Commit d5a512b0 authored by Tim-Philipp Müller's avatar Tim-Philipp Müller 🐠

Release 1.14.1

parent 8fbfc4f3
=== release 1.14.1 ===
2018-05-17 13:21:47 +0100 Tim-Philipp Müller <>
* ChangeLog:
* gst-plugins-base.doap:
Release 1.14.1
2018-05-17 13:21:47 +0100 Tim-Philipp Müller <>
* docs/plugins/inspect/plugin-adder.xml:
* docs/plugins/inspect/plugin-alsa.xml:
* docs/plugins/inspect/plugin-app.xml:
* docs/plugins/inspect/plugin-audioconvert.xml:
* docs/plugins/inspect/plugin-audiomixer.xml:
* docs/plugins/inspect/plugin-audiorate.xml:
* docs/plugins/inspect/plugin-audioresample.xml:
* docs/plugins/inspect/plugin-audiotestsrc.xml:
* docs/plugins/inspect/plugin-cdparanoia.xml:
* docs/plugins/inspect/plugin-encoding.xml:
* docs/plugins/inspect/plugin-gio.xml:
* docs/plugins/inspect/plugin-libvisual.xml:
* docs/plugins/inspect/plugin-ogg.xml:
* docs/plugins/inspect/plugin-opengl.xml:
* docs/plugins/inspect/plugin-opus.xml:
* docs/plugins/inspect/plugin-pango.xml:
* docs/plugins/inspect/plugin-pbtypes.xml:
* docs/plugins/inspect/plugin-playback.xml:
* docs/plugins/inspect/plugin-rawparse.xml:
* docs/plugins/inspect/plugin-subparse.xml:
* docs/plugins/inspect/plugin-tcp.xml:
* docs/plugins/inspect/plugin-theora.xml:
* docs/plugins/inspect/plugin-typefindfunctions.xml:
* docs/plugins/inspect/plugin-videoconvert.xml:
* docs/plugins/inspect/plugin-videorate.xml:
* docs/plugins/inspect/plugin-videoscale.xml:
* docs/plugins/inspect/plugin-videotestsrc.xml:
* docs/plugins/inspect/plugin-volume.xml:
* docs/plugins/inspect/plugin-vorbis.xml:
* docs/plugins/inspect/plugin-ximagesink.xml:
* docs/plugins/inspect/plugin-xvimagesink.xml:
Update docs
2018-05-17 12:37:27 +0100 Tim-Philipp Müller <>
* po/hr.po:
Update translations
2018-05-17 13:54:35 +0900 hoonhee.lee <>
* gst/playback/gstplaybin3.c:
playbin3: Collect appropriate stream-type when doing stream selection
2017-12-13 12:30:54 +0100 Georg Lippitsch <>
* gst-libs/gst/video/gstvideotimecode.c:
videotimecode: Allow 24000/1001 frame rate
2018-05-02 18:39:31 +0300 Sebastian Dröge <>
* gst-libs/gst/app/gstappsink.c:
appsink: Handle unlock in drain query handling too
And also handle flushing, we might otherwise wait here forever when
flushing too.
2018-05-02 18:35:23 +0300 Sebastian Dröge <>
* gst-libs/gst/app/gstappsink.c:
appsink: Make sure to also handle unlock when waiting for EOS to be handled
Otherwise shutting down during EOS waiting will cause a deadlock.
2018-05-02 18:11:58 +0300 Sebastian Dröge <>
* gst-libs/gst/app/gstappsink.c:
* gst-libs/gst/app/gstappsrc.c:
appsrc/sink: Fix optimization for only signalling waiters if someone is actually waiting
It is possible that both application and the stream are waiting
currently, if for example the following happens:
1) app is waiting because no buffer in appsink
2) appsink providing a buffer and waking up app
3) appsink getting another buffer and waiting because it's full now
4) app thread getting back control
Previously step 4 would overwrite that the appsink is currently waiting,
so it would never be signalled again.
