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authorTim-Philipp Müller <tim@centricular.com>2019-12-03 11:03:11 +0000
committerTim-Philipp Müller <tim@centricular.com>2019-12-03 11:03:12 +0000
commit9d3581b2e6f12f0b7e790d1ebb63b90cf5b1ef4e (patch)
tree3f8a55c89ac89a790a50361b6db5cd0000f445f0
parent7c885c3a8155e565730a08f1ecd4f0c818975365 (diff)
Release 1.16.21.16.2
-rw-r--r--ChangeLog245
-rw-r--r--NEWS206
-rw-r--r--RELEASE2
-rw-r--r--configure.ac6
-rw-r--r--gst-plugins-base.doap10
-rw-r--r--meson.build2
6 files changed, 451 insertions, 20 deletions
diff --git a/ChangeLog b/ChangeLog
index 2d37dfa46..2a63b446a 100644
--- a/ChangeLog
+++ b/ChangeLog
@@ -1,3 +1,248 @@
+=== release 1.16.2 ===
+
+2019-12-03 11:03:11 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * gst-plugins-base.doap:
+ * meson.build:
+ Release 1.16.2
+
+2019-12-03 11:03:11 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * 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-compositor.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-overlaycomposition.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
+
+2019-12-03 11:03:09 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * po/hu.po:
+ Update translations
+
+2019-10-02 23:21:09 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * ext/gl/effects/gstgleffectblur.c:
+ * ext/gl/gstglbumper.c:
+ * ext/gl/gstglbumper.h:
+ * ext/gl/gstglfiltercube.c:
+ * ext/gl/gstglfiltercube.h:
+ * ext/gl/gstglfilterglass.c:
+ * ext/gl/gstglfilterglass.h:
+ * ext/gl/gstglmosaic.c:
+ * ext/gl/gstglmosaic.h:
+ glfilters: Don't use static variables for storing per-element state
+
+2019-11-08 17:28:44 +0900 Seungha Yang <seungha.yang@navercorp.com>
+
+ * sys/xvimage/xvimagepool.c:
+ * sys/xvimage/xvimagepool.h:
+ xvimagepool: Update size, stride, and offset with allocated XvImage
+ Memory layout of XvImage might be different from that of GstVideoInfo.
+ If so, the image size, stride, and offset would be wrongly informed.
+ Fixes: https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/issues/677
+
+2019-11-08 17:33:17 +0900 Seungha Yang <seungha.yang@navercorp.com>
+
+ * sys/xvimage/xvimagepool.c:
+ xvimagepool: Fix confusing debug message for padding size
+
+2019-11-08 15:13:59 +0100 Edward Hervey <edward@centricular.com>
+
+ * gst-libs/gst/video/video-converter.c:
+ video-converter: Fix RGB-XYZ-RGB conversion
+ The matrices were in the wrong order.
+ Instead of the conversion matrix being
+ _ XYZ_TO_RGB_output * RGB_TO_XYZ_input * input_RGB
+ It was
+ _ RGB_TO_XYZ_input * XYZ_TO_RGB_output * input_RGB
+
+2019-11-07 22:00:03 +0900 Seungha Yang <seungha.yang@navercorp.com>
+
+ * gst/audiorate/gstaudiorate.c:
+ audiorate: Update next_offset per rate change
+ To support runtime audio samplerate change, re-calculate next target offset
+ per caps. Calculating the next buffer offset using the previous
+ offset seems to be tricky and rounding error prone.
+ Fixes: https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/issues/693
+
+2019-11-07 21:42:25 +0900 Seungha Yang <seungha.yang@navercorp.com>
+
+ * gst/audiorate/gstaudiorate.c:
+ * gst/audiorate/gstaudiorate.h:
+ Revert "audiorate: accumulate offset by time diff"
+ This reverts commit 4fa850e3e6c039000fc7f648de238af6c2278469.
+ The commit would break an constant rate audio stream with gap.
+
+2019-08-26 15:03:48 +0200 Jochen Henneberg <jh@henneberg-systemdesign.com>
+
+ * gst-libs/gst/audio/gstaudioringbuffer.c:
+ audioringbuffer: Reset reorder flag before check
+ This function might be revisited with different channel position mapping
+ while audio source goes into play so the reorder flag needs to be reset
+ before the checks happen.
