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authorTim-Philipp Müller <tim@centricular.com>2019-02-26 11:43:43 +0000
committerTim-Philipp Müller <tim@centricular.com>2019-02-26 11:43:44 +0000
commitaf1730252672d1ed68dfb7e1416af20eaa335156 (patch)
treee0476144680751494e78890ef5764965c77d92af
parent2864c635f7f5efd0a2c3d69e423fd2f9b3db73cb (diff)
Release 1.15.21.15.2
-rw-r--r--ChangeLog541
-rw-r--r--NEWS307
-rw-r--r--RELEASE2
-rw-r--r--configure.ac6
-rw-r--r--gst-plugins-base.doap10
-rw-r--r--meson.build2
6 files changed, 767 insertions, 101 deletions
diff --git a/ChangeLog b/ChangeLog
index 7ad00e47b..1088a5a2e 100644
--- a/ChangeLog
+++ b/ChangeLog
@@ -1,3 +1,544 @@
+=== release 1.15.2 ===
+
+2019-02-26 11:43:43 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * gst-plugins-base.doap:
+ * meson.build:
+ Release 1.15.2
+
+2019-02-26 11:43:42 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * docs/plugins/gst-plugins-base-plugins.args:
+ * 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
+
+2019-02-26 11:43:40 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * 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/fur.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/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:
+ Update translations
+
+2019-02-19 16:59:34 +0200 Vivia Nikolaidou <vivia@ahiru.eu>
+
+ * gst/videorate/gstvideorate.c:
+ * gst/videorate/gstvideorate.h:
+ * tests/check/elements/videorate.c:
+ videorate: Add max-duplication-time property
+ This will only duplicate buffers if the gap between two consecutive
+ buffers is up to fill-until nsec. If it's larger, it will only output
+ the new buffer and mark it as discont.
+
+2019-02-21 19:18:18 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
+
+ * ext/meson.build:
+ meson: Remove outdated msvc-specific disabling code
+ This was done ages ago when the meson build files were newly added
+ but now we do the appropriate disabling in Cerbero instead since this
+ does not apply to gst-build.
+ https://gitlab.freedesktop.org/gstreamer/cerbero/issues/121
+
+2019-02-20 09:46:30 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/check/libs/video.c:
+ tests: video: add basic sanity check of pstrides for formats
+ https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/merge_requests/117
+
+2019-02-09 17:21:13 +0000 James Cowgill <jcowgill@jcowgill.uk>
+
+ * gst-libs/gst/video/video-format.c:
+ video-format: Fix GBRA_10/12 alpha channel pixel strides
+ These formats have 4 components, so they should also have 4 components
+ of pixel stride.
+
+2019-01-17 15:38:40 +0100 Victor Toso <me@victortoso.com>
+
+ * tests/check/libs/video.c:
+ tests: use GPOINTER_TO_INT to avoid warnings with mingw
+ New casts to avoid the the warnings mentioned below. While at it, move
+ some existing casts (introduced at 61bc9091894062b9) to use
+ GPOINTER_TO_INT too.
+ [458/673] Compiling C object 'tests/check/7d01337@@libs_video@exe/libs_video.c.obj'.
+ ../tests/check/libs/video.c: In function 'fourcc_get_size':
+ ../tests/check/libs/video.c:160:10: warning: cast from pointer to integer of different size [-Wpointer-to-int-cast]
+ return (unsigned long) p->endptr;
+ ^
+ In file included from ../tests/check/libs/video.c:32:
+ ../tests/check/libs/video.c: In function 'test_video_formats':
+ ../tests/check/libs/video.c:563:39: warning: cast from pointer to integer of different size [-Wpointer-to-int-cast]
+ fail_unless_equals_int (size, (unsigned long) paintinfo.endptr);
+ ^
+ And more.
+ https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/merge_requests/94
+
+2019-01-17 15:25:58 +0100 Victor Toso <me@victortoso.com>
+
+ * tests/check/libs/profile.c:
+ tests: fix compiler warnings on Windows with mingw
+ With commit 3f184c3abc55, the gst_dir variable becomes unusable in
+ windows build. Moving it to linux scope to avoid warning:
+ [433/673] Compiling C object 'tests/check/7d01337@@libs_profile@exe/libs_profile.c.obj'.
+ ../tests/check/libs/profile.c: In function 'profile_suite':
+ ../tests/check/libs/profile.c:688:10: warning: unused variable 'gst_dir' [-Wunused-variable]
+ gchar *gst_dir;
+ ^~~~~~~
+ Also fix a typo in the comment.
+
+2019-02-18 15:24:18 +0100 Kristofer Bjorkstrom <kristofb@axis.com>
+
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ rtspconnection: Fix GError set over the top of a previous GError
+ The function fill_bytes could sometimes return a value greater than zero
+ and in the same time set the GError.
+ Function read_bytes calls fill_bytes in a while loop. In the special
+ case above it would call fill_bytes with error already set.
+ Thus resulting in "GError set over the top of a previous GError".
+ Solved this by clearing GError when return value is greater than zero.
+ Actions are taken depending on error type by caller of read_bytes. Eg.
+ with EWOULDBLOCK gst_rtsp_source_dispatch_read will try to read the
+ missing bytes again (GST_RTSP_EINTR )
+ https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/issues/445
+
+2019-02-18 13:28:49 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst-libs/gst/gl/egl/gsteglimage.c:
+ gl: eglimage: fix build on RPi by adding more fallback defines for EGL_*_EXT
+
+2018-11-16 23:51:44 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * tests/check/libs/video.c:
+ tests: video: Test video format enum stability
+ It is really easy to break the API and insert a new video format in the
+ middle of the enum instead of at the end. This minimal test should catch
+ the most obvious errors. Ideally, this test should be updated after new
+ format have been added, so that it won't allow further modification to
+ the enumeration API.