2018-05-13 23:31:22 +0100 Pierre Labastie <>
* tests/examples/gl/sdl/
examples: gl: sdl: link to the right in-tree libgstvideo
2018-05-12 17:19:50 +0100 Philippe Normand <>
* gst/subparse/gstsubparse.c:
subparse: follow-up build fix after d871b1205
2018-05-12 13:53:02 +0100 Philippe Normand <>
* gst/subparse/gstsubparse.c:
* tests/check/elements/subparse.c:
subparse: support for more than 32 unclosed markup tags
2018-05-10 01:54:36 +0900 Seungha Yang <>
* gst/playback/gstdecodebin3-parse.c:
* gst/playback/gstdecodebin3.c:
decodebin3: Do not modify structure of EOS event
2018-05-10 01:33:55 +0900 Seungha Yang <>
* gst/playback/gsturisourcebin.c:
urisourcebin: Do not modify structure of EOS event
2018-05-09 10:39:23 +0900 Seungha Yang <>
* gst/playback/gsturidecodebin3.c:
uridecodebin3: Fix GList leak
2018-05-08 23:44:38 +0900 Seungha Yang <>
* gst/playback/gsturidecodebin3.c:
uridecodebin3: Fix string leak
uri and suburi should be free'd
2018-05-04 10:35:36 +0200 Edward Hervey <>
* gst-libs/gst/video/video-chroma.c:
* gst-libs/gst/video/video-converter.c:
video: Silence "restrict" issues with ORC code
The problem is that even though the functions we are calling are
in-place transformation, orc automatically puts the restrict keyword
on all arguments. To silence that warning just create yet-another
variable containing the same value.
2018-04-30 17:17:22 +0200 Thibault Saunier <>
* gst-libs/gst/pbutils/descriptions.c:
pbutils: Avoid assertion describing raw audio caps without format
We used to get:
gst_audio_format_from_string: assertion 'format != NULL' failed
2018-04-22 10:49:29 -0300 Thibault Saunier <>
* gst/encoding/gstencodebin.c:
encodebin: Also lock input caps when dynamic output is disabled
With the way caps negotiation work in encoders, the only way to ensure
that no downstream renegotiation is done in the encoder is to also lock
upstream caps. Anyway with the current behavior upstream of encoders
*require* to handle any file format so locking upstream format should
be safe.
2018-04-30 19:49:20 +0900 Seungha Yang <>
* gst-libs/gst/tag/gsttagmux.c:
tagmux: Reset final tags for reusing element
If the output tag had been exposed, it never ever updated
even if we reset the tagmux using state change.
2018-04-20 12:30:22 +0200 Michael Olbrich <>
* tests/check/libs/videodecoder.c:
videodecoder: add test for event order
When frames are dropped or reordered then the serialized events are
collected and pushed with the next frame. This test verifies that the
order is preserved.
2018-03-08 11:28:58 +0100 Matthias Fend <>
* gst-libs/gst/video/gstvideodecoder.c:
videodecoder: keep event order
Since events are pushed out in reverse order, newer events need to
be added at the front of event lists
2018-04-25 09:28:53 +0900 hoonhee.lee <>
* gst/playback/gsturidecodebin3.c:
uridecodebin3: don't segfault if a pad is not a source pad when it is removed
Ignore to handling a pad of decodebin3 which doesn't have corresponding output
when it is removed.
2018-04-25 01:33:43 +0300 Sebastian Dröge <>
* gst-libs/gst/gl/cocoa/gstglwindow_cocoa.m:
gl/cocoa: Let ARC clean up our dispatch queue if ARC is used, and otherwise do it manually
Also don't use __bridge casts if ARC is not used, as is the case on 32
bit systems.
2018-04-25 01:08:58 +0300 Sebastian Dröge <>
* gst-libs/gst/gl/cocoa/gstglwindow_cocoa.m:
gl/cocoa: Use NSRect instead of CGRect
On 64 bit systems they're typedefs of each other but on 32 bit systems
not, and we pass the rect to an API that expects a NSRect
2018-04-20 21:54:23 +0200 Mark Nauwelaerts <>
* gst-libs/gst/gl/gstgldisplay.c:
* gst-libs/gst/gl/gstglmemory.c:
* gst-libs/gst/gl/gstglslstage.c:
* gst-libs/gst/gl/gstglupload.c:
gl: fix some GIR annotations
Mostly related to out and array parameters
2018-04-20 21:53:17 +0200 Mark Nauwelaerts <>
* gst-libs/gst/pbutils/codec-utils.c:
pbutils: fix some GIR annotations
Mostly related to out and array parameters
2018-04-20 21:53:16 +0200 Mark Nauwelaerts <>
* gst-libs/gst/video/gstvideometa.c:
* gst-libs/gst/video/video-color.c:
* gst-libs/gst/video/video-event.c:
* gst-libs/gst/video/video-info.c:
* gst-libs/gst/video/videoorientation.c:
video: fix some GIR annotations
Mostly related to out and array parameters
2018-04-20 21:53:16 +0200 Mark Nauwelaerts <>
* gst-libs/gst/audio/audio-converter.c:
* gst-libs/gst/audio/audio-info.c:
* gst-libs/gst/audio/gstaudiodecoder.c:
* gst-libs/gst/audio/gstaudioringbuffer.c:
audio: fix some GIR annotations
Mostly related to out and array parameters
2018-04-20 21:53:15 +0200 Mark Nauwelaerts <>
* gst-libs/gst/rtsp/gstrtspmessage.c:
* gst-libs/gst/rtsp/gstrtsprange.c:
* gst-libs/gst/rtsp/gstrtsptransport.c:
* gst-libs/gst/rtsp/gstrtspurl.c:
rtsp: fix some GIR annotations
Mostly related to out and array parameters.
