@@ -1,4 +1,9 @@

 -------------------------------------------------------------------

+Thu May 25 17:06:42 UTC 2023 - Bjørn Lie <zaitor@opensuse.org>

+

+- Update to version 1.22.3

+

+-------------------------------------------------------------------

 Mon Apr 24 13:16:44 UTC 2023 - Bjørn Lie <zaitor@opensuse.org>

 - Add pkgconfig(libopenaptx) BuildRequires, build the

​

@@ -7,7 +7,7 @@

 %define _version 1.22.0

 Name:           gstreamer-plugins-bad-codecs

-Version:        1.22.2

+Version:        1.22.3

 Release:        0

 Summary:        Codecs/plugins for gstreamer-plugins-bad

 License:        LGPL-2.1-or-later

​

@@ -1,3 +1,376 @@

+=== release 1.22.3 ===

+

+2023-05-19 09:23:19 +0100  Tim-Philipp Müller <tim@centricular.com>

+

+   * NEWS:

+   * README.md:

+   * RELEASE:

+   * gst-plugins-bad.doap:

+   * meson.build:

+     Release 1.22.3

+

+2023-04-20 16:41:11 +0800  Haihua Hu <jared.hu@nxp.com>

+

+   * gst-libs/gst/play/gstplay.c:

+     gstplay: fix critical log when enable playbin3

+     when play rtsp stream with playbin3 enabled, there are some critical logs:

+     g_object_get_is_valid_property: object class 'GstPlayBin3' has no property named 'n-video'

+     g_object_get_is_valid_property: object class 'GstPlayBin3' has no property named 'n-audio'

+     g_object_get_is_valid_property: object class 'GstPlayBin3' has no property named 'n-text'

+     self->collection could be NULL when READY->PAUSED if the pipeline

+     is live, then it will fallback to query playbin2's property,

+     we can call gst_play_streams_info_create_from_collection

+     directly, it will check self->collection internal.

+     Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4666>

+

+2023-03-02 18:09:54 +0800  Elliot Chen <elliot.chen@nxp.com>

+

+   * gst-libs/gst/play/gstplay.c:

+     gstplay: avoid getting property of playbin2 if subtitle_sid is null

+     There is a probability of getting "current-text" property

+     when play with playbin3, and this property is available

+     only in playbin2.

+     Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4661>

+

+2023-05-16 22:56:15 +0900  Seungha Yang <seungha@centricular.com>

+

+   * sys/d3d11/gstd3d11convert.cpp:

+     d3d11convert: Fix for runtime property update

+     Every setup happens in set_caps() method but basetransform will not

+     call the set_caps() if in/out caps were not changed

+     Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4652>

+

+2023-03-17 20:44:30 +0100  Aleksandr Slobodeniuk <aslobodeniuk@fluendo.com>

+

+   * sys/d3d11/gstd3d11convert.cpp:

+     d3d11convert: protect 'add-borders' with mutex

+     Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4652>

+

+2023-05-15 13:47:16 +0200  Carlos Rafael Giani <crg7475@mailbox.org>

+

+   * sys/bluez/gstavdtputil.c:

+     avdtputil: Use int instead of int range for fixed bitpool values

+     Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/1698

+     Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4655>

+

+2023-05-16 16:17:05 +0100  Tim-Philipp Müller <tim@centricular.com>

+

+   * gst-libs/gst/webrtc/nice/nice.c:

+     Revert "webrtc/nice: support consent-freshness RFC7675"

+     This reverts commit 0161687505156f54ff38b976668804ac2763a863.

+     This causes problems when the connection is congested because

+     libnice uses a too agressive timeout contrary to the spec, so

+     it's easy for consent to lapse and streaming to stop if there's

+     packet loss.

+     Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4649>

+

+2023-05-10 12:00:15 +0200  Johan Sternerup <johast@axis.com>

+

+   * ext/sctp/gstsctpenc.c:

+     sctpenc: Fix potential shutdown deadlock

+     When transitioning from state PAUSED to READY, the sctpenc element

+     could previously be stuck in an endless loop trying to resend data

+     in case the underlying sctp stream was in the process of

+     resetting. usrsctp_sendv() would repeatedly return EAGAIN with the

+     result that 0 bytes were sent and then sctpenc would retry forever.

+     To bring sctpenc out of the resend loop we just need to inform the

+     sink pad that it is flushing, which is already done for the associated

+     data queue, but we also need to set the bools associated with the

+     sinkpads that are used as the loop criterion.

+     Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4636>

+

+2023-04-18 11:21:05 +0200  Daniel Moberg <daniemob@axis.com>

+

+   * ext/webrtc/webrtcdatachannel.c:

+     webrtc: do not tear down data channel before data is flushed

+     Current implementation can in some cases detect

+     that all data is sent but in reality it is not,

+     leading to a push to an unlinked pad.

+     This is a race between the probe used to track data sent and a

+     call to close.

+     This patch sends an EOS before starting the close procedure

+     and then waits for the EOS event to come through to the

+     src pad before commencing with tear down.

+     This ensures that any queued data before EOS is flushed.

+     Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4633>

+

+2023-04-04 21:50:01 +1000  Jan Schmidt <jan@centricular.com>

+

+   * gst/mpegdemux/gstmpegdemux.c:

+   * gst/mpegdemux/gstmpegdemux.h:

+     mpegpsdemux: Rework gap sending

+     Take the gap logic from mpegtsdemux, and don't

+     send gap events on a stream that's outputting buffers with

+     no timestamps. Time isn't advancing, but the stream has

+     buffers - so it's not sparse.