+
+2019-02-16 15:29:57 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst-libs/gst/pbutils/descriptions.c:
+ pbutils: add description for AV1 codec
+ Fixes #558
+
+2019-02-15 16:45:09 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * ext/gl/gstglimagesink.c:
+ glimagesink: Don't call set_property helper in get_property
+
+2019-02-13 11:59:10 +0100 Edward Hervey <edward@centricular.com>
+
+ * gst-libs/gst/gl/wayland/Makefile.am:
+ wayland: Also dist the private header
+
+2019-02-11 10:01:55 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * gst-libs/gst/gl/egl/gsteglimage.c:
+ eglimage: Add some more defines
+ This allow building on advertised version of libdrm drm_fourcc.h files.
+ Fixes #549
+
+2019-02-11 10:01:50 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * gst-libs/gst/gl/egl/gsteglimage.c:
+ Revert "fix issue"
+ This reverts commit 5e0c458e0ef544f1afae13c5eb047bc0826b011a.
+
+2019-02-11 16:13:15 +0800 yanle.zhang <yanle.zhang@hobot.cc>
+
+ * gst-libs/gst/gl/egl/gsteglimage.c:
+ fix issue 549."https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/issues/549".
+
+2019-01-30 10:49:37 -0300 Thibault Saunier <tsaunier@igalia.com>
+
+ * tools/gst-device-monitor.c:
+ tools: device-monitor: Add support for modified devices
+
+2019-02-08 21:38:04 +0900 Seungha Yang <seungha.yang@navercorp.com>
+
+ * gst-libs/gst/gl/gstglupload.c:
+ glupload: Don't leak caps features
+ Create caps features when it is required.
+
+2018-12-14 16:33:50 +0100 Niels De Graef <niels.degraef@barco.com>
+
+ * gst-libs/gst/gl/meson.build:
+ * gst-libs/gst/gl/wayland/Makefile.am:
+ * gst-libs/gst/gl/wayland/gstgldisplay_wayland.c:
+ * gst-libs/gst/gl/wayland/gstgldisplay_wayland.h:
+ * gst-libs/gst/gl/wayland/gstgldisplay_wayland_private.h:
+ * gst-libs/gst/gl/wayland/gstglwindow_wayland_egl.c:
+ * gst-libs/gst/gl/wayland/gstglwindow_wayland_egl.h:
+ * m4/gst-gl.m4:
+ gl/wayland: add support for XDG-shell
+ [wl_shell] is officially [deprecated], so provide support for the
+ XDG-shell protocol should be provided by all desktop-like compositors.
+ (In case they don't, we can of course fall back to wl_shell).
+ Note that the [XML spec] is provided by the `wayland-protocols`
+ git repository, which is provided by the Wayland project.
+ [wl_shell]: https://people.freedesktop.org/~whot/wayland-doxygen/wayland/Client/group__iface__wl__shell.html
+ [deprecated]: https://github.com/wayland-project/wayland/commit/698dde195837f3d0844b2725ba4ea8ce9ee7518c
+ [XML spec]: https://github.com/wayland-project/wayland-protocols/blob/master/stable/xdg-shell/xdg-shell.xml
+
+2018-12-14 14:54:24 +0100 Niels De Graef <niels.degraef@barco.com>
+
+ * gst-libs/gst/gl/wayland/gstglwindow_wayland_egl.c:
+ gl/wayland: extract code to create wl_shell_surface
+ This is just a cosmetic change that will make it easier to differentiate
+ between wl_shell and xdg_wm_base later.
+
+2018-12-14 14:28:26 +0100 Niels De Graef <niels.degraef@barco.com>
+
+ * gst-libs/gst/gl/wayland/gstgldisplay_wayland.c:
+ * gst-libs/gst/gl/wayland/gstgldisplay_wayland.h:
+ * gst-libs/gst/gl/wayland/gstglwindow_wayland_egl.c:
+ * gst-libs/gst/gl/wayland/gstglwindow_wayland_egl.h:
+ gl/wayland: prefix shell(_surface) with wl_
+ This will help us make the distinction later with xdg-shell and other
+ possible protocols that need to be supported.
+
+2019-02-05 22:06:15 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
+
+ * tests/check/elements/videoscale.c:
+ * tests/check/libs/profile.c:
+ * tests/check/libs/rtpbasedepayload.c:
+ misc: Fix compiler warnings on Cerbero's MinGW
+ rtpbasedepayload.c:126:5: error: unknown conversion type character 'z' in format [-Werror=format]
+ profile.c:688:10: error: unused variable 'gst_dir' [-Werror=unused-variable]
+
+2019-02-04 11:48:25 +0100 Guillaume Desmottes <guillaume.desmottes@collabora.com>
+
+ * gst-libs/gst/video/gstvideodecoder.c:
+ videodecoder: remove useless code in negotiate_default_caps()
+ gst_video_decoder_negotiate_default_caps() is meant to pick a default output
+ format when we need one earlier because of an incoming GAP.
+ It tries to use the input caps as a base if available and fallback to a default
+ format (I420 1280x720@30) for the missing fields.
+ But the framerate and pixel-aspect were not explicitly passed to
+ gst_video_decoder_set_output_state() which is solely relying on the input format
+ as reference to get the framerate anx pixel-aspect-ratio.
+ So there is no need to manually handling those two fields as
+ gst_video_decoder_set_output_state() will already use the ones from
+ upstream if available, and they will be ignored anyway if there are not.
