pipelines (in -bad for now)
• EGL helper library (in -bad for now)
• MPEG-TS data structure library (in -bad for now)
• New GstVideoRegionOfInterestMeta to describe a region of interest on
video frames.
• GstVideoDecoder/Encoder has new ::flush() vfunc to replace the
ill-defined ::reset() vfunc.
• The URI query allows to query the redirected URI now.
• GstMessageType has GST_MESSAGE_EXTENDED added. All types before
that can be used together as a flags type as before, but from
that message onwards the types are just counted incrementally.
This was necessary to be able to add more message types.
In 2.0 GstMessageType will just become an enum and not a flags
type anymore.
• GstDeviceMonitor for device probing, e.g. to list all available
audio or video capture devices. This is the replacement for
GstPropertyProbe from 0.10.
• Events accumulate the running-time offset now when travelling
through pads, as set by the gst_pad_set_offset() function. This
allows to compensate for this in the QOS event for example.
• GstBuffer has a new flag "tag-memory" that is set automatically
when memory is added or removed to a buffer. This allows buffer
pools to detect if they can recycle a buffer or need to reset
it first.
• GstToc has new API to mark GstTocEntries as loops.
• A not-authorized resource error has been defined to notify
applications that accessing the resource has failed because
of missing authorization and to distinguish this case from others.
This change is actually already in 1.2.4.
• GstPad has a new flag "accept-intersect", that will let the default
ACCEPT_CAPS query handler do an intersection instead of subset check.
This is interesting for parser elements that can handle incomplete
caps.
• GstCollectPads has support for flushing and a default handler for
SEEK events now.
• GstSegment has new API to offset the running time by a specific
value and this is used in GstPad to allow positive and negative
offsets in gst_pad_set_offset() in all situations.
• Support for h265/HEVC and VP8 has been added to the codec utils and codec
parsers library, and was integrated into various elements.
• API for adjusting the TLS validation of RTSP connection has been added.
• The RTSP and SDP library has MIKEY (RFC 3830) support now, and
there is API to distinguish between the different RTSP profiles.
• API to access RTP time information and statistics.
• Support for auxiliary streams was added to rtpbin.
• Support for tiled, raw video formats has been added.
• GstVideoDecoder and GstAudioDecoder have API to help aggregating tag
events and merge custom tags into them consistently.
• playbin/playsink has support for application provided audio and video
filters.
• The GL library was merged from gst-plugins-gl to gst-plugins-bad,
providing a generic infrastructure for handling GL inside GStreamer
pipelines and a plugin with some elements using these, especially
a video sink. Supported platforms currently are Android, Cocoa (OS X),
DispManX (Raspberry Pi), EAGL (iOS), WGL (Windows) and generic X11,
Wayland and EGL platforms.
This replaces eglglessink and also is supposed to replace osxvideosink.
Major changes:
• New tool: gst-play-1.0 in gst-plugins-base for basic playback
testing on the command line.
• New plugins:
∘ mssdemux for Microsoft Smooth Streaming
∘ dashdemux for DASH adaptive streaming protocol
∘ bluez for interaction with Bluetooth devices
∘ openjpeg for JPEG2000 decoding and encoding
∘ daala for experimental Daala decoding and encoding
∘ vpx plugin has experimental VP9 decoding and encoding support
∘ webp plugin for WebP decoding (encoding to be added later)
∘ Various others: yadif, srtp, sbc, fluidsynth, midiparse,
mfc, ivtv, accuraterip and audiofxbad
• Moved plugins:
∘ dtmf, vp8rtp, scaletempo and rtpmux plugins are in
gst-plugins-good now
• Video:
∘ Fix handling of interlaced video in converters such as videoscale
and videoconvert (e.g. scale both fields independently)
∘ videoconvert will try harder to minimise quality losses when
conversion is necessary
∘ The experimental GstSurfaceConverter, GstSurfaceMeta and
GstVideoContext APIs from the (confusingly-named)
libgstbasevideo-1.0 library in gst-plugins-bad have now been
removed and been replaced by new APIs in GStreamer Core and
gst-plugins-base (see above). Since that was all that was left in
this library, the entire experimental libgstbasevideo-1.0 library
has been removed from gst-plugins-bad
∘ Chroma subsampling and chroma siting conversion is better handled
in videoconvert and the support for interlaced video was improved.