+
+2019-11-14 12:37:58 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/audio/audio-buffer.c:
+ audio-buffer: Don't fail to map buffers with zero samples
+ Instead initialize the map infos, etc to NULL like gst_buffer_map()
+ would be doing on a zero-sized buffer.
+ This fixes a crash in audioresample if the first output buffer would
+ contain zero samples.
+
+2019-10-28 14:43:50 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/videorate/gstvideorate.c:
+ * tests/check/elements/videorate.c:
+ videorate: Fix max-duplication-time handling
+ Previously this would've only set discont=TRUE and then for all future
+ buffers simply returned immediately.
+ Instead we also need to
+ a) drain previous input until its buffer time
+ b) update next_ts and base_ts accordingly for the gap
+ c) actually store the new buffer after the gap so it can be used in
+ the future and so the old buffer before the gap is gone
+ Also update the unit test accordingly so that it actually tests for this
+ behaviour. Previously it only tested that after the gap we got no output
+ at all.
+
+2019-11-04 00:38:18 +1100 Matthew Waters <matthew@centricular.com>
+
+ * ext/gl/gstglimagesink.c:
+ * gst-libs/gst/gl/gbm/gstglwindow_gbm_egl.c:
+ gl/gbm: ensure we call the resize callback before attempting to draw
+ Without this, sinks will not be notified about size changes or even the
+ initial size and would render at 0x0.
+
+2019-10-21 16:43:32 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/video/video-converter.c:
+ video-converter: Use G_N_ELEMENTS instead of writing it out manually
+
+2019-10-21 17:40:35 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/video/video-converter.c:
+ video-converter: Scaling-only fast-paths for planar formats are keeping interlacing
+ So we can also use them for interlaced content and speed up scaling a
+ bit.
+
+2019-10-21 17:28:46 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/video/video-scaler.c:
+ video-scaler: Correctly handle interlaced vertical scalers in gst_video_scaler_2d()
+ We need to provide twice as many lines as usual to the scaling function
+ as every second lines would be skipped.
+ Without this we read from random memory and produce colorful output and
+ crashes.
+
+2019-10-21 16:43:02 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/video/video-converter.c:
+ video-converter: Don't use fast-path halfing/doubling rescalers for interlaced planar formats
+ They will mix lines from the top and bottom field and by that
+ effectively deinterlace the video, badly.
+
+2019-10-21 16:43:02 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/video/video-converter.c:
+ video-converter: Use interlaced scalers for vertical scaling if needed
+ Without this, scaling e.g. interlaced UYVY causes corrupted output with
+ lines as follows: f1 f1 f2 f2, i.e. two lines of each field and only
+ then the other field.
+
+2019-10-03 12:03:09 +0200 Joakim Johansson <joakimj@axis.com>
+
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ gstrtspconnection: messages_bytes not decreased
+ The watch->messages_bytes is not decreased when the write operation
+ from the backlog is only partly successfull.
+ This commit decreases the watch->messages_bytes for the successfully
+ sent messages.
+ Fixes #679
+
+2019-10-10 16:58:26 +0200 Edward Hervey <edward@centricular.com>
+
+ * tests/check/elements/adder.c:
+ * tests/check/elements/audiomixer.c:
+ check: Don't use real audio devices for tests
+ When checking the behaviour of live seeking on audiomixer or
+ adder we don't *really* need real audio devices. audiotestsrc
+ in live mode is enough to test the behaviour of those elements.
+ Also avoids people repeatedly wasting hours trying to figure out
+ whether that failing behaviour is due to their code or not.
+
+2019-10-08 19:35:49 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst-libs/gst/pbutils/descriptions.c:
+ pbutils: add description for CineForm codec
+
+2019-10-08 19:35:23 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst-libs/gst/riff/riff-media.c:
+ riff: add mapping for CineForm codec
+ Makes CineForm-in-AVI work.
+
+2019-09-27 11:10:43 -0300 Thibault Saunier <tsaunier@igalia.com>
+
+ * gst-libs/gst/gl/gstglupload.c:
+ * tests/check/libs/gstglupload.c:
+ glupload: Add VideoMetas and GLSyncMeta to the raw uploaded buffers
+ This is done by reusing `gst_gl_memory_setup_buffer` avoiding to
+ duplicate code.