2018-04-20 21:53:10 +0200 Mark Nauwelaerts <>
* gst-libs/gst/rtp/gstrtcpbuffer.c:
* gst-libs/gst/rtp/gstrtpbaseaudiopayload.c:
* gst-libs/gst/rtp/gstrtpbuffer.c:
* gst-libs/gst/rtp/gstrtphdrext.c:
rtp: fix some GIR annotations
Mostly related to out and array parameters.
2018-04-20 11:42:16 +0100 Tim-Philipp Müller <>
* ext/gl/
* gst-libs/gst/gl/
meson: gl: fix 'invalid keyword argument' meson warnings
Required is not a valid kwarg for cc.has_header()
2018-04-18 10:28:42 -0400 Omar Akkila <>
* gst-libs/gst/gl/egl/gsteglimage.c:
egl: fix build when using RPi EGL
2018-04-16 11:10:45 +0200 Víctor Manuel Jáquez Leal <>
* gst-libs/gst/gl/
gl: Define default value for GST_GL_HAVE_WINDOW_GBM
Thus, silent compiler's warning:
"GST_GL_HAVE_WINDOW_GBM" is not defined, evaluates to 0 [-Wundef]
2018-04-23 16:32:41 +0200 Mathieu Duponchelle <>
* gst-libs/gst/audio/gstaudioaggregator.c:
audioaggregator: fix filtered getcaps
In the situation described in,
downstream_caps consists of two structures, the first with
the preferred rate, if at all possible (44100), the second
containing the full range of allowed rates, as audioresample
correctly tries to negotiate passthrough caps.
As audioaggregator cannot perform rate conversion, it wants
to return a fixated rate in its getcaps implementation,
however it previously directly used the first structure in
the caps allowed downstream, without taking the filter into
consideration, to determine the rate to fixate to.
With this, we first intersect our downstream caps with the
filter, in order not to fixate to an unsupported rate.
2018-04-13 20:18:56 +0200 Mark Nauwelaerts <>
* gst-libs/gst/tag/gstid3tag.c:
* gst-libs/gst/tag/gstvorbistag.c:
* gst-libs/gst/tag/gstxmptag.c:
* gst-libs/gst/tag/tags.c:
tag: fix some GIR annotations
Mostly related to out and array parameters.
2018-04-12 22:24:26 +0200 Mathieu Duponchelle <>
* gst-libs/gst/audio/gstaudioaggregator.c:
audioaggregator: unref converted buffer after gst_buffer_replace
2018-04-12 22:23:50 +0200 Mathieu Duponchelle <>
* ext/alsa/gstalsamidisrc.c:
alsamidisrc: unref buffer_list before early return
2018-04-10 09:31:32 -0300 Thibault Saunier <>
* gst/playback/gsturisourcebin.c:
urisourcebin: Avoid unreffing a pad we are not owning
expose_output_pad takes ownership of the pad.
2018-04-12 19:33:18 +0200 Mathieu Duponchelle <>
* gst/playback/gsturidecodebin3.c:
uridecodebin3: free_play_items when READY_TO_PAUSED failed.
We will never go through the PAUSED_TO_READY transition if
that is the case, and thus never free the play items.
2018-04-12 18:12:49 +0100 Tim-Philipp Müller <>
* gst/playback/gstplaybin3.c:
playbin3: fix leak of recursive mutex
2018-04-11 22:56:34 +0200 Mathieu Duponchelle <>
* ext/vorbis/gstvorbisenc.c:
vorbisenc: do not map input buffer in WRITE mode
2018-04-11 21:40:23 +0200 Sebastian Dröge <>
* gst-libs/gst/pbutils/gstaudiovisualizer.c:
audiovisualizer: Only fixate pixel-aspect-ratio if the field exists
It's optional.