+ This also prevent confusing debugging output where we claim to use a
+ specific framerate while actually none was set.
+
+2019-01-31 15:22:21 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
+
+ * tests/check/meson.build:
+ meson: orc-test is not required
+ This is especially never available on iOS.
+
+2019-01-30 14:32:50 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ rtspconnection: Fix uninitialized variable warning when compiling with pre-2.59.1 GLib
+ gstrtspconnection.c: In function ‘writev_bytes’:
+ gstrtspconnection.c:1348:10: error: ‘res’ may be used uninitialized in this function [-Werror=maybe-uninitialized]
+ return res;
+ ^
+
+2019-01-30 20:41:13 +0900 Seungha Yang <seungha.yang@navercorp.com>
+
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ rtspconnection: Fix broken build on GLib 2.59.0
+ GPollableReturn enum was introduced after GLib 2.59.0 release.
+
+2019-01-29 10:38:15 +0900 Seungha Yang <seungha.yang@navercorp.com>
+
+ * meson.build:
+ * tests/check/meson.build:
+ meson: Add support orc fallback
+ Allow fallback to orc subproject if any.
+ Additionally 'dependencies' keyword is removed from find_library,
+ because it's invalid keyword for find_library.
+
+2019-01-17 18:04:11 -0300 Thibault Saunier <tsaunier@igalia.com>
+
+ * gst/typefind/gsttypefindfunctions.c:
+ typefindfunctions: Add a function to typefind xges files
+
+2019-01-27 12:35:12 +0900 mrk501 <mrk501e@outlook.com>
+
+ * gst-libs/gst/audio/gstaudioringbuffer.c:
+ audioringbuffer: Fix wrong memcpy address when reordering channels
+ When using multichannel audio data and being needed to reorder channels,
+ audio data is not copied correctly because destination address of
+ memcpy is wrong.
+ For example, the following command
+ $ gst-launch-1.0 pulsesrc ! audio/x-raw,channels=6,format=S16LE ! filesink location=test.raw
+ will reproduce this issue if there is 6-ch audio input device.
+ This commit fixes that.
+ The detailed process of this issue is as follows:
+ 1. gst-launch-1.0 calls gst_pulsesrc_prepare (gst-plugins-good/ext/pulse/pulsesrc.c)
+ 1466 gst_pulsesrc_prepare (GstAudioSrc * asrc, GstAudioRingBufferSpec * spec)
+ 1467 {
+ (skip...)
+ 1480 {
+ 1481 GstAudioRingBufferSpec s = *spec;
+ 1482 const pa_channel_map *m;
+ 1483
+ 1484 m = pa_stream_get_channel_map (pulsesrc->stream);
+ 1485 gst_pulse_channel_map_to_gst (m, &s);
+ 1486 gst_audio_ring_buffer_set_channel_positions (GST_AUDIO_BASE_SRC
+ 1487 (pulsesrc)->ringbuffer, s.info.position);
+ 1488 }
+ In my environment, after line 1485 is processed, position of spec and s are
+ spec->info.position[0] = 0
+ spec->info.position[1] = 1
+ spec->info.position[2] = 2
+ spec->info.position[3] = 6
+ spec->info.position[4] = 7
+ spec->info.position[5] = 8
+ s.info.position[0] = 0
+ s.info.position[1] = 6
+ s.info.position[2] = 2
+ s.info.position[3] = 1
+ s.info.position[4] = 7
+ s.info.position[5] = 8
+ The values of spec->info.positions equal
+ GST_AUDIO_BASE_SRC(pulsesrc)->ringbuffer->spec->info.positions.
+ 2. gst_audio_ring_buffer_set_channel_positions calls
+ gst_audio_get_channel_reorder_map.
+ 3. Arguments of gst_audio_get_channel_reorder_map are
+ from = s.info.position
+ to = GST_AUDIO_BASE_SRC(pulsesrc)->ringbuffer->spec->info.positions
+ At the end of this function, reorder_map is set to
+ reorder_map[0] = 0
+ reorder_map[1] = 3
+ reorder_map[2] = 2
+ reorder_map[3] = 1
+ reorder_map[4] = 4
+ reorder_map[5] = 5
+ 4. Go back to gst_audio_ring_buffer_set_channel_positions and
+ 2065 buf->need_reorder = TRUE;
+ is processed.
+ 5. Finally, in gst_audio_ring_buffer_read,
+ 1821 if (need_reorder) {
+ (skip...)
+ 1829 memcpy (data + i * bpf + reorder_map[j] * bps, ptr + j * bps, bps);
+ is processed and makes this issue.
+
+2019-01-24 17:52:50 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ rtspconnection: Update to merged GOutputStream::writev() API
+
+2018-11-30 12:47:57 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ rtspconnection: Handle EOF on writev() after checking for all other error conditions
+ Otherwise we would return EOF if nothing was written in any case, even
+ if this was actually a case of TIMEOUT or EWOULDBLOCK for example.
+ Thanks to Edward Hervey for debugging and finding this issue.
+
+2018-10-24 11:32:22 +0200 Ognyan Tonchev <ognyan@axis.com>
+
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ rtspconnection: Fixes for corrupt RTP packets in dispatch_write()
+ Fixes 2 problems:
+ 1) Number of unmapped memories does not always match number of mmaped ones in
+ dispatch_write().
+ 2) When dispatch_write() is dispatched second time after an incomplete write,
+ already set offsets will not be taken into account, thus corrupt RTP data will
+ be sent.
+
+2018-09-17 17:03:45 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ * gst-libs/gst/rtsp/gstrtspconnection.h:
+ rtsp-connection: Make use of new GstRTSPMessage API for directly storing a body buffer and add API for writing multiple messages
+ By doing so we can send a whole GstBufferList and each memory in the
+ contained buffers without copying into a single memory area and with a
+ single writev() call. This improves performance considerably for
+ high-packet-rate streams.