∘ New pinwheel and spoke patterns in videotestsrc
∘ videomixer can now accept different video formats on its sinkpads
and converts to a common format during mixing
• Audio:
∘ audioconvert will try harder to minimise quality losses when
conversion is necessary
∘ adder now allows muting/unmuting of its input streams, and also
per-input stream volume
∘ pulseaudio elements can switch between devices during playback now
∘ aacparse can convert between ADTS←→RAW
• Platform specific changes:
∘ Caps, events, etc. are now printed in the GStreamer debug logs
with their content instead of just the pointer address even on
non-glibc platforms (e.g. Windows, OSX, Android).
∘ Network elements (UDP/TCP) now work better with platforms,
where IPv6 sockets can't handle IPv4 (e.g. Windows)
∘ Linux/BSD: v4l2 had many improvements and cleanups
Major changes:
• New plugins and elements:
∘ v4l2videodec element for accessing hardware codecs on
platforms that make them accessible via V4L2, e.g.
Samsung Exynos. This comes together with major refactoring
of the existing V4L2 elements and the corresponding
infrastructure.
The v4l2videodec element replaces the mfcdec element.
∘ rtpstreampay and rtpstreamdepay elements for transmitting
RTP packets over a stream API (e.g. TCP) according to
RFC 4571.
∘ rtprtx elements for standard compliant implementation of
retransmissions, integrated into the rtpmanager plugin.
∘ audiomixer element that mixes multiple audio streams together
into a single one while keeping synchronization. This is
planned to become the replacement of the adder element.
∘ OpenNI2 plugin for 3D cameras like the Kinect camera.
∘ OpenEXR plugin for decoding high-dynamic-range EXR images.
∘ curlsshsink and curlsftpsink to write files via SSH/SFTP.
∘ videosignal, ivfparse and sndfile plugins ported from 0.10.
∘ avfvideosrc, vtdec and other elements were ported from 0.10 and
are available on OS X and iOS now.
• Other changes:
∘ gst-libav now uses libav 9
∘ Static linking of plugins is supported now (also in 1.0.7)
∘ rtspsrc: add support for NetClientClock: when the server suggests a
GstNetTimeProvider in the SDP, set up a GstNetClientClock that
slaves to the remote clock and suggest this clock in provide_clock.
Simplifies synchronized playback of a resource from an RTSP server.
gst-rtsp-server now supports adding this to the SDP and can provide
a network clock
∘ RTP retransmission / NACK support and big RTP jitterbuffer improvements
∘ SRTP and DTLS support
∘ Changes to many elements and core to use the correct sticky event
order and also not lose any important sticky events during flushing
∘ >1000 fixed bug reports, and many other bug fixes and other
improvements everywhere that had no bug report
∘ gst-libav now uses libav 10, and gained support for H265/HEVC.
∘ Support for hardware codecs and special memory types has been
improved with bugfixes and feature additions in various plugins
and base classes.
∘ Various bugfixes and improvements to buffering in queue2 and
multiqueue elements.
∘ dvbsrc supports more delivery mechanisms and other features
now, including DVB S2 and T2 support.
∘ The MPEGTS library has support for many more descriptors.
∘ Major improvements to tsdemux, especially time related.
∘ souphttpsrc now has support for keep-alive connections,
compression, configurable number of retries and configuration
for SSL certificate validation.
∘ hlsdemux has undergone major refactoring and works more
reliable now and supports more HLS features like trick modes.
Also fragments are pushed downstream while they're downloaded
now instead of waiting for each fragment to finish.
∘ videoflip can automatically flip based on the orientation tag.
∘ openjpeg supports the OpenJPEG2 API.
∘ gst-rtsp-server supports SRTP and MIKEY now.
∘ Lots of fixes for coverity warnings all over the place.
∘ 400+ fixed bug reports, and many other bug fixes and other
improvements everywhere that had no bug report.
