Release 1.19.11.19.1

7 files changed, 4122 insertions, 1997 deletions

diff --git a/ChangeLog b/ChangeLog
index 252d6223c..8f54dc5c4 100644
--- a/ChangeLog
+++ b/ChangeLog
@@ -1,11 +1,4037 @@
+=== release 1.19.1 ===
+
+2021-06-01 00:11:44 +0100  Tim-Philipp Müller <tim@centricular.com>
+
+	* ChangeLog:
+	* NEWS:
+	* README:
+	* RELEASE:
+	* gst-plugins-good.doap:
+	* meson.build:
+	  Release 1.19.1
+
+2021-05-29 12:54:22 +0100  Tim-Philipp Müller <tim@centricular.com>
+
+	* gst/rtp/gstrtpjpegpay.c:
+	  rtpjpegpay: fix image corruption when compiled with MSVC on Windows
+	  On Windows with MSVC, jpeg_header_size would end up 2 bytes larger
+	  than it should be. This then leads to the first 2 bytes of the
+	  actual jpeg image data to be dropped, because we think those
+	  belong to the header, which results in an undecodable image when
+	  reconstructed in the depayloader.
+	  What happens is that when the compiler evaluates
+	  jpeg_header_size = mem.offset + read_u16_and_inc_offset_by_2(&mem);
+	  it actually uses the mem.offset value after it has been increased
+	  by the function call on the right hand size of the equation.
+	  From section 6.5 of the C99 spec:
+	  3. The grouping of operators and operands is indicated by the syntax [74].
+	  Except as specified later (for the function-call (), &&, ||, ?:, and
+	  comma operators), the order of evaluation of subexpressions and the
+	  order in which side effects take place are both unspecified.
+	  Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/889
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/999>
+
+2021-05-25 16:19:20 +0800  Hou Qi <qi.hou@nxp.com>
+
+	* sys/v4l2/gstv4l2videoenc.c:
+	  v4l2videoenc: Set default latency if the frame duration is invalid
+	  If the duration of the v4l2object is invalid, use default 25fps instead.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/996>
+
+2021-05-26 00:23:56 +0900  Seungha Yang <seungha@centricular.com>
+
+	* gst/deinterlace/gstdeinterlace.c:
+	  deinterlace: Drop "field-order" field while transforming caps
+	  Like other basetransform subclasses are doing, drop field
+	  which can be converted by deinterlace.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/997>
+
+2021-05-25 20:10:34 +0900  Seungha Yang <seungha@centricular.com>
+
+	* gst/deinterlace/gstdeinterlace.c:
+	  deinterlace: Drop field-order field if outputting progressive
+	  Progressive with field-order doesn't make sense
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/997>
+
+2021-05-21 14:19:29 +0200  Havard Graff <havard@pexip.com>
+
+	* gst/rtpmanager/gstrtpssrcdemux.c:
+	* tests/check/elements/rtpssrcdemux.c:
+	  rtpssrcdemux: fix "data flow before segment event" crash
+	  This crash could happen at any time a RTP and RTCP buffer arrived
+	  simultaneously in ssrcdemux.
+	  The problem was that sticky-event arriving while the rtp and rtcp pads
+	  were being set up could arrive just too late to be included in the initial
+	  forwarding.
+	  The fix checks if the stickies have been sent on the srcpad about to be
+	  pushed on, and if not sends them. It also blocks any stickes from
+	  being forwarded *prior* to this happening, to avoid them arriving on
+	  the srcpad multiple times.
+	  Since the test loops 1000 times, this will make running under valgrind
+	  take forever, so use the RUNNING_ON_VALGRIND variable to detect we
+	  are running under valgrind, and reduce the loop-count to 2 in that case.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/992>
+
+2021-05-21 18:45:17 +0200  Havard Graff <havard@pexip.com>
+
+	* gst/rtpmanager/gstrtpssrcdemux.c:
+	  rtpssrcdemux: refactor destruction of GstRtpSsrcDemuxPads
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/992>
+
+2021-05-21 18:30:28 +0200  Havard Graff <havard@pexip.com>
+
+	* gst/rtpmanager/gstrtpssrcdemux.c:
+	* gst/rtpmanager/gstrtpssrcdemux.h:
+	  rtpssrcdemux: make naming consistent
+	  Use plural for GstRtpSsrcDemuxPads, since it contains two pads, and
+	  use the variable-name 'dpads' everywhere.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/992>
+
+2021-05-23 15:14:11 +0100  Tim-Philipp Müller <tim@centricular.com>
+
+	* gst/wavparse/gstwavparse.c:
+	  wavparse: use g_strndup() for copying text data
+	  So we don't rely on NUL terminators inside the data.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/994>
+
+2021-05-23 13:29:07 +0100  Tim-Philipp Müller <tim@centricular.com>
+
+	* gst/wavparse/gstwavparse.c:
+	  wavparse: clean up adtl/note/labl chunk parsing
+	  We were passing the size of the adtl chunk to the note/labl
+	  sub-chunk parsing function, which means we may memdup lots of
+	  data after the chunk string's NUL terminator that doesn't
+	  really belong to it.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/994>
+
+2021-05-23 13:24:21 +0100  Tim-Philipp Müller <tim@centricular.com>
+
+	* gst/wavparse/gstwavparse.c:
+	  wavparse: guard against overflow when comparing chunk sizes
+	  Could be rewritten as lsize > (size - 8) a well, but the
+	  extra check seems clearer. Doesn't look like it was problematic,
+	  lsize wasn't actually used when parsing the sub-chunks.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/994>
+
+2021-05-21 13:31:12 -0300  Daniel Almeida <daniel.almeida@collabora.com>
+
+	* docs/gst_plugins_cache.json:
+	  doc: update gst_plugins_cache.json
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/991>
+
+2021-05-05 13:20:04 +0200  Stéphane Cerveau <scerveau@collabora.com>
+
+	* gst/matroska/matroska-demux.c:
+	  matroskademux: fix decoder glitches with H264 content
+	  To avoid decoder starvation causing glitches on screen,
+	  the demuxer shall clip only when the buffer is a key frame
+	  and the lace time is greater than the stop time.
+	  Fixes gst-editing-services#128
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/973>
+
+2021-05-11 20:41:38 +1000  Matthew Waters <matthew@centricular.com>
+
+	* ext/qt/gstqtoverlay.cc:
+	  qml: don't use buffers that have invalid contents
+	  If the GL context is not shareable, ignore it.
+	  A future change may also not output the relevant output either.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/983>
+
+2021-05-11 20:38:52 +1000  Matthew Waters <matthew@centricular.com>
+
+	* ext/qt/gstqsgtexture.cc:
+	  qml: also use the dummy texture when no buffer has been set
+	  Fixes corrupted texture output when changing OpenGL display/contexts.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/983>
+
+2021-05-11 17:20:00 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+	* docs/gst_plugins_cache.json:
+	  doc: Update cache for RGBP format addition
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/968>
+
+2021-04-23 14:37:46 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+	* gst/matroska/matroska-demux.c:
+	* gst/matroska/matroska-ids.c:
+	* gst/matroska/matroska-ids.h:
+	  matroskademux: Advertise codec-alpha in caps
+	  This will be used to select the appropriate decoders. We also only attach the
+	  GstVideoCodecAlphaMeta if the AlphaMode element is set, this is to stay on the
+	  safe side and mimic what browsers (verified in Firefox and Chromium code) do.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/968>
+
+2021-03-22 16:58:26 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+	* gst/matroska/matroska-demux.c:
+	  matroskademux: Store alpha stream in VideoCodecAlphaMeta
+	  This generalize the feature over using mini object quark data. If
+	  that feature was Matroska specifc, using the new CustomMeta would have
+	  been enough and arguably cleaner then QData, though it seems that
+	  similar technique is use with AV1 Image Format (AVIF).
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/968>
+
+2016-12-03 14:27:57 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* gst/matroska/matroska-demux.c:
+	  matroska-demux: extract VP8 alpha from BlockAdditionals
+	  And put it on buffers as qdata (which is easier in this
+	  case than a private custom meta because it can be picked
+	  up easily in other modules).
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/968>
+
+2021-05-03 17:39:05 +1000  Matthew Waters <matthew@centricular.com>
+
+	* ext/qt/gstqtglutility.cc:
+	* ext/qt/gstqtglutility.h:
+	* ext/qt/gstqtoverlay.cc:
+	* ext/qt/qtitem.cc:
+	* ext/qt/qtwindow.cc:
+	  qt: return a different GstGLDisplay object when the first sink requests
+	  This allows the 'replace-gstreamer-opengl-context' context machinery to
+	  correctly replace the OpenGL context used by the pipeline when the first
+	  qmlglsink is added to the pipeline.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/982>
+
+2021-05-07 11:16:47 +0200  Jan Alexander Steffens (heftig) <jan.steffens@ltnglobal.com>
+
+	* gst/udp/gstudpsrc.c:
+	  udpsrc: Plug leaks of saddr in error cases
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/977>
+
+2021-05-07 11:16:21 +0200  Jan Alexander Steffens (heftig) <jan.steffens@ltnglobal.com>
+
+	* gst/udp/gstudpsrc.c:
+	  udpsrc: Whitespace
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/977>
+
+2021-05-07 00:43:44 +0200  Jan Alexander Steffens (heftig) <jan.steffens@ltnglobal.com>
+
+	* gst/deinterlace/gstdeinterlace.c:
+	  deinterlace: Plug a method subobject leak
+	  Changing the method would leak the previous method.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/976>
+
+2021-05-06 15:04:42 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+	* ext/vpx/gstvp9enc.c:
+	  vp9enc: Add color range support
+	  When setting the colorspace, we now clear the range to reduced range,
+	  the default, and then we also set the range so the VP9 encoder encodes
+	  the right information in the bitstream.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/975>
+
+2021-05-06 14:51:31 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+	* ext/vpx/gstvp9enc.c:
+	* ext/vpx/gstvpxenc.c:
+	  vp9enc: Move colorspace configuration in VP9 enc
+	  This is not supported by VP8 and was causing a warning.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/975>
+
+2021-05-06 14:48:36 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+	* ext/vpx/gstvp9enc.c:
+	* ext/vpx/gstvpxenc.c:
+	* ext/vpx/gstvpxenc.h:
+	  vpxdenc: Add a GstVideoCodecState to configure_encoder virtual
+	  This will be needed to configure the VP9 specific colorimetry, which is
+	  currently configured for VP8 casing warning.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/975>
+
+2021-05-05 16:48:10 +0200  Bastien Nocera <hadess@hadess.net>
+
+	* ext/gtk/gtkgstbasewidget.c:
+	  gtk: Remove coordinates double-translation
+	  Remove our own translation in the mouse event capture code, as that
+	  translation will be done through the navigation interface.
+	  Tested by resizing the window created by:
+	  gst-launch-1.0 -v videotestsrc ! navigationtest ! glupload ! glcolorconvert ! tee name=t  ! gtkglsink
+	  and checking that the cursor follows the mouse as expected.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/974>
+
+2021-05-05 14:28:15 +0200  Bastien Nocera <hadess@hadess.net>
+
+	* ext/gtk/gstgtkbasesink.c:
+	  gtk: Translate navigation events coordinates
+	  If the application passed down some pointer coordinates, translate those
+	  from display coordinates to stream coordinates, so things work as
+	  expected even if the video is resized.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/974>
+
+2021-05-05 14:24:31 +0200  Bastien Nocera <hadess@hadess.net>
+
+	* ext/gtk/gtkgstbasewidget.c:
+	* ext/gtk/gtkgstbasewidget.h:
+	  gtk: Export _display_size_to_stream_size()
+	  Export _display_size_to_stream_size() so that GstNavigation implementors
+	  can translate from display coordinates to stream coordinates before
+	  pushing the events upstream to the DVD source, for example.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/974>
+
+2018-02-26 17:26:41 +0100  David Fernandez <d.fernandezlop@gmail.com>
+
+	* docs/gst_plugins_cache.json:
+	* gst/matroska/matroska-mux.c:
+	  matroska-mux: Change accepted caps width and height from [16, MAX] to [1, MAX]
+	  There are cases where the video size might be less than 16x16.
+	  This change allows the Matroska muxer to accept this cases.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/539>
+
+2021-04-20 22:08:23 +0200  François Laignel <fengalin@free.fr>
+
+	* gst/multifile/gstsplitmuxsink.c:
+	* gst/rtpmanager/gstrtpbin.c:
+	* gst/rtsp/gstrtspsrc.c:
+	* tests/check/elements/avimux.c:
+	* tests/check/elements/flvmux.c:
+	* tests/check/elements/interleave.c:
+	* tests/check/elements/qtmux.c:
+	* tests/check/elements/rtpbin.c:
+	* tests/check/elements/rtpcollision.c:
+	* tests/check/elements/rtpmux.c:
+	* tests/check/elements/splitmuxsink.c:
+	* tests/check/elements/videomixer.c:
+	* tests/examples/rtp/client-PCMA.c:
+	* tests/examples/rtp/server-alsasrc-PCMA.c:
+	  Use gst_element_request_pad_simple
+	  Instead of the deprecated gst_element_get_request_pad.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/958>
+
+2021-04-30 08:12:47 +1000  Jan Schmidt <jan@centricular.com>
+
+	* gst/isomp4/atoms.c:
+	* gst/isomp4/atoms.h:
+	  qtmux: Make sure to write 64-bit STCO table when needed.
+	  qtmux attempts to choose between writing a 32-bit stco chunk offset table
+	  when it can, but switch to a 64-bit co64 table when file offsets go over
+	  4GB.
+	  This patch fixes a problem where the atom handling code was checking
+	  mdat-relative offsets instead of the final file offset (computed by
+	  adding the mdat position plus the mdat-relative offset) - leading to
+	  problems where files with a size between 4GB and 4GB+offset-of-the-mdat
+	  would write incorrect STCO tables with some samples having truncated
+	  32-bit offsets.
+	  Smaller files write STCO correctly, larger files would switch to
+	  co64 and also output correctly.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/970>
+
+2021-04-22 15:01:32 +0800  Hou Qi <qi.hou@nxp.com>
+
+	* sys/v4l2/gstv4l2object.c:
+	  v4l2object: Add interlace-mode back to caps for camera
+	  skip_try_fmt_probes is set to TRUE for v4l2src to skip interlace-mode and
+	  colorimetry when probe caps. gst_v4l2_object_set_format_full() will add
+	  colorimetry back to caps when iterating over the negotiated caps. There is
+	  one case that v4l2src is first in preview state then starts recording.