+     Fixes #2374

+     Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4623>

+

+2023-03-05 20:15:19 +0900  Seungha Yang <seungha@centricular.com>

+

+   * sys/va/gstvavp8dec.c:

+     vavp8dec: Fix return type of decode_picture()

+     Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4614>

+

+2023-03-05 20:14:06 +0900  Seungha Yang <seungha@centricular.com>

+

+   * sys/va/gstjpegdecoder.h:

+     vajpegdec: Hide gst_jpeg_decoder_get_type() symbol

+     It's not a public symbol yet

+     Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4614>

+

+2023-03-05 20:01:44 +0900  Seungha Yang <seungha@centricular.com>

+

+   * sys/va/gstvabaseenc.h:

+     vabaseenc: Fix return type of encode_frame vfunc

+     Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4614>

+

+2023-03-05 19:53:37 +0900  Seungha Yang <seungha@centricular.com>

+

+   * sys/va/gstvaav1dec.c:

+   * sys/va/gstvafilter.c:

+   * sys/va/gstvah264enc.c:

+   * sys/va/gstvah265enc.c:

+   * sys/va/gstvavp8dec.c:

+   * sys/va/gstvavpp.c:

+     va: Fix struct empty initialization syntax

+     "struct Foo bar; bar = {};" is not a valid syntax. Also remove use

+     of __typeof__ which is GCC specific

+     Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4614>

+

+2023-05-10 00:51:31 +0900  Seungha Yang <seungha@centricular.com>

+

+   * sys/d3d11/gstd3d11videosink.cpp:

+     d3d11videosink: Don't clear prepared buffer on unlock_stop()

+     That can be called between prepare() and render() which results in

+     unexpected error flow return

+     Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4596>

+

+2023-05-06 04:14:49 +0900  Seungha Yang <seungha@centricular.com>

+

+   * gst-libs/gst/codecs/gsth264decoder.c:

+     h264decoder: Drop nonexisting picture silently without error

+     If end_picture() was not successful, we do drop corresponding

+     GstVideoCodecFrame and therefore gst_video_decoder_get_frame()

+     will return nullptr which is expected behavior.

+     Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4586>

+

+2023-03-12 14:55:22 +0000  Philippe Normand <philn@igalia.com>

+

+   * ext/webrtc/gstwebrtcbin.c:

+   * ext/webrtc/webrtcdatachannel.c:

+   * ext/webrtc/webrtcdatachannel.h:

+     webrtcdatachannel: Bind to parent webrtcbin using a weak reference

+     The previous approach of using a simple pointer could lead to a use-after-free

+     in case a data-channel was created and its parent webrtcbin was disposed soon

+     after.

+     Fixes #2103

+     Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4580>

+

+2023-05-04 16:30:09 +1000  Matthew Waters <matthew@centricular.com>

+

+   * gst-libs/gst/webrtc/nice/nice.c:

+     webrtc/nice: support consent-freshness RFC7675

+     As is supported by libwebrtc already.  This allows ICE components to

+     transition to failed if consent to send from the peer is revoked or if

+     multiple consent packets are lost.

+     Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4575>

+

+2023-05-05 21:53:44 +0900  Seungha Yang <seungha@centricular.com>

+

+   * gst-libs/gst/d3d11/gstd3d11memory.cpp:

+     d3d11memory: Don't clear wrapped texture memory

+     The external texture may hold already rendered scene and therefore

+     it should not be cleared in alloc method

+     Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4558>

+

+2023-05-04 04:44:31 +0900  Seungha Yang <seungha@centricular.com>

+

+   * sys/mediafoundation/gstmfvideoencoder.cpp:

+     mfvideoenc: Allow only even resolution numbers

+     Some H/W vendors support odd resolution if D3D11 texture is used

+     or via IMF2DBuffer, but not all vendors support it.

+     Also software MFT does not allow odd resolution.

+     Fixes: https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1165

+     Fixes: https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/2537

+     Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4548>

+

+2023-05-02 18:14:20 +0100  Philippe Normand <philn@igalia.com>

+

+   * ext/webrtc/gstwebrtcbin.c:

+     webrtcbin: Fix potential deadlock when closing before any data was sent

+     A blocking pad probe is added on new sink pads, it's usually removed after the

+     caps have been negotiated or the signaling state switched to stable, but if that

+     never happens and the pad is released we kept the pad probe active, leaving the

+     pad blocked, preventing clean disposal.

+     Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4533>

+

+2023-05-01 17:46:25 +0100  Philippe Normand <philn@igalia.com>

+

+   * gst-libs/gst/webrtc/dtlstransport.c:

+     dtlstransport: Keep strong ref of dtls encoder/decoder

+     Otherwise get_property() calls for the client, certificate and/or

+     remote-certificate properties might access moved objects, since the encoder and

+     decoder are added to the transportsendbin.

+     Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4526>

+

+2023-05-01 10:24:54 +0000  Mihail Ivanchev <contact@ivanchev.net>

+

+   * gst/codectimestamper/meson.build:

+     gstcodectimestamper: remove PC file generation from plugin's own meson.build

+     The file generated here is incomplete; it is generated for all plugins in a loop at an upper level.

+     Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4520>

+

+2023-04-28 04:28:24 +0900  Seungha Yang <seungha@centricular.com>

+

+   * sys/wasapi2/gstwasapi2util.c:

+     wasapi2: Allows process loopback capture on Windows 10

+     As per MS documentation1, it requires Windows 10 Build 20348

+     but it seems to be supported by old versions too

+     1 https://learn.microsoft.com/en-us/windows/win32/api/audioclientactivationparams/

+     ns-audioclientactivationparams-audioclient_process_loopback_params

+     Fixes: https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/2524

+     Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4517>

+

+2023-04-28 18:52:48 +0900  Seungha Yang <seungha@centricular.com>

+

+   * gst-libs/gst/cuda/gstcudamemory.c:

+     cudamemory: Fix for semi planar YUV memory size decision

+     UV plan of the semi planar format requires only half of Y plane size

+     Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4506>

+

+2023-04-25 04:59:33 +0900  Seungha Yang <seungha@centricular.com>

+

+   * sys/d3d11/gstd3d11window_win32.cpp:

+     d3d11videosink: Fix for ignored initial render rectangle

+     Application can set target render rect before internal HWND

+     configuration

+     Fixes: https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/2518

+     Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4496>

+

+2023-04-25 10:14:47 +0200  Stéphane Cerveau <scerveau@collabora.com>

+

+   * tests/check/meson.build:

+     bad: disable dtls test if openssl is not present

+     On MacOS with homebrew, the openssl library is not

+     properly detected with pkg-config.