+ This depends on https://gitlab.gnome.org/GNOME/glib/merge_requests/333
+ to be efficient, otherwise each chunk of memory is a separate write()
+ call.
+ https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/issues/370
+
+2018-08-17 12:51:31 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * docs/libs/gst-plugins-base-libs-sections.txt:
+ * gst-libs/gst/rtsp/gstrtspmessage.c:
+ * gst-libs/gst/rtsp/gstrtspmessage.h:
+ rtsp-message: Add support for storing GstBuffers directly as body payload of messages
+ This makes it unnecessary for callers to first merge together all
+ memories, and it allows API like GstRTSPConnection to write them out
+ without first copying all memories together or using writev()-style API
+ to write multiple memories out in one go.
+ Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/issues/370
+
+2019-01-28 15:16:06 +0100 Andrew Gall <a.gall@activevideo.com>
+
+ * gst-libs/gst/video/video-anc.c:
+ video-anc: Fix glib version check for G_GNUC_CHECK_VERSION macro
+ Fixes #544
+
+2019-01-28 13:54:43 +0900 Seungha Yang <seungha.yang@navercorp.com>
+
+ * tests/check/libs/discoverer.c:
+ tests: discoverer: Add async API test cases
+ Add more test cases for async APIs such as gst_discoverer_{start,stop},
+ and gst_discoverer_discover_uri_async()
+
+2019-01-28 18:13:27 +0900 Seungha Yang <seungha.yang@navercorp.com>
+
+ * gst-libs/gst/pbutils/gstdiscoverer.c:
+ discoverer: Hold GSource object instead of source id
+ g_source_remove() works only for a GSource which was attached
+ to default GMainContext, but the GSource might be attached to
+ custom context depending on how gst_discoverer_start() was called.
+ Whatever the attached context was, g_source_destroy() can clean it up.
+
+2019-01-24 10:14:36 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * ext/gl/gstglcolorbalance.c:
+ glcolorbalance: Copy caps in transform_internal_caps()
+ We don't get ownership of the caps that are passed in, and doing so
+ causes crashes at a later time.
+ Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/issues/546
+
+2019-01-22 13:24:29 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst-libs/gst/gl/meson.build:
+ meson: opengl: fix enabled_gl_apis in pkg-config file
+ Make consistent with what autotools puts into enabled_gl_apis
+ variable. Autotools puts 'gl' in there instead of 'opengl'.
+ This would cause problems when building -bad glmixers plugin
+ in meson against a -base that was built with autotools.
+ See https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/issues/871
+
+2018-12-19 10:59:09 +0800 Haihao Xiang <haihao.xiang@intel.com>
+
+ * gst-libs/gst/gl/x11/gstglwindow_x11.c:
+ gstglwindow_x11: require a resize event at once after XResizeWindow
+ Otherwise surface_width/surface_height stored in GstGLWindowPrivate
+ isn't changed, sometimes an unnecessary reconfigure event is sent on
+ sinkpad, then result in upstream reconfiguring.
+ Example pipeline:
+ gst-launch-1.0 videotestsrc ! msdkvpp ! glimagesink
+
+2019-01-18 11:39:02 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * ext/alsa/Makefile.am:
+ * ext/alsa/gstalsadeviceprobe.c:
+ * ext/alsa/gstalsadeviceprobe.h:
+ * ext/alsa/gstalsadeviceprovider.c:
+ * ext/alsa/gstalsadeviceprovider.h:
+ * ext/alsa/gstalsaplugin.c:
+ * ext/alsa/gstalsasink.c:
+ * ext/alsa/gstalsasrc.c:
+ * ext/alsa/meson.build:
+ Revert "alsa: Implement a DeviceProvider"
+ This reverts commit 69c3c31608ecebfadd9717e950d8c708988563e3.
+ All devices have the same name, they are duplicated with pulseaudio one
+ and the provided does not respond to HW being plugged/unplugged. I think
+ it's not ready for 1.16.
+
+2018-08-31 18:33:43 -0300 Thibault Saunier <tsaunier@igalia.com>
+
+ * ext/alsa/Makefile.am:
+ * ext/alsa/gstalsadeviceprobe.c:
+ * ext/alsa/gstalsadeviceprobe.h:
+ * ext/alsa/gstalsadeviceprovider.c:
+ * ext/alsa/gstalsadeviceprovider.h:
+ * ext/alsa/gstalsaplugin.c:
+ * ext/alsa/gstalsasink.c:
+ * ext/alsa/gstalsasrc.c:
+ * ext/alsa/meson.build:
+ alsa: Implement a DeviceProvider
+ Removing gstalsadeviceprobe.[ch] as it was a relique from the 0.10
+ century.
+
+2018-12-07 18:07:42 +0200 George Kiagiadakis <george.kiagiadakis@collabora.com>
+
+ * gst-libs/gst/video/gstvideoaggregator.c:
+ videoaggregator: remove broken rate adjustment
+ The start_time and end_time in this context have already
+ been adjusted for the input's rate by converting them to running
+ time above. What is needed afterwards is to compare these
+ with the output's start/stop running time, which also takes
+ into account the rate, so we are comparing equal things.
+ Multiplying these with the output's rate here is only breaking
+ this logic. In most cases the input and output rate is the same,
+ so this multiplication effectively reverses the rate adjustment
+ that happened while converting to running time, which is why
+ we see the video playing with the original rate in tests.