+	  v4l2src caps will change with an additional interlace-mode structure after
+	  renegotiation, then v4l2src needs to reset. But this camera driver can't
+	  orphan buffer pool, it causes require buffer failed as streaming is still
+	  in active state.
+	  To fix this, also need to add interlace-mode back to caps for camera to
+	  avoid reset.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/961>
+
+2021-04-02 18:41:28 +0200  Guillaume Desmottes <guillaume.desmottes@collabora.com>
+
+	* gst/rtp/gstrtpopuspay.c:
+	* gst/rtp/gstrtpopuspay.h:
+	* tests/check/elements/rtp-payloading.c:
+	  rtpopuspay: set MARKER flag
+	  Set MARKER flag on first buffer after DTX.
+	  According to RFC 3551 section 4.1 the marker bit needs to be set on
+	  "the first packet after a silence period during which packets have
+	  not been transmitted contiguously".
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/967>
+
+2021-03-31 11:18:30 +0200  Guillaume Desmottes <guillaume.desmottes@collabora.com>
+
+	* docs/gst_plugins_cache.json:
+	* gst/rtp/gstrtpopuspay.c:
+	* gst/rtp/gstrtpopuspay.h:
+	* tests/check/elements/rtp-payloading.c:
+	  rtpopuspay: add DTX support
+	  If enabled, the payloader won't transmit empty frames.
+	  Can be tested using:
+	  opusenc dtx=true bitrate-type=vbr ! rtpopuspay dtx=true
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/967>
+
+2021-04-24 11:15:50 -0400  Doug Nazar <nazard@nazar.ca>
+
+	* ext/taglib/gstid3v2mux.cc:
+	  taglib: Update createFrame() to non-deprecated version.
+	  ID3v2::FrameFactory::createFrame() versions not taking a Header have
+	  been deprecated since v1.5 (Jan 2008).
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/965>
+
+2021-04-25 02:16:45 +0200  Havard Graff <havard@pexip.com>
+
+	* gst/rtpmanager/gstrtpjitterbuffer.c:
+	* tests/check/elements/rtpjitterbuffer.c:
+	  rtpjitterbuffer: fix divide-by-zero
+	  The estimated packet-duration can sometimes end up as zero, and dividing
+	  by that is never a good idea...
+	  The test reproduces the scenario, and the fix is easy.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/966>
+
+2020-06-02 19:38:33 +0200  Havard Graff <havard@pexip.com>
+
+	  rtpjitterbuffer: clean up and improve missing packets handling
+	  * Try to make variable and function names more clear.
+	  * Add plenty of comments describing the logic step-by-step.
+	  * Improve the logging around this, making the logs easier to read and
+	  understand when debugging these issues.
+	  * Revise the logic of packets that are actually beyond saving in doing
+	  the following:
+	  1. Do an optimistic estimation of which packets can still arrive.
+	  2. Based on this, find which packets (and duration) are now hopelessly
+	  lost.
+	  3. Issue an immediate lost-event for the hopelessly lost and then add
+	  lost/rtx timers for the ones we still hope to save, meaning that if
+	  they are to arrive, they will not be discarded.
+	  * Revise the use of rtx-delay:
+	  Earlier the rtx-delay would vary, depending on the pts of the latest
+	  packet and the estimated pts of the packet it being issued a RTX for,
+	  but now that we aim to estimate the PTS of the missing packet accurately,
+	  the RTX delay should remain the same for all packets.
+	  Meaning: If the packet have a PTS of X, the delay in asked for a RTX
+	  for this packet is always a constant X + delay, not a variable one.
+	  * Finally ensure that the chaotic "check-for-stall" tests uses timestamps
+	  that starts from 0 to make them easier to debug.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/952>
+
+2021-04-23 12:07:52 +0200  Guillaume Desmottes <guillaume.desmottes@collabora.com>
+
+	* gst/level/gstlevel.c:
+	* gst/level/gstlevel.h:
+	  level: make properties thread-safe
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/962>
+
+2021-04-22 14:11:09 +0200  Guillaume Desmottes <guillaume.desmottes@collabora.com>
+
+	* gst/level/gstlevel.c:
+	  level: disable passthrough when audio-level-meta is enabled
+	  Ensure we receive a writable buffer to add the meta.
+	  Fix #878
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/962>
+
+2021-04-23 08:28:06 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/matroska/matroska-mux.c:
+	  matroskamux: Don't pass a non-GObject pointer to GST_DEBUG_OBJECT and similar
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/964>
+
+2021-04-22 08:57:23 +0200  Edward Hervey <edward@centricular.com>
+
+	* gst/rtpmanager/gstrtpjitterbuffer.c:
+	  rtpjitterbuffer: Avoid generation of invalid timestamps
+	  When updating timestamps and timer timeouts with a new offset, make sure that
+	  the resulting value is valid (and not a negative (signed) value which ends up in
+	  a massive (unsigned) value).
+	  Fixes #571
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/960>
+
+2021-04-21 18:41:08 +0100  Philippe Normand <philn@igalia.com>
+
+	* sys/v4l2/v4l2_calls.c:
+	  v4l2: Fix glib warning emitted when attribute query fails
+	  The v4l2object is not a GstObject. Logging has to go through its dbg_obj
+	  specially meant for this.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/959>
+
+2021-03-25 13:20:38 +0100  VaL Doroshchuk <valentyn.doroshchuk@qt.io>
+
+	* ext/qt/gstqtoverlay.cc:
+	* tests/examples/qt/qmloverlay/overlay.py:
+	  qmloverlay: Use first found GstGLVideoItem as widget property
+	  GstGLVideoItem is required to render input video in the overlay's qml.
+	  And currently qmlgloverlay requires to set this GstGLVideoItem to its widget property.
+	  Instead of fetching GstGLVideoItem from the overlay's root object (root-item prop),
+	  and setting it back as a widget (widget prop),
+	  proposing to use found GstGLVideoItem in the current object hierarchy (passed in qml-scene) by default.
+	  Also useful in Python, which solves the issue when casting gpointer <=> QQuickItem* is required.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/919>
+
+2021-04-19 16:39:03 +0100  Tim-Philipp Müller <tim@centricular.com>
+
+	* sys/v4l2/gstv4l2.c:
+	  v4l2: fix debug category initialisation again
+	  Would spew warnings on the rpi4 when calling into
+	  gst_v4l2_object_get_codec_caps() from the probe_and_register()
+	  function since the v4l2_debug category initialisation would
+	  only be done later as part of the element/device provider
+	  registration.
+	  Also log things in the probe function to the v4l2 category
+	  instead of the default category while we're at it.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/956>
+
+2021-04-19 01:29:33 -0400  Doug Nazar <nazard@nazar.ca>
+
+	* gst/rtsp/gstrtspsrc.c:
+	  rtspsrc: Fix race saving seek event seqnum.
+	  We need to save the seek seqnum before the flush stop event
+	  since that will start the basesrc task which may send the segment
+	  event before we're ready.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/954>
+
+2021-03-31 10:52:14 +0200  Marco Felsch <m.felsch@pengutronix.de>
+
+	* ext/qt/qtitem.cc:
+	* ext/qt/qtitem.h:
+	  qmlglsink: allow to set force-aspect-ratio property
+	  Add the forceAspectRatio Q_PROPERTY to allow changing the aspect ratio
+	  from QML code as well.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/937>
+
+2021-04-19 11:14:00 +0100  Tim-Philipp Müller <tim@centricular.com>
+
+	* sys/v4l2/v4l2_calls.c:
+	  v4l2src: fix spurious SOURCE_CHANGED error-level log messages
+	  They're harmless, and some drivers at least return EINVAL
+	  instead of ENOTTY for unsupported events (here: uvcvideo).
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/955>
+
+2021-04-14 16:32:06 -0400  Doug Nazar <nazard@nazar.ca>
+
+	* gst/rtp/gstrtpsbcpay.c:
+	  rtpsbcpay: remove use of packed struct for payload
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/950>
+
+2021-04-14 11:13:45 -0400  Doug Nazar <nazard@nazar.ca>
+
+	* gst/dtmf/gstdtmfcommon.h:
+	* gst/dtmf/gstrtpdtmfdepay.c:
+	* gst/dtmf/gstrtpdtmfsrc.c:
+	  dtmf: convert to bit accessors
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/949>
+
+2021-04-13 09:23:12 +0530  Nirbheek Chauhan <nirbheek@centricular.com>
+
+	* gst/rtsp/gstrtspsrc.c:
+	  rtspsrc: Remove some dead code
+	  stop is not used after this point, nor do we create a new segment
+	  here since 84725d62b57bc74ce34abde755f35bf8f948f94d
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/940>
+
+2021-04-10 02:53:51 +0530  Nirbheek Chauhan <nirbheek@centricular.com>
+
+	* gst/rtsp/gstrtspsrc.c:
+	  rtspsrc: Do not overwrite the known duration after a seek
+	  This breaks the duration query and also the seeking query.
+	  Broke in 5f1a732bc7b76a6f1b8aa5f26b6e76fbca0261c7
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/940>
+
+2021-04-10 04:40:46 +0530  Nirbheek Chauhan <nirbheek@centricular.com>
+
+	* gst/rtsp/gstrtspsrc.c:
+	  rtspsrc: Just assign the segment instead of memcpy
+	  Assignments copy by value, we don't need to memcpy...
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/940>
+
+2021-04-13 11:30:51 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/rtpmanager/gstrtpjitterbuffer.c:
+	  rtpjitterbuffer: Check srcresult before waiting on the condition variable too
+	  It might've been set to FLUSHING between the last check and the waiting,
+	  and in that case we'd be waiting here forever now.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/944>
+
+2021-04-12 23:15:17 -0400  Doug Nazar <nazard@nazar.ca>
+
+	* tests/check/elements/rtpsession.c:
+	  rtp: fix test_twcc_header_and_run to support big endian.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/942>
+
+2021-04-12 23:13:15 -0400  Doug Nazar <nazard@nazar.ca>
+
+	* gst/rtpmanager/rtptwcc.c:
+	  rtp: fix rtptwcc to support big endian.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/942>
+
+2021-04-12 21:59:45 -0400  Doug Nazar <nazard@nazar.ca>
+
+	* gst/rtpmanager/gstrtphdrext-rfc6464.c:
+	  rtp: fix rtphdrextrfc6464 to support big endian.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/942>
+
+2021-04-12 21:36:58 -0400  Doug Nazar <nazard@nazar.ca>
+
+	* tests/check/elements/alpha.c:
+	  tests: Fix alpha test on big endian machines.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/943>
+
+2021-03-19 02:51:20 +1100  Jan Schmidt <jan@centricular.com>
+
+	* gst/isomp4/gstqtmux.c:
+	* gst/isomp4/gstqtmux.h:
+	  qtmux: Protect against writing absurd sample durations
+	  If the input DTS goes backward or is missing, the calculated
+	  sample duration goes negative and wraps around to a very big
+	  number. In that case, just write a sample with a duration of
+	  0 and hope the problem is transient.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/914>
+
+2021-04-10 03:09:44 +0530  Nirbheek Chauhan <nirbheek@centricular.com>
+
+	* gst/rtsp/gstrtspsrc.c:
+	  rtspsrc: De-dup seek event seqnums to avoid multiple seeks
+	  Seek events are sent upstream on each sink, so if we receive multiple
+	  seeks with the same seqnum, we must only perform one seek, not N seeks
+	  where N = the number of sinks in the pipeline connected to rtspsrc.
+	  This is the same thing done by demuxers like qtdemux or matrsokademux.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/938>
+
+2021-04-10 01:55:28 +0530  Nirbheek Chauhan <nirbheek@centricular.com>
+
+	* gst/rtsp/gstrtspsrc.c:
+	  rtspsrc: Using multicast UDP has no relation to seekability
+	  The transport has no relation to whether a media can be seeked. The
+	  range response having a duration is the correct thing to check for.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/939>
+
+2021-04-10 01:54:48 +0530  Nirbheek Chauhan <nirbheek@centricular.com>
+
+	* gst/rtsp/gstrtspsrc.c:
+	  rtspsrc: Add more logging for range parsing and seekable
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/939>
+
+2021-04-10 14:47:23 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* docs/gst_plugins_cache.json:
+	  videocrop: Update documentation cache
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/515>
+
+2021-04-07 21:57:11 +0200  Markus Ebner <info@ebner-markus.de>
+
+	* gst/videocrop/gstvideocrop-private.h:
+	* gst/videocrop/gstvideocrop.c:
+	* gst/videocrop/gstvideocrop.h:
+	  videocrop: Add support for GBR* video formats
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/515>
+
+2021-04-07 18:54:49 +0200  Markus Ebner <info@ebner-markus.de>
+
+	* gst/videocrop/gstvideocrop-private.h:
+	* gst/videocrop/gstvideocrop.c:
+	* gst/videocrop/gstvideocrop.h:
+	  videocrop: Added support for planar pixel formats > 8bits
+	  - Added support for planar pixel formats with depths greater than 8bits
+	  to transform_planar implementation
+	  - Added a whole lot of new pixel formats to the support-list
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/515>
+
+2021-04-07 17:52:34 +0200  Markus Ebner <info@ebner-markus.de>
+
+	* gst/videocrop/gstaspectratiocrop.c:
+	* gst/videocrop/gstvideocrop-private.h:
+	* gst/videocrop/gstvideocrop.c:
+	  videocrop: Move supported format list into private header
+	  - Moved declaration of supported pixel formats to private header, which
+	  can be shared between videocrop and aspectvideocrop
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/515>
+
+2021-04-06 17:02:34 +0530  Nirbheek Chauhan <nirbheek@centricular.com>
+
+	* gst/rtpmanager/rtpjitterbuffer.c:
+	  rtpjitterbuffer: More logging when calculating rfc7273 timestamps
+	  This code can be fragile, since it is very exacting in the timestamps
+	  that it will accept. Add more logging so it's easier to debug issues
+	  and figure out whether it's a bug in the calculation or something
+	  wrong in the incoming buffers.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/934>
+
+2021-04-08 13:29:10 +0200  Stéphane Cerveau <scerveau@collabora.com>
+
+	* gst/rtp/gstrtph264depay.c:
+	* gst/rtp/gstrtph265depay.c:
+	* gst/rtp/gstrtpsv3vdepay.c:
+	  rtp: missing debug init after element splitting
+	  - h264depay
+	  - h265depay
+	  - sv3vdepay
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/936>
+
+2020-03-30 09:29:07 +0200  Michal Dzik <michal.dzik@streamunlimited.com>
+
+	* gst/rtp/gstrtpsbcpay.c:
+	  rtp: rename gst_rtp_sbc_pay_flush_buffers()
+	  gst_rtp_sbc_pay_flush_buffers() is a misleading name. A better name would
+	  be gst_rtp_sbc_pay_drain_buffers(), because that's what it does, it drains
+	  any leftover queued data and pushes it downstream. "Flushing" in GStreamer
+	  typically means to throw away any queued data and not process/push it
+	  downstream.
