Commit 365962fe authored by Sebastian Dröge's avatar Sebastian Dröge 🍵

Release 1.2.0

parent e679d1a1
=== release 1.2.0 ===
2013-09-24 Sebastian Dröge <>
releasing 1.2.0
2013-09-20 19:43:21 +0200 Andoni Morales Alastruey <>
* sys/osxvideo/osxvideosink.m:
osxvideosink: fix segfault releasing the sink
show_frame is deferred to the main thread and can be called
when the sink has been released, so we need to keep an extra ref
on ObjectiveC object helper.
2013-09-19 17:11:34 -0400 Robert Krakora <>
* sys/v4l2/gstv4l2bufferpool.c:
v4l2bufferpool: Restore original GstMemory in buffer if it has been changed
2013-09-23 16:34:15 +0200 Wim Taymans <>
* gst/rtpmanager/gstrtpjitterbuffer.c:
* gst/rtpmanager/gstrtpsession.c:
rtpmanager: update docs
2013-09-23 15:36:32 +0200 Wim Taymans <>
* gst/rtpmanager/gstrtpbin.c:
* gst/rtpmanager/gstrtpjitterbuffer.c:
* gst/rtpmanager/gstrtpptdemux.c:
* gst/rtpmanager/gstrtpsession.c:
* gst/rtpmanager/gstrtpssrcdemux.c:
docs: update docs with 1.0 element names
2013-09-23 14:13:30 +0200 Wim Taymans <>
* tests/check/elements/rtpjitterbuffer.c:
tests: add test for retransmission because of reordering
2013-09-23 14:12:03 +0200 Wim Taymans <>
* gst/rtpmanager/gstrtpjitterbuffer.c:
rtpjitterbuffer: always store lost event in jitterbuffer
Always prepare a lost event in the jitterbuffer, it is to wake up and make the
pushing thread continue. We drop the event when we are not supposed to push lost
events downstream.
2013-09-23 11:18:46 +0200 Wim Taymans <>
* gst/rtpmanager/gstrtpjitterbuffer.c:
rtpjitterbuffer: schedule lost event differently
Schedule the lost event by placing it inside the jitterbuffer with the seqnum
that was lost so that the pushing thread can interleave and push it properly.
2013-09-23 11:17:34 +0200 Wim Taymans <>
* tests/check/elements/rtpjitterbuffer.c:
tests: remove timeouts from check
Timeouts make the test unreliable and are not needed.
2013-09-23 11:15:30 +0200 Wim Taymans <>
* gst/rtpmanager/rtpjitterbuffer.c:
rtpjitterbuffer: remove list debug
2013-09-23 11:14:01 +0200 Wim Taymans <>
* gst/rtpmanager/gstrtpjitterbuffer.c:
* gst/rtpmanager/rtpjitterbuffer.h:
rtpjitterbuffer: add type to the item
So that the upper layer can know what data is contained in the item.
2013-09-23 09:58:32 +0200 Wim Taymans <>
* gst/rtpmanager/gstrtpjitterbuffer.c:
* gst/rtpmanager/rtpjitterbuffer.c:
* gst/rtpmanager/rtpjitterbuffer.h:
rtpjitterbuffer: fix flush
Pass function to flush to properly free the queue items.
2013-09-21 00:08:20 +0200 Wim Taymans <>
* gst/rtpmanager/rtpjitterbuffer.c:
rtpjitterbuffer: append seqnum -1 packets
2013-09-20 23:48:20 +0200 Wim Taymans <>
* gst/rtpmanager/gstrtpjitterbuffer.c:
* gst/rtpmanager/rtpjitterbuffer.c:
* gst/rtpmanager/rtpjitterbuffer.h:
rtpjitterbuffer: use structure to hold packet information
Make the jitterbuffer operate on a structure containing all the packet
information. This avoids mapping the buffer multiple times just to get the RTP
information. It will also make it possible to store other miniobjects such as
events later.