+ Fixes #541
+
=== release 1.15.1 ===
2019-01-17 01:50:25 +0000 Tim-Philipp Müller <tim@centricular.com>
diff --git a/NEWS b/NEWS
index 1e860c47a..6457a5d99 100644
--- a/NEWS
+++ b/NEWS
@@ -15,7 +15,7 @@ the git master branch and which will eventually result in 1.16.
See https://gstreamer.freedesktop.org/releases/1.16/ for the latest
version of this document.
-_Last updated: Monday 14 January 2019, 13:00 UTC (log)_
+_Last updated: Monday 25 January 2019, 15:00 UTC (log)_
Introduction
@@ -24,8 +24,8 @@ 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
framework!
-As always, this release is again packed with new features, bug fixes and
-other improvements.
+As always, this release is again packed with many new features, bug
+fixes and other improvements.
Highlights
@@ -40,7 +40,7 @@ Highlights
- Support for Closed Captions and other Ancillary Data in video
-- Spport for planar (non-interleaved) raw audio
+- Support for planar (non-interleaved) raw audio
- GstVideoAggregator, compositor and OpenGL mixer elements are now in
-base
@@ -98,14 +98,17 @@ Noteworthy new API
to process the media in a live pipeline before it reaches the sink.
This is on top of the systemic latency that is normally reported by
the latency query. This defaults to 20ms and should make pipelines
- such as “v4lsrc ! xvimagesink” not claim that all frames are late in
- the QoS events. Ideally, this should replace max_lateness for most
- applications.
+ such as v4l2src ! xvimagesink not claim that all frames are late in
+ the QoS events. Ideally, this should replace the "max-lateness"
+ property for most applications.
- RTCP Extended Reports (XR) parsing according to RFC 3611:
Loss/Duplicate RLE, Packet Receipt Times, Receiver Reference Time,
Delay since the last Receiver (DLRR), Statistics Summary, and VoIP
- Metrics reports.
+ Metrics reports. This only provides the ability to parse such
+ packets, generation of XR packets is not supported yet and XR
+ packets are not automatically parsed by rtpbin / rtpsession but must
+ be actively handled by the application.
- a new mode for interlaced video was added where each buffer carries
a single field of interlaced video, with buffer flags indicating
@@ -146,9 +149,10 @@ or planar arrangement in memory would look like
|LEFT|LEFT|LEFT| and |RIGHT|RIGHT|RIGHT| residing in separate memory
chunks or separated by some padding.
-GStreamer has always had signalling for non-interleaved audio, but it
-was never actually properly implemented in any elements. audioconvert
-would advertise support for it, but wasn’t actually able to handle it.
+GStreamer has always had signalling for non-interleaved audio since
+version 1.0, but it was never actually properly implemented in any
+elements. audioconvert would advertise support for it, but wasn’t
+actually able to handle it correctly.
With this release we now have full support for non-interleaved audio as
well, which means more efficient integration with external APIs that
@@ -177,18 +181,18 @@ The video support library has gained support for detecting and
extracting Ancillary Data from videos as per the SMPTE S291M
specification, including:
-- a VBI (Video Blanking Interval) parser that can detect and extract
- Ancillary Data from Vertical Blanking Interval lines of component
- signals. This is currently supported for videos in v210 and UYVY
- format.
+- a VBI (Vertical Blanking Interval) parser that can detect and
+ extract Ancillary Data from Vertical Blanking Interval lines of
+ component signals. This is currently supported for videos in v210
+ and UYVY format.
- a new GstMeta for closed captions: GstVideoCaptionMeta. This
supports the two types of closed captions, CEA-608 and CEA-708,
along with the four different ways they can be transported (other
systems are a superset of those).
-- a VBI (Video Blanking Interval) encoder for writing ancillary data
- to the Vertical Blanking Interval lines of component signals.
+- a VBI (Vertical Blanking Interval) encoder for writing ancillary
+ data to the Vertical Blanking Interval lines of component signals.
The new closedcaption plugin in gst-plugins-bad then makes use of all
this new infrastructure and provides the following elements:
@@ -222,6 +226,9 @@ support:
- playbin and playbin3 learned how to autoplug CEA 608/708 CC overlay
elements
+- the externally maintained ajavideosrc element for AJA capture cards
+ has support for extracting closed captions
+
The rsclosedcaption plugin in the Rust plugins collection includes a
MacCaption (MCC) file parser and encoder.
@@ -239,7 +246,7 @@ New Elements
- gloverlaycompositor: New OpenGL-based compositor element that
flattens any overlays from GstVideoOverlayCompositionMetas into the
- video stream.
+ video stream. This element is also always part of glimagesink.
- glalpha: New element that adds an alpha channel to a video stream.
The values of the alpha channel can either be set to a constant or
@@ -248,7 +255,7 @@ New Elements
done in floating point so results may not be identical to the output
of the existing alpha element.
-- rtpfunnel funnels together rtp-streams into a single session. Use
+- rtpfunnel funnels together RTP streams into a single session. Use
cases include multiplexing and bundle. webrtcbin uses it to
implement BUNDLE support.
@@ -264,10 +271,12 @@ New Elements
WPE
- Two new OpenCV-based elements: cameracalibrate and cameraundistort
- who can communicate to figure out distortion correction parameters
+ that can communicate to figure out distortion correction parameters
for a camera and correct for the distortion.
-- new sctp plugin based on usrsctp with sctpenc and sctpdec elements
+- New sctp plugin based on usrsctp with sctpenc and sctpdec elements.
+ These elements are used inside webrtcbin for implementing data
+ channels.
New element features and additions
@@ -348,12 +357,12 @@ New element features and additions
- rtspsrc now allows applications to send RTSP SET_PARAMETER and
GET_PARAMETER requests using action signals.