+ Without a VideoMeta, mapping those buffers lead to GstBuffer mapping the
+ buffer in system memory even when specifying the GL flags (through the
+ buffer merging mechanism) making the result totally broken.
+
+2019-09-20 09:48:30 +0100 Charlie Turner <cturner@igalia.com>
+
+ * gst/playback/gststreamsynchronizer.c:
+ streamsynchronizer: avoid pad destruction races.
+ Due to the use of {set/get}-element_private methods being used to store
+ the GstSyncStream in the src and sink pads, and the racey nature of pad
+ destruction, there are numerous ways we can be bitten by race conditions
+ in the stream synchronizer. Fix that by tying the pads toghether with
+ references.
+
+2019-09-24 15:32:33 +0900 Seungha Yang <seungha.yang@navercorp.com>
+
+ * tools/gst-play.c:
+ gst-play: Use gst_print* to avoid broken stdout string on Windows
+ Equvalant to https://gitlab.freedesktop.org/gstreamer/gstreamer/merge_requests/258
+ When debug enabled, the debug string might be broken on Windows.
+
=== release 1.16.1 ===
2019-09-23 11:06:23 +0100 Tim-Philipp Müller <tim@centricular.com>
diff --git a/NEWS b/NEWS
index c93be83e6..98dc512e6 100644
--- a/NEWS
+++ b/NEWS
@@ -5,13 +5,13 @@ GSTREAMER 1.16 RELEASE NOTES
GStreamer 1.16.0 was originally released on 19 April 2019.
-The latest bug-fix release in the 1.16 series is 1.16.1 and was released
-on 23 September 2019.
+The latest bug-fix release in the 1.16 series is 1.16.2 and was released
+on 3 December 2019.
See https://gstreamer.freedesktop.org/releases/1.16/ for the latest
version of this document.
-_Last updated: Sunday 22 September 2019, 21:00 UTC (log)_
+_Last updated: Tuesday 03 December 2019, 08:00 UTC (log)_
Introduction
@@ -142,9 +142,9 @@ Support for Planar (Non-Interleaved) Raw Audio
Raw audio samples are usually passed around in interleaved form in
GStreamer, which means that if there are multiple audio channels the
-samples for each channel are interleaved in memory, e.g.
-|LEFT|RIGHT|LEFT|RIGHT|LEFT|RIGHT| for stereo audio. A non-interleaved
-or planar arrangement in memory would look like
+samples for each channel are interleaved in memory,
+e.g. |LEFT|RIGHT|LEFT|RIGHT|LEFT|RIGHT| for stereo audio. A
+non-interleaved or planar arrangement in memory would look like
|LEFT|LEFT|LEFT|RIGHT|RIGHT|RIGHT| instead, possibly with
|LEFT|LEFT|LEFT| and |RIGHT|RIGHT|RIGHT| residing in separate memory
chunks or separated by some padding.
@@ -243,7 +243,7 @@ New Elements
GstVideoOverlayCompositions on a stream. The element will emit the
"draw" signal for each video buffer, and the application then
generates an overlay for that frame (or not). This is much more
- performant than e.g. cairooverlay for many use cases, e.g. because
+ performant than e.g. cairooverlay for many use cases, e.g. because
pixel format conversions can be avoided or the blitting of the
overlay can be delegated to downstream elements (such as
gloverlaycompositor). It’s particularly useful for cases where only
@@ -336,7 +336,7 @@ New element features and additions
- "reset-muxer" property: when unset, the muxer is reset using
flush events instead of setting its state to NULL and back. This
- means the muxer can keep state across resets, e.g. mpegtsmux
+ means the muxer can keep state across resets, e.g. mpegtsmux
will keep the continuity counter continuous across segments as
required by hlssink2.
@@ -700,9 +700,9 @@ Tracing framework and debugging improvements
- There is now a GDB PRETTY PRINTER FOR VARIOUS GSTREAMER TYPES: For
GstObject pointers the type and name is added, e.g.