2013-09-20 17:48:52 +0200 Wim Taymans <>
* gst/rtpmanager/gstrtpjitterbuffer.c:
rtpjitterbuffer: update expected timer when possible
When we receive a packet and we have some missing packets, we can update their
estimated arrival times based on the timestamp difference.
2013-09-20 17:18:27 +0200 Wim Taymans <>
* gst/rtpmanager/gstrtpjitterbuffer.c:
rtpjitterbuffer: fix order of timeout events
Improve the order of the timeout events, if there are timers with the same
timeout, we want to trigger the lowest seqnum first. For this we need to loop
over the complete array of timers to find the best one before triggering the
2013-09-20 16:58:38 +0200 Wim Taymans <>
* gst/rtpmanager/gstrtpjitterbuffer.c:
rtpjitterbuffer: send lost event before signaling next buffer
First send the lost event, then update the next_seqnum counter and then
send the signal to the pushing thread that it can retry to push a buffer. This
avoids pushing out buffers before the lost event is pushed.
2013-09-20 15:35:25 +0200 Wim Taymans <>
* gst/rtpmanager/gstrtpjitterbuffer.c:
* gst/rtpmanager/rtpjitterbuffer.c:
* gst/rtpmanager/rtpjitterbuffer.h:
jitterbuffer: configure clock-rate on jitterbuffer
Add a get and setter to configure the clock-rate in the jitterbuffer instead of
passing it as an argument to the insert method.
2013-09-20 12:29:39 +0200 Wim Taymans <>
* tests/check/elements/rtpjitterbuffer.c:
tests: add test for packet delay and retransmission
2013-09-20 12:27:26 +0200 Wim Taymans <>
* gst/rtpmanager/gstrtpjitterbuffer.c:
rtpjitterbuffer: add option to reset retransmission timers
2013-09-20 12:25:43 +0200 Wim Taymans <>
* gst/rtpmanager/gstrtpjitterbuffer.c:
rtpjitterbuffer: stop the timer thread
The timeout code could release the lock so we need to check if we are allowed to
wait for the clock some more.
2013-09-20 12:25:12 +0200 Wim Taymans <>
* gst/rtpmanager/gstrtpjitterbuffer.c:
rtpjitterbuffer: unlock only once
2013-09-20 11:30:04 +0200 Wim Taymans <>
* tests/check/elements/rtpjitterbuffer.c:
tests: check both PTS and DTS
2013-09-20 10:55:03 +0200 Wim Taymans <>
* tests/check/elements/rtpjitterbuffer.c:
tests: add unit-test for multiple missing packets
Check if multiple missing packets generate retransmission events and that the
retranmission requests are canceled when the missing packet arrives.
2013-09-20 10:53:29 +0200 Wim Taymans <>
* gst/rtpmanager/gstrtpjitterbuffer.c:
rtpjitterbuffer: improve flush and shutdown
There is no need to unschedule the timer in flush-start, flush-stop will remove
the timers and unschedule.
Unschedule the current timer before attempting to join the timer thread.
2013-09-20 10:43:53 +0200 Wim Taymans <>
* tests/check/elements/rtpjitterbuffer.c:
tests: improve debug
2013-09-20 10:42:27 +0200 Wim Taymans <>
* gst/rtpmanager/gstrtpjitterbuffer.c:
rtpjitterbuffer: set correct expected time
When we already have a timer for a packet, skip it but don't forget to adjust
the dts to the expected dts of the next packet.
2013-09-20 10:41:59 +0200 Wim Taymans <>
* gst/rtpmanager/gstrtpjitterbuffer.c:
jitterbuffer: improve debug
2013-09-19 16:55:25 +0200 Wim Taymans <>
* gst/alpha/gstalpha.c:
alpha: use POFFSET instead of OFFSET
Use the more correct POFFSET macro to get the offset of a component in its
plane. The offset macro gives the offset of the component relative to the start
of the frame.
