Commit f1eed72c authored by Tim-Philipp Müller's avatar Tim-Philipp Müller 🐠

Release 1.14.1

parent a1204282
=== release 1.14.1 ===
2018-05-17 13:25:00 +0100 Tim-Philipp Müller <>
* ChangeLog:
* gst-plugins-good.doap:
Release 1.14.1
2018-05-17 13:25:00 +0100 Tim-Philipp Müller <>
* docs/plugins/gst-plugins-good-plugins.args:
* 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-gtk.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-lame.xml:
* docs/plugins/inspect/plugin-level.xml:
* docs/plugins/inspect/plugin-matroska.xml:
* docs/plugins/inspect/plugin-mpg123.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-qmlgl.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-shout2.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-twolame.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:
Update docs
2018-05-17 12:37:35 +0100 Tim-Philipp Müller <>
* po/hr.po:
Update translations
2018-05-15 14:56:04 -0400 Thibault Saunier <>
* gst/isomp4/qtdemux.c:
Revert "qtdemux: also push buffers without encryption info instead of dropping them"
This reverts commit 762e9c645ec13513c62eb5a3800d7406e01cdcb7.
This was pushed by mistake
2018-05-15 14:55:58 -0400 Thibault Saunier <>
* gst/isomp4/qtdemux.c:
* gst/isomp4/qtdemux.h:
Revert "qtdemux: add context for a preferred protection"
This reverts commit 0ba62ba4805e2cdbed17fa9934762d685be42fd4.
This was pushed by mistake
2018-05-15 14:55:46 -0400 Thibault Saunier <>
* ext/soup/gstsouphttpsrc.c:
* ext/soup/gstsouphttpsrc.h:
Revert "souphttpsrc: cookie jar and context query support"
This reverts commit 6715af9933a6607e5d86ac6fc1bcf476761cbf10.
This was pushed by mistake
2018-04-22 10:40:19 -0300 Thibault Saunier <>
* ext/jpeg/gstjpegenc.c:
jpegenc: Accept sof-marker=4
sof-marker is 4 when input is in the RGB colorspace.
2017-04-24 17:22:02 +0000 Enrique Ocaña González <>
* gst/isomp4/qtdemux.c:
qtdemux: also push buffers without encryption info instead of dropping them
2017-06-21 17:59:21 +0200 Xabier Rodriguez Calvar <>
* gst/isomp4/qtdemux.c:
* gst/isomp4/qtdemux.h:
qtdemux: add context for a preferred protection
qtdemux selected the first system corresponding to a working GStreamer
decryptor. With this change, before selecting that decryptor, qtdemux
will check if it has context (a preferred decryptor id) and if not, it
will request it.
The request includes track-id, available key system ids for the
available decryptors and even the events so that the init data is
[ select the preferred protection system even if not available]
Test "4. ClearKeyVideo" in YouTube leanback EME conformance tests 2016 for
H.264[1] uses a media file[2] with cenc encryption which embeds 'pssh' boxes
with the init data for the Playready and Widevine encryption systems, but not
for the ClearKey encryption system (as defined by the EMEv0.1b spec[3] and with
the encryption system id defined in [4]).
Instead, the ClearKey encryption system is manually selected by the web page
code (even if not originally detected by qtdemux) and the proper decryption key
is dispatched to the decryptor, which can then decrypt the video successfully.
2015-10-28 12:00:09 +0100 Philippe Normand <>
* ext/soup/gstsouphttpsrc.c:
* ext/soup/gstsouphttpsrc.h:
souphttpsrc: cookie jar and context query support
Use a volatile Cookie jar to store cookies and handle the context
query so that session data can be shared with other elements (like
2017-08-25 11:59:00 +0200 Mikhail Fludkov <>
* gst/rtpmanager/rtpsession.c:
* tests/check/elements/rtpsession.c:
rtpsession: Fix on-feedback-rtcp race
If there is an external source which is about to timeout and be removed
from the source hashtable and we receive feedback RTCP packet with the
media ssrc of the source, we unlock the session in
rtp_session_process_feedback before emitting 'on-feedback-rtcp' signal
allowing rtcp timer to kick in and grab the lock. It will get rid of
the source and rtp_session_process_feedback will be left with RTPSource
with ref count 0.
The fix is to grab the ref to the RTPSource object in
2017-11-27 10:56:47 +0100 Stian Selnes <>
* gst/rtpmanager/rtpsession.c:
rtpsession: Add missing lock around sess->ssrcs iteration
2014-10-05 15:51:18 +0200 Matej Knopp <>
* gst/audioparsers/gstdcaparse.c:
dcaparse: do not accept header with invalid channel count
2018-05-10 13:57:30 +0200 Edward Hervey <>
* gst/isomp4/qtdemux.c:
qtdemux: Initialize riff library
Avoids debugging message issues. Also just use the main riff header
2018-05-05 16:32:59 +0200 Tim-Philipp Müller <>
* gst/rtp/gstrtpvrawpay.c:
rtpvrawpay: don't use buffer lists if everything fits into one buffer
People might use very large mtu sizes where every payload
fits into a single output packet.