Things to look out for:
• Single header includes for all libraries, e.g. #include
<gst/video/video.h> - this was needed for some bindings.
• Stricter (correct) caps subset checking in some cases where this was
not correct before. Caps will now always fail to be a compatible
subset of another set of caps if the subset caps are missing some
fields that the superset caps have. This might lead to not-negotiated
errors if caps are incomplete now. However, it also prevents possible
data corruption caused by piping data formatted in an
incompatible/unexpected way into some elements. Check your h264 caps
for stream-format and alignment fields and AAC caps for the
stream-format field. This change will also be included in the next
stable 1.0.8 release.
• Stricter checking for missing events and correct sticky event order
(stream-start, caps, segment) in some places; this is not enabled in
stable releases by default, but you may get warnings when using git
builds, development releases or when compiling with
-UG_DISABLE_ASSERT in CFLAGS
• x264enc now outputs data in byte-stream by default if downstream has
ANY caps (e.g. appsink without caps set, filesink, udpsink,
tcpserversink etc.)
• The MPEG TS demuxer posts messages contain the PMT, PAT, etc. in a
different format now. This new format uses the data structures from
the new MPEGTS library
• The GstContext API has changed between 1.1.4 and 1.1.90
• The eglglessink element was removed and replaced by the glimagesink
element.
• The mfcdec element was removed and replaced by v4l2videodec.
• osxvideosink is only available in OS X 10.6 or newer.
Release notes for GStreamer Good Plugins 1.2.0
Release notes for GStreamer Good Plugins 1.3.1
The GStreamer team is proud to announce a new feature release
in the 1.x stable series of the
core of the GStreamer streaming media framework.
The GStreamer team is pleased to announce the first release of the unstable
1.3 release series. The 1.3 release series is adding new features on top of
the 1.0 and 1.2 series and is part of the API and ABI-stable 1.x release
series of the GStreamer multimedia framework. The unstable 1.3 release series
will lead to the stable 1.4 release series in the next weeks, and newly added
API can still change until that point.
The 1.x series is a stable series targeted at end users.
It is not API or ABI compatible with the stable 0.10.x series.
It is, however, parallel installable with the 0.10.x series and
will not affect an existing 0.10.x installation.
Binaries for Android, iOS, Mac OS X and Windows will be provided separately
during the unstable 1.3 release series.
The versioning scheme that is used in general is that 1.x.y is API and
ABI backwards compatible with previous 1.x.y releases. If x is an even
number it is a stable release series and all releases in this series
will only contain important bugfixes, e.g. the 1.0 series with 1.0.7. If
x is odd it is a development release series that will lead to the next
stable release series 1.x+1 and contains new features and bigger
changes. During the development release series, new API can still
change.
......@@ -57,11 +70,101 @@ contains a set of codecs plugins based on libav (formerly gst-ffmpeg)
Bugs fixed in this release
* 706083 : v4l2src: UVC Allocated buffers wrapped in GstBuffer get orphaned by GstBuffer API
* 707242 : qtmux: streamable and faststart property have no effect
* 707933 : matroskademux: Wrong UTF8 detection causes wrong detection of subtitle encoding
* 708501 : osxvideosink: fix segfault releasing the element
* 708622 : rtpjitterbuffer: fix various regressions
* 728501 : rtpaux/rtprtx: Unit tests are racy and take very long sometimes
* 719636 : deinterlace: alters caps in passthrough mode preventing hardware decode
* 727305 : matroskademux: Add support for A_OPUS
* 725632 : v4l2: Normalise control names in the same way as v4l2-ctl
* 345830 : qtdemux: better edit lists handling
* 636143 : avidemux: report creation date/time via GST_TAG_DATE_TIME
* 652986 : rtpjitterbuffer: events are not serialized
* 664339 : matroskamux: support for audio/x-adpcm
* 691570 : [isomp4/qtdemux] lots of critical warnings on this sample file
* 692787 : rtph264pay: No way to clear SPS and PPS in case of a new stream
* 705024 : aacparse: does not propagate downstream sample rate restriction upstream
* 705982 : mp4mux: HDLR box name string is not NULL terminated
* 708165 : videomixer: Store and forward tag events