-- rtspsrc also has a small (100ms) configurable teardown delay by
- default to try and make sure an RTSP TEARDOWN request gets sent out
- when the source element shuts down. This will block the downward
- PAUSED to READY state change for a short time, but can be unset
- where it’s a problem. Some servers only allow a limited number of
- concurren clients, so if no proper TEARDOWN is sent clients may have
+- rtspsrc has a small (100ms) configurable teardown delay by default
+ to try and make sure an RTSP TEARDOWN request gets sent out when the
+ source element shuts down. This will block the downward PAUSED to
+ READY state change for a short time, but can be disabled where it’s
+ a problem. Some servers only allow a limited number of concurrent
+ clients, so if no proper TEARDOWN is sent new clients may have
problems connecting to the server for a while.
- souphttpsrc behaves better with low bitrate streams now. Before it
@@ -364,6 +373,7 @@ New element features and additions
- filesink: do internal buffering to avoid performance regression with
small writes since we bypass libc buffering by using writev()
+ instead of fwrite()
- identity: add "eos-after" property and fix "error-after" property
when the element is reused
@@ -405,9 +415,9 @@ New element features and additions
relays (TURN servers).
- The removesilence element has received various new features and
- properties, such as a
- "threshold"1 property, detecting silence only after minimum silence time/buffers, a“silent”property to control bus message notifications as well as a“squash”`
- property.
+ properties, such as a "threshold" property, detecting silence only
+ after minimum silence time/buffers, a "silent" property to control
+ bus message notifications as well as a "squash" property.
- AOMedia AV1 decoder gained support for 10/12bit decoding whilst the
AV1 encoder supports more image formats and subsamplings now and
@@ -430,15 +440,15 @@ Plugin and library moves
- The stereo element was moved from -bad into the existing audiofx
plugin in -good. If you get duplicate type registration warnings
- when upgrading, check that you don’t have a stale gststereo plugin
- lying about somewhere.
+ when upgrading, check that you don’t have a stale stereoplugin lying
+ about somewhere.
GstVideoAggregator, compositor, and OpenGL mixer elements moved from -bad to -base
GstVideoAggregator is a new base class for raw video mixers and muxers
-and is based on [GstAggregator][aggregator]. It provides defined-latency
-mixing of raw video inputs and ensures that the pipeline won’t stall
-even if one of the input streams stops producing data.
+and is based on GstAggregator. It provides defined-latency mixing of raw
+video inputs and ensures that the pipeline won’t stall even if one of
+the input streams stops producing data.
As part of the move to stabilise the API there were some last-minute API
changes and clean-ups, but those should mostly affect internal elements.
@@ -456,14 +466,15 @@ would expected in most scenarios.
The compositor element has gained support for per-pad blending mode
operators (SOURCE, OVER, ADD) which determines what operator to use for
blending this pad over the previous ones. This can be used to implement
-crossfading.
+crossfading and the available operators can be extended in the future as
+needed.
A number of OpenGL-based video mixer elements (glvideomixer, glmixerbin,
glvideomixerelement, glstereomix, glmosaic) which are built on top of
GstVideoAggregator have also been moved from -bad to -base now. These
elements have been merged into the existing OpenGL plugin, so if you get
duplicate type registration warnings when upgrading, check that you
-don’t have a stale gstopenglmixers plugin lying about somewhere.
+don’t have a stale openglmixers plugin lying about somewhere.
Plugin removals
@@ -477,11 +488,11 @@ The following plugins have been removed from gst-plugins-bad:
plugin.
- The acmmp3dec and acmenc plugins for Windows have been removed. ACM
- is an ancient legacy API and there was no point in keeping them
- around for a licensed mp3 decoder now that mp3 patents have expired
- and we have a decoder in -good. We also didn’t ship these in our
- cerbero-built Windows packages, so it’s unlikely that they’ll be
- missed.
+ is an ancient legacy API and there was no point in keeping the
+ plugins around for a licensed MP3 decoder now that the MP3 patents
+ have expired and we have a decoder in -good. We also didn’t ship
+ these in our cerbero-built Windows packages, so it’s unlikely that
+ they’ll be missed.
Miscellaneous API additions
@@ -506,7 +517,8 @@ Miscellaneous API additions
one might need to put such elements into READY state to test if the
hardware is present in the system for example.
-- protection: Add a new definition for unspecified system protection
+- protection: Add a new definition for unspecified system protection,
+ GST_PROTECTION_UNSPECIFIED_SYSTEM_ID
- take functions for various mini objects that didn’t have them yet:
gst_query_take(), gst_message_take(), gst_tag_list_take(),
@@ -522,11 +534,18 @@ Miscellaneous API additions
gst_clear_mini_object(), gst_clear_object()
- miniobject: new API gst_mini_object_add_parent() and
- gst_mini_object_remove_parent()to set parent pointers on mini objects to ensure correct writability: Every container of miniobjects now needs to store itself as parent in the child object, and remove itself again later. A mini object is then only writable if there is at most one parent, that parent is writable itself, and the reference count of the mini object is 1.GstBuffer(for memories),GstBufferList(for buffers),GstSample(for caps, buffer, bufferlist), andGstVideoOverlayComposition`
- were updated accordingly. Without this it was possible to have
- e.g. a buffer list with a refcount of 2 used in two places at once
- that both modify the same buffer with refcount 1 at the same time
- wrongly thinking it is writable even though it’s really not.