+	  Signed-off-by: Michal Dzik <michal.dzik@streamunlimited.com>
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/700>
+
+2020-03-24 13:31:00 +0100  Michal Dzik <michal.dzik@streamunlimited.com>
+
+	* gst/rtp/gstrtpsbcpay.c:
+	  rtp: fix adapter flushing in sbc payloader
+	  GstAdapter must be flushed in some cases (flush, new segment, state change)
+	  Without it, it may, for example, push some leftover buffer from old
+	  segment in new segment. This, in general, breaks timestamps.
+	  See GstAdapter documentation for more.
+	  Signed-off-by: Michal Dzik <michal.dzik@streamunlimited.com>
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/700>
+
+2020-08-18 20:16:06 +0200  Jakub Adam <jakub.adam@collabora.com>
+
+	* ext/vpx/gstvpxenc.c:
+	  vpxenc: add colorspace information into VP9 bitstream
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/871>
+
+2021-03-26 16:26:22 +0800  Hou Qi <qi.hou@nxp.com>
+
+	* sys/v4l2/gstv4l2object.c:
+	  v4l2object: Use default colorimetry if that in caps is unknown
+	  Some streams have unknown colorimetry in caps, but v4l2object sets
+	  default values for each primaries. It will cause check colorimetry
+	  fail when do gst_v4l2_video_colorimetry_matches().
+	  To fix this, need to keep the unknown colorimetry in caps same as
+	  the default value set by v4l2object.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/923>
+
+2021-03-31 16:37:56 +0300  Vivia Nikolaidou <vivia@ahiru.eu>
+
+	* gst/matroska/matroska-demux.c:
+	  matroskademux: Take segment stop into account when need_segment
+	  Otherwise, in the case of e.g. a deferred seek event, the segment stop
+	  would be replaced with GST_CLOCK_TIME_NONE.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/929>
+
+2021-03-29 16:45:26 +0200  Val Doroshchuk <val@sevendof.com>
+
+	* ext/qt/gstqtoverlay.cc:
+	* ext/qt/gstqtoverlay.h:
+	  gstqtoverlay: Add initialization and finalization to qml-scene prop
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/924>
+
+2021-03-31 10:21:59 +1100  Matthew Waters <matthew@centricular.com>
+
+	* ext/qt/gstqtglutility.h:
+	  qt: fix build warning with clang and c-linkage of user defined type
+	  In file included from ../subprojects/gst-plugins-good/ext/qt/qtglrenderer.cc:17:
+	  ../subprojects/gst-plugins-good/ext/qt/gstqtglutility.h:35:16: error: 'qt_opengl_native_context_from_gst_gl_context' has C-linkage specified, but returns user-defined type 'QVariant' which is incompatible with C [-Werror,-Wreturn-type-c-linkage]
+	  QVariant       qt_opengl_native_context_from_gst_gl_context     (GstGLContext * context);
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/928>
+
+2021-03-30 09:45:45 +0200  Stéphane Cerveau <scerveau@collabora.com>
+
+	* ext/qt/gstqtelement.cc:
+	* ext/qt/gstqtelements.h:
+	* ext/qt/gstqtoverlay.cc:
+	* ext/qt/gstqtsink.cc:
+	* ext/qt/gstqtsrc.cc:
+	  qt: hotfix: allow per feature registration
+	  Fixes #869
+	  Split plugin into features including
+	  dynamic types which can be indiviually
+	  registered during a static build.
+	  More details here:
+	  https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
+	  https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/925>
+
+2021-02-17 08:52:40 +0100  Stéphane Cerveau <scerveau@collabora.com>
+
+	* ext/twolame/gsttwolamemp2enc.c:
+	* ext/twolame/gsttwolamemp2enc.h:
+	  twolame: allow per feature registration
+	  Split plugin into features including
+	  dynamic types which can be indiviually
+	  registered during a static build.
+	  More details here:
+	  https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
+	  https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>
+
+2021-02-16 17:49:03 +0100  Stéphane Cerveau <scerveau@collabora.com>
+
+	* ext/shout2/gstshout2.c:
+	* ext/shout2/gstshout2.h:
+	  shout2: allow per feature registration
+	  Split plugin into features including
+	  dynamic types which can be indiviually
+	  registered during a static build.
+	  More details here:
+	  https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
+	  https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>
+
+2021-02-16 17:38:46 +0100  Stéphane Cerveau <scerveau@collabora.com>
+
+	* ext/cairo/gstcairo.c:
+	* ext/cairo/gstcairooverlay.c:
+	* ext/cairo/gstcairooverlay.h:
+	  cairo: allow per feature registration
+	  Split plugin into features including
+	  dynamic types which can be indiviually
+	  registered during a static build.
+	  More details here:
+	  https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
+	  https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>
+
+2021-02-16 17:34:34 +0100  Stéphane Cerveau <scerveau@collabora.com>
+
+	* gst/y4m/gsty4mencode.c:
+	* gst/y4m/gsty4mencode.h:
+	  y4m: allow per feature registration
+	  Split plugin into features including
+	  dynamic types which can be indiviually
+	  registered during a static build.
+	  More details here:
+	  https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
+	  https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>
+
+2021-02-16 17:32:26 +0100  Stéphane Cerveau <scerveau@collabora.com>
+
+	* gst/wavparse/gstwavparse.c:
+	* gst/wavparse/gstwavparse.h:
+	  wavparse: allow per feature registration
+	  Split plugin into features including
+	  dynamic types which can be indiviually
+	  registered during a static build.
+	  More details here:
+	  https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
+	  https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>
+
+2021-02-16 17:29:40 +0100  Stéphane Cerveau <scerveau@collabora.com>
+
+	* gst/wavenc/gstwavenc.c:
+	* gst/wavenc/gstwavenc.h:
+	  wavenc: allow per feature registration
+	  Split plugin into features including
+	  dynamic types which can be indiviually
+	  registered during a static build.
+	  More details here:
+	  https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
+	  https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>
+
+2021-02-16 17:27:24 +0100  Stéphane Cerveau <scerveau@collabora.com>
+
+	* gst/spectrum/gstspectrum.c:
+	* gst/spectrum/gstspectrum.h:
+	  spectrum: allow per feature registration
+	  Split plugin into features including
+	  dynamic types which can be indiviually
+	  registered during a static build.
+	  More details here:
+	  https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
+	  https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>
+
+2021-02-16 17:27:12 +0100  Stéphane Cerveau <scerveau@collabora.com>
+
+	* gst/monoscope/gstmonoscope.c:
+	* gst/monoscope/gstmonoscope.h:
+	  monoscope: allow per feature registration
+	  Split plugin into features including
+	  dynamic types which can be indiviually
+	  registered during a static build.
+	  More details here:
+	  https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
+	  https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>
+
+2021-02-16 17:22:47 +0100  Stéphane Cerveau <scerveau@collabora.com>
+
+	* gst/imagefreeze/gstimagefreeze.c:
+	* gst/imagefreeze/gstimagefreeze.h:
+	  imagefreeze: allow per feature registration
+	  Split plugin into features including
+	  dynamic types which can be indiviually
+	  registered during a static build.
+	  More details here:
+	  https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
+	  https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>
+
+2021-02-16 17:19:52 +0100  Stéphane Cerveau <scerveau@collabora.com>
+
+	* gst/id3demux/gstid3demux.c:
+	* gst/id3demux/gstid3demux.h:
+	  id3demux: allow per feature registration
+	  Split plugin into features including
+	  dynamic types which can be indiviually
+	  registered during a static build.
+	  More details here:
+	  https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
+	  https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>
+
+2021-02-16 17:16:33 +0100  Stéphane Cerveau <scerveau@collabora.com>
+
+	* gst/icydemux/gsticydemux.c:
+	* gst/icydemux/gsticydemux.h:
+	  icydemux: allow per feature registration
+	  Split plugin into features including
+	  dynamic types which can be indiviually
+	  registered during a static build.
+	  More details here:
+	  https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
+	  https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>
+
+2021-02-16 17:14:26 +0100  Stéphane Cerveau <scerveau@collabora.com>
+
+	* gst/goom2k1/gstgoom.c:
+	* gst/goom2k1/gstgoom.h:
+	  goom2k1: allow per feature registration
+	  Split plugin into features including
+	  dynamic types which can be indiviually
+	  registered during a static build.
+	  More details here:
+	  https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
+	  https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>
+
+2021-02-16 17:11:26 +0100  Stéphane Cerveau <scerveau@collabora.com>
+
+	* gst/cutter/gstcutter.c:
+	* gst/cutter/gstcutter.h:
+	  cutter: allow per feature registration
+	  Split plugin into features including
+	  dynamic types which can be indiviually
+	  registered during a static build.
+	  More details here:
+	  https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
+	  https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>
+
+2021-02-16 17:11:14 +0100  Stéphane Cerveau <scerveau@collabora.com>
+
+	* gst/goom/gstgoom.c:
+	* gst/goom/gstgoom.h:
+	  goom: allow per feature registration
+	  Split plugin into features including
+	  dynamic types which can be indiviually
+	  registered during a static build.
+	  More details here:
+	  https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
+	  https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>
+
+2021-02-16 17:10:33 +0100  Stéphane Cerveau <scerveau@collabora.com>
+
+	* gst/deinterlace/gstdeinterlace.c:
+	* gst/deinterlace/gstdeinterlace.h:
+	  deinterlace: allow per feature registration
+	  Split plugin into features including
+	  dynamic types which can be indiviually
+	  registered during a static build.
+	  More details here:
+	  https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
+	  https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>
+
+2021-02-16 16:34:48 +0100  Stéphane Cerveau <scerveau@collabora.com>
+
+	* sys/oss4/gstoss4audioplugin.c:
+	* sys/oss4/meson.build:
+	* sys/oss4/oss4-audio.c:
+	* sys/oss4/oss4-audio.h:
+	* sys/oss4/oss4-sink.c:
+	* sys/oss4/oss4-sink.h:
+	* sys/oss4/oss4-source.c:
+	* sys/oss4/oss4-source.h:
+	  oss4: allow per feature registration
+	  Split plugin into features including
+	  dynamic types which can be indiviually
+	  registered during a static build.
+	  More details here:
+	  https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
+	  https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>
+
+2021-02-16 16:11:36 +0100  Stéphane Cerveau <scerveau@collabora.com>
+
+	* sys/oss/gstossaudio.c:
+	* sys/oss/gstossaudioelement.c:
+	* sys/oss/gstossaudioelements.h:
+	* sys/oss/gstosssink.c:
+	* sys/oss/gstosssrc.c:
+	* sys/oss/meson.build:
+	  oss: allow per feature registration
+	  Split plugin into features including
+	  dynamic types which can be indiviually
+	  registered during a static build.
+	  More details here:
+	  https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
+	  https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>
+
+2021-02-16 15:56:35 +0100  Stéphane Cerveau <scerveau@collabora.com>
+
+	* gst/auparse/gstauparse.c:
+	* gst/auparse/gstauparse.h:
+	  auparse: allow per feature registration
+	  Split plugin into features including
+	  dynamic types which can be indiviually
+	  registered during a static build.
+	  More details here:
+	  https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
+	  https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>
+
+2021-02-16 15:29:06 +0100  Stéphane Cerveau <scerveau@collabora.com>
+
+	* sys/v4l2/gstv4l2.c:
+	* sys/v4l2/gstv4l2deviceprovider.c:
+	* sys/v4l2/gstv4l2element.c:
+	* sys/v4l2/gstv4l2elements.h:
+	* sys/v4l2/gstv4l2radio.c:
+	* sys/v4l2/gstv4l2sink.c:
+	* sys/v4l2/gstv4l2src.c:
+	* sys/v4l2/meson.build:
+	  v4l2: allow per feature registration
+	  Split plugin into features including
+	  dynamic types which can be indiviually
+	  registered during a static build.
+	  More details here:
+	  https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
+	  https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>
+
+2021-02-16 15:05:43 +0100  Stéphane Cerveau <scerveau@collabora.com>
+
+	* gst/videofilter/gstgamma.c:
+	* gst/videofilter/gstgamma.h:
+	* gst/videofilter/gstvideobalance.c:
+	* gst/videofilter/gstvideobalance.h:
+	* gst/videofilter/gstvideoflip.c:
+	* gst/videofilter/gstvideoflip.h:
+	* gst/videofilter/gstvideomedian.c:
+	* gst/videofilter/gstvideomedian.h:
+	* gst/videofilter/plugin.c:
+	  videofilter: allow per feature registration
+	  Split plugin into features including
+	  dynamic types which can be indiviually
+	  registered during a static build.
+	  More details here:
+	  https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
+	  https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>
+
+2021-02-16 14:58:57 +0100  Stéphane Cerveau <scerveau@collabora.com>
+
+	* gst/videocrop/gstaspectratiocrop.c:
+	* gst/videocrop/gstvideocrop.c:
+	* gst/videocrop/gstvideocropelement.c:
+	* gst/videocrop/gstvideocropelements.h:
+	* gst/videocrop/gstvideocropplugin.c:
+	* gst/videocrop/meson.build:
+	  videocrop: allow per feature registration
+	  Split plugin into features including
+	  dynamic types which can be indiviually
+	  registered during a static build.