* 709079 : rtpgstpay: Leaks memory
* 709093 : qtdemux: add HEVC support
* 709266 : matroska-demux leaks memory
* 709312 : videoflip: Add an automatic method that flip base on image-orientation tag
* 710415 : hdv1394src: Not possible to select a HDV camera from GUID
* 710762 : qtdemux: fails reading some MOV files with problematic jpeg frames
* 711010 : videomixer: remove unneeded guint comparison
* 711011 : y4mencode: fix uninitialized variable warning
* 711013 : osxvideosink: fix missing selector name warning
* 711084 : rtpmanager: add new rtprtxsend and rtprtxreceive elements for retransmission
* 711087 : rtpbin: Support Auxiliary streams
* 711270 : check: add rtpsession test
* 711411 : rtpjitterbuffer: implement RTX statistics
* 711412 : rtpjitterbuffer: Automatically calculate RTX properties based on RTT
* 711560 : rtpsession: ssrc collision improvements
* 711693 : rtpsession: Implement various session statistics
* 712206 : v4l2: print FOURCC before enumerating
* 712254 : multifilesrc: Implement seeking in case of multiple images
* 712303 : qtdemux: playback regression after commit ae1150e85cf99d7482933aa6f7e4f012fe45a3ec
* 712567 : rtpsession: RBs are not included in SRs after the first RTCP timeout when there are multiple internal senders
* 712612 : v4l2bufferpool: take over the floating reference for the new allocator
* 712754 : v4l2: add support for multi-planar V4L2 API
* 719434 : rtph264pay maps and unmaps inbuffer twice
* 719497 : videoflip: crashes on tag list without orientation tag in git master
* 719783 : qtdemux: regression with mp4-main-multi-mpd-AV-NBS.mpd
* 719829 : rtp: Add RFC4571 framing/de-framing element
* 719938 : rtpbin: allow dynamic RTP/RTCP encoders and decoders
* 720371 : rtpbin: Impossible to set jitterbuffer rtx-* properties when using rtpbin
* 720512 : flacparse: Doesn't set the codec tag
* 720568 : v4l2: Various changes to allow using M2M decoders
* 720995 : matroskamux: add g726 adpcm support
* 721245 : osxvideosink: Fails to build on OS X Leopard 10.5.8
* 721342 : shout2send: Some minor cleanups
* 722175 : rtpmanager: improve code of rtprtx* elements
* 722370 : rtprtxsend: push rtx buffers from a different thread to avoid long retransmission delays
* 722372 : rtpjitterbuffer: Got data flow before stream-start event
* 722394 : v4l2: set GST_BUFFER_FLAG_DELTA_UNIT when appropriate
* 722396 : avimux: don't make the buffer writable unless absolutely necessary
* 722866 : rtspsrc: add rtpjitterbuffer do-retransmission property
* 722981 : autodetect elements have inconsistent handling of autoplugin
* 723166 : qtdemux: incorrect tag character
* 723269 : matroskamux: used uid list grows forever
* 723289 : cairooverlay: add RGB16 support
* 723502 : gst-plugins-good: Do not build check tests for disabled plugins
* 723849 : matroska: add support for GRAY8, BGR and RGB video colourspaces in V_UNCOMPRESSED codec
* 724085 : gst-plugins-good/docs: Rebalance docbook < para > tags in comments
* 724213 : rtph264pay: shouldn't update time for sending SPS and PPS if we failed to send SPS or PPS
* 724396 : rtspsrc: add tls-database property
* 724636 : v4l2videodec: VP8 KO with playbin
* 724705 : videomixer: Port to new collectpads API
* 724712 : rtspsrc doesn't set caps on pads before adding them
* 724899 : v4l2src does not set interlaced flag on buffers
* 725008 : matroskademux: crash with 24bit raw audio
* 725159 : rtpjitterbuffer: RTP sequence number rollover problems
* 725361 : [regression] rtpsession: setting the " internal-ssrc " property does nothing
* 725480 : gst-plugins-good: Ignore gcov intermediate files
* 725723 : osxvideo: GetCurrentProcess not available on Mavericks
* 725948 : videomixer2: crash after renegotiating with different resolution
* 726106 : matroskademux: does not handle fps lower than 1
* 726161 : png plugin handles interlaced png files incorrectly
* 726696 : rtspsrc memleaks
* 726737 : osxvideosink: advertize for the video meta API support
* 726738 : osxvideosink: use the video frame API instead of the video meta API
* 726833 : ximagesrc: Add alpha channel support
* 726837 : rtspsrc segfault
* 727821 : souphttpsrc: Regression in push mode
* 727867 : qtdemux: Does not return stream flags from trex atom
* 727878 : qtdemux: replace duplicated variable when parsing trex atom
* 729223 : wavparse: drops upstream tags for .wav files that are ID3 tagged
* 728987 : qtdemux: 'caps' may be used uninitialized in this function.