+ gst_mini_object_remove_parent() to set parent pointers on mini
+ objects to ensure correct writability: Every container of
+ miniobjects now needs to store itself as parent in the child object,
+ and remove itself again later. A mini object is then only writable
+ if there is at most one parent, that parent is writable itself, and
+ the reference count of the mini object is 1. GstBuffer (for
+ memories), GstBufferList (for buffers), GstSample (for caps, buffer,
+ bufferlist), and GstVideoOverlayComposition were updated
+ accordingly. Without this it was possible to have e.g. a buffer list
+ with a refcount of 2 used in two places at once that both modify the
+ same buffer with refcount 1 at the same time wrongly thinking it is
+ writable even though it’s really not.
- poll: add API to watch for POLLPRI and stop treating POLLPRI as a
read. This is useful to wait for video4linux events which are
@@ -596,7 +615,7 @@ GstPlayer
Miscellaneous changes
-- As a result of moving to different FFmpeg APIs, encoder and decoder
+- As a result of moving to newer FFmpeg APIs, encoder and decoder
elements exposed by the GStreamer FFmpeg wrapper plugin (gst-libav)
may have seen possibly incompatible changes to property names and/or
types, and not all properties exposed might be functional. We are
@@ -649,48 +668,48 @@ Tracing framework and debugging improvements
object. This is currently limited to pads for GstElements and
events for the pads. The output may look like this:
- (gdb) gst-print pad.object.parent
- GstMatroskaDemux (matroskademux0) {
- SinkPad (sink, pull) {
- }
- SrcPad (video_0, push) {
- events:
- stream-start:
- stream-id: 0463ccb080d00b8689bf569a435c4ff84f9ff753545318ae2328ea0763fd0bec/001:1274058367
- caps: video/x-theora
- width: 1920
- height: 800
- pixel-aspect-ratio: 1/1
- framerate: 24/1
- streamheader: < 0x5555557c7d30 [GstBuffer], 0x5555557c7e40 [GstBuffer], 0x7fffe00141d0 [GstBuffer] >
- segment: time
- rate: 1
- tag: global
- container-format: Matroska
- }
- SrcPad (audio_0, push) {
- events:
- stream-start:
- stream-id: 0463ccb080d00b8689bf569a435c4ff84f9ff753545318ae2328ea0763fd0bec/002:1551204875
- caps: audio/mpeg
- mpegversion: 4
- framed: true
- stream-format: raw
- codec_data: 0x7fffe0014500 [GstBuffer]
- level: 2
- base-profile: lc
- profile: lc
- channels: 2
- rate: 44100
- segment: time
- rate: 1
- tag: global
- container-format: Matroska
- tag: stream
- audio-codec: MPEG-4 AAC audio
- language-code: en
+ (gdb) gst-print pad.object.parent
+ GstMatroskaDemux (matroskademux0) {
+ SinkPad (sink, pull) {
+ }
+ SrcPad (video_0, push) {
+ events:
+ stream-start:
+ stream-id: 0463ccb080d00b8689bf569a435c4ff84f9ff753545318ae2328ea0763fd0bec/001:1274058367
+ caps: video/x-theora
+ width: 1920
+ height: 800
+ pixel-aspect-ratio: 1/1
+ framerate: 24/1
+ streamheader: < 0x5555557c7d30 [GstBuffer], 0x5555557c7e40 [GstBuffer], 0x7fffe00141d0 [GstBuffer] >
+ segment: time
+ rate: 1
+ tag: global
+ container-format: Matroska
+ }
+ SrcPad (audio_0, push) {
+ events:
+ stream-start:
+ stream-id: 0463ccb080d00b8689bf569a435c4ff84f9ff753545318ae2328ea0763fd0bec/002:1551204875
+ caps: audio/mpeg
+ mpegversion: 4
+ framed: true
+ stream-format: raw
+ codec_data: 0x7fffe0014500 [GstBuffer]
+ level: 2
+ base-profile: lc
+ profile: lc
+ channels: 2
+ rate: 44100
+ segment: time
+ rate: 1
+ tag: global
+ container-format: Matroska
+ tag: stream
+ audio-codec: MPEG-4 AAC audio
+ language-code: en
+ }
}
- }
- gst_structure_to_string() now serialises the actual value of
pointers when serialising GstStructures instead of claiming they’re
@@ -704,10 +723,10 @@ Tools
- gst-inspect-1.0 has coloured output now and will automatically use a
pager if the output does not fit on a page. This only works in a
- unix environment and if the output is not piped. If you don’t like
- the colours you can disable them by setting the
- GST_INSPECT_NO_COLORS=1 environment variable or passing the
- --no-colors command line option.
+ UNIX environment and if the output is not piped, and on Windows 10
+ build 16257 or newer. If you don’t like the colours you can disable
+ them by setting the GST_INSPECT_NO_COLORS=1 environment variable or
+ passing the --no-color command line option.
GStreamer RTSP server
@@ -736,6 +755,29 @@ GStreamer VAAPI
- this section will be filled in in due course
+GStreamer OMX
+
+- Add support of NV16 format to video encoders input.
+
+- Video decoders now handle the ALLOCATION query to tell upstream
+ about the number of buffers they require. Video encoders will also
+ use this query to adjust their number of allocated buffers
+ preventing starvation when using dynamic buffer mode.
+
+- The OMX_PERFORMANCE debug category has been renamed to OMX_API_TRACE
+ and can now be used to track a widder variety of interactions
+ between OMX and GStreamer.
+
+- Video encoders will now detect frame rate only changes and will
+ inform OMX about it rather than doing a full format reset.
+
+- Various Zynq UltraScale+ specific improvements:
+ - Video encoders are now able to import dmabuf from upstream.
+ - Support for HEVC range extension profiles and more AVC profiles.
+ - We can now request video encoders to generate an IDR using the
+ force key unit event.