+     So disable the test compilation if openssl

+     is not properly detected along with libcrypto.

+     libcrypto is detected but it uses the system one

+     which leads to the error:

+     your binary is not an allowed client of /usr/lib/libcrypto.dylib for

+     architecture x86_64

+     See more details from Apple:

+     https://developer.apple.com/forums/thread/124782

+     Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4483>

+

+2023-04-19 16:59:01 +0200  Martin Nordholts <martn@axis.com>

+

+   * gst-libs/gst/webrtc/nice/nice.c:

+     webrtc: Plug leaks of resolved ICE addresses

+     The addresses we get from `resolve_host_finish()` (via

+     `resolve_host_async()`, `resolve_host_main_cb()`, `on_resolve_host()`,

+     `g_resolver_lookup_by_name_finish()`) must be freed. Otherwise we leak

+     memory.

+     Leak found and confirmed fixed with GCC AddressSanitizer.

+     Change-Id: If32d24452d626234f01b253b77a7d6d16eac1cee

+     Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4482>

+

+2023-04-17 21:34:19 +0900  Seungha Yang <seungha@centricular.com>

+

+   * sys/d3d11/gstd3d11compositor.cpp:

+     d3d11compositor: Skip zero alpha input

+     Blending such input is a waste of resource since nothing will

+     be rendered

+     Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4456>

+

+2023-03-30 10:11:11 +1100  Matthew Waters <matthew@centricular.com>

+

+   * ext/closedcaption/gstccconverter.c:

+     ccconverter: reintroduce frame count reset on cycle completion

+     Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4453>

+

+2023-04-18 14:55:23 +0300  Sebastian Dröge <sebastian@centricular.com>

+

+   * gst/mpegtsdemux/tsdemux.c:

+     tsdemux: Set number of channels to 2 for dual mono Opus

+     Instead of leaving it at 0, which will then cause caps creation to fail.

+     Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4446>

+

+2023-04-17 11:36:55 +0200  Guillaume Desmottes <guillaume.desmottes@onestream.live>

+

+   * ext/dash/gstmpdclient.c:

+     dash: mpdclient: fix divide by 0 if segment has no duration

+     Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4441>

+

+2023-04-17 23:06:54 +0900  Seungha Yang <seungha@centricular.com>

+

+   * sys/d3d11/gstd3d11compositor.cpp:

+     d3d11compositor: Reconfigure resource only when output caps is changed

+     GstD3D11Converter setup is heavy operation since it requires

+     shader compile, GPU resource allocation, some math, mutex, etc.

+     We can avoid it if negotiated caps is not changed.

+     Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4440>

+

+2018-07-19 17:21:22 +0200  Michael Olbrich <m.olbrich@pengutronix.de>

+

+   * ext/srtp/gstsrtpdec.c:

+     srtpdec: fix "srtp-key" check

+     The original code was:

+     if (!gst_structure_get (s, "srtp-key", GST_TYPE_BUFFER, &buf, NULL) || !buf) {

+     goto error;

+     } else {

+     stream->key = buf;

+     }

+     So use "srtp-key" if it is set so a non NULL buffer. The condition was

+     incorrectly inverted in ad7ffe64a66ab48d81671651031c449149db4973 to:

+     if (gst_structure_get (s, "srtp-key", GST_TYPE_BUFFER, &buf, NULL) || !buf) {

+     stream->key = buf;

+     } ...

+     Fix the condition so it works as originally intended and avoid accessing

+     'buf' uninitialised.

+     Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4423>

+

+2023-03-27 09:20:27 +0200  Aleksandr Slobodeniuk <aslobodeniuk@fluendo.com>

+

+   * sys/d3d11/gstd3d11videosink.cpp:

+   * sys/d3d11/gstd3d11window_win32.cpp:

+     d3d11videosink: fix race conditions in win32 window

+     One race condition is the fact that the window object

+     can be destroyed while running some routine in the UI

+     thread (such as resizing). To avoid that situation we make

+     UI thread hold a reference on the window object while it's

+     running.

+     Other probpematic case is when the window handle is reused:

+     if we stop and start the pipeline very fast,

+     so the sink creates a new window object that is going to use

+     the same window handle as the previous one.

+     And finally the case when the pipeline is stopped immediatelly

+     right after starting, this one is also handled in this commit.

+     NOTE: a unit test that reproduces this cases have been added

+     in the previous commit.

+     Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4417>

+

+2023-03-21 18:17:12 +0100  Aleksandr Slobodeniuk <aslobodeniuk@fluendo.com>

+

+   * tests/check/elements/d3d11videosink.c:

+   * tests/check/meson.build:

+     d3d11videosink: add gstcheck test for win32 window

+     Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4417>

+

+2023-04-02 00:29:24 +0900  Seungha Yang <seungha@centricular.com>

+

+   * sys/d3d11/gstd3d11window_win32.cpp:

+     d3d11videosink: Enhancement for initial window size decision

+     Use AdjustWindowRect() method to calculate window size so that

+     video scene can be rendered on client area without black borders

+     Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4416>

+

+2023-04-12 00:31:17 +0100  Tim-Philipp Müller <tim@centricular.com>

+

+   * meson.build:

+     Back to development

+

 === release 1.22.2 ===

 2023-04-11 17:29:28 +0100  Tim-Philipp Müller <tim@centricular.com>

​

@@ -2,13 +2,13 @@

 GStreamer 1.22.0 was originally released on 23 January 2023.

-The latest bug-fix release in the stable 1.22 series is 1.22.2 and was

-released on 11 April 2023.

+The latest bug-fix release in the stable 1.22 series is 1.22.3 and was

+released on 19 May 2023.

 See https://gstreamer.freedesktop.org/releases/1.22/ for the latest

 version of this document.