2013-09-21 18:46:29 +0200 Sebastian Dröge <>
* gst/goom/mmx.h:
goom: Fix MMX assembly compilation with clang
clang does not want or need a clobber list for emms:
error: clobbers must be last on the x87 stack
Patch taken from the FreeBSD ports, provided by
Dan McGregor <>
2013-09-20 16:16:57 +0200 Edward Hervey <>
* common:
Automatic update of common submodule
From b613661 to 6b03ba7
2013-09-20 10:19:22 +0200 Sebastian Dröge <>
* gst/matroska/matroska-demux.c:
matroska-demux: Make sure that subtitle buffers are \0-terminated
2013-09-17 12:17:54 +0200 Andoni Morales Alastruey <>
* gst/isomp4/gstqtmux.c:
qtmux: handle issues correctly when downstream is not seekable
The streamable property only make sense for fragmented formats.
For regular MP4, when downstream is not seekable we can't rewrite
the headers, so qtmux can only work with fast-start=TRUE, where
the headers are written finishing the file.
For fragmented MP4, when streamable is not seekable and the streamable
property is FALSE, we must enforce streamable=TRUE warning the user
about this change
2013-09-17 12:06:06 +0200 Andoni Morales Alastruey <>
* gst/isomp4/gstqtmux.c:
qtmux: make "streamable" TRUE as default
The most common use case for fragmented MP4 (Dash and Smooth Streaming)
is producing streamable content (even for VOD). streamable=FALSE would only
be used to generate fragmented MP4 with and index of MOOF's that could
be reproduced without a playlist/manifest
2013-09-17 12:01:30 +0200 Andoni Morales Alastruey <>
* gst/isomp4/gstqtmux.c:
qtmux: deprecate the streamable property for non-fragmented MP4
The streamable property only makes sense for fragmented MP4.
2013-09-19 17:08:19 -0400 Olivier Crête <>
* sys/v4l2/gstv4l2bufferpool.h:
v4l2: Remove commented out line
2013-09-19 18:43:08 +0100 Tim-Philipp Müller <>
* common:
Automatic update of common submodule
From 74a6857 to b613661
2013-09-19 17:35:27 +0100 Tim-Philipp Müller <>
* common:
Automatic update of common submodule
From 098c0d7 to 74a6857
2013-09-19 16:50:44 +0200 Wim Taymans <>
* gst/alpha/gstalpha.c:
alpha: don't assume planar formats have just 1 block
Don't assume planar formats have just one memory block with the data but use the
macros to access the right memory block where a component can be found.
2013-09-19 14:14:52 +0200 Wim Taymans <>
* tests/check/elements/rtpjitterbuffer.c:
tests: add retransmission jitterbuffer test
Store both DTS and PTS on buffers.
Make a queue for srcpad events.
Activate pads after linking so that we don't get RECONFIGURE events.
Add test for retransmission.
2013-09-19 14:12:18 +0200 Wim Taymans <>
* gst/rtpmanager/gstrtpjitterbuffer.c:
rtpjitterbuffer: keep delay as a separate variable in timer
Keep a separate delay in the timer so that we still know the original timestamp
of the packet that this timer refers to. We can then place the correct
running-time in the Retransmission event.