0x5555557e4110 [GstDecodeBin|decodebin0]. For GstMiniObject pointers
- the object type is added, e.g. 0x7fffe001fc50 [GstBuffer]. For
+ the object type is added, e.g. 0x7fffe001fc50 [GstBuffer]. For
GstClockTime and GstClockTimeDiff the time is also printed in human
- readable form, e.g. 150116219955 [+0:02:30.116219955].
+ readable form, e.g. 150116219955 [+0:02:30.116219955].
- GDB EXTENSION WITH TWO CUSTOM GDB COMMANDS gst-dot AND gst-print:
@@ -1044,7 +1044,7 @@ Changed
- Rust 1.31 is the minimum supported Rust version now
- Update to latest gir code generator and glib bindings
-- Functions returning e.g. gst::FlowReturn or other “combined” enums
+- Functions returning e.g. gst::FlowReturn or other “combined” enums
were changed to return split enums like
Result<gst::FlowSuccess, gst::FlowError> to allow usage of the
standard Rust error handling.
@@ -1388,7 +1388,8 @@ Highlighted bugfixes in 1.16.1
- decklinkaudiosink: Drop late buffers
- openh264enc: Fix compilation with openh264 v2.0
- wasapisrc: fix segtotal value being always 2
-- Fix issues on Android Q
+- android: Fix gnutls issue causing a FORTIFY crash on Android Q
+- windows: Fix two crashes due to cross-CRT free when using MSVC
gstreamer core
@@ -1698,6 +1699,182 @@ List of merge requests and issues fixed in 1.16.1
- List of Merge Requests applied in 1.16
- List of Issues fixed in 1.16.1
+1.16.2
+
+The second 1.16 bug-fix release (1.16.2) was released on 03 December
+2019.
+
+This release only contains bugfixes and it _should_ be safe to update
+from 1.16.1.
+
+Highlighted bugfixes in 1.16.2
+
+- Interlaced video scaling fixes
+- CineForm video support in AVI
+- audioresample: avoid glitches due to rounding errors after changing
+ rate
+- Command line tool output printing improvements on Windows
+- various performance improvements, memory leak fixes and security
+ fixes
+- VP9 decoding fixes
+- avfvideosrc: Explicitly request video permission on macOS 10.14+
+- wasapi: bug fixes and stability improvements
+- webrtc-audio-processing: fix segmentation fault on 32-bit windows
+- tsdemux: improved handling of certain discontinuities
+- vaapi h265 decoder: wait for I-frame before trying to decode
+
+gstreamer
+
+- gst-launch: Fix ugly stdout on Windows
+- tee: Make sure to actually deactivate pads that are released
+- bin: Drop need-context messages without source instead of crashing
+- gst: Don’t pass miniobjects to GST_DEBUG_OBJECT() and similar macros
+- tracers: Don’t leak temporary GstStructure
+
+gst-plugins-base
+
+- xvimagepool: Update size, stride, and offset with allocated XvImage
+- video-converter: Fix RGB-XYZ-RGB conversion
+- audiorate: Update next_offset on rate change
+- audioringbuffer: Reset reorder flag before check
+- audio-buffer: Don’t fail to map buffers with zero samples
+- videorate: Fix max-duplication-time handling
+- gl/gbm: ensure we call the resize callback before attempting to draw
+- video-converter: Various fixes for interlaced scaling
+- gstrtspconnection: messages_bytes not decreased
+- check: Don’t use real audio devices for tests
+- riff: add CineForm mapping
+- glfilters: Don’t use static variables for storing per-element state
+- glupload: Add VideoMetas and GLSyncMeta to the raw uploaded buffers
+- streamsynchronizer: avoid pad release race during logging.