-Last updated: Tuesday 11 April 2023, 16:45 UTC (log)

+Last updated: Friday 19 May 2023, 01:00 UTC (log)

 Introduction

@@ -1648,6 +1648,189 @@

 -   List of Merge Requests applied in 1.22.2

 -   List of Issues fixed in 1.22.2

+1.22.3

+

+The third 1.22 bug-fix release (1.22.3) was released on 19 May 2023.

+

+This release only contains bugfixes and it should be safe to update from

+1.22.x.

+

+Highlighted bugfixes in 1.22.3

+

+-   avdec: fix occasional video decoder deadlock on seeking with FFmpeg

+    6.0

+-   decodebin3: fix regression handling input streams without CAPS or

+    TIME segment such as e.g. udpsrc or `pushfilesrc

+-   bluez: a2dpsink: fix Bluetooth SIG Certification test failures

+-   osxvideosink: fix deadlock upon closing output window

+-   qtdemux: fix edit list handling regression and AV1 codec box parsing

+-   qtmux: fix extraction of CEA608 closed caption data from S334-1A

+    packets

+-   rtspsrc: Fix handling of * control path

+-   splitmux: timestamp handling improvements

+-   v4l2videodec: Rework dynamic resolution change handling (needed for

+    IMX6 mainline codec)

+-   videoflip: fix regression with automatically rotating video based on

+    tags

+-   d3d11: many d3d11videosink and d3d11compositor fixes

+-   webrtc, rtp: numerous data race fixes and stability fixes

+-   various bug fixes, memory leak fixes, and other stability and

+    reliability improvements

+

+gstreamer

+

+-   tracing: Initialize tracing infrastructure even if the debug system

+    is not compiled in

+-   parse-launch: fix missing unref of looked-up child element

+-   gstutils: Add category and object to most logging messages

+

+gst-plugins-base

+

+-   allocators: Fix fdmem unit test with recent GLib versions

+-   audiotestsrc: Initialize all samples in wave=ticks mode

+-   decodebin3: Handle input streams without CAPS or TIME segment such

+    as e.g. udpsrc or pushfilesrc

+-   decodebin3: fix regression handling streams without caps

+-   decodebin3: fix random hang when remove failing stream

+-   uridecodebin3: Ensure atomic urisourcebin state change

+-   glvideoflip: fix leaked caps

+-   glcontext_wgl: fix missing unref

+-   playsink: Fix volume leak

+

+gst-plugins-good

+

+-   adaptivedemux2: fix critical when using an unsupported URI

+-   dashdemux2: mpdclient: fix divide by 0 if segment has no duration

+-   imagesequencesrc: Properly set default location

+-   multifile: error out if no filename was set

+-   osxvideosink: fix deadlock upon closing output window

+-   rtpmanager: rtpsession: data race leading to critical warnings

+-   rtpmanager: rtpsession: race conditions leading to critical warnings

+-   rtspsrc: Fix handling of * control path

+-   splitmuxsink: Catch invalid DTS to avoid running into problems later

+-   splitmuxsrc: Make PTS contiguous by preference

+-   qtdemux: emit no-more-pads after pruning old pads

+-   Revert “qtdemux: fix conditions for end of segment in reverse

+    playback” to fix edit list regression

+-   qtdemux: Fix av1C parsing

+-   qtmux: Fix extraction of CEA608 data from S334-1A packets

+-   qtwindow: unref caps in destructor

+-   v4l2: device provider: Fix GMainLoop leak

+-   v4l2: videodec: Rework dynamic resolution change handling

+-   v4l2: videodec: Prefer acquired caps over anything downstream

+-   videoflip: fix setting of method property at construction time

+-   videoflip 1.22.2 not rotating video when extracting frames

+

+gst-plugins-bad

+

+-   a2dpsink: Fails many tests in Bluetooth SIG Certification

+-   avdtputil: Use int instead of int range for fixed bitpool values

+-   ccconverter: reintroduce frame count reset on cycle completion

+-   ccconverter: integer overflow & crashing

+-   codectimestamper: remove PC file generation from plugin’s own

+    meson.build

+-   cudamemory: Fix for semi planar YUV memory size decision

+-   d3d11compositor: Reconfigure resource only when output caps is

+    changed

+-   d3d11compositor: Skip zero alpha input

+-   d3d11convert: Fix for runtime property update

+-   d3d11memory: Don’t clear wrapped texture memory

+-   d3d11videosink: Fix for ignored initial render rectangle

+-   d3d11videosink: fix race condition in window unprepare

+-   d3d11videosink: Enhancement for initial window size decision

+-   d3d11videosink: Don’t clear prepared buffer on unlock_stop()