+	  More details here:
+	  https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
+	  https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>
+
+2021-02-16 14:54:15 +0100  Stéphane Cerveau <scerveau@collabora.com>
+
+	* gst/videobox/gstvideobox.c:
+	* gst/videobox/gstvideobox.h:
+	  videobox: allow per feature registration
+	  Split plugin into features including
+	  dynamic types which can be indiviually
+	  registered during a static build.
+	  More details here:
+	  https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
+	  https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>
+
+2021-02-16 14:49:56 +0100  Stéphane Cerveau <scerveau@collabora.com>
+
+	* gst/udp/gstdynudpsink.c:
+	* gst/udp/gstmultiudpsink.c:
+	* gst/udp/gstudp.c:
+	* gst/udp/gstudpelement.c:
+	* gst/udp/gstudpelements.h:
+	* gst/udp/gstudpsink.c:
+	* gst/udp/gstudpsrc.c:
+	* gst/udp/meson.build:
+	  udp: allow per feature registration
+	  Split plugin into features including
+	  dynamic types which can be indiviually
+	  registered during a static build.
+	  More details here:
+	  https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
+	  https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>
+
+2021-02-16 14:43:32 +0100  Stéphane Cerveau <scerveau@collabora.com>
+
+	* gst/smpte/gstsmpte.c:
+	* gst/smpte/gstsmpte.h:
+	* gst/smpte/gstsmptealpha.c:
+	* gst/smpte/gstsmptealpha.h:
+	* gst/smpte/plugin.c:
+	  smpte: allow per feature registration
+	  Split plugin into features including
+	  dynamic types which can be indiviually
+	  registered during a static build.
+	  More details here:
+	  https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
+	  https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>
+
+2021-02-16 14:38:37 +0100  Stéphane Cerveau <scerveau@collabora.com>
+
+	* gst/shapewipe/gstshapewipe.c:
+	* gst/shapewipe/gstshapewipe.h:
+	  shapewipe: allow per feature registration
+	  Split plugin into features including
+	  dynamic types which can be indiviually
+	  registered during a static build.
+	  More details here:
+	  https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
+	  https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>
+
+2021-02-16 14:35:51 +0100  Stéphane Cerveau <scerveau@collabora.com>
+
+	* gst/rtsp/gstrtpdec.c:
+	* gst/rtsp/gstrtsp.c:
+	* gst/rtsp/gstrtspelement.c:
+	* gst/rtsp/gstrtspelements.h:
+	* gst/rtsp/gstrtspsrc.c:
+	* gst/rtsp/meson.build:
+	  rtsp: allow per feature registration
+	  Split plugin into features including
+	  dynamic types which can be indiviually
+	  registered during a static build.
+	  More details here:
+	  https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
+	  https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>
+
+2021-02-16 14:24:33 +0100  Stéphane Cerveau <scerveau@collabora.com>
+
+	* gst/rtpmanager/gstrtpbin.c:
+	* gst/rtpmanager/gstrtpbin.h:
+	* gst/rtpmanager/gstrtpdtmfmux.c:
+	* gst/rtpmanager/gstrtpdtmfmux.h:
+	* gst/rtpmanager/gstrtpfunnel.c:
+	* gst/rtpmanager/gstrtpfunnel.h:
+	* gst/rtpmanager/gstrtphdrext-rfc6464.c:
+	* gst/rtpmanager/gstrtphdrext-rfc6464.h:
+	* gst/rtpmanager/gstrtphdrext-twcc.c:
+	* gst/rtpmanager/gstrtphdrext-twcc.h:
+	* gst/rtpmanager/gstrtpjitterbuffer.c:
+	* gst/rtpmanager/gstrtpjitterbuffer.h:
+	* gst/rtpmanager/gstrtpmanager.c:
+	* gst/rtpmanager/gstrtpmux.c:
+	* gst/rtpmanager/gstrtpmux.h:
+	* gst/rtpmanager/gstrtpptdemux.c:
+	* gst/rtpmanager/gstrtpptdemux.h:
+	* gst/rtpmanager/gstrtprtxqueue.c:
+	* gst/rtpmanager/gstrtprtxqueue.h:
+	* gst/rtpmanager/gstrtprtxreceive.c:
+	* gst/rtpmanager/gstrtprtxreceive.h:
+	* gst/rtpmanager/gstrtprtxsend.c:
+	* gst/rtpmanager/gstrtprtxsend.h:
+	* gst/rtpmanager/gstrtpsession.c:
+	* gst/rtpmanager/gstrtpsession.h:
+	* gst/rtpmanager/gstrtpssrcdemux.c:
+	* gst/rtpmanager/gstrtpssrcdemux.h:
+	* gst/rtpmanager/gstrtpst2022-1-fecdec.c:
+	* gst/rtpmanager/gstrtpst2022-1-fecdec.h:
+	* gst/rtpmanager/gstrtpst2022-1-fecenc.c:
+	* gst/rtpmanager/gstrtpst2022-1-fecenc.h:
+	  rtpmanager: allow per feature registration
+	  Split plugin into features including
+	  dynamic types which can be indiviually
+	  registered during a static build.
+	  More details here:
+	  https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
+	  https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>
+
+2021-02-16 13:49:15 +0100  Stéphane Cerveau <scerveau@collabora.com>
+
+	* gst/replaygain/gstrganalysis.c:
+	* gst/replaygain/gstrglimiter.c:
+	* gst/replaygain/gstrglimiter.h:
+	* gst/replaygain/gstrgvolume.c:
+	* gst/replaygain/gstrgvolume.h:
+	* gst/replaygain/replaygain.c:
+	* gst/replaygain/rganalysis.h:
+	  replaygain: allow per feature registration
+	  Split plugin into features including
+	  dynamic types which can be indiviually
+	  registered during a static build.
+	  More details here:
+	  https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
+	  https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>
+
+2021-02-16 13:43:44 +0100  Stéphane Cerveau <scerveau@collabora.com>
+
+	* gst/multipart/multipart.c:
+	* gst/multipart/multipartdemux.c:
+	* gst/multipart/multipartdemux.h:
+	* gst/multipart/multipartmux.c:
+	* gst/multipart/multipartmux.h:
+	  multipart: allow per feature registration
+	  Split plugin into features including
+	  dynamic types which can be indiviually
+	  registered during a static build.
+	  More details here:
+	  https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
+	  https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>
+
+2021-02-16 12:04:26 +0100  Stéphane Cerveau <scerveau@collabora.com>
+
+	* gst/multifile/gstimagesequencesrc.c:
+	* gst/multifile/gstimagesequencesrc.h:
+	* gst/multifile/gstmultifile.c:
+	* gst/multifile/gstmultifilesink.c:
+	* gst/multifile/gstmultifilesink.h:
+	* gst/multifile/gstmultifilesrc.c:
+	* gst/multifile/gstmultifilesrc.h:
+	* gst/multifile/gstsplitfilesrc.c:
+	* gst/multifile/gstsplitfilesrc.h:
+	* gst/multifile/gstsplitmuxsink.c:
+	* gst/multifile/gstsplitmuxsink.h:
+	* gst/multifile/gstsplitmuxsrc.c:
+	* gst/multifile/gstsplitmuxsrc.h:
+	  multifile: allow per feature registration
+	  Split plugin into features including
+	  dynamic types which can be indiviually
+	  registered during a static build.
+	  More details here:
+	  https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
+	  https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>
+
+2021-02-16 11:14:17 +0100  Stéphane Cerveau <scerveau@collabora.com>
+
+	* gst/matroska/gstmatroskaelement.c:
+	* gst/matroska/gstmatroskaelements.h:
+	* gst/matroska/matroska-demux.c:
+	* gst/matroska/matroska-demux.h:
+	* gst/matroska/matroska-mux.c:
+	* gst/matroska/matroska-parse.c:
+	* gst/matroska/matroska-parse.h:
+	* gst/matroska/matroska.c:
+	* gst/matroska/meson.build:
+	* gst/matroska/webm-mux.c:
+	  matroska: allow per feature registration
+	  Split plugin into features including
+	  dynamic types which can be indiviually
+	  registered during a static build.
+	  More details here:
+	  https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
+	  https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>
+
+2021-02-16 10:59:34 +0100  Stéphane Cerveau <scerveau@collabora.com>
+
+	* gst/level/gstlevel.c:
+	* gst/level/gstlevel.h:
+	  level: allow per feature registration
+	  Split plugin into features including
+	  dynamic types which can be indiviually
+	  registered during a static build.
+	  More details here:
+	  https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
+	  https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>
+
+2021-02-16 10:57:58 +0100  Stéphane Cerveau <scerveau@collabora.com>
+
+	* gst/law/alaw-decode.c:
+	* gst/law/alaw-decode.h:
+	* gst/law/alaw-encode.c:
+	* gst/law/alaw-encode.h:
+	* gst/law/alaw.c:
+	* gst/law/mulaw-decode.c:
+	* gst/law/mulaw-decode.h:
+	* gst/law/mulaw-encode.c:
+	* gst/law/mulaw-encode.h:
+	* gst/law/mulaw.c:
+	  law: allow per feature registration
+	  Split plugin into features including
+	  dynamic types which can be indiviually
+	  registered during a static build.
+	  More details here:
+	  https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
+	  https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>
+
+2021-02-16 10:26:40 +0100  Stéphane Cerveau <scerveau@collabora.com>
+
+	* gst/isomp4/gstisomp4element.c:
+	* gst/isomp4/gstisomp4elements.h:
+	* gst/isomp4/gstqtmoovrecover.c:
+	* gst/isomp4/gstqtmux.c:
+	* gst/isomp4/gstrtpxqtdepay.c:
+	* gst/isomp4/isomp4-plugin.c:
+	* gst/isomp4/meson.build:
+	* gst/isomp4/qtdemux.c:
+	  isomp4: allow per feature registration
+	  Split plugin into features including
+	  dynamic types which can be indiviually
+	  registered during a static build.
+	  More details here:
+	  https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
+	  https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>
+
+2021-02-16 09:57:27 +0100  Stéphane Cerveau <scerveau@collabora.com>
+
+	* gst/interleave/deinterleave.c:
+	* gst/interleave/gstinterleaveelements.h:
+	* gst/interleave/interleave.c:
+	* gst/interleave/plugin.c:
+	  interleave: allow per feature registration
+	  Split plugin into features including
+	  dynamic types which can be indiviually
+	  registered during a static build.
+	  More details here:
+	  https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
+	  https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>
+
+2021-02-16 09:51:16 +0100  Stéphane Cerveau <scerveau@collabora.com>
+
+	* gst/flx/gstflxdec.c:
+	* gst/flx/gstflxdec.h:
+	  flx: allow per feature registration
+	  Split plugin into features including
+	  dynamic types which can be indiviually
+	  registered during a static build.
+	  More details here:
+	  https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
+	  https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>
+
+2021-02-15 17:37:09 +0100  Stéphane Cerveau <scerveau@collabora.com>
+
+	* gst/flv/gstflvdemux.c:
+	* gst/flv/gstflvelement.c:
+	* gst/flv/gstflvelements.h:
+	* gst/flv/gstflvmux.c:
+	* gst/flv/gstflvplugin.c:
+	* gst/flv/meson.build:
+	  flv: allow per feature registration
+	  Split plugin into features including
+	  dynamic types which can be indiviually
+	  registered during a static build.
+	  More details here:
+	  https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
+	  https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>
+
+2021-02-15 17:27:51 +0100  Stéphane Cerveau <scerveau@collabora.com>
+
+	* gst/equalizer/gstiirequalizer.c:
+	* gst/equalizer/gstiirequalizer.h:
+	* gst/equalizer/gstiirequalizer10bands.c:
+	* gst/equalizer/gstiirequalizer3bands.c:
+	* gst/equalizer/gstiirequalizernbands.c:
+	* gst/equalizer/gstiirequalizerplugin.c:
+	* gst/equalizer/meson.build:
+	  equalizer: allow per feature registration
+	  Split plugin into features including
+	  dynamic types which can be indiviually
+	  registered during a static build.
+	  More details here:
+	  https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
+	  https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>
+
+2021-02-15 15:37:52 +0100  Stéphane Cerveau <scerveau@collabora.com>
+
+	* gst/effectv/gstaging.c:
+	* gst/effectv/gstdice.c:
+	* gst/effectv/gstedge.c:
+	* gst/effectv/gsteffectv.c:
+	* gst/effectv/gsteffectv.h:
+	* gst/effectv/gstop.c:
+	* gst/effectv/gstquark.c:
+	* gst/effectv/gstradioac.c:
+	* gst/effectv/gstrev.c:
+	* gst/effectv/gstripple.c:
+	* gst/effectv/gstshagadelic.c:
+	* gst/effectv/gststreak.c:
+	* gst/effectv/gstvertigo.c:
+	* gst/effectv/gstwarp.c:
+	  effectv: allow per feature registration
+	  Split plugin into features including
+	  dynamic types which can be indiviually
+	  registered during a static build.
+	  More details here:
+	  https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
+	  https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>
+
+2021-02-15 15:03:10 +0100  Stéphane Cerveau <scerveau@collabora.com>
+
+	* gst/dtmf/gstdtmf.c:
+	* gst/dtmf/gstdtmfsrc.c:
+	* gst/dtmf/gstdtmfsrc.h:
+	* gst/dtmf/gstrtpdtmfdepay.c:
+	* gst/dtmf/gstrtpdtmfdepay.h:
+	* gst/dtmf/gstrtpdtmfsrc.c:
+	* gst/dtmf/gstrtpdtmfsrc.h:
+	  dtmf: allow per feature registration
+	  Split plugin into features including
+	  dynamic types which can be indiviually
+	  registered during a static build.
+	  More details here:
+	  https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
+	  https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>
+
+2021-02-15 14:55:15 +0100  Stéphane Cerveau <scerveau@collabora.com>
+
+	* gst/debugutils/breakmydata.c:
+	* gst/debugutils/cpureport.c:
+	* gst/debugutils/gstcapsdebug.c:
+	* gst/debugutils/gstcapssetter.c:
+	* gst/debugutils/gstdebug.c:
+	* gst/debugutils/gstdebugutilselements.h:
+	* gst/debugutils/gstnavigationtest.c:
+	* gst/debugutils/gstnavigationtest.h:
+	* gst/debugutils/gstnavseek.c:
+	* gst/debugutils/gstpushfilesrc.c:
+	* gst/debugutils/gsttaginject.c:
+	* gst/debugutils/progressreport.c:
+	* gst/debugutils/rndbuffersize.c:
+	* gst/debugutils/testplugin.c:
+	  debugutils: allow per feature registration
+	  Split plugin into features including
+	  dynamic types which can be indiviually
+	  registered during a static build.