+- gst-play: Use gst_print* to avoid broken stdout string on Windows
+
+gst-plugins-good
+
+- vp9dec: Fix broken 4:4:4 8bits decoding
+- rtpsession: add locking for clear-pt-map
+- rtpL16depay: don’t crash if data is not modulo channels*width
+- wavparse: Fix push mode ignoring audio with a size smaller than
+ segment buffer
+- wavparse: Fix push mode ignoring last audio payload chunk
+- aacparse: fix wrong offset of the channel number in adts header
+- jpegdec: Fix incorrect logic in EOI tag detection
+- videocrop: Also update the coordinate when in-place
+- jpegdec: don’t overwrite the last valid line
+- vpx: Error out if enabled and no features found
+- v4l2videodec: ensure pool exists before orphaning it
+- v4l2videoenc: fix type conversion errors
+- v4l2bufferpool: Queue number of allocated buffers to capture
+- v4l2object: fix mpegversion number typo
+- v4l2object: Work around bad TRY_FMT colorimetry implementations
+
+gst-plugins-bad
+
+- avfvideosrc: Explicitly request video permission on macOS 10.14+
+- wasapi: Various fixes and a workaround for a specific driver bug
+- wasapi: Move to CoInitializeEx for COM initialization
+- wasapi: Fix runtime/build warnings
+- waylandsink: Commit the parent after creating subsurface
+- msdkdec: fix surface leak in msdkdec_handle_frame
+- tsmux: Fix copying of buffer region
+- tsdemux: Handle continuity mismatch in more cases
+- tsdemux: Always issue a DTS even when it’s equal to PTS
+- openexr: Fix build with OpenEXR 2.4 (and also OpenEXR 2.2 on Ubuntu
+ 18.04)
+- ccextractor: Always forward all sticky events to the caption pad
+- pnmdec: Return early on ::finish() if we have no actual data to
+ parse
+- ass: avoid infinite unref loop with bad data
+- fluidsynth: add sf3 to soundfont search path
+- webrtcdsp/webrtcechoprobe segmentation fault on windows (1.16.0 x86)
+
+gst-libav
+
+- avvidenc: Fix error propagation
+- avdemux: Fix segmentation fault if long_name is NULL
+- avviddec: Fix huge leak caused by circular reference
+- avviddec: Enforce allocate new AVFrame per input frame
+- avdec_mpeg2video (and probably more): Huge memory leak in git master
+
+gst-rtsp-server
+
+- rtsp-media: Use lock in gst_rtsp_media_is_receive_only
+- rtsp-client: RTP Info when completed_sender
+- rtsp-client: fix location uri-format by getting uri directly from
+ context instead
+
+gstreamer-vaapi
+
+- meson build: halt configuration if no renderer API
+- libs: decoder: h265: skip all pictures prior the first I-frame
+- libs: window: x11: Avoid usage of deprecated API
+
+gst-editing-services
+
+- Initialize debug categories before usage
+
+gst-build
+
+- gst-env: Use locally built GStreamer utility programs
+
+Cerbero build tool and packaging changes in 1.16.2
+
+General
+
+- openssl: Update to 1.1.1d
+- Updated ffmpeg, expat, flac, freetype, croco, ogg, xml2, mpg123,
+ openjpeg, opus, pixman, speex, tiff recipes
+- Fix setting of git credentials in local source repos
+
+Windows
+
+- webrtc-audio-processing: fix segmentation fault on 32-bit windows
+ with webrtcdsp/webrtcechoprobe elemens
+- vpx plugin has no features when built with Visual Studio 2019
+- libvpx: Add support for Visual Studio 2019
+- mingw-runtime.recipe: Correctly package pkg-config in the MSI
+- GIO doesn’t load any modules on Windows with MSVC, which breaks TLS
+ support since glib-networking’s giognutls module isn’t loaded
+- Make the instructions for running Cerbero the same on all platforms
+
+macOS + iOS
+
+- Add support for macOS 10.15 Catalina
+- Updates for Xcode 11
+- macos/ios: expose objc++ compilers in env variables
+- srt.recipe: Fix crash in constructor on iOS
+- osx-framework.recipe: Dynamically generate the list of libraries and
+ ship pkg-config
+- macos: add -mmacosx-version-min for framework
+- gstreamer-1.0-osx-framework.recipe contains an outdated hard-coded
+ list of libraries
+- We need to ship pkg-config with macOS
+
+Linux
+
+- Fix filesprovider.find_shlib_regex when a lib_suffix is used in the
+ cerbero config file
+
+Contributors to 1.16.2
+
+Adam Nilsson, Amr Mahdi, Angus Ao, Charlie Turner, Edward Hervey, Fabian
+Greffrath, Fuwei Tang, Havard Graff, Hu Qian, James Cowgill, Jan
+Alexander Steffens (heftig), Jeffy Chen, Jeremy Lempereur, Joakim
+Johansson, Jochen Henneberg, Julien Isorce, Kevin Joly, Kristofer
+Bjorkstrom, Kyrylo Polezhaiev, Matthew Waters, Michael Olbrich, Muhammet
+Ilendemli, Nicolas Dufresne, Nirbheek Chauhan, Pablo Marcos Oltra, Roman
+Shpuntov, Ruben Gonzalez, Scott Kanowitz, Sebastian Dröge, Seungha Yang,
+Thibault Saunier, Tim-Philipp Müller, Víctor Manuel Jáquez Leal, Vivia
+Nikolaidou,
+
+… and many others who have contributed bug reports, translations, sent
+suggestions or helped testing. Thank you all!