+-   dashdemux: mpdclient: fix divide by 0 if segment has no duration

+-   dtlstransport: Keep strong ref of dtls encoder/decoder

+-   GstPlay: avoid getting property of playbin2 if subtitle_sid is null

+-   GstPlay: fix critical log when using playbin3

+-   h264decoder: Drop nonexisting picture silently without error

+-   dtmf: element classification improvements

+-   mfvideoenc: Allow only even resolution numbers

+-   sctpenc: Fix potential shutdown deadlock

+-   srtpdec: fix “srtp-key” check

+-   tests: disable dtls test if openssl is not present

+-   tsdemux: Set number of channels to 2 for dual mono Opus

+-   va: Various fixes for defects found with MSVC

+-   wasapi2: Allows process loopback capture on Windows 10

+-   webrtcdatachannel: Bind to parent webrtcbin using a weak reference

+-   webrtcbin: Fix potential deadlock when closing before any data was

+    sent

+-   webrtc: Plug leaks of resolved ICE addresses

+-   webrtc: do not tear down data channel before data is flushed

+

+gst-plugins-ugly

+

+-   mpegpsdemux: Rework gap sending

+

+gst-libav

+

+-   avviddec: Temporarily unlock stream lock while flushing buffers

+-   Random freeze and deadlock in ffmpegviddec flush and get_buffer

+    while seeking

+

+gst-rtsp-server

+

+-   No changes

+

+gstreamer-vaapi

+

+-   No changes

+

+gstreamer-sharp

+

+-   No changes

+

+gst-omx

+

+-   No changes

+

+gst-python

+

+-   No changes

+

+gst-editing-services

+

+-   ges: base-xml-formatter: Don’t pass non-GObject pointers to

+    GST_DEBUG_OBJECT

+

+gst-validate + gst-integration-testsuites

+

+-   No changes

+

+gst-examples

+

+-   No changes

+

+Development build environment

+

+-   No changes

+

+Cerbero build tool and packaging changes in 1.22.3

+

+-   glib: Ship Windows process spawning helpers

+-   recipes: add recipe for libltc for timecodestamper element

+-   Add support for RHEL9 and Rocky Linux

+

+Contributors to 1.22.3

+

+Aleksandr Slobodeniuk, Antonio Kevo, Arun Raghavan, Carlos Rafael Giani,

+Daniel Moberg, Edward Hervey, Elliot Chen, François Laignel, Guillaume

+Desmottes, Haihua Hu, Jan Alexander Steffens (heftig), Jan Beich, Jan

+Schmidt, Johan Sternerup, John King, Jordan Petridis, Juan Navarro, Lily

+Foster, Martin Nordholts, Mathieu Duponchelle, Matthew Waters, Matthias

+Fuchs, Michael Olbrich, Mihail Ivanchev, Nick Steel, Nicolas Dufresne,

+Nirbheek Chauhan, Patricia Muscalu, Philippe Normand, Piotr Brzeziński,

+Sanchayan Maity, Sebastian Dröge, Seungha Yang, Stéphane Cerveau,

+Thibault Saunier, Tim-Philipp Müller, Xabier Rodriguez Calvar,

+

+… 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 1.22.3

+

+-   List of Merge Requests applied in 1.22.3

+-   List of Issues fixed in 1.22.3

+

 Schedule for 1.24

 Our next major feature release will be 1.24, and 1.23 will be the

​

@@ -1,4 +1,4 @@

-GStreamer 1.20.x stable series

+GStreamer 1.22.x stable series

 WHAT IT IS

 ----------

​

@@ -1,4 +1,4 @@

-This is GStreamer gst-plugins-bad 1.22.2.

+This is GStreamer gst-plugins-bad 1.22.3.

 The GStreamer team is thrilled to announce a new major feature release

 of your favourite cross-platform multimedia framework!

​

@@ -591,6 +591,11 @@

   output_time_cmp = gst_util_fraction_compare (input_frame_n, input_frame_d,

       output_frame_n, output_frame_d);

+  if (output_time_cmp == 0) {

+    self->output_frames = 0;

+    self->input_frames = 0;

+  }

+

   in_fps_entry = cdp_fps_entry_from_fps (self->in_fps_n, self->in_fps_d);

   if (!in_fps_entry || in_fps_entry->fps_n == 0)

     g_assert_not_reached ();

@@ -1455,6 +1460,11 @@

   output_time_cmp = gst_util_fraction_compare (input_frame_n, input_frame_d,

       output_frame_n, output_frame_d);

+  if (output_time_cmp == 0) {

+    self->output_frames = 0;

+    self->input_frames = 0;

+  }

+

   /* if the next output frame is at or before the current input frame */

   if (output_time_cmp >= 0)

     return TRUE;

​

@@ -1768,7 +1768,7 @@

     g_return_val_if_fail (GST_MPD_MULT_SEGMENT_BASE_NODE

         (stream->cur_seg_template)->SegmentTimeline == NULL, FALSE);

-    if (!GST_CLOCK_TIME_IS_VALID (duration)) {

+    if (!GST_CLOCK_TIME_IS_VALID (duration) || duration == 0) {

       return FALSE;

     }

​

@@ -331,6 +331,34 @@

   }

 }

+static void

+flush_sinkpad (const GValue * item, gpointer user_data)

+{

+  GstSctpEncPad *sctpenc_pad = g_value_get_object (item);

+  gboolean flush = GPOINTER_TO_INT (user_data);

+

+  if (flush) {

+    g_mutex_lock (&sctpenc_pad->lock);

+    sctpenc_pad->flushing = TRUE;

+    g_cond_signal (&sctpenc_pad->cond);

+    g_mutex_unlock (&sctpenc_pad->lock);

+  } else {

+    sctpenc_pad->flushing = FALSE;

+  }

+}

+

+static void

+flush_sinkpads (GstSctpEnc * self, gboolean state)

+{

+  GstIterator *it;

+

+  it = gst_element_iterate_sink_pads (GST_ELEMENT (self));

+  while (gst_iterator_foreach (it, flush_sinkpad,

+          GINT_TO_POINTER (state)) == GST_ITERATOR_RESYNC)

+    gst_iterator_resync (it);

+  gst_iterator_free (it);

+}

+

 static GstStateChangeReturn

 gst_sctp_enc_change_state (GstElement * element, GstStateChange transition)

 {

@@ -351,6 +379,7 @@

       break;

     case GST_STATE_CHANGE_PAUSED_TO_READY:

       stop_srcpad_task (self->src_pad, self);

+      flush_sinkpads (self, TRUE);

       self->src_ret = GST_FLOW_FLUSHING;

       break;

     case GST_STATE_CHANGE_READY_TO_NULL:

@@ -764,22 +793,6 @@

   return ret;

 }

-static void

-flush_sinkpad (const GValue * item, gpointer user_data)

-{

-  GstSctpEncPad *sctpenc_pad = g_value_get_object (item);

-  gboolean flush = GPOINTER_TO_INT (user_data);

-

-  if (flush) {

-    g_mutex_lock (&sctpenc_pad->lock);

-    sctpenc_pad->flushing = TRUE;

-    g_cond_signal (&sctpenc_pad->cond);

-    g_mutex_unlock (&sctpenc_pad->lock);

-  } else {

-    sctpenc_pad->flushing = FALSE;

-  }

-}

-

 static gboolean

 gst_sctp_enc_src_event (GstPad * pad, GstObject * parent, GstEvent * event)