2018-04-10 21:18:11 +0200 Sebastian Dröge <>
* gst-libs/gst/pbutils/gstaudiovisualizer.c:
audiovisualizer: Fixate pixel-aspect-ratio to the closest value to 1/1
2018-04-07 11:07:45 +0530 Nirbheek Chauhan <>
* gst-libs/gst/audio/gstaudioringbuffer.c:
audioringbuffer: Don't spam INFO for every buffer
This makes GST_DEBUG=4 outputs too spammy, and such frequent messages
are meant to go into DEBUG or TRACE anyway.
2018-04-05 16:41:57 +0200 Zeeshan Ali <>
* tests/check/
tests: Enable tests for videodecoder
The tests pass fine here so don't see any reason to keep them disabled.
2018-04-04 19:30:55 -0300 Thibault Saunier <>
* gst/encoding/gstencodebin.c:
encodebin: Always respect encoding profile preset factory name
And fail if it is not present.
2018-03-04 16:41:14 +0100 Carlos Rafael Giani <>
* gst-libs/gst/gl/egl/gstglcontext_egl.c:
* gst-libs/gst/gl/gbm/gstglwindow_gbm_egl.c:
* gst-libs/gst/gl/gbm/gstglwindow_gbm_egl.h:
gl/gbm: Initialize window handle (= gbm surface) like other window systems
2018-03-27 10:43:16 +0100 Tim-Philipp Müller <>
* gst-libs/gst/gl/
gl: pick up GstVideo-1.0.gir from local build dir
2018-03-22 11:12:20 +0100 Antonio Ospite <>
* tools/gst-play-kb.c:
tools: play: fix leaving STDIN in non-blocking mode after exit
gst-play-1.0 sets STDIN to non-blocking mode to have the input
characters read as soon as they arrive.
However, when gst_play_kb_set_key_handler() gets called from
restore_terminal() it forgets to restore the STDIN blocking status.
This can result in broken behavior for cli command executed in the same
terminal after gst-play-1.0 exited.
It turns out that putting STDIN in non-blocking mode is not even the
proper way to achieve the desired effect, instead VMIN and VTIME in
struct termios should be set to 0.
Let's do that, and don't mess with the STDIN blocking mode now that it's
not necessary.
2018-03-25 12:48:12 +0300 Sebastian Dröge <>
* ext/gl/
gl: Disable glmixerbin for the time being too
Otherwise we have one copy in gst-plugins-bad and one (unused) here,
which makes static linking unhappy.
2018-03-23 14:24:38 +0100 Edward Hervey <>
* gst-libs/gst/audio/gstaudioaggregator.c:
audio-aggregator: Check return values
And copy over already-parsed information
CID #1427140
2018-03-22 07:56:28 +0100 Carlos Rafael Giani <>
* gst-libs/gst/gl/gstglupload.c:
* gst-libs/gst/gl/viv-fb/gstglwindow_viv_fb_egl.c:
viv-fb: Include gstglfuncs.h to fix cross compilation errors
2018-03-21 10:27:04 +0200 Sebastian Dröge <>
* gst-libs/gst/video/video-tile.h:
video: Set correct value in g-i annotations for tile related mask constants
2018-03-21 10:25:43 +0200 Sebastian Dröge <>
* gst-libs/gst/video/video.h:
video: Include gstvideoaffinetransformationmeta.h in video.h
2018-03-21 10:21:41 +0200 Sebastian Dröge <>
* gst-libs/gst/pbutils/pbutils.h:
pbutils: Include gstaudiovisualizer.h in pbutils.h
=== release 1.14.0 ===
2018-03-19 20:15:02 +0000 Tim-Philipp Müller <>
......@@ -3,19 +3,15 @@
The GStreamer team is proud to announce a new major feature release in
the stable 1.x API series of your favourite cross-platform multimedia
GStreamer 1.14.0 was originally released on 19 March 2018.
As always, this release is again packed with new features, bug fixes and
other improvements.
GStreamer 1.14.0 was released on 19 March 2018.
The latest bug-fix release in the 1.14 series is 1.14.1 and was released
on 17 May 2018.
See for the latest
version of this document.