2018-04-23 11:26:12 -0400 Olivier Crête <>
* gst/flv/gstflvmux.c:
flvmux: Don't wake up the muxer unless there is data
2018-04-23 11:19:18 -0400 Olivier Crête <>
* gst/flv/gstflvmux.c:
flvmux: Save the current position in the output segment
2018-04-19 17:53:51 -0400 Olivier Crête <>
* gst/flv/gstflvmux.c:
* tests/check/elements/flvmux.c:
flvmux: Wait for caps from both srcs before writing header
Wait for caps on all pads to start writing data even when source is live.
Includes unit test by Havard Graff that simulates it.
2018-04-16 21:27:47 +0300 Sebastian Dröge <>
* gst/audioparsers/gstflacparse.c:
flacparse: Drain the parser when a CAPS event is received
After a CAPS event, in theory a new stream can start and it might start
with the FLAC headers again. We can't detect FLAC headers in the middle
of the stream, so we drain the parser to be able to detect either FLAC
headers after the CAPS event or the continuation of the previous stream.
This fixes for example
gst-launch-1.0 audiotestsrc num-buffers=200 ! flacenc ! c. \
audiotestsrc num-buffers=200 freq=880 ! flacenc ! c. \
concat name=c ! rtpgstpay ! udpsink host= port=5000
gst-launch-1.0 udpsrc multicast-group= port=5000 \
caps=application/x-rtp,media=application,clock-rate=90000,encoding-name=X-GST ! \
rtpgstdepay ! flacparse ! flacdec ! audioconvert ! pulsesin
2018-04-04 15:50:55 +0200 Kirill Marinushkin <>
configure: Fix hard-coded enabled v4l2 probe on Linux/ARM
Currently, enable_v4l2_probe is hard-coded to "yes" on linux, platforms
arm and aarch64. This even overrides the --disable-v4l2-probe argument.
As a result, it is impossible to disable v4l2_probe. It becomes a problem
for use-cases, when startup time is critical, because the v4l2_probe
feature increases the initialization time.
This commit makes the v4l2_probe feature configurable.
On linux, platforms arm and aarch64, the default value is still "yes".
But now it can be disabled by the --disable-v4l2-probe argument.
2018-04-13 13:29:06 +0200 Guillaume Desmottes <>
* sys/v4l2/gstv4l2transform.c:
* sys/v4l2/gstv4l2videodec.c:
* sys/v4l2/gstv4l2videoenc.c:
* sys/v4l2/v4l2_calls.c:
v4l2: rely on gst_v4l2_dup() to set no_initial_format and keep_aspect
gst_v4l2_dup() will now take care of setting
v4l2capture->no_initial_format and keep_aspect instead of doing it
Fix a typo as keep_aspect was set twice on v4l2output but never on
2018-04-17 17:57:16 +0300 Sebastian Dröge <>
* gst/rtsp/gstrtspsrc.c:
* tests/examples/rtsp/test-onvif.c:
Revert "rtspsrc: Fix up sendonly/recvonly attribute handling"
This reverts commit af273b4de9eb292c0b6af63665e10ca015895902.
While RFC 3264 (SDP) says that sendonly/recvonly are from the point of view of
the requester, the actual RTSP RFCs (RFC 2326 / 7826) disagree and say
the opposite, just like the ONVIF standard.
Let's follow those RFCs as we're doing RTSP here, and add a property at
a later time if needed to switch to the SDP RFC behaviour.
2018-04-13 22:49:43 +0200 Mathieu Duponchelle <>
* gst/isomp4/gstqtmux.c:
qtmux: Fix leak
gst_qt_mux_can_renegotiate () gets called everywhere following
that pattern:
return gst_qt_mux_can_renegotiate (ref(self));
This means the reference must be released both in the success
and failure cases, it was only done in the success case.
2018-04-13 22:44:14 +0200 Mathieu Duponchelle <>
* gst/flv/gstflvmux.c:
flvmux: aggregate should not push EOS itself
Instead it is expected to return GST_FLOW_EOS, and let the
base class handle that.
2018-04-13 23:01:20 +0200 Mathieu Duponchelle <>
* gst/flv/gstflvmux.c:
flvmux: unref return of aggregator_pad_peek_buffer
We ended up leaking every single buffer going through the
muxer, which is far from ideal
2018-04-11 20:28:00 +0000 Whoopie <>
* sys/v4l2/gstv4l2object.c:
v4l2object: Disable DMABuf for emulated formats
libv4l2 does not prevent exporting DMABuf even when emulated formats are
in use. As a side effect, userspace ends up with buffers of the original
formats which will cause issues.