 {

@@ -788,29 +801,17 @@

   switch (GST_EVENT_TYPE (event)) {

     case GST_EVENT_FLUSH_START:{

-      GstIterator *it;

-

       gst_data_queue_set_flushing (self->outbound_sctp_packet_queue, TRUE);

       gst_data_queue_flush (self->outbound_sctp_packet_queue);

-      it = gst_element_iterate_sink_pads (GST_ELEMENT (self));

-      while (gst_iterator_foreach (it, flush_sinkpad,

-              GINT_TO_POINTER (TRUE)) == GST_ITERATOR_RESYNC)

-        gst_iterator_resync (it);

-      gst_iterator_free (it);

+      flush_sinkpads (self, TRUE);

       ret = gst_pad_event_default (pad, parent, event);

       break;

     }

     case GST_EVENT_RECONFIGURE:

     case GST_EVENT_FLUSH_STOP:{

-      GstIterator *it;

-

-      it = gst_element_iterate_sink_pads (GST_ELEMENT (self));

-      while (gst_iterator_foreach (it, flush_sinkpad,

-              GINT_TO_POINTER (FALSE)) == GST_ITERATOR_RESYNC)

-        gst_iterator_resync (it);

-      gst_iterator_free (it);

+      flush_sinkpads (self, FALSE);

       gst_data_queue_set_flushing (self->outbound_sctp_packet_queue, FALSE);

       self->need_segment = TRUE;

@@ -977,7 +978,6 @@

   g_signal_handler_disconnect (self->sctp_association,

       self->signal_handler_state_changed);

-  stop_srcpad_task (self->src_pad, self);

   gst_sctp_association_force_close (self->sctp_association);

   g_object_unref (self->sctp_association);

   self->sctp_association = NULL;

​

@@ -606,7 +606,7 @@

     goto error;

   }

-  if (gst_structure_get (s, "srtp-key", GST_TYPE_BUFFER, &buf, NULL) || !buf) {

+  if (gst_structure_get (s, "srtp-key", GST_TYPE_BUFFER, &buf, NULL) && buf) {

 #ifdef HAVE_SRTP2

     GstBuffer *mki = NULL;

     guint i;

​

@@ -2543,7 +2543,7 @@

   if (!channel) {

     channel = g_object_new (WEBRTC_TYPE_DATA_CHANNEL, NULL);

     channel->parent.id = stream_id;

-    channel->webrtcbin = webrtc;

+    webrtc_data_channel_set_webrtcbin (channel, webrtc);

     g_signal_emit (webrtc, gst_webrtc_bin_signalsPREPARE_DATA_CHANNEL_SIGNAL,

         0, channel, FALSE);

@@ -7166,7 +7166,7 @@

   gst_element_sync_state_with_parent (ret->sink_bin);

   ret = gst_object_ref (ret);

-  ret->webrtcbin = webrtc;

+  webrtc_data_channel_set_webrtcbin (ret, webrtc);

   g_ptr_array_add (webrtc->priv->data_channels, ret);

   DC_UNLOCK (webrtc);

@@ -8184,6 +8184,11 @@

   gst_caps_replace (&webrtc_pad->received_caps, NULL);

   PC_UNLOCK (webrtc);

+  if (webrtc_pad->block_id) {

+    gst_pad_remove_probe (GST_PAD (pad), webrtc_pad->block_id);

+    webrtc_pad->block_id = 0;

+  }

+

   _remove_pad (webrtc, webrtc_pad);

   PC_LOCK (webrtc);

​

@@ -51,6 +51,7 @@

 struct task

 {

+  GstWebRTCBin *webrtcbin;

   GstWebRTCDataChannel *channel;

   ChannelTask func;

   gpointer user_data;

@@ -69,6 +70,7 @@

 static void

 _free_task (struct task *task)

 {

+  g_object_unref (task->webrtcbin);

   gst_object_unref (task->channel);

   if (task->notify)

@@ -80,14 +82,22 @@

 _channel_enqueue_task (WebRTCDataChannel * channel, ChannelTask func,

     gpointer user_data, GDestroyNotify notify)

 {

-  struct task *task = g_new0 (struct task, 1);

+  GstWebRTCBin *webrtcbin = NULL;

+  struct task *task = NULL;

+  webrtcbin = g_weak_ref_get (&channel->webrtcbin_weak);

+  if (!webrtcbin)

+    return;

+

+  task = g_new0 (struct task, 1);

+

+  task->webrtcbin = webrtcbin;

   task->channel = gst_object_ref (channel);

   task->func = func;

   task->user_data = user_data;

   task->notify = notify;

-  gst_webrtc_bin_enqueue_task (channel->webrtcbin,

+  gst_webrtc_bin_enqueue_task (task->webrtcbin,

       (GstWebRTCBinFunc) _execute_task, task, (GDestroyNotify) _free_task,

       NULL);

 }

@@ -427,11 +437,15 @@

     GST_WEBRTC_DATA_CHANNEL_UNLOCK (channel);

     g_object_notify (G_OBJECT (channel), "ready-state");

-    GST_WEBRTC_DATA_CHANNEL_LOCK (channel);

-    if (channel->parent.buffered_amount <= 0) {

-      _channel_enqueue_task (channel, (ChannelTask) _close_sctp_stream,

-          NULL, NULL);

+    /* Make sure that all data enqueued gets properly sent before data channel is closed. */

+    GstFlowReturn ret =

+        gst_app_src_end_of_stream (GST_APP_SRC (WEBRTC_DATA_CHANNEL

+            (channel)->appsrc));

+    if (ret != GST_FLOW_OK) {

+      GST_WARNING_OBJECT (channel, "Send end of stream returned %i, %s", ret,

+          gst_flow_get_name (ret));

     }

+    return;

   }

   GST_WEBRTC_DATA_CHANNEL_UNLOCK (channel);

@@ -1035,6 +1049,16 @@

   } else if (GST_PAD_PROBE_INFO_TYPE (info) & GST_PAD_PROBE_TYPE_BUFFER_LIST) {

     GstBufferList *list = GST_PAD_PROBE_INFO_BUFFER_LIST (info);

     size = gst_buffer_list_calculate_size (list);