_Last updated: Monday 19 March 2018, 12:00 UTC (log)_
_Last updated: Thursday 17 May 2018, 12:00 UTC (log)_
......@@ -482,6 +478,9 @@ New element features
passing through data (e.g. because target-timecode and end-timecode
respectively have been reached).
- 'alsamidisrc' element has been broken for many many years and has
now been repaired allowing live capture from your MIDI HW.
- h265parse and h265parse will try harder to make upstream output the
same caps as downstream requires or prefers, thus avoiding
unnecessary conversion. The parsers also expose chroma format and
......@@ -668,7 +667,7 @@ multiple sharing contexts in different threads; on Linux Nouveau is
known to be broken in this respect, whilst NVIDIA's proprietary drivers
and most other drivers generally work fine, and the experience with
Intel's driver seems to be mixed; some proprietary embedded Linux
drivers don't work; macOS works).
drivers don't work; macOS works.
GstPhysMemoryAllocator interface moved from -bad to -base
......@@ -763,7 +762,7 @@ Tracing framework and debugging improvements
of GStreamer.
- 'fakevideosink is a null sink for video data that advertises
video-specific metas ane behaves like a video sink. See above for
video-specific metas and behaves like a video sink. See above for
more details.
- gst_util_dump_buffer() prints the content of a buffer to stdout.
......@@ -925,6 +924,8 @@ GStreamer VAAPI
- vaapisink was demoted to marginal rank on Wayland because COGL
cannot display YUV surfaces.
More details in Víctor's blog post _GStreamer VA-API 1.14: what’s new?_.
GStreamer Editing Services and NLE
......@@ -1045,7 +1046,7 @@ Android
macOS and iOS
- this section will be filled in shortly {FIXME!}
- no major changes in macOS and iOS support, only bugfixes
......@@ -1076,6 +1077,9 @@ Windows
latency compared to shared mode where WASAPI's engine period is
10ms. This can be activated via the "exclusive" property.
- Also see Nirbheek's blog post _Low Latency Audio on Windows with
- There are now GstDeviceProvider implementations for the wasapi and
directsound plugins, so it's now possible to discover both audio
sources and audio sinks on Windows via the GstDeviceMonitor API
......@@ -1167,12 +1171,141 @@ the git 1.14 branch, which is a stable branch.
The first 1.14 bug-fix release (1.14.1) is scheduled to be released
around the end of March or beginning of April.
The first 1.14 bug-fix release (1.14.1) was released on 17 May 2018.
This release only contains bugfixes and it should be safe to update from
Noteworthy bugfixes in 1.14.1
- GstPad: Fix race condition causing the same probe to be called
multiple times
- Fix occasional deadlocks on windows when outputting debug logging
- Fix debug levels being applied in the wrong order
- GIR annotation fixes for bindings
- audiomixer, audioaggregator: fix some negotiation issues
- gst-play-1.0: fix leaving stdin in non-blocking mode after exit
- flvmux: wait for caps on all input pads before writing header even
if source is live
- flvmux: don't wake up the muxer unless there is data, fixes busy
looping if there's no input data
- flvmux: fix major leak of input buffers
- rtspsrc, rtsp-server: revert to RTSP RFC handling of
sendonly/recvonly attributes
- rtpvrawpay: fix payloading with very large mtu sizes where
everything fits into a single RTP packet
- v4l2: Fix hard-coded enabled v4l2 probe on Linux/ARM
- v4l2: Disable DMABuf for emulated formats when using libv4l2
- v4l2: Always set colorimetry in S_FMT
- asfdemux: Set stream-format field for H264 streams and handle H.264
in bytestream format
- x265enc: Fix tagging of keyframes on output buffers
- ladspa: Fix critical during plugin load on Windows
- decklink: Fix COM initialisation on Windows
- h264parse: fix re-use across pipeline stop/restart
- mpegtsmux: fix force-keyframe event handling and PCR/PMT changes
that would confuse some players with generated HLS streams
- adaptivedemux: Support period change in live playlist
- rfbsrc: Fix support for applevncserver and support NULL pool in
- jpegparse: Fix APP1 marker segment parsing
- h265parse: Make caps writable before modifying them, fixes criticals
- fakevideosink: request an extra buffer if enable-last-sample is
- wasapisrc: Don't provide a clock based on WASAPI's clock
- wasapi: Only use audioclient3 when low-latency, as it might
otherwise glitch with slow CPUs or VMs
- wasapi: Don't derive device period from latency time, should make it
more robust against glitches
- audiolatency: Fix wave detection in buffers and avoid bogus pts
values while starting
- msdk: fix plugin load on implementations with only HW support
- msdk: dec: set framerate to the driver only if provided, not in 0/1
- msdk: Don't set extended coding options for JPEG encode
- rtponviftimestamp: fix state change function init/reset causing
races/crashes on shutdown
- decklink: fix initialization failure in windows binary
- ladspa: Fix critical warnings during plugin load on Windows and fix
dependencies in meson build
- gl: fix cross-compilation error with viv-fb
- qmlglsink: make work with eglfs_kms
- rtspclientsink: Don't deadlock in preroll on early close
- rtspclientsink: Fix client ports for the RTCP backchannel
- rtsp-server: Fix session timeout when streaming data to client over
- vaapiencode: h264: find best profile in those available, fixing
negotiation errors
- vaapi: remove custom GstGL context handling, use GstGL instead.