+
+List of merge requests and issues fixed in 1.16.2
+
+- List of Merge Requests applied in 1.16
+- List of Issues fixed in 1.16.2
+
Known Issues
@@ -1721,9 +1898,8 @@ unstable development version leading up to the stable 1.18 release. The
development of 1.17/1.18 will happen in the git master branch.
The plan for the 1.18 development cycle is yet to be confirmed, but it
-is now expected that feature freeze will take place shortly after the
-GStreamer conference/hackfest in early November 2019, with the first
-1.18 stable release ready in late November or early December.
+is now expected that feature freeze will take place in December 2019,
+with the first 1.18 stable release ready in late January or February.
1.18 will be backwards-compatible to the stable 1.16, 1.14, 1.12, 1.10,
1.8, 1.6, 1.4, 1.2 and 1.0 release series.
diff --git a/RELEASE b/RELEASE
index 1e0814b15..662aafac8 100644
--- a/RELEASE
+++ b/RELEASE
@@ -1,4 +1,4 @@
-This is GStreamer gst-plugins-base 1.16.1.
+This is GStreamer gst-plugins-base 1.16.2.
The GStreamer team is pleased to announce another bug-fix release in the
stable 1.x API series of your favourite cross-platform multimedia framework!
diff --git a/configure.ac b/configure.ac
index 5bb5fe163..180f7aad9 100644
--- a/configure.ac
+++ b/configure.ac
@@ -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/prerelease
-AC_INIT([GStreamer Base Plug-ins],[1.16.1],[http://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer],[gst-plugins-base])
+AC_INIT([GStreamer Base Plug-ins],[1.16.2],[http://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer],[gst-plugins-base])
AG_GST_INIT
@@ -56,10 +56,10 @@ dnl 1.2.5 => 205
dnl 1.10.9 (who knows) => 1009
dnl
dnl sets GST_LT_LDFLAGS
-AS_LIBTOOL(GST, 1601, 0, 1601)
+AS_LIBTOOL(GST, 1602, 0, 1602)
dnl *** required versions of GStreamer stuff ***
-GST_REQ=1.16.1
+GST_REQ=1.16.2
dnl *** autotools stuff ****
diff --git a/gst-plugins-base.doap b/gst-plugins-base.doap
index 5924fef31..1d2fdf29e 100644
--- a/gst-plugins-base.doap
+++ b/gst-plugins-base.doap
@@ -36,6 +36,16 @@ A wide range of video and audio decoders, encoders, and filters are included.
<release>
<Version>
+ <revision>1.16.2</revision>
+ <branch>1.16</branch>
+ <name></name>
+ <created>2019-12-03</created>
+ <file-release rdf:resource="https://gstreamer.freedesktop.org/src/gst-plugins-base/gst-plugins-base-1.16.2.tar.xz" />
+ </Version>
+ </release>
+
+ <release>
+ <Version>
<revision>1.16.1</revision>
<branch>1.16</branch>
<name></name>
diff --git a/meson.build b/meson.build
index 874b3e467..767f98f11 100644
--- a/meson.build
+++ b/meson.build
@@ -1,5 +1,5 @@
project('gst-plugins-base', 'c',
- version : '1.16.1',
+ version : '1.16.2',
meson_version : '>= 0.47',
default_options : [ 'warning_level=1',
'buildtype=debugoptimized' ])