* 729067 : goom filter: left shift of 24 places cannot be represented in type 'int'
* 712333 : regression: videoflip: aborts with gst_video_flip_transform_caps: code should not be reached
* 722077 : v4l2: compile error - 'V4L2_CAP_VIDEO_M2M_MPLANE' undeclared
* 722127 : v4l2: Add NV12_64Z32 support
* 722128 : v4l2: Implement video decoder
* 723446 : v4l2src: Should detect support for mplanar formats during runtime
* 726453 : v4l2 plugin broken due to bundeling of videodev2.h
* 721764 : souphttpsrc: Add ability to do HTTP session logging
* 722311 : matroskaparse: should try to identify data on stream header before going with a blind
* 722705 : Factor out common init/reset code from matroska parse/demux
* 712643 : qtdemux: couple of issues with vobsub
* 581295 : mp4mux: Add support for embedded subtitles
==== Download ====
......@@ -98,11 +201,70 @@ subscribe to the gstreamer-devel list.
Contributors to this release
* Akihiro Tsukada
* Aleix Conchillo Flaque
* Aleix Conchillo Flaqué
* Alessandro Decina
* Alexander Zallesov
* Andoni Morales Alastruey
* Benjamin Gaignard
* Branislav Katreniak
* Brendan Long
* Christian Fredrik Kalager Schaller
* Dan Kegel
* Darryl Gamroth
* Djalma Lúcio Soares da Silva
* Edward Hervey
* George Kiagiadakis
* Göran Jönsson
* Hans Månsson
* Hugues Fruchet
* Jake Foytik
* Jan Schmidt
* Jeremy Huddleston
* Jeremy Huddleston Sequoia
* Jimmy Ohn
* Jonas Holmberg
* Josep Torra
* Julien Isorce
* Justin Joy
* Luis de Bethencourt
* Marc Leeman
* Mark Nauwelaerts
* Matej Knopp
* Mathieu Duponchelle
* MathieuDuponchelle
* Matthieu Bouron
* Michael Olbrich
* Mike Sheldon
* Nicola Murino
* Nicolas Dufresne
* Ognyan Tonchev
* Olivier Crête
* Robert Krakora
* Paul HENRYS
* Per x Johansson
* Peter Korsgaard
* Philippe Normand
* Rafał Mużyło
* Reynaldo H. Verdejo Pinochet
* Rico Tzschichholz
* Ryan Lortie
* Santiago Carot-Nemesio
* Sebastian Dröge
* Sebastian Rasmussen
* Simon Farnsworth
* Sreerenj Balachandran
* Stefan Sauer
* Stéphane Cerveau
* Thiago Santos
* Thibault Saunier
* Thijs Vermeir
* Tim-Philipp Müller
* Todd Agulnick
* Torrie Fischer
* Vincent Penquerc'h
* William Jon McCann
* William Manley
* Wim Taymans
* divhaere
 
\ No newline at end of file
......@@ -5,7 +5,7 @@ dnl please read gstreamer/docs/random/autotools before changing this file
dnl initialize autoconf
dnl releases only do -Wall, git and prerelease does -Werror too
dnl use a three digit version number for releases, and four for git/pre
AC_INIT([GStreamer Good Plug-ins],[1.3.0.1],[http://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer],[gst-plugins-good])
AC_INIT([GStreamer Good Plug-ins],[1.3.1],[http://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer],[gst-plugins-good])
AG_GST_INIT
......@@ -43,11 +43,11 @@ AC_DEFINE_UNQUOTED(GST_API_VERSION, "$GST_API_VERSION",
[GStreamer API Version])
AG_GST_LIBTOOL_PREPARE
AS_LIBTOOL(GST, 300, 0, 300)
AS_LIBTOOL(GST, 301, 0, 301)
dnl *** required versions of GStreamer stuff ***
GST_REQ=1.3.0.1
GSTPB_REQ=1.3.0.1
GST_REQ=1.3.1
GSTPB_REQ=1.3.1
dnl *** autotools stuff ****
......