+  } else if (GST_PAD_PROBE_INFO_TYPE (info) &

+      GST_PAD_PROBE_TYPE_EVENT_DOWNSTREAM) {

+    GstEvent *event = GST_PAD_PROBE_INFO_EVENT (info);

+    if (GST_EVENT_TYPE (event) == GST_EVENT_EOS

+        && channel->parent.ready_state ==

+        GST_WEBRTC_DATA_CHANNEL_STATE_CLOSING) {

+      _channel_enqueue_task (channel, (ChannelTask) _close_sctp_stream, NULL,

+          NULL);

+      return GST_PAD_PROBE_DROP;

+    }

   }

   if (size > 0) {

@@ -1053,11 +1077,6 @@

           NULL);

     }

-    if (channel->parent.ready_state == GST_WEBRTC_DATA_CHANNEL_STATE_CLOSING

-        && channel->parent.buffered_amount <= 0) {

-      _channel_enqueue_task (channel, (ChannelTask) _close_sctp_stream, NULL,

-          NULL);

-    }

     GST_WEBRTC_DATA_CHANNEL_UNLOCK (channel);

     g_object_notify (G_OBJECT (&channel->parent), "buffered-amount");

   }

@@ -1130,6 +1149,8 @@

   g_clear_object (&channel->appsrc);

   g_clear_object (&channel->appsink);

+  g_weak_ref_clear (&channel->webrtcbin_weak);

+

   G_OBJECT_CLASS (parent_class)->finalize (object);

 }

@@ -1155,6 +1176,8 @@

   G_LOCK (outstanding_channels_lock);

   outstanding_channels = g_list_prepend (outstanding_channels, channel);

   G_UNLOCK (outstanding_channels_lock);

+

+  g_weak_ref_init (&channel->webrtcbin_weak, NULL);

 }

 static void

@@ -1204,3 +1227,10 @@

     }

   }

 }

+

+void

+webrtc_data_channel_set_webrtcbin (WebRTCDataChannel * channel,

+    GstWebRTCBin * webrtcbin)

+{

+  g_weak_ref_set (&channel->webrtcbin_weak, webrtcbin);

+}

​

@@ -51,7 +51,7 @@

   GstElement                       *sink_bin;

   GstElement                       *appsink;

-  GstWebRTCBin                     *webrtcbin;

+  GWeakRef                          webrtcbin_weak;

   gboolean                          opened;

   gulong                            src_probe;

   GError                           *stored_error;

@@ -72,6 +72,10 @@

 void    webrtc_data_channel_link_to_sctp (WebRTCDataChannel                 *channel,

                                           WebRTCSCTPTransport               *sctp_transport);

+G_GNUC_INTERNAL

+void    webrtc_data_channel_set_webrtcbin (WebRTCDataChannel                *channel,

+                                           GstWebRTCBin                     *webrtcbin);

+

 G_DECLARE_FINAL_TYPE (WebRTCErrorIgnoreBin, webrtc_error_ignore_bin, WEBRTC, ERROR_IGNORE_BIN, GstBin);

 G_END_DECLS

​

@@ -1792,10 +1792,16 @@

       picture->system_frame_number);

   if (!frame) {

-    GST_ERROR_OBJECT (self,

-        "No available codec frame with frame number %d",

-        picture->system_frame_number);

-    UPDATE_FLOW_RETURN (ret, GST_FLOW_ERROR);

+    /* The case where the end_picture() got failed and corresponding

+     * GstVideoCodecFrame was dropped already */

+    if (picture->nonexisting) {

+      GST_DEBUG_OBJECT (self, "Dropping non-existing picture %p", picture);

+    } else {

+      GST_ERROR_OBJECT (self,

+          "No available codec frame with frame number %d",

+          picture->system_frame_number);

+      UPDATE_FLOW_RETURN (ret, GST_FLOW_ERROR);

+    }

     gst_h264_picture_unref (picture);

​

@@ -518,11 +518,13 @@

     case GST_VIDEO_FORMAT_I420:

     case GST_VIDEO_FORMAT_YV12:

     case GST_VIDEO_FORMAT_I420_10LE:

+      alloc_height *= 2;

+      break;

     case GST_VIDEO_FORMAT_NV12:

     case GST_VIDEO_FORMAT_NV21:

     case GST_VIDEO_FORMAT_P010_10LE:

     case GST_VIDEO_FORMAT_P016_LE:

-      alloc_height *= 2;

+      alloc_height += alloc_height / 2;

       break;

     case GST_VIDEO_FORMAT_Y42B:

     case GST_VIDEO_FORMAT_I422_10LE:

​

@@ -1473,6 +1473,7 @@

   GstD3D11Memory *dmem;

   ID3D11RenderTargetView *rtv = nullptr;

   GstD3D11ClearRTVFunc clear_func = nullptr;

+  gboolean is_new_texture = TRUE;

   device_handle = gst_d3d11_device_get_device_handle (device);

@@ -1482,6 +1483,8 @@

       GST_ERROR_OBJECT (self, "Couldn't create texture");

       return nullptr;

     }

+  } else {

+    is_new_texture = FALSE;

   }

   mem =

@@ -1489,6 +1492,10 @@

   if (!mem)

     return nullptr;

+  /* Don't clear external texture */

+  if (!is_new_texture)

+    return mem;

+

   /* Clear with YUV black if needed and possible

    * TODO: do this using UAV if RTV is not allowed (e.g., packed YUV formats) */

   if ((desc->BindFlags & D3D11_BIND_RENDER_TARGET) == 0)

​

@@ -1762,9 +1762,11 @@

       g_object_get (G_OBJECT (self->playbin), "current-suburi", &suburi, NULL);

       if (suburi) {

-        if (self->use_playbin3 && self->subtitle_sid) {

-          if (g_str_equal (self->subtitle_sid, stream_info->stream_id))