+	  More details here:
+	  https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
+	  https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>
+
+2021-02-15 13:38:21 +0100  Stéphane Cerveau <scerveau@collabora.com>
+
+	* gst/avi/gstavi.c:
+	* gst/avi/gstavidemux.c:
+	* gst/avi/gstavielement.c:
+	* gst/avi/gstavielements.h:
+	* gst/avi/gstavimux.c:
+	* gst/avi/gstavisubtitle.c:
+	* gst/avi/meson.build:
+	  avi: allow per feature registration
+	  Split plugin into features including
+	  dynamic types which can be indiviually
+	  registered during a static build.
+	  More details here:
+	  https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
+	  https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>
+
+2021-02-15 13:02:59 +0100  Stéphane Cerveau <scerveau@collabora.com>
+
+	* gst/autodetect/gstautoaudiosink.c:
+	* gst/autodetect/gstautoaudiosrc.c:
+	* gst/autodetect/gstautodetect.c:
+	* gst/autodetect/gstautodetect.h:
+	* gst/autodetect/gstautodetectelement.c:
+	* gst/autodetect/gstautodetectelements.h:
+	* gst/autodetect/gstautodetectplugin.c:
+	* gst/autodetect/gstautovideosink.c:
+	* gst/autodetect/gstautovideosrc.c:
+	* gst/autodetect/meson.build:
+	  autodetect: allow per feature registration
+	  Split plugin into features including
+	  dynamic types which can be indiviually
+	  registered during a static build.
+	  More details here:
+	  https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
+	  https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>
+
+2021-02-15 13:00:38 +0100  Stéphane Cerveau <scerveau@collabora.com>
+
+	* gst/audioparsers/gstaacparse.c:
+	* gst/audioparsers/gstac3parse.c:
+	* gst/audioparsers/gstamrparse.c:
+	* gst/audioparsers/gstaudioparserselements.h:
+	* gst/audioparsers/gstdcaparse.c:
+	* gst/audioparsers/gstflacparse.c:
+	* gst/audioparsers/gstmpegaudioparse.c:
+	* gst/audioparsers/gstsbcparse.c:
+	* gst/audioparsers/gstwavpackparse.c:
+	* gst/audioparsers/plugin.c:
+	  audioparsers: allow per feature registration
+	  Split plugin into features including
+	  dynamic types which can be indiviually
+	  registered during a static build.
+	  More details here:
+	  https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
+	  https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>
+
+2021-02-15 12:44:31 +0100  Stéphane Cerveau <scerveau@collabora.com>
+
+	* gst/apetag/gstapedemux.c:
+	* gst/apetag/gstapedemux.h:
+	  apetag: allow per feature registration
+	  Split plugin into features including
+	  dynamic types which can be indiviually
+	  registered during a static build.
+	  More details here:
+	  https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
+	  https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>
+
+2021-02-15 11:00:46 +0100  Stéphane Cerveau <scerveau@collabora.com>
+
+	* ext/vpx/gstvp8dec.c:
+	* ext/vpx/gstvp8enc.c:
+	* ext/vpx/gstvp9dec.c:
+	* ext/vpx/gstvp9enc.c:
+	* ext/vpx/gstvpxelement.c:
+	* ext/vpx/gstvpxelements.h:
+	* ext/vpx/meson.build:
+	* ext/vpx/plugin.c:
+	  vpx: allow per feature registration
+	  Split plugin into features including
+	  dynamic types which can be indiviually
+	  registered during a static build.
+	  More details here:
+	  https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
+	  https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>
+
+2021-02-12 17:26:36 +0100  Stéphane Cerveau <scerveau@collabora.com>
+
+	* ext/taglib/gstapev2mux.cc:
+	* ext/taglib/gstid3v2mux.cc:
+	* ext/taglib/gsttaglibelement.c:
+	* ext/taglib/gsttaglibelements.h:
+	* ext/taglib/gsttaglibplugin.c:
+	* ext/taglib/meson.build:
+	  taglib: allow per feature registration
+	  Split plugin into features including
+	  dynamic types which can be indiviually
+	  registered during a static build.
+	  More details here:
+	  https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
+	  https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>
+
+2021-02-12 17:09:19 +0100  Stéphane Cerveau <scerveau@collabora.com>
+
+	* ext/qt/gstplugin.cc:
+	* ext/qt/gstqtelement.cc:
+	* ext/qt/gstqtelements.h:
+	* ext/qt/gstqtoverlay.cc:
+	* ext/qt/gstqtsink.cc:
+	* ext/qt/gstqtsrc.cc:
+	* ext/qt/meson.build:
+	* ext/qt/qtplugin.pro:
+	  qt: allow per feature registration
+	  Split plugin into features including
+	  dynamic types which can be indiviually
+	  registered during a static build.
+	  More details here:
+	  https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
+	  https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>
+
+2021-02-12 16:09:53 +0100  Stéphane Cerveau <scerveau@collabora.com>
+
+	* ext/speex/gstspeex.c:
+	* ext/speex/gstspeexdec.c:
+	* ext/speex/gstspeexelement.c:
+	* ext/speex/gstspeexelements.h:
+	* ext/speex/gstspeexenc.c:
+	* ext/speex/meson.build:
+	  speex: allow per feature registration
+	  Split plugin into features including
+	  dynamic types which can be indiviually
+	  registered during a static build.
+	  More details here:
+	  https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
+	  https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>
+
+2021-02-12 16:04:16 +0100  Stéphane Cerveau <scerveau@collabora.com>
+
+	* ext/soup/gstsoup.c:
+	* ext/soup/gstsoupelement.c:
+	* ext/soup/gstsoupelements.h:
+	* ext/soup/gstsouphttpclientsink.c:
+	* ext/soup/gstsouphttpsrc.c:
+	* ext/soup/meson.build:
+	  soup: allow per feature registration
+	  Split plugin into features including
+	  dynamic types which can be indiviually
+	  registered during a static build.
+	  More details here:
+	  https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
+	  https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>
+
+2021-02-12 15:53:19 +0100  Stéphane Cerveau <scerveau@collabora.com>
+
+	* ext/raw1394/gst1394.c:
+	* ext/raw1394/gstdv1394src.c:
+	* ext/raw1394/gstdv1394src.h:
+	* ext/raw1394/gsthdv1394src.c:
+	* ext/raw1394/gsthdv1394src.h:
+	  raw1394: allow per feature registration
+	  Split plugin into features including
+	  dynamic types which can be indiviually
+	  registered during a static build.
+	  More details here:
+	  https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
+	  https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>
+
+2021-02-12 15:47:46 +0100  Stéphane Cerveau <scerveau@collabora.com>
+
+	* ext/wavpack/gstwavpack.c:
+	* ext/wavpack/gstwavpackdec.c:
+	* ext/wavpack/gstwavpackelement.c:
+	* ext/wavpack/gstwavpackelements.h:
+	* ext/wavpack/gstwavpackenc.c:
+	* ext/wavpack/meson.build:
+	  wavpack: allow per feature registration
+	  Split plugin into features including
+	  dynamic types which can be indiviually
+	  registered during a static build.
+	  More details here:
+	  https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
+	  https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>
+
+2021-02-12 15:35:11 +0100  Stéphane Cerveau <scerveau@collabora.com>
+
+	* gst/alpha/gstalpha.c:
+	* gst/alpha/gstalpha.h:
+	  alpha: allow per feature registration
+	  Split plugin into features including
+	  dynamic types which can be indiviually
+	  registered during a static build.
+	  More details here:
+	  https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
+	  https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>
+
+2021-02-12 15:27:31 +0100  Stéphane Cerveau <scerveau@collabora.com>
+
+	* gst/audiofx/audioamplify.c:
+	* gst/audiofx/audioamplify.h:
+	* gst/audiofx/audiochebband.c:
+	* gst/audiofx/audiochebband.h:
+	* gst/audiofx/audiocheblimit.c:
+	* gst/audiofx/audiocheblimit.h:
+	* gst/audiofx/audiodynamic.c:
+	* gst/audiofx/audiodynamic.h:
+	* gst/audiofx/audioecho.c:
+	* gst/audiofx/audioecho.h:
+	* gst/audiofx/audiofirfilter.c:
+	* gst/audiofx/audiofirfilter.h:
+	* gst/audiofx/audiofx.c:
+	* gst/audiofx/audioiirfilter.c:
+	* gst/audiofx/audioiirfilter.h:
+	* gst/audiofx/audioinvert.c:
+	* gst/audiofx/audioinvert.h:
+	* gst/audiofx/audiokaraoke.c:
+	* gst/audiofx/audiokaraoke.h:
+	* gst/audiofx/audiopanorama.c:
+	* gst/audiofx/audiopanorama.h:
+	* gst/audiofx/audiowsincband.c:
+	* gst/audiofx/audiowsincband.h:
+	* gst/audiofx/audiowsinclimit.c:
+	* gst/audiofx/audiowsinclimit.h:
+	* gst/audiofx/gstscaletempo.c:
+	* gst/audiofx/gstscaletempo.h:
+	* gst/audiofx/gststereo.c:
+	* gst/audiofx/gststereo.h:
+	  audiofx: allow per feature registration
+	  Split plugin into features including
+	  dynamic types which can be indiviually
+	  registered during a static build.
+	  More details here:
+	  https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
+	  https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>
+
+2021-02-12 13:16:28 +0100  Stéphane Cerveau <scerveau@collabora.com>
+
+	* gst/rtp/gstasteriskh263.c:
+	* gst/rtp/gstasteriskh263.h:
+	* gst/rtp/gstrtp.c:
+	* gst/rtp/gstrtpL16depay.c:
+	* gst/rtp/gstrtpL16depay.h:
+	* gst/rtp/gstrtpL16pay.c:
+	* gst/rtp/gstrtpL16pay.h:
+	* gst/rtp/gstrtpL24depay.c:
+	* gst/rtp/gstrtpL24depay.h:
+	* gst/rtp/gstrtpL24pay.c:
+	* gst/rtp/gstrtpL24pay.h:
+	* gst/rtp/gstrtpL8depay.c:
+	* gst/rtp/gstrtpL8depay.h:
+	* gst/rtp/gstrtpL8pay.c:
+	* gst/rtp/gstrtpL8pay.h:
+	* gst/rtp/gstrtpac3depay.c:
+	* gst/rtp/gstrtpac3depay.h:
+	* gst/rtp/gstrtpac3pay.c:
+	* gst/rtp/gstrtpac3pay.h:
+	* gst/rtp/gstrtpamrdepay.c:
+	* gst/rtp/gstrtpamrdepay.h:
+	* gst/rtp/gstrtpamrpay.c:
+	* gst/rtp/gstrtpamrpay.h:
+	* gst/rtp/gstrtpbvdepay.c:
+	* gst/rtp/gstrtpbvdepay.h:
+	* gst/rtp/gstrtpbvpay.c:
+	* gst/rtp/gstrtpbvpay.h:
+	* gst/rtp/gstrtpceltdepay.c:
+	* gst/rtp/gstrtpceltdepay.h:
+	* gst/rtp/gstrtpceltpay.c:
+	* gst/rtp/gstrtpceltpay.h:
+	* gst/rtp/gstrtpdvdepay.c:
+	* gst/rtp/gstrtpdvdepay.h:
+	* gst/rtp/gstrtpdvpay.c:
+	* gst/rtp/gstrtpdvpay.h:
+	* gst/rtp/gstrtpelement.c:
+	* gst/rtp/gstrtpelements.h:
+	* gst/rtp/gstrtpg722depay.c:
+	* gst/rtp/gstrtpg722depay.h:
+	* gst/rtp/gstrtpg722pay.c:
+	* gst/rtp/gstrtpg722pay.h:
+	* gst/rtp/gstrtpg723depay.c:
+	* gst/rtp/gstrtpg723depay.h:
+	* gst/rtp/gstrtpg723pay.c:
+	* gst/rtp/gstrtpg723pay.h:
+	* gst/rtp/gstrtpg726depay.c:
+	* gst/rtp/gstrtpg726depay.h:
+	* gst/rtp/gstrtpg726pay.c:
+	* gst/rtp/gstrtpg726pay.h:
+	* gst/rtp/gstrtpg729depay.c:
+	* gst/rtp/gstrtpg729depay.h:
+	* gst/rtp/gstrtpg729pay.c:
+	* gst/rtp/gstrtpg729pay.h:
+	* gst/rtp/gstrtpgsmdepay.c:
+	* gst/rtp/gstrtpgsmdepay.h:
+	* gst/rtp/gstrtpgsmpay.c:
+	* gst/rtp/gstrtpgsmpay.h:
+	* gst/rtp/gstrtpgstdepay.c:
+	* gst/rtp/gstrtpgstdepay.h:
+	* gst/rtp/gstrtpgstpay.c:
+	* gst/rtp/gstrtpgstpay.h:
+	* gst/rtp/gstrtph261depay.c:
+	* gst/rtp/gstrtph261depay.h:
+	* gst/rtp/gstrtph261pay.c:
+	* gst/rtp/gstrtph261pay.h:
+	* gst/rtp/gstrtph263depay.c:
+	* gst/rtp/gstrtph263depay.h:
+	* gst/rtp/gstrtph263pay.c:
+	* gst/rtp/gstrtph263pay.h:
+	* gst/rtp/gstrtph263pdepay.c:
+	* gst/rtp/gstrtph263pdepay.h:
+	* gst/rtp/gstrtph263ppay.c:
+	* gst/rtp/gstrtph263ppay.h:
+	* gst/rtp/gstrtph264depay.c:
+	* gst/rtp/gstrtph264depay.h:
+	* gst/rtp/gstrtph264pay.c:
+	* gst/rtp/gstrtph264pay.h:
+	* gst/rtp/gstrtph265depay.c:
+	* gst/rtp/gstrtph265depay.h:
+	* gst/rtp/gstrtph265pay.c:
+	* gst/rtp/gstrtph265pay.h:
+	* gst/rtp/gstrtpilbcdepay.c:
+	* gst/rtp/gstrtpilbcdepay.h:
+	* gst/rtp/gstrtpilbcpay.c:
+	* gst/rtp/gstrtpilbcpay.h:
+	* gst/rtp/gstrtpisacdepay.c:
+	* gst/rtp/gstrtpisacdepay.h:
+	* gst/rtp/gstrtpisacpay.c:
+	* gst/rtp/gstrtpisacpay.h:
+	* gst/rtp/gstrtpj2kdepay.c:
+	* gst/rtp/gstrtpj2kdepay.h:
+	* gst/rtp/gstrtpj2kpay.c:
+	* gst/rtp/gstrtpj2kpay.h:
+	* gst/rtp/gstrtpjpegdepay.c:
+	* gst/rtp/gstrtpjpegdepay.h:
+	* gst/rtp/gstrtpjpegpay.c:
+	* gst/rtp/gstrtpjpegpay.h:
+	* gst/rtp/gstrtpklvdepay.c:
+	* gst/rtp/gstrtpklvdepay.h:
+	* gst/rtp/gstrtpklvpay.c:
+	* gst/rtp/gstrtpklvpay.h:
+	* gst/rtp/gstrtpldacpay.c:
+	* gst/rtp/gstrtpmp1sdepay.c:
+	* gst/rtp/gstrtpmp1sdepay.h:
+	* gst/rtp/gstrtpmp2tdepay.c:
+	* gst/rtp/gstrtpmp2tdepay.h:
+	* gst/rtp/gstrtpmp2tpay.c:
+	* gst/rtp/gstrtpmp2tpay.h:
+	* gst/rtp/gstrtpmp4adepay.c:
+	* gst/rtp/gstrtpmp4adepay.h:
+	* gst/rtp/gstrtpmp4apay.c:
+	* gst/rtp/gstrtpmp4apay.h:
+	* gst/rtp/gstrtpmp4gdepay.c:
+	* gst/rtp/gstrtpmp4gdepay.h:
+	* gst/rtp/gstrtpmp4gpay.c:
+	* gst/rtp/gstrtpmp4gpay.h:
+	* gst/rtp/gstrtpmp4vdepay.c:
+	* gst/rtp/gstrtpmp4vdepay.h:
+	* gst/rtp/gstrtpmp4vpay.c:
+	* gst/rtp/gstrtpmp4vpay.h:
+	* gst/rtp/gstrtpmpadepay.c:
+	* gst/rtp/gstrtpmpadepay.h:
+	* gst/rtp/gstrtpmpapay.c:
+	* gst/rtp/gstrtpmpapay.h:
+	* gst/rtp/gstrtpmparobustdepay.c:
+	* gst/rtp/gstrtpmparobustdepay.h:
+	* gst/rtp/gstrtpmpvdepay.c:
+	* gst/rtp/gstrtpmpvdepay.h:
+	* gst/rtp/gstrtpmpvpay.c:
+	* gst/rtp/gstrtpmpvpay.h:
+	* gst/rtp/gstrtpopusdepay.c:
+	* gst/rtp/gstrtpopusdepay.h:
+	* gst/rtp/gstrtpopuspay.c:
+	* gst/rtp/gstrtpopuspay.h:
+	* gst/rtp/gstrtppcmadepay.c:
+	* gst/rtp/gstrtppcmadepay.h:
+	* gst/rtp/gstrtppcmapay.c:
+	* gst/rtp/gstrtppcmapay.h:
+	* gst/rtp/gstrtppcmudepay.c:
+	* gst/rtp/gstrtppcmudepay.h:
+	* gst/rtp/gstrtppcmupay.c:
+	* gst/rtp/gstrtppcmupay.h:
+	* gst/rtp/gstrtpqcelpdepay.c:
+	* gst/rtp/gstrtpqcelpdepay.h:
+	* gst/rtp/gstrtpqdmdepay.c:
+	* gst/rtp/gstrtpqdmdepay.h:
+	* gst/rtp/gstrtpreddec.c:
+	* gst/rtp/gstrtpredenc.c:
+	* gst/rtp/gstrtpsbcdepay.c:
+	* gst/rtp/gstrtpsbcdepay.h:
+	* gst/rtp/gstrtpsbcpay.c:
+	* gst/rtp/gstrtpsbcpay.h:
+	* gst/rtp/gstrtpsirendepay.c:
+	* gst/rtp/gstrtpsirendepay.h:
+	* gst/rtp/gstrtpsirenpay.c:
+	* gst/rtp/gstrtpsirenpay.h:
+	* gst/rtp/gstrtpspeexdepay.c:
+	* gst/rtp/gstrtpspeexdepay.h:
+	* gst/rtp/gstrtpspeexpay.c:
+	* gst/rtp/gstrtpspeexpay.h:
+	* gst/rtp/gstrtpstorage.c:
+	* gst/rtp/gstrtpstreamdepay.c:
+	* gst/rtp/gstrtpstreamdepay.h:
+	* gst/rtp/gstrtpstreampay.c:
+	* gst/rtp/gstrtpstreampay.h:
+	* gst/rtp/gstrtpsv3vdepay.c:
+	* gst/rtp/gstrtpsv3vdepay.h:
+	* gst/rtp/gstrtptheoradepay.c:
+	* gst/rtp/gstrtptheoradepay.h:
+	* gst/rtp/gstrtptheorapay.c:
+	* gst/rtp/gstrtptheorapay.h:
+	* gst/rtp/gstrtpulpfecdec.c:
+	* gst/rtp/gstrtpulpfecenc.c:
+	* gst/rtp/gstrtpvorbisdepay.c:
+	* gst/rtp/gstrtpvorbisdepay.h:
+	* gst/rtp/gstrtpvorbispay.c:
+	* gst/rtp/gstrtpvorbispay.h:
+	* gst/rtp/gstrtpvp8depay.c:
+	* gst/rtp/gstrtpvp8depay.h:
+	* gst/rtp/gstrtpvp8pay.c:
+	* gst/rtp/gstrtpvp8pay.h:
+	* gst/rtp/gstrtpvp9depay.c:
+	* gst/rtp/gstrtpvp9depay.h:
+	* gst/rtp/gstrtpvp9pay.c:
+	* gst/rtp/gstrtpvp9pay.h:
+	* gst/rtp/gstrtpvrawdepay.c:
+	* gst/rtp/gstrtpvrawdepay.h:
+	* gst/rtp/gstrtpvrawpay.c:
+	* gst/rtp/gstrtpvrawpay.h:
+	* gst/rtp/meson.build:
+	* tests/check/meson.build:
+	  rtp: allow per feature registration
+	  Split plugin into features including
+	  dynamic types which can be indiviually
+	  registered during a static build.
+	  More details here:
+	  https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
+	  https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>
+
+2021-02-12 11:12:34 +0100  Stéphane Cerveau <scerveau@collabora.com>
+
+	* ext/pulse/gstpulseelement.c:
+	* ext/pulse/gstpulseelements.h:
+	* ext/pulse/meson.build:
+	* ext/pulse/plugin.c:
+	* ext/pulse/pulsesink.c:
+	* ext/pulse/pulsesrc.c:
+	  pulse: allow per feature registration
+	  Split plugin into features including
+	  dynamic types which can be indiviually
+	  registered during a static build.
+	  More details here:
+	  https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
+	  https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>
+
+2021-02-12 10:41:29 +0100  Stéphane Cerveau <scerveau@collabora.com>
+
+	* ext/mpg123/gstmpg123audiodec.c:
+	* ext/mpg123/gstmpg123audiodec.h:
+	  mpeg123: allow per feature registration
+	  Split plugin into features including
+	  dynamic types which can be indiviually
+	  registered during a static build.
+	  More details here:
+	  https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
+	  https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>
+
+2021-02-12 10:33:50 +0100  Stéphane Cerveau <scerveau@collabora.com>
+
+	* ext/libpng/gstpng.c:
+	* ext/libpng/gstpngdec.c:
+	* ext/libpng/gstpngdec.h:
+	* ext/libpng/gstpngenc.c:
+	* ext/libpng/gstpngenc.h:
+	  libpng: allow per feature registration
+	  Split plugin into features including
+	  dynamic types which can be indiviually
+	  registered during a static build.
+	  More details here:
+	  https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
+	  https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>
+
+2021-02-12 10:27:18 +0100  Stéphane Cerveau <scerveau@collabora.com>
+
+	* ext/lame/gstlamemp3enc.c:
+	* ext/lame/gstlamemp3enc.h:
+	* ext/lame/plugin.c:
+	  lame: allow per feature registration
+	  Split plugin into features including
+	  dynamic types which can be indiviually
+	  registered during a static build.
+	  More details here:
+	  https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
+	  https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>
+
+2021-02-12 10:26:26 +0100  Stéphane Cerveau <scerveau@collabora.com>
+
+	* ext/libcaca/gstcacaplugin.c:
+	* ext/libcaca/gstcacasink.c:
+	* ext/libcaca/gstcacasink.h:
+	* ext/libcaca/gstcacatv.c:
+	* ext/libcaca/gstcacatv.h:
+	* ext/libcaca/meson.build:
+	  libcaca: allow per feature registration
+	  Split plugin into features including
+	  dynamic types which can be indiviually
+	  registered during a static build.
+	  More details here:
+	  https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
+	  https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>
+
+2021-02-12 10:09:46 +0100  Stéphane Cerveau <scerveau@collabora.com>
+
+	* ext/jpeg/gstjpeg.c:
+	* ext/jpeg/gstjpegdec.c:
+	* ext/jpeg/gstjpegelements.h:
+	* ext/jpeg/gstjpegenc.c:
+	* ext/jpeg/gstjpegplugin.c:
+	* ext/jpeg/gstsmokedec.c:
+	* ext/jpeg/gstsmokeenc.c:
+	* ext/jpeg/meson.build:
+	  jpeg: allow per feature registration
+	  Split plugin into features including
+	  dynamic types which can be indiviually
+	  registered during a static build.
+	  More details here:
+	  https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
+	  https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>
+
+2021-02-12 09:56:36 +0100  Stéphane Cerveau <scerveau@collabora.com>
+
+	* ext/jack/gstjack.c:
+	* ext/jack/gstjack.h:
+	* ext/jack/gstjackaudiosink.c:
+	* ext/jack/gstjackaudiosrc.c:
+	  jack: allow per feature registration
+	  Split plugin into features including
+	  dynamic types which can be indiviually
+	  registered during a static build.
+	  More details here:
+	  https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
+	  https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>
+
+2021-02-12 08:57:55 +0100  Stéphane Cerveau <scerveau@collabora.com>
+
+	* ext/gdk_pixbuf/gstgdkpixbufdec.c:
+	* ext/gdk_pixbuf/gstgdkpixbufelement.c:
+	* ext/gdk_pixbuf/gstgdkpixbufelements.h:
+	* ext/gdk_pixbuf/gstgdkpixbufoverlay.c:
+	* ext/gdk_pixbuf/gstgdkpixbufplugin.c:
+	* ext/gdk_pixbuf/gstgdkpixbufsink.c:
+	* ext/gdk_pixbuf/meson.build:
+	  gdk_pixbuf: allow per feature registration
+	  Split plugin into features including
+	  dynamic types which can be indiviually
+	  registered during a static build.
+	  More details here:
+	  https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
+	  https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>
+
+2021-02-12 08:48:21 +0100  Stéphane Cerveau <scerveau@collabora.com>
+
+	* ext/gtk/gstgtkglsink.c:
+	* ext/gtk/gstgtkglsink.h:
+	* ext/gtk/gstgtksink.c:
+	* ext/gtk/gstgtksink.h:
+	* ext/gtk/gstplugin.c:
+	  gtk: allow per feature registration
+	  Split plugin into features including
+	  dynamic types which can be indiviually
+	  registered during a static build.
+	  More details here:
+	  https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
+	  https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>
+
+2021-02-11 19:53:30 +0100  Stéphane Cerveau <scerveau@collabora.com>
+
+	* ext/flac/gstflac.c:
+	* ext/flac/gstflacdec.c:
+	* ext/flac/gstflacelement.c:
+	* ext/flac/gstflacelements.h:
+	* ext/flac/gstflacenc.c:
+	* ext/flac/gstflactag.c:
+	* ext/flac/meson.build:
+	  flac: allow per feature registration
+	  Split plugin into features including
+	  dynamic types which can be indiviually
+	  registered during a static build.
+	  More details here:
+	  https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
+	  https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>
+
+2021-02-11 18:57:03 +0100  Stéphane Cerveau <scerveau@collabora.com>
+
+	* ext/dv/gstdv.c:
+	* ext/dv/gstdvdec.c:
+	* ext/dv/gstdvdemux.c:
+	* ext/dv/gstdvelement.c:
+	* ext/dv/gstdvelements.h:
+	* ext/dv/meson.build:
+	  dv: allow per feature registration
+	  Split plugin into features including
+	  dynamic types which can be indiviually
+	  registered during a static build.
+	  More details here:
+	  https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
+	  https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>
+
+2020-08-14 15:27:31 -0400  Julian Bouzas <julian.bouzas@collabora.com>
+
+	* ext/aalib/gstaaplugin.c:
+	* ext/aalib/gstaasink.c:
+	* ext/aalib/gstaasink.h:
+	* ext/aalib/gstaatv.c:
+	* ext/aalib/gstaatv.h:
+	* ext/aalib/meson.build:
+	  aalib: allow per feature registration
+	  Split plugin into features including
+	  dynamic types which can be indiviually
+	  registered during a static build.
+	  More details here:
+	  https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
+	  https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/876>
+
+2021-03-19 17:19:43 +0100  Víctor Manuel Jáquez Leal <vjaquez@igalia.com>
+
+	* docs/gst_plugins_cache.json:
+	* gst/videocrop/gstvideocrop.c:
+	* gst/videocrop/gstvideocrop.h:
+	* tests/check/elements/videocrop.c:
+	  videocrop: handle non raw caps features
+	  Currently, videocrop, only negotiates raw caps (system memory) because
+	  it's the type of memory it can modify. Nonetheless, it's also possible
+	  for the element to handle non-raw caps when only adding the crop meta
+	  is possible, in other words, when downstream buffer pools expose the
+	  crop API.