......@@ -988,6 +988,26 @@
<DEFAULT></DEFAULT>
</ARG>
<ARG>
<NAME>GstRTSPSrc::tls-database</NAME>
<TYPE>GTlsDatabase*</TYPE>
<RANGE></RANGE>
<FLAGS>rw</FLAGS>
<NICK>TLS database</NICK>
<BLURB>TLS database with anchor certificate authorities used to validate the server certificate.</BLURB>
<DEFAULT></DEFAULT>
</ARG>
<ARG>
<NAME>GstRTSPSrc::tls-validation-flags</NAME>
<TYPE>GTlsCertificateFlags</TYPE>
<RANGE></RANGE>
<FLAGS>rw</FLAGS>
<NICK>TLS validation flags</NICK>
<BLURB>TLS certificate validation flags used to validate the server certificate.</BLURB>
<DEFAULT>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</DEFAULT>
</ARG>
<ARG>
<NAME>GstRTPDec::skip</NAME>
<TYPE>gint</TYPE>
......@@ -1264,7 +1284,7 @@
<RANGE></RANGE>
<FLAGS>rw</FLAGS>
<NICK>ip</NICK>
<BLURB>ip.</BLURB>
<BLURB>IP address or hostname.</BLURB>
<DEFAULT>"127.0.0.1"</DEFAULT>
</ARG>
......@@ -1334,7 +1354,7 @@
<RANGE></RANGE>
<FLAGS>rw</FLAGS>
<NICK>url</NICK>
<BLURB>url.</BLURB>
<BLURB>the stream's homepage URL.</BLURB>
<DEFAULT>""</DEFAULT>
</ARG>
......@@ -2432,7 +2452,7 @@
<NAME>GstVideoFlip::method</NAME>
<TYPE>GstVideoFlipMethod</TYPE>
<RANGE></RANGE>
<FLAGS>rw</FLAGS>
<FLAGS>rwx</FLAGS>
<NICK>method</NICK>
<BLURB>method.</BLURB>
<DEFAULT>Identity (no rotation)</DEFAULT>
......@@ -3214,7 +3234,7 @@
<RANGE>>= G_MAXULONG</RANGE>
<FLAGS>rw</FLAGS>
<NICK>Maximum Consecutive Decoding Errors</NICK>
<BLURB>Error out after receiving N consecutive decoding errors (-1 = never fail, 0 = automatic, 1 = fail on first error).</BLURB>
<BLURB>(Deprecated) Error out after receiving N consecutive decoding errors (-1 = never fail, 0 = automatic, 1 = fail on first error).</BLURB>
<DEFAULT>0</DEFAULT>
</ARG>
......@@ -4165,7 +4185,77 @@
<FLAGS>rw</FLAGS>
<NICK>timeout</NICK>
<BLURB>Value in seconds to timeout a blocking I/O (0 = No timeout).</BLURB>
<DEFAULT>0</DEFAULT>
<DEFAULT>15</DEFAULT>
</ARG>
<ARG>
<NAME>GstSoupHTTPSrc::compress</NAME>
<TYPE>gboolean</TYPE>
<RANGE></RANGE>
<FLAGS>rw</FLAGS>
<NICK>Compress</NICK>
<BLURB>Allow compressed content encodings.</BLURB>
<DEFAULT>FALSE</DEFAULT>
</ARG>
<ARG>
<NAME>GstSoupHTTPSrc::http-log-level</NAME>
<TYPE>SoupLoggerLogLevel</TYPE>
<RANGE></RANGE>
<FLAGS>rw</FLAGS>
<NICK>HTTP log level</NICK>
<BLURB>Set log level for soup's HTTP session log.</BLURB>