2013-09-19 14:08:56 +0200 Wim Taymans <>
* gst/rtpmanager/gstrtpjitterbuffer.c:
rtpjitterbuffer: fix writability of properties
2013-09-19 11:34:57 +0200 Sebastian Dröge <>
Back to development
=== release 1.1.90 ===
2013-09-19 Sebastian Dröge <>
2013-09-19 10:50:23 +0200 Sebastian Dröge <>
* ChangeLog:
releasing 1.1.90
* docs/plugins/gst-plugins-good-plugins.args:
* docs/plugins/gst-plugins-good-plugins.hierarchy:
* docs/plugins/inspect/plugin-1394.xml:
* docs/plugins/inspect/plugin-aasink.xml:
* docs/plugins/inspect/plugin-alaw.xml:
* docs/plugins/inspect/plugin-alpha.xml:
* docs/plugins/inspect/plugin-alphacolor.xml:
* docs/plugins/inspect/plugin-apetag.xml:
* docs/plugins/inspect/plugin-audiofx.xml:
* docs/plugins/inspect/plugin-audioparsers.xml:
* docs/plugins/inspect/plugin-auparse.xml:
* docs/plugins/inspect/plugin-autodetect.xml:
* docs/plugins/inspect/plugin-avi.xml:
* docs/plugins/inspect/plugin-cacasink.xml:
* docs/plugins/inspect/plugin-cairo.xml:
* docs/plugins/inspect/plugin-cutter.xml:
* docs/plugins/inspect/plugin-debug.xml:
* docs/plugins/inspect/plugin-deinterlace.xml:
* docs/plugins/inspect/plugin-dtmf.xml:
* docs/plugins/inspect/plugin-dv.xml:
* docs/plugins/inspect/plugin-effectv.xml:
* docs/plugins/inspect/plugin-equalizer.xml:
* docs/plugins/inspect/plugin-flac.xml:
* docs/plugins/inspect/plugin-flv.xml:
* docs/plugins/inspect/plugin-flxdec.xml:
* docs/plugins/inspect/plugin-gdkpixbuf.xml:
* docs/plugins/inspect/plugin-goom.xml:
* docs/plugins/inspect/plugin-goom2k1.xml:
* docs/plugins/inspect/plugin-icydemux.xml:
* docs/plugins/inspect/plugin-id3demux.xml:
* docs/plugins/inspect/plugin-imagefreeze.xml:
* docs/plugins/inspect/plugin-interleave.xml:
* docs/plugins/inspect/plugin-isomp4.xml:
* docs/plugins/inspect/plugin-jack.xml:
* docs/plugins/inspect/plugin-jpeg.xml:
* docs/plugins/inspect/plugin-level.xml:
* docs/plugins/inspect/plugin-matroska.xml:
* docs/plugins/inspect/plugin-mulaw.xml:
* docs/plugins/inspect/plugin-multifile.xml:
* docs/plugins/inspect/plugin-multipart.xml:
* docs/plugins/inspect/plugin-navigationtest.xml:
* docs/plugins/inspect/plugin-oss4.xml:
* docs/plugins/inspect/plugin-ossaudio.xml:
* docs/plugins/inspect/plugin-png.xml:
* docs/plugins/inspect/plugin-pulseaudio.xml:
* docs/plugins/inspect/plugin-replaygain.xml:
* docs/plugins/inspect/plugin-rtp.xml:
* docs/plugins/inspect/plugin-rtpmanager.xml:
* docs/plugins/inspect/plugin-rtsp.xml:
* docs/plugins/inspect/plugin-shapewipe.xml:
* docs/plugins/inspect/plugin-shout2send.xml:
* docs/plugins/inspect/plugin-smpte.xml:
* docs/plugins/inspect/plugin-soup.xml:
* docs/plugins/inspect/plugin-spectrum.xml:
* docs/plugins/inspect/plugin-speex.xml:
* docs/plugins/inspect/plugin-taglib.xml:
* docs/plugins/inspect/plugin-udp.xml:
* docs/plugins/inspect/plugin-video4linux2.xml:
* docs/plugins/inspect/plugin-videobox.xml:
* docs/plugins/inspect/plugin-videocrop.xml:
* docs/plugins/inspect/plugin-videofilter.xml:
* docs/plugins/inspect/plugin-videomixer.xml:
* docs/plugins/inspect/plugin-vpx.xml:
* docs/plugins/inspect/plugin-wavenc.xml:
* docs/plugins/inspect/plugin-wavpack.xml:
* docs/plugins/inspect/plugin-wavparse.xml:
* docs/plugins/inspect/plugin-ximagesrc.xml:
* docs/plugins/inspect/plugin-y4menc.xml:
* gst-plugins-good.doap:
* win32/common/config.h:
Release 1.1.90
2013-09-19 10:21:42 +0200 Sebastian Dröge <>
* po/af.po:
* po/az.po:
* po/bg.po:
* po/ca.po:
* po/cs.po:
* po/da.po:
* po/de.po:
* po/el.po:
* po/en_GB.po:
* po/eo.po:
* po/es.po:
* po/eu.po:
* po/fi.po:
* po/fr.po:
* po/gl.po:
* po/hr.po:
* po/hu.po:
* po/id.po:
* po/it.po:
* po/ja.po:
* po/lt.po:
* po/lv.po:
* po/mt.po:
* po/nb.po:
* po/nl.po:
* po/or.po:
* po/pl.po:
* po/pt_BR.po:
* po/ro.po:
* po/ru.po:
* po/sk.po:
* po/sl.po:
* po/sq.po:
* po/sr.po:
* po/sv.po:
* po/tr.po:
* po/uk.po:
* po/vi.po:
* po/zh_CN.po:
* po/zh_HK.po:
* po/zh_TW.po:
Update .po files
2013-09-19 09:45:18 +0200 Sebastian Dröge <>
This is GStreamer Good Plugins 1.1.90
This is GStreamer Good Plugins 1.2.0
Changes since 1.0:
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
• 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.