+	  This patch enable non-raw caps negotiation. If downstream doesn't
+	  expose crop API and negotiated caps are featured, the negotiation
+	  fails.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/915>
+
+2021-03-19 10:35:09 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/rtpmanager/gstrtpbin.c:
+	  rtpbin: Don't special-case G_SIGNAL_RUN_CLEANUP stage in signal accumulators
+	  All these signals don't run the class handler in the CLEANUP stage.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/913>
+
+2021-03-19 10:34:33 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* ext/shout2/gstshout2.c:
+	  shout2: Don't register signal without class handler with G_SIGNAL_RUN_CLEANUP
+	  There is no class handler to run during the CLEANUP stage.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/913>
+
+2021-03-23 16:59:28 +0800  Hou Qi <qi.hou@nxp.com>
+
+	* sys/v4l2/gstv4l2object.c:
+	  v4l2object: Avoid colorimetry mismatch for streams with invalid colorimetry
+	  video-info sets gst colorimetry to default value when colorimetry in caps
+	  is unparsable or invalid. Then v4l2object uses this gst colorimetry to do
+	  mapping with v4l2 colorimetry. This may cause colorimetry mismatch when
+	  check mapped gst colorimetry with that read from caps directly.
+	  To fix this, need to correct gst colorimetry as that parsed from video-info
+	  when check gst_v4l2_video_colorimetry_matches().
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/917>
+
+2021-03-19 10:52:26 +0800  Hou Qi <qi.hou@nxp.com>
+
+	* sys/v4l2/gstv4l2object.c:
+	  v4l2object: Add support for hdr10 stream playback
+	  Colorimetry of hdr10 video is bt2100-pq with transfer as
+	  GST_VIDEO_TRANSFER_SMPTE2084. So map GST_VIDEO_TRANSFER_SMPTE2084
+	  to V4L2_XFER_FUNC_SMPTE2084 to support hdr10 stream playback.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/888>
+
+2021-03-20 10:41:29 -0500  Sid Sethupathi <sid.sethupathi@gmail.com>
+
+	* gst/shapewipe/gstshapewipe.c:
+	  shapewipe: fix broken link in docs
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/916>
+
+2021-03-18 17:42:02 +0000  Alba Mendez <me@alba.sh>
+
+	* docs/gst_plugins_cache.json:
+	* gst/rtsp/gstrtspsrc.c:
+	  rtspsrc: Fix more signals
+	  Behaviour change in GLib causes select-stream signal to discard
+	  the value returned by handlers. See !909 for more info.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/912>
+
+2021-03-18 19:52:53 +1100  Matthew Waters <matthew@centricular.com>
+
+	* ext/jack/gstjack.c:
+	* ext/jack/gstjackaudiosink.c:
+	* ext/jack/gstjackaudiosrc.c:
+	* ext/pulse/pulsesink.h:
+	* ext/qt/gstqsgtexture.cc:
+	* ext/qt/gstqtglutility.cc:
+	* ext/qt/qtglrenderer.cc:
+	* ext/qt/qtitem.cc:
+	* ext/qt/qtwindow.cc:
+	* ext/vpx/gstvpxdec.c:
+	* ext/vpx/gstvpxenc.c:
+	* gst/audioparsers/gstac3parse.h:
+	* sys/rpicamsrc/gstrpicamsrc.c:
+	* sys/ximage/ximageutil.c:
+	  gst: don't use volatile to mean atomic
+	  volatile is not sufficient to provide atomic guarantees and real atomics
+	  should be used instead.  GCC 11 has started warning about using volatile
+	  with atomic operations.
+	  https://gitlab.gnome.org/GNOME/glib/-/merge_requests/1719
+	  Discovered in https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/868
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/911>
+
+2021-03-17 15:54:59 +0530  Nirbheek Chauhan <nirbheek@centricular.com>
+
+	* docs/gst_plugins_cache.json:
+	* gst/rtsp/gstrtspsrc.c:
+	  Update docs cache and fix before-send signal doc syntax
+	  The docs for before-send were missing because of this
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/909>
+
+2021-03-17 13:18:34 +0530  Nirbheek Chauhan <nirbheek@centricular.com>
+
+	* gst/rtsp/gstrtspsrc.c:
+	  rtspsrc: Fix accumulation of before-send signal return values
+	  Since glib 2.62, the accumulated return values in RUN_CLEANUP override the
+	  accumulated return values in RUN_FIRST. Since:
+	  1. We have a default handler that always returns TRUE, and
+	  2. User handlers are only run in RUN_FIRST, and
+	  3. Our accumulator just takes the latest return value
+	  We were discarding the return value from the user handler and always
+	  sending messages even if the user handler said not to. See
+	  https://gitlab.gnome.org/GNOME/glib/-/issues/2352 for more details.
+	  This signal does not need RUN_CLEANUP or RUN_FIRST, so just change it
+	  to RUN_LAST so that it's emitted exactly once and accumulated once.
+	  With this fix, this signal can now be used to intercept PAUSE when
+	  going to GST_STATE_NULL so that the server does a TEARDOWN (if
+	  necessary) and not a PAUSE, which will confuse other RTSP clients when
+	  playing shared media.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/909>
+
+2021-03-17 11:32:08 +0530  Nirbheek Chauhan <nirbheek@centricular.com>
+
+	* gst/rtsp/gstrtspsrc.c:
+	  Revert unusable workaround for PAUSE being sent when going NULL
+	  Directly setting rtspsrc to the NULL state before putting the pipeline
+	  in the NULL state usually works, but it can cause a deadlock in some
+	  cases, so it's not a reliable mechanism to fix this.
+	  This reverts commit f37afdafff1fd0a339966116261f5cd0de53f5d1:
+	  "rtspsrc: Fix state changes from PAUSED to PLAYING"
+	  and commit 76d624b2df5594a82269b94dffe8766a372d059d:
+	  "rtspsrc: Do not send PAUSE command when going to GST_STATE_NULL"
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/908>
+
+2021-03-16 19:25:36 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/rtpmanager/gstrtpjitterbuffer.c:
+	  rtpjitterbuffer: Fix parsing of the mediaclk:direct= field
+	  Due to an off-by-one when parsing the string, the most significant digit
+	  or the clock offset was skipped when parsing the offset.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/907>
+
+2021-03-16 00:08:43 +0530  Nirbheek Chauhan <nirbheek@centricular.com>
+
+	* gst/rtsp/gstrtspsrc.c:
+	  rtspsrc: Fix state changes from PAUSED to PLAYING
+	  This was accidentally broken in the last commit that touched this
+	  because I missed the fall-through in the case immediately above this.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/906>
+
+2021-03-04 13:05:19 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/matroska/matroska-demux.c:
+	* gst/matroska/matroska-ids.h:
+	  matroskademux: Fix extraction of multichannel WavPack
+	  The old code had a couple of issues that all lead to potential memory
+	  safety bugs.
+	  - Use a constant for the Wavpack4Header size instead of using sizeof.
+	  It's written out into the data and not from the struct and who knows
+	  what special alignment/padding requirements some C compilers have.
+	  - gst_buffer_set_size() does not realloc the buffer when setting a
+	  bigger size than allocated, it only allows growing up to the maximum
+	  allocated size. Instead use a GstAdapter to collect all the blocks
+	  and take out everything at once in the end.
+	  - Check that enough data is actually available in the input and
+	  otherwise handle it an error in all cases instead of silently
+	  ignoring it.
+	  Among other things this fixes out of bounds writes because the code
+	  assumed gst_buffer_set_size() can grow the buffer and simply wrote after
+	  the end of the buffer.
+	  Thanks to Natalie Silvanovich for reporting.
+	  Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/859
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/902>
+
+2021-03-03 11:31:52 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/matroska/matroska-demux.c:
+	  matroskademux: Initialize track context out parameter to NULL before parsing
+	  Various error return paths don't set it to NULL and callers are only
+	  checking if the pointer is NULL. As it's allocated on the stack this
+	  usually contains random stack memory, and more often than not the memory
+	  of a previously parsed track.
+	  This then causes all kinds of memory corruptions further down the line.
+	  Thanks to Natalie Silvanovich for reporting.
+	  Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/858
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/902>
+
+2021-03-15 12:57:19 +0530  Nirbheek Chauhan <nirbheek@centricular.com>
+
+	* gst/rtsp/gstrtspsrc.c:
+	  rtspsrc: Do not send PAUSE command when going to GST_STATE_NULL
+	  This usually doesn't matter, but it is disruptive when streaming from
+	  a shared media since it will pause all other clients when any client
+	  exits.
+	  This new behaviour is opt-in and should be safe because you need to
+	  set the NULL state on rtspsrc directly, instead of just on the
+	  pipeline. See the updated documentation for an explanation.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/901>
+
+2021-01-18 15:54:43 +0100  Philipp Zabel <p.zabel@pengutronix.de>
+
+	* sys/v4l2/gstv4l2object.c:
+	  v4l2object: handle GST_VIDEO_TRANSFER_BT601
+	  V4L2 makes no difference between the BT.601 and BT.709 transfer
+	  functions [1], but GStreamer does since 1.18 [2].
+	  Adapt gst_v4l2_object_get_colorspace() and
+	  gst_v4l2_object_set_format_full().
+	  [1] https://linuxtv.org/downloads/v4l-dvb-apis-new/userspace-api/v4l/colorspaces-details.html#colorspace-smpte-170m-v4l2-colorspace-smpte170m
+	  [2] https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/724
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/856>
+
+2021-03-11 22:22:15 +0100  Mathieu Duponchelle <mathieu@centricular.com>
+
+	* gst/rtsp/gstrtspsrc.c:
+	  rtspsrc: fix title of a few properties docstrings
+	  GstRtspSrc -> GstRTSPSrc
+	  This would have been noticed by the since checker, but those
+	  properties were introduced prior to that.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/899>
+
+2021-03-07 21:25:01 +0000  Vladimir Menshakov <vladimir.menshakov@gmail.com>
+
+	* docs/gst_plugins_cache.json:
+	* ext/wavpack/gstwavpackdec.c:
+	* ext/wavpack/gstwavpackdec.h:
+	  wavpackdec: Add floating point format support
+	  This commit negotiate F32 audio format if MODE_FLOAT used in wavpack file.
+	  Wavpack float mode is always in 32-bit IEEE format.
+	  The following pipeline plays distorted audio if source file is encoded in float mode:
+	  gst-launch-1.0 filesrc ... ! wavpackparse ! wavpackdec ! pulsesink
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/894>
+
+2021-03-04 16:40:06 +1100  Matthew Waters <matthew@centricular.com>
+
+	* gst/matroska/matroska-demux.c:
+	  matroska: also support push-mode from seek events sent to the element
+	  Otherwise sending seek events would fail to actually seek.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/891>
+
+2021-02-26 10:49:10 +0100  Marc Leeman <m.leeman@televic.com>
+
+	* gst/rtsp/gstrtspsrc.c:
+	  gstrtspsrc: 551 should not result in an unhandled error
+	  Some cameras (e.g. HikVision DS-2CD2732F-IS) return "551 Option
+	  not supported" when a command is sent that is not implemented
+	  (e.g. PAUSE). Instead; it should return "501 Not Implemented".
+	  This is wrong, as previously, the camera did announce support for PAUSE
+	  in the OPTIONS.
+	  In this case, handle the 551 as if it was 501 to avoid throwing errors
+	  to application level. */
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/885>
+
+2021-03-01 14:32:40 +0800  Hou Qi <qi.hou@nxp.com>
+
+	* sys/v4l2/gstv4l2videodec.c:
+	  v4l2videodec: Do not expose profiles/levels in vp8/vp9 template caps
+	  Vp8/vp9 supported profiles/levels are listed in decoder sink caps, but
+	  there is no parser for these two formats and the demuxers also don't have
+	  these information. It causes negotiation fail between demuxers and decoder
+	  when check caps "accept = gst_caps_is_subset (caps, template_caps);".
+	  To fix this, need to remove profiles/levels for vp8/vp9 formats in decoder
+	  sink caps.
+	  Fix #854
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/887>
+
+2021-03-03 18:30:39 +0900  Seungha Yang <seungha@centricular.com>
+
+	* gst/rtpmanager/gstrtphdrext-twcc.h:
+	  rtpmanager: Fix an MSVC compile warning
+	  We don't expect this object is a part of public library.
+	  gstrtphdrext-twcc.c(45): warning C4273: 'gst_rtp_header_extension_twcc_get_type': inconsistent dll linkage
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/889>
+
+2021-02-24 13:25:43 +0100  Philipp Zabel <p.zabel@pengutronix.de>
+
+	* sys/v4l2/gstv4l2videodec.c:
+	  v4l2videodec: fix src side frame rate negotiation
+	  Negotiating v4l2h264dec ! v4l2h264enc transcoding pipelines fails in
+	  case the encoder does not accept framerate=(fraction)0/1.
+	  The acquired caps used for downstream negotiation are determined from
+	  gst_v4l2_object_acquire_format(), which sets the GstVideoInfo::fps_n
+	  and ::fps_d fields to 0.