<DEFAULT>SOUP_LOGGER_LOG_HEADERS</DEFAULT>
</ARG>
<ARG>
<NAME>GstSoupHTTPSrc::keep-alive</NAME>
<TYPE>gboolean</TYPE>
<RANGE></RANGE>
<FLAGS>rw</FLAGS>
<NICK>keep-alive</NICK>
<BLURB>Use HTTP persistent connections.</BLURB>
<DEFAULT>FALSE</DEFAULT>
</ARG>
<ARG>
<NAME>GstSoupHTTPSrc::retries</NAME>
<TYPE>gint</TYPE>
<RANGE>>= G_MAXULONG</RANGE>
<FLAGS>rw</FLAGS>
<NICK>Retries</NICK>
<BLURB>Maximum number of retries until giving up (-1=infinite).</BLURB>
<DEFAULT>3</DEFAULT>
</ARG>
<ARG>
<NAME>GstSoupHTTPSrc::ssl-ca-file</NAME>
<TYPE>gchar*</TYPE>
<RANGE></RANGE>
<FLAGS>rw</FLAGS>
<NICK>SSL CA File</NICK>
<BLURB>Location of a SSL anchor CA file to use.</BLURB>
<DEFAULT>NULL</DEFAULT>
</ARG>
<ARG>
<NAME>GstSoupHTTPSrc::ssl-strict</NAME>
<TYPE>gboolean</TYPE>
<RANGE></RANGE>
<FLAGS>rw</FLAGS>
<NICK>SSL Strict</NICK>
<BLURB>Strict SSL certificate checking.</BLURB>
<DEFAULT>TRUE</DEFAULT>
</ARG>
<ARG>
<NAME>GstSoupHTTPSrc::ssl-use-system-ca-file</NAME>
<TYPE>gboolean</TYPE>
<RANGE></RANGE>
<FLAGS>rw</FLAGS>
<NICK>Use System CA File</NICK>
<BLURB>Use system CA file.</BLURB>
<DEFAULT>TRUE</DEFAULT>
</ARG>
<ARG>
......@@ -21335,7 +21425,7 @@
<FLAGS>rw</FLAGS>
<NICK>RTX Delay</NICK>
<BLURB>Extra time in ms to wait before sending retransmission event (-1 automatic).</BLURB>
<DEFAULT>20</DEFAULT>
<DEFAULT>-1</DEFAULT>
</ARG>
<ARG>
......@@ -21355,7 +21445,7 @@
<FLAGS>rw</FLAGS>
<NICK>RTX Retry Period</NICK>
<BLURB>Try to get a retransmission for this many ms (-1 automatic).</BLURB>
<DEFAULT>160</DEFAULT>
<DEFAULT>-1</DEFAULT>
</ARG>
<ARG>
......@@ -21365,7 +21455,17 @@
<FLAGS>rw</FLAGS>
<NICK>RTX Retry Timeout</NICK>
<BLURB>Retry sending a transmission event after this timeout in ms (-1 automatic).</BLURB>
<DEFAULT>40</DEFAULT>
<DEFAULT>-1</DEFAULT>
</ARG>
<ARG>
<NAME>GstRtpJitterBuffer::stats</NAME>
<TYPE>GstStructure*</TYPE>
<RANGE></RANGE>
<FLAGS>r</FLAGS>
<NICK>Statistics</NICK>
<BLURB>Various statistics.</BLURB>
<DEFAULT></DEFAULT>
</ARG>
<ARG>
......@@ -21488,6 +21588,16 @@
<DEFAULT>2</DEFAULT>
</ARG>
<ARG>
<NAME>GstRtpSession::stats</NAME>
<TYPE>GstStructure*</TYPE>
<RANGE></RANGE>
<FLAGS>r</FLAGS>
<NICK>Statistics</NICK>
<BLURB>Various statistics.</BLURB>
<DEFAULT></DEFAULT>
</ARG>
<ARG>
<NAME>GstRtpRtxSend::rtx-payload-type</NAME>
<TYPE>guint</TYPE>
......@@ -21503,7 +21613,7 @@