+
+
GStreamer Editing Services and NLE
- this section will be filled in in due course
@@ -858,7 +900,7 @@ Added
value per tag. The old ::iter_tag_list() function was renamed to
::iter_generic() and still provides access to each value for a tag
- Bus::iter() and Bus::iter_timed() iterators around the corresponding
- ::pop*() functions
+ ::pop\*() functions
- serde serialization of Value can also handle Buffer now
@@ -995,11 +1037,78 @@ Build and Dependencies
- New sctp plugin based on usrsctp (for WebRTC data channels)
+Cerbero
+
+Cerbero is a meta build system used to build GStreamer plus dependencies
+on platforms where dependencies are not readily available, such as
+Windows, Android, iOS and macOS.
+
+Cerbero has seen a number of improvements:
+
+- Cerbero has been ported to Python 3 and requires Python 3.5 or newer
+ now
+
+- Source tarballs are now protected by checksums in the recipes to
+ guard against download errors and malicious takeover of projects or
+ websites. In addition, downloads are only allowed via secure
+ transports now and plain HTTP, FTP and git:// transports are not
+ allowed anymore.
+
+- There is now a new fetch-bootstrap command which downloads sources
+ required for bootstrapping, with an optional --build-tools-only
+ argument to match the bootstrap --build-tools-only command.
+
+- The bootstrap, build, package and bundle-source commands gained a
+ new --offline switch that ensures that only sources from the cache
+ are used and never downloaded via the network. This is useful in
+ combination with the fetch and fetch-bootstrap commands that acquire
+ sources ahead of time before any build steps are executed. This
+ allows more control over the sources used and when sources are
+ updated, and is particularly useful for build environments that
+ don’t have network access.
+
+- bootstrap --assume-yes will automatically say ‘yes’ to any
+ interactive prompts during the bootstrap stage, such as those from
+ apt-get or yum.
+
+- bootstrap --system-only will only bootstrap the system without build
+ tools.
+
+- Manifest support: The build manifest can be used in continuous
+ integration (CI) systems to fixate the Git revision of certain
+ projects so that all builds of a pipeline are on the same reference.
+ This is used in GStreamer’s gitlab CI for example. It can also be
+ used in order to re-produce a specific build. To set a manifest, you
+ can set manifest = 'my_manifest.xml' in your configuration file, or
+ use the --manifest command line option. The command line option will
+ take precendence over anything specific in the configuration file.
+
+- The new build-deps command can be used to build only the
+ dependencies of a recipe, without the recipe itself.
+
+- new --list-variants command to list available variants
+
+- variants can now be set on the command line via the -v option as a
+ comma-separated list. This overrides any variants set in any
+ configuration files.
+
+- new qt5, intelmsdk and nvidia variants for enabling Qt5 and hardware
+ codec support. See the Enabling Optional Features with Variants
+ section in the Cerbero documentation for more details how to enable
+ and use these variants.
+
+- A new -t / --timestamp command line switch makes commands print
+ timestamps
+
Platform-specific changes and improvements
Android
+- toolchain: update compiler to clang and NDKr18. NDK r18 removed the
+ armv5 target and only has Android platforms that target at least
+ armv7 so the armv5 target is not useful anymore.
+
- The way that GIO modules are named has changed due to upstream GLib
natively adding support for loading static GIO modules. This means
that any GStreamer application using gnutls for SSL/TLS on the
@@ -1010,9 +1119,15 @@ Android
library. Look at this commit for the necessary change in the
examples.
+- various build issues on Android have been fixed.
+
macOS and iOS
-- macOS binaries should be fully relocatable now
+- various build issues on iOS have been fixed.
+
+- the minimum required iOS version is now 9.0. The difference in
+ adoption between 8.0 and 9.0 is 0.1% and the bump to 9.0 fixes some
+ build issues.
- The way that GIO modules are named has changed due to upstream GLib
natively adding support for loading static GIO modules. This means
diff --git a/RELEASE b/RELEASE
index 24c5e024a..f07687c6d 100644
--- a/RELEASE
+++ b/RELEASE
@@ -1,4 +1,4 @@
-This is GStreamer gst-plugins-base 1.15.1.
+This is GStreamer gst-plugins-base 1.15.2.
GStreamer 1.15 is the development branch leading up to the next major
stable version which will be 1.16.
diff --git a/configure.ac b/configure.ac
index df9896623..5cd407cd1 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],[1.15.1],[http://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer],[gst-plugins-base])
+AC_INIT([GStreamer Base Plug-ins],[1.15.2],[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, 1501, 0, 1501)
+AS_LIBTOOL(GST, 1502, 0, 1502)
dnl *** required versions of GStreamer stuff ***
-GST_REQ=1.15.1
+GST_REQ=1.15.2
dnl *** autotools stuff ****
diff --git a/gst-plugins-base.doap b/gst-plugins-base.doap
index e735042ea..b793a918a 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>1.15.2</revision>
+ <branch>master</branch>
+ <name></name>
+ <created>2019-02-26</created>
+ <file-release rdf:resource="https://gstreamer.freedesktop.org/src/gst-plugins-base/gst-plugins-base-1.15.2.tar.xz" />
+ </Version>
+ </release>
+
+ <release>
+ <Version>
<revision>1.15.1</revision>
<branch>master</branch>
<name></name>
diff --git a/meson.build b/meson.build
index 8ea22656f..e6af5bc8b 100644
--- a/meson.build
+++ b/meson.build
@@ -1,5 +1,5 @@
project('gst-plugins-base', 'c',
- version : '1.15.1',
+ version : '1.15.2',
meson_version : '>= 0.47',
default_options : [ 'warning_level=1',
'buildtype=debugoptimized' ])