Fixes GL Context sharing with WebkitGtk on wayland
- gst-editing-services: various fixes
- gst-python: bump pygobject req to 3.8; fix
GstPad.set_query_function(); dist and in
- g-i: pick up GstVideo-1.0.gir from local build directory in GstGL
- g-i: update constant values for bindings
- avoid duplicate symbols in plugins across modules in static builds
- ... and many, many more!
Cerbero build tool and packaging changes in 1.14.1
Toolchain updates on iOS and Android necessitated a fairly large number
of changes in our cerbero build tool used to create our binary packages
for the various platforms we support:
- Add support for Ubuntu 18.04 in cerbero
- Fix generation of fat shared libraries on macOS
- gnutls: also rename assembly functions on macos/ios to fix link
- gnutls: fix assembly symbol names for windows x86
- openssl: fix linking on android/armv7
- openssl: fix linker issue with Android NDK's r16 binutils
- ffmpeg: disable asm for android x86 to fix issues when linking with
- x264: disable asm for android x86 to fix issues when linking with
- gnutls: rename private symbols for armv8, x86 to not conflict with
- mpg123: disable assembly on android/x86 to fix linker problems with
- Check built version while loading recipe and rebuild if needed
- Fix packaging of libgcc_s_sjlj which was missing in Windows packages
- Make not-found in library search fatal so we don't accidentally ship
broken packages
- ship the proxy plugin which was new in 1.14
- Fix git commands accidentally pulling in locally built libraries and
Contributors to 1.14.1
Antonio Ospite, Aurélien Zanelli, Brendan Shanks, Carlos Rafael Giani,
Edward Hervey, Emilio Pozuelo Monfort, Enrique Ocaña González, Garima
Gaur, Georg Lippitsch, Guillaume Desmottes, Havard Graff, Hoonhee Lee,
Hyunjun Ko, James Stevenson, Jan Alexander Steffens (heftig), Jan
Schmidt, Joakim Johansson, Jun Xie, Kai Kang, Kirill Marinushkin, Mark
Nauwelaerts, Matej Knopp, Mathieu Duponchelle, Matthew Waters, Matthias
Fend, Michael Olbrich, Mikhail Fludkov, Nicolas Dufresne, Nirbheek
Chauhan, Olivier Crête, Omar Akkila, Patrik Nilsson, Philippe Normand,
Pierre Labastie, Sebastian Dröge, Seungha Yang, Sreerenj Balachandran,
Stian Selnes, Takeshi Sato, Thibault Saunier, Tim-Philipp Müller, U.
Artie Eoff, Víctor Manuel Jáquez Leal, Vivia Nikolaidou, Whoopie, Xabier
Rodriguez Calvar, Xavier Claessens, Zeeshan Ali, and countless others.
List of bugs fixed in 1.14.1
For a full list of bugfixes see Bugzilla. Note that this is not the full
list of changes. For the full list of changes please refer to the GIT
logs or ChangeLogs of the particular modules.
The second 1.14 bug-fix release (1.14.2) is scheduled to be released
around mid-June 2018.
This release only contains bugfixes and it should be safe to update from
Known Issues
......@@ -1180,6 +1313,10 @@ Known Issues
GStreamer webrtc support) is currently not shipped as part of the
Windows binary packages due to a build system issue.
- The gst-libav module currently won't build against the
newly-released ffmpeg 4.0 (as in F28). Use the internal ffmpeg copy