<TYPE>guint</TYPE>
<RANGE></RANGE>
<FLAGS>rw</FLAGS>
<NICK>Max Size Times</NICK>
<NICK>Max Size Time</NICK>
<BLURB>Amount of ms to queue (0 = unlimited).</BLURB>
<DEFAULT>0</DEFAULT>
</ARG>
......@@ -21511,11 +21621,11 @@
<ARG>
<NAME>GstRtpRtxSend::max-size-packets</NAME>
<TYPE>guint</TYPE>
<RANGE></RANGE>
<RANGE><= 32767</RANGE>
<FLAGS>rw</FLAGS>
<NICK>Max Size Packets</NICK>
<BLURB>Amount of packets to queue (0 = unlimited).</BLURB>
<DEFAULT>0</DEFAULT>
<DEFAULT>100</DEFAULT>
</ARG>
<ARG>
......@@ -21533,11 +21643,31 @@
<TYPE>guint</TYPE>
<RANGE></RANGE>
<FLAGS>r</FLAGS>
<NICK>Num RTX Packet</NICK>
<BLURB>Number of retransmission packets sent.</BLURB>
<NICK>Num RTX Packets</NICK>
<BLURB> Number of retransmission packets sent.</BLURB>
<DEFAULT>0</DEFAULT>
</ARG>
<ARG>
<NAME>GstRtpRtxSend::payload-type-map</NAME>
<TYPE>GstStructure*</TYPE>
<RANGE></RANGE>
<FLAGS>rw</FLAGS>
<NICK>Payload Type Map</NICK>
<BLURB>Map of original payload types to their retransmission payload types.</BLURB>
<DEFAULT></DEFAULT>
</ARG>
<ARG>
<NAME>GstRtpRtxSend::ssrc-map</NAME>
<TYPE>GstStructure*</TYPE>
<RANGE></RANGE>
<FLAGS>w</FLAGS>
<NICK>SSRC Map</NICK>
<BLURB>Map of SSRCs to their retransmission SSRCs for SSRC-multiplexed mode (default = random).</BLURB>
<DEFAULT></DEFAULT>
</ARG>
<ARG>
<NAME>GstRtpRtxReceive::rtx-payload-types</NAME>
<TYPE>string</TYPE>
......@@ -21563,8 +21693,8 @@
<TYPE>guint</TYPE>
<RANGE></RANGE>
<FLAGS>r</FLAGS>
<NICK>Num RTX Packet</NICK>
<BLURB>Number of retransmission packets received.</BLURB>
<NICK>Num RTX Packets</NICK>
<BLURB> Number of retransmission packets received.</BLURB>
<DEFAULT>0</DEFAULT>
</ARG>
......@@ -21574,10 +21704,20 @@
<RANGE></RANGE>
<FLAGS>r</FLAGS>
<NICK>Num RTX Associated Packets</NICK>
<BLURB>correctly associated with retransmission requests.</BLURB>
<BLURB>Number of retransmission packets correctly associated with retransmission requests.</BLURB>
<DEFAULT>0</DEFAULT>
</ARG>
<ARG>
<NAME>GstRtpRtxReceive::payload-type-map</NAME>
<TYPE>GstStructure*</TYPE>
<RANGE></RANGE>
<FLAGS>rw</FLAGS>
<NICK>Payload Type Map</NICK>
<BLURB>Map of original payload types to their retransmission payload types.</BLURB>
<DEFAULT></DEFAULT>
</ARG>
<ARG>
<NAME>GstV4l2Sink::device</NAME>
<TYPE>gchar*</TYPE>
......@@ -23088,6 +23228,16 @@
<DEFAULT>""</DEFAULT>
</ARG>