2018-04-08 20:42:16 -0400 Nicolas Dufresne <>
* sys/v4l2/gstv4l2object.c:
v4l2object: Only use BT2020_12 for BT2020 v4l2 colorspace
BT2020_12 is not represented in V4L2, so drivers providing full colority
for BT2020 will set V4L2_XFER_FUNC_709 transfer function. To fix the
issue, we bump this to BT2020_12 if the resoltion is 4K, but we should
only do that if the colorspace is BT2020 to start with, otherwise it's
not possible to use normal BT709 for 4K 8bit formats.
2018-04-08 13:43:56 -0400 Nicolas Dufresne <>
* sys/v4l2/gstv4l2object.c:
v4l2object: Always set the colorimetry in S_FMT
So far we were only setting colorimetry for OUTPUT devices (v4l2sink or
m2m sink pad). This prevented selecting through caps negotiation the
colorimetry for CAPTURE devices (v4l2src or m2m src pad). This is rarely
selectable, but trying is harmless.
2018-04-11 17:54:38 +0300 Vivia Nikolaidou <>
* gst/multifile/gstsplitmuxsink.c:
splitmuxsink: Don't send fragment-opened-closed message if the reference ctx is NULL
It can happen during teardown that the reference context becomes NULL.
In that case, trying to send the fragment-opened-closed message would
lead to a crash.
2018-04-11 21:41:58 +0200 Sebastian Dröge <>
* gst/monoscope/gstmonoscope.c:
monoscope: Only fixate pixel-aspect-ratio if the field exists
2018-04-10 21:15:48 +0200 Sebastian Dröge <>
* gst/monoscope/gstmonoscope.c:
monoscope: Fixate pixel-aspect-ratio too and make sure the final caps are completely fixated
Otherwise e.g. this fails with assertions:
gst-launch-1.0 audiotestsrc ! audioconvert ! monoscope ! videoconvert ! \
videoscale ! video/x-raw,width=800,height=600 ! ximagesink
2018-03-08 10:10:01 +0100 Edward Hervey <>
* gst/isomp4/gstqtmux.c:
qtmux: Add comments and doc about prefill mode
2018-04-04 01:48:44 +0200 Mathieu Duponchelle <>
* gst/rtpmanager/gstrtprtxsend.c:
rtxsend: fix wrong memory layout assumption
The code responsible for creating retransmitted buffers
assumed the stored buffer had been created with
rtp_buffer_new_allocate when copying the extension data,
which isn't necessarily the case, for example when
the rtp buffers come from a udpsrc.
2018-03-04 15:14:08 +0100 Carlos Rafael Giani <>
* ext/qt/
qt: Get EGL native display from QPA if platform header is available
2018-03-21 00:19:37 +0900 Seungha Yang <>
* sys/v4l2/gstv4l2object.c:
* sys/v4l2/gstv4l2object.h:
v4l2: Fix unknown type name ‘off_t’ error
Fix following build error
gstv4l2object.h:197:17: error: unknown type name ‘off_t’
gint fd, off_t offset);
2018-03-22 15:20:47 +0100 Edward Hervey <>
* gst/isomp4/qtdemux.c:
qtdemux: Check sample count is valid in PIFF parsing
The value stored in cenc_aux_sample_count wasn't in sync with the
parsing code that followed which checks whether all entries are
valid and present.
Only write the actual sample count when we know for sure.
CID #1427087
2018-03-20 11:36:32 +0200 Sebastian Dröge <>
* gst/rtp/gstrtpreddec.c:
* gst/rtp/gstrtpredenc.c:
* gst/rtp/gstrtpulpfecdec.c:
* gst/rtp/gstrtpulpfecenc.c:
* gst/rtp/rtpstoragestream.c:
* tests/check/elements/rtpred.c:
* tests/check/elements/rtpulpfec.c:
rtp: Fix compilation with non-C99 compilers
By moving variable declarations out of loop headers.
=== release 1.14.0 ===
2018-03-19 20:18:22 +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
instead, if you build using autotools.
Schedule for 1.16
This is GStreamer gst-plugins-good 1.14.0.
This is GStreamer gst-plugins-good 1.14.1.
The GStreamer team is thrilled to announce a new major feature release in the
The GStreamer team is pleased to announce a new bug-fix release in the
stable 1.x API series of your favourite cross-platform multimedia framework!
As always, this release is again packed with new features, bug fixes and
other improvements.
The 1.14 release series adds new features on top of the 1.12 series and is
part of the API and ABI-stable 1.x release series of the GStreamer multimedia
......@@ -60,7 +57,7 @@ You can find source releases of gstreamer in the download
The git repository and details how to clone it can be found at
==== Homepage ====
......@@ -69,7 +66,7 @@ The project's website is