+        if (self->use_playbin3) {

+          if (self->subtitle_sid &&

+              g_str_equal (self->subtitle_sid, stream_info->stream_id)) {

             info->language = g_path_get_basename (suburi);

+          }

         } else {

           g_object_get (G_OBJECT (self->playbin), "current-text", &text_index,

               NULL);

@@ -2387,7 +2389,7 @@

     gst_query_parse_seeking (query, NULL, &media_info->seekable, NULL, NULL);

   gst_query_unref (query);

-  if (self->use_playbin3 && self->collection) {

+  if (self->use_playbin3) {

     gst_play_streams_info_create_from_collection (self, media_info,

         self->collection);

   } else {

​

@@ -112,23 +112,35 @@

   switch (prop_id) {

     case PROP_SESSION_ID:

+      GST_OBJECT_LOCK (webrtc);

       g_value_set_uint (value, webrtc->session_id);

+      GST_OBJECT_UNLOCK (webrtc);

       break;

     case PROP_TRANSPORT:

+      GST_OBJECT_LOCK (webrtc);

       g_value_set_object (value, webrtc->transport);

+      GST_OBJECT_UNLOCK (webrtc);

       break;

     case PROP_STATE:

+      GST_OBJECT_LOCK (webrtc);

       g_value_set_enum (value, webrtc->state);

+      GST_OBJECT_UNLOCK (webrtc);

       break;

     case PROP_CLIENT:

+      GST_OBJECT_LOCK (webrtc);

       g_object_get_property (G_OBJECT (webrtc->dtlssrtpenc), "is-client",

           value);

+      GST_OBJECT_UNLOCK (webrtc);

       break;

     case PROP_CERTIFICATE:

+      GST_OBJECT_LOCK (webrtc);

       g_object_get_property (G_OBJECT (webrtc->dtlssrtpdec), "pem", value);

+      GST_OBJECT_UNLOCK (webrtc);

       break;

     case PROP_REMOTE_CERTIFICATE:

+      GST_OBJECT_LOCK (webrtc);

       g_object_get_property (G_OBJECT (webrtc->dtlssrtpdec), "peer-pem", value);

+      GST_OBJECT_UNLOCK (webrtc);

       break;

     default:

       G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);

@@ -182,9 +194,13 @@

 static void

 gst_webrtc_dtls_transport_constructed (GObject * object)

 {

-  GstWebRTCDTLSTransport *webrtc = GST_WEBRTC_DTLS_TRANSPORT (object);

+  GstWebRTCDTLSTransport *webrtc = NULL;

   gchar *connection_id;

+  G_OBJECT_CLASS (parent_class)->constructed (object);

+

+  webrtc = GST_WEBRTC_DTLS_TRANSPORT (object);

+

   /* XXX: this may collide with another connection_id however this is only a

    * problem if multiple dtls element sets are being used within the same

    * process */

@@ -192,17 +208,28 @@

       g_random_int ());

   webrtc->dtlssrtpenc = gst_element_factory_make ("dtlssrtpenc", NULL);

+  gst_object_ref_sink (webrtc->dtlssrtpenc);

   g_object_set (webrtc->dtlssrtpenc, "connection-id", connection_id,

       "is-client", webrtc->client, "rtp-sync", FALSE, NULL);

   webrtc->dtlssrtpdec = gst_element_factory_make ("dtlssrtpdec", NULL);

+  gst_object_ref_sink (webrtc->dtlssrtpdec);

   g_object_set (webrtc->dtlssrtpdec, "connection-id", connection_id, NULL);

   g_free (connection_id);

   g_signal_connect (webrtc->dtlssrtpenc, "notify::connection-state",

       G_CALLBACK (on_connection_state_changed), webrtc);

+}

-  G_OBJECT_CLASS (parent_class)->constructed (object);

+static void

+gst_webrtc_dtls_transport_dispose (GObject * object)

+{

+  GstWebRTCDTLSTransport *webrtc = GST_WEBRTC_DTLS_TRANSPORT (object);

+

+  gst_clear_object (&webrtc->dtlssrtpdec);

+  gst_clear_object (&webrtc->dtlssrtpenc);

+

+  G_OBJECT_CLASS (parent_class)->dispose (object);

 }

 static void

@@ -211,6 +238,7 @@

   GObjectClass *gobject_class = (GObjectClass *) klass;

   gobject_class->constructed = gst_webrtc_dtls_transport_constructed;

+  gobject_class->dispose = gst_webrtc_dtls_transport_dispose;

   gobject_class->get_property = gst_webrtc_dtls_transport_get_property;

   gobject_class->set_property = gst_webrtc_dtls_transport_set_property;

   gobject_class->finalize = gst_webrtc_dtls_transport_finalize;

​

@@ -750,6 +750,8 @@

   }

   new_addr = g_inet_address_to_string (addresses->data);

+  g_resolver_free_addresses (addresses);

+  addresses = NULL;

   new_candv0 = rc->prefix;

   new_candv1 = new_addr;

@@ -1317,6 +1319,8 @@

   /* XXX: only the first IP is used */

   ip = g_inet_address_to_string (addresses->data);

+  g_resolver_free_addresses (addresses);

+  addresses = NULL;

   if (!ip) {

     GST_ERROR_OBJECT (ice, "failed to resolve host for proxy");

​

@@ -35,6 +35,16 @@

  <release>

   <Version>

+   <revision>1.22.3</revision>

+   <branch>1.22</branch>

+   <name></name>

+   <created>2023-05-19</created>

+   <file-release rdf:resource="https://gstreamer.freedesktop.org/src/gst-plugins-bad/gst-plugins-bad-1.22.3.tar.xz" />

+  </Version>

+ </release>

+

+ <release>

+  <Version>

    <revision>1.22.2</revision>

    <branch>1.22</branch>

    <name></name>

​

@@ -13,5 +13,4 @@

   install : true,

   install_dir : plugins_install_dir,

 )

-pkgconfig.generate(gstcodectimestamper, install_dir : plugins_pkgconfig_install_dir)

 plugins += gstcodectimestamper

​