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
• 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
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
• 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
Release notes for GStreamer Good Plugins 1.1.90
Release notes for GStreamer Good Plugins 1.2.0
The GStreamer team is proud to announce a new bug-fix release
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.
......@@ -57,24 +57,11 @@ contains a set of codecs plugins based on libav (formerly gst-ffmpeg)
Bugs fixed in this release
* 646963 : rtpmanager: Only update last_rtcp_send_time when actually sending a report
* 687657 : rtpsession: Demux RTCP buffers from the RTP stream
* 695090 : rganalysis: fix tests
* 702004 : qtdemux: add support for the avc3 sample entry format of the AVC file format
* 704950 : videomixer: add colorspace conversion
* 705371 : souphttpsrc: Does network operations from the state change thread
* 706955 : autoaudiosink/autovideosink: Introduce " sync " property
* 707042 : udpsrc binds to INADDR_ANY instead of multicast group address
* 707303 : flacenc: 'Got data flow before stream-start event' warnings
* 707340 : qtmux: should NOT use PTS if DTS is missing
* 707411 : qtmux: what is purpose of MAX_TOLERATED_LATENESS?
* 707530 : qtdemux: Handle segments correctly in push mode seeks
* 707688 : matroskademux: leaking buffer and caps when parsing attachments
* 707734 : rtp: add payloader and depayloader for 24bit raw audio
* 707758 : v4l2: Incorrect UV plane stride value for NV12/NV21 formats
* 707886 : videomixer: build fails due to unrenamed files if orc is not installed
* 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
* 708131 : videomixer: undefined reference to `videomixer_video_convert_orc_convert_I420_BGRA'
* 708501 : osxvideosink: fix segfault releasing the element
* 708622 : rtpjitterbuffer: fix various regressions
==== Download ====
......@@ -111,20 +98,11 @@ subscribe to the gstreamer-devel list.
Contributors to this release
* Alex Ashley
* Andreea Fulger
* Bernhard Miller
* Christoph Reiter
* David Holroyd
* Jan Schmidt
* Matej Knopp
* Mathieu Duponchelle
* Andoni Morales Alastruey
* Edward Hervey
* Olivier Crête
* Rico Tzschichholz
* Robert Krakora
* Sebastian Dröge
* Seán de Búrca
* Thiago Santos
* Thibault Saunier
* Tim-Philipp Müller
* Wim Taymans
\ 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],[],[],[gst-plugins-good])
AC_INIT([GStreamer Good Plug-ins],[1.2.0],[],[gst-plugins-good])
[GStreamer API Version])
AS_LIBTOOL(GST, 190, 0, 190)
AS_LIBTOOL(GST, 200, 0, 200)
dnl *** required versions of GStreamer stuff ***
dnl *** autotools stuff ****
......@@ -21332,7 +21332,7 @@
<BLURB>Extra time in ms to wait before sending retransmission event (-1 automatic).</BLURB>
......@@ -21342,7 +21342,7 @@
<NICK>RTX Delay Reorder</NICK>
<BLURB>Sending retransmission event when this much reordering (-1 automatic).</BLURB>
......@@ -21352,7 +21352,7 @@