+	  To fix this, copy the frame rate from the sink side.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/882>
+
+2021-02-16 16:20:05 +0200  Jordan Petridis <jpetridis@gnome.org>
+
+	* sys/rpicamsrc/meson.build:
+	  rpicamsrc: depend on posix threads and vchiq_arm
+	  Could only test on rpi 3b+
+	  Close #839
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/875>
+
+2021-02-11 14:48:07 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+	* sys/v4l2/gstv4l2bufferpool.c:
+	  v4l2bufferpool: Silence traces around unsupported source change
+	  Don't be too spamy about unsupported source change flags as these will be
+	  commonly extended in the future.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/870>
+
+2021-02-11 14:24:29 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+	* sys/v4l2/gstv4l2src.c:
+	  v4l2src: Move preferred resolution query before the probe
+	  As we lock the DV_TIMINGS (and standards in the future), we need to probe the
+	  caps after, otherwise, we may endup fixating to an unsupported resolution,
+	  which would lead to a not-negotiated error.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/870>
+
+2021-02-10 16:37:01 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+	* sys/v4l2/gstv4l2src.c:
+	* sys/v4l2/v4l2_calls.c:
+	  v4l2src: Calculate framerate from DV timings
+	  And use this framerate in our preference. Note that we also flush
+	  the probed caps as it seems that the format enumeration may change
+	  when a new source change event get triggered.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/870>
+
+2021-02-10 15:52:55 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+	* sys/v4l2/gstv4l2bufferpool.h:
+	* sys/v4l2/gstv4l2object.h:
+	* sys/v4l2/gstv4l2src.c:
+	* sys/v4l2/v4l2_calls.c:
+	  v4l2rc: Add DV_TIMINGS query and locking
+	  This adds support to DV_TIMINGS query and locking. The timing width and
+	  height is then used as a preference.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/870>
+
+2021-02-10 15:49:03 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+	* sys/v4l2/gstv4l2src.c:
+	  v4l2src: Force renegotiation on resolution change
+	  As mandated by the specification, make sure to cycle through streamoff
+	  / streamon regardless if the caps have changed or not.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/870>
+
+2021-02-10 14:52:14 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+	* sys/v4l2/gstv4l2object.h:
+	  v4l2object: Remove unused streaming member
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/870>
+
+2021-02-10 10:48:48 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+	* sys/v4l2/gstv4l2src.c:
+	  v4l2src: Refactor to use PreferredCapsInfo structure
+	  Avoid passing around a bare structure for the preference, this removes
+	  the need to copy and free that structure and simplify the code. Also
+	  fix a type in the structure name, Prefered -> Preferred.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/870>
+
+2021-02-08 17:27:20 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+	* sys/v4l2/gstv4l2src.c:
+	  v4l2src: Stub preferred resolution support
+	  This stubs the ability to use preferred resolution from digital
+	  video timings, analog TV standards or driver reported native
+	  resolution.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/870>
+
+2021-02-09 14:44:02 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+	* sys/v4l2/gstv4l2bufferpool.c:
+	* sys/v4l2/gstv4l2object.h:
+	* sys/v4l2/v4l2_calls.c:
+	  v4l2: Subscribe source_change for the current input
+	  When we subscribe for source-change event, we need to specify for which
+	  input. Make sure we subscribe for the current input.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/870>
+
+2021-02-08 17:26:20 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+	* sys/v4l2/gstv4l2src.c:
+	* sys/v4l2/gstv4l2src.h:
+	  v4l2src: Add input signal status detection
+	  As part of the support to select a preferred size, we can also
+	  detect the signal status. This is a split patch so that feature
+	  is separated to ease review.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/870>
+
+2021-02-08 17:24:00 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+	* sys/v4l2/gstv4l2object.h:
+	* sys/v4l2/v4l2_calls.c:
+	  v4l2: Add helper to query input status
+	  This is a wrapper around ENUM_INPUT renamed for readability.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/870>
+
+2021-02-08 17:22:37 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+	* sys/v4l2/gstv4l2object.h:
+	* sys/v4l2/gstv4l2radio.c:
+	* sys/v4l2/gstv4l2tuner.c:
+	* sys/v4l2/v4l2_calls.c:
+	  v4l2: Fix input/output index sign
+	  This is an unsigned integer in the kernel API.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/870>
+
+2021-02-04 16:59:44 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+	* sys/v4l2/gstv4l2src.c:
+	  v4l2src: Add source resolution change support
+	  This patch adds support for source resolution change detection.
+	  Resolution change is signaled by drivers when a change in the detected
+	  signal have been detected. This is notably seen on HDMI receivers.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/870>
+
+2021-02-04 14:13:32 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+	* sys/v4l2/gstv4l2bufferpool.c:
+	* sys/v4l2/gstv4l2bufferpool.h:
+	  v4l2bufferpool: Handle resolution change event
+	  This patch adds the detection, dequeuing and reporting of the SOURCE_CHANGE
+	  event when the CH_RESOLUTION flag is set. The acquire function will now return
+	  a new custom success called GST_V4L2_FLOW_RESOLUTION_CHANGE. In order to use
+	  this new feature, elements must enable it by calling:
+	  gst_v4l2_buffer_pool_enable_resolution_change (pool);
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/870>
+
+2021-02-04 11:01:38 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+	* sys/v4l2/gstv4l2object.h:
+	* sys/v4l2/v4l2_calls.c:
+	  v4l2object: Add event helpers
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/870>
+
+2021-02-04 10:10:34 -0500  Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+	* sys/v4l2/gstv4l2bufferpool.c:
+	  v4l2bufferpool: use FLOW_LAST_BUFFER
+	  This uses the GST_V4L2_FLOW_LAST_BUFFER alias instead of
+	  GST_FLOW_CUSTOM_SUCCESS to make the code more readable.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/870>
+
+2018-12-10 14:10:05 +0100  Lucas Stach <l.stach@pengutronix.de>
+
+	* sys/v4l2/gstv4l2object.c:
+	  v4l2object: prefer NV12 over I420
+	  Considering NV12 an 'odd' format is a historical artifact. This format
+	  is now quite common, and usually preferable to I420 due to more memory
+	  friendly access patterns.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/857>
+
+2021-02-18 10:34:25 +0100  Guillaume Desmottes <guillaume.desmottes@collabora.com>
+
+	* gst/wavparse/gstwavparse.c:
+	* tests/check/elements/wavparse.c:
+	  wavparse: fix seeking in READY state
+	  wavparse claims to be able to support seeking in the READY state by
+	  saving the pending seek event and actually seeking later after having parsed the
+	  header.
+	  Problem was that this seek event was reset on the READY to PAUSED
+	  transition, making all this code useless. Fixing it by stop resetting
+	  on READY to PAUSED transition as we already reset on PAUSED to READY
+	  and when initiating the element.
+	  Note that DTS marker detection isn't support in such scenario as
+	  gst_type_find_helper_for_buffer() needs a buffer containing the
+	  beginning of the stream.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/879>
+
+2021-02-18 10:05:03 +0100  Guillaume Desmottes <guillaume.desmottes@collabora.com>
+
+	* tests/check/elements/wavparse.c:
+	  tests: wavparse: factor out create_pipeline()
+	  No semantic change.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/879>
+
+2021-02-18 00:34:02 +0100  Mathieu Duponchelle <mathieu@centricular.com>
+
+	* docs/gst_plugins_cache.json:
+	  docs: update plugins cache with new h264 / vp8 depay properties
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/834>
+
+2020-12-09 01:40:45 +0100  Mathieu Duponchelle <mathieu@centricular.com>
+
+	* gst/rtp/gstrtph264depay.c:
+	* gst/rtp/gstrtph264depay.h:
+	  rtph264depay: expose request-keyframe property
+	  When set, the depayloader will request new keyframes on packet
+	  loss
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/834>
+
+2020-12-09 01:34:20 +0100  Mathieu Duponchelle <mathieu@centricular.com>
+
+	* gst/rtp/gstrtpvp8depay.c:
+	* gst/rtp/gstrtpvp8depay.h:
+	  rtpvp8depay: expose request-keyframe property
+	  When set, the depayloader will request new keyframes on packet
+	  loss
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/834>
+
+2020-12-09 01:24:57 +0100  Mathieu Duponchelle <mathieu@centricular.com>
+
+	* gst/rtp/gstrtph264depay.c:
+	* gst/rtp/gstrtph264depay.h:
+	  rtph264depay: expose wait-for-keyframe property
+	  Similar to rtpvp8depay, when packet loss occurs, the depayloader
+	  starts waiting for a keyframe.
+	  We try to only stop waiting when all the packets for the new keyframe
+	  have been received, by only resetting waiting_for_keyframe when
+	  encountering the first packet of a keyframe, this is slightly
+	  fragile because there is no bit that explicitly marks the start
+	  of an access unit, so we rely on the existing picture_start
+	  detection code.
+	  As a consequence, the property is only meaningful when outputting
+	  access units, and is ignored when outputting NALs directly.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/834>
+
+2021-02-18 00:36:43 +0100  Mathieu Duponchelle <mathieu@centricular.com>
+
+	* docs/gst_plugins_cache.json:
+	* gst/videomixer/videomixer2.c:
+	  videomixer: document as deprecated
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/878>
+
+2021-02-16 22:20:17 +1100  Ashley Brighthope <ashley.b@reddegrees.com>
+
+	* gst/wavenc/gstwavenc.c:
+	  wavenc: Fixed INFO chunk corruption, caused by odd sized data not being padded. Code style was updated.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/873>
+
+2020-12-07 19:51:35 +0100  Jakub Adam <jakub.adam@collabora.com>
+
+	* gst/rtp/gstrtpopuspay.c:
+	  rtpopuspay: add info regarding (non-standard) multichannel support
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/832>
+
+2020-12-07 16:50:01 +0100  Jakub Adam <jakub.adam@collabora.com>
+
+	* docs/gst_plugins_cache.json:
+	  docs: update plugins cache for rtpopus
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/832>
+
+2020-12-01 20:09:58 +0100  Jakub Adam <jakub.adam@collabora.com>
+
+	* tests/check/elements/rtpopus.c:
+	  tests: add rtpopus multichannel test cases
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/832>
+
+2020-12-01 16:43:32 +0100  Jakub Adam <jakub.adam@collabora.com>
+
+	* gst/rtp/gstrtpopusdepay.c:
+	  rtpopusdepay: support libwebrtc-compatible multichannel payload
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/832>
+
+2020-11-30 21:49:48 +0100  Jakub Adam <jakub.adam@collabora.com>
+
+	* gst/rtp/gstrtpopuspay.c:
+	  rtpopuspay: support libwebrtc-compatible multichannel payload
+	  When the audio has more than 2 channels, add optional fields to output
+	  caps from which webrtcbin can generate SDP in the syntax recognized by
+	  "multiopus" codec present in libwebrtc [1].
+	  e.g. for 5.1 audio:
+	  a=rtpmap:96 multiopus/48000/6
+	  a=fmtp:96 num_streams=4;coupled_streams=2;channel_mapping=0,4,1,2,3,5
+	  [1] https://webrtc-review.googlesource.com/c/src/+/129768
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/832>
+
+2020-11-30 22:10:14 +0100  Jakub Adam <jakub.adam@collabora.com>
+
+	* gst/rtp/gstrtpopuspay.c:
+	  rtpopuspay: make use of gst_rtp_base_payload_set_outcaps_structure()
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/832>
+
+2021-02-09 19:31:28 -0500  Olivier Crête <olivier.crete@collabora.com>
+
+	* gst/effectv/LICENSE:
+	  effectv: Remove redundant license file
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/869>
+
+2021-02-05 00:55:12 +0000  Kevin Song <kevinbing.song@gmail.com>
+
+	* sys/v4l2/gstv4l2videoenc.c:
+	  Apply 1 suggestion(s) to 1 file(s)
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/868>
+
+2021-02-05 00:55:04 +0000  Kevin Song <kevinbing.song@gmail.com>
+
+	* sys/v4l2/gstv4l2videoenc.c:
+	  Apply 1 suggestion(s) to 1 file(s)
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/868>
+
+2021-02-04 13:43:17 +0800  Bing Song <bing.song@nxp.com>
+
+	* sys/v4l2/gstv4l2videoenc.c:
+	  v4l2videoenc: support resolution change stream encode.
+	  Resolution change stream transcoding will drain before send new video
+	  frame buffer. Need encode video frame after process EOS.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/868>
+
+2021-02-04 11:44:53 +0100  Xabier Rodriguez Calvar <calvaris@igalia.com>
+
+	* gst/isomp4/fourcc.h:
+	* gst/isomp4/qtdemux.c:
+	  qtdemux: added support for cbcs encryption scheme
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/865>
+
+2021-01-21 18:04:58 +0100  Guillaume Desmottes <guillaume.desmottes@collabora.com>
+
+	* docs/gst_plugins_cache.json:
+	* gst/rtpmanager/gstrtphdrext-rfc6464.c:
+	* gst/rtpmanager/gstrtphdrext-rfc6464.h:
+	* gst/rtpmanager/gstrtpmanager.c:
+	* gst/rtpmanager/meson.build:
+	* tests/check/elements/rtphdrextrfc6464.c:
+	* tests/check/meson.build:
+	  rtp: add rtphdrextrfc6464
+	  Header Extension for Client-to-Mixer Audio Level Indication as
+	  defined in RFC 6464.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/630>
+
+2020-06-16 12:01:30 +0200  Guillaume Desmottes <guillaume.desmottes@collabora.com>
+
+	* docs/gst_plugins_cache.json:
+	* gst/level/gstlevel.c:
+	* gst/level/gstlevel.h:
+	* tests/check/elements/level.c:
+	  level: add GstRTPAudioLevelMeta on buffers
+	  This meta can be used by a RTP payloader to send the level information
+	  to the peer.
+	  Part of https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/issues/446
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/630>
+
+2021-02-03 17:10:20 +0200  Robert Swain <robert.swain@gmail.com>
+
+	* gst/deinterlace/gstdeinterlace.c:
+	  deinterlace: Provide documentation for GST_DEINTERLACE_BUFFER_STATE
+	  More information available in
+	  https://gstconf.ubicast.tv/videos/interlacing-and-telecine-in-gstreamer/
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/866>
+
+2021-01-30 16:16:13 +0200  Vivia Nikolaidou <vivia@ahiru.eu>
+
+	* gst/deinterlace/gstdeinterlacemethod.c:
+	  deinterlace: Fix telecine/onefield mixup
+	  https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/838
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/866>
+
+2021-01-30 15:49:23 +0200  Vivia Nikolaidou <vivia@ahiru.eu>
+
+	* gst/deinterlace/gstdeinterlace.c:
+	* gst/deinterlace/gstdeinterlacemethod.c:
+	  deinterlace: Better alternate support
+	  Improve line offset halving based on whether this field is top or
+	  bottom.
+	  Also handle the buffer state the same as mixed.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/866>
+
+2021-01-14 01:12:06 +0800  Bing Song <bing.song@nxp.com>
+
+	* sys/v4l2/gstv4l2h265codec.c:
+	  v4l2h265codec: fix HEVC profile string issue.
+	  Keep HEVC profile compatible with other module.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/850>
+
+2020-12-15 10:41:40 +0800  Bing Song <bing.song@nxp.com>
+
+	* sys/v4l2/gstv4l2object.c:
+	* sys/v4l2/gstv4l2object.h:
+	  v4l2object: Need keep same transfer as input caps.
+	  GST_VIDEO_TRANSFER_BT2020_12 and GST_VIDEO_TRANSFER_BT2020_10 will
+	  be mapped to V4L2_XFER_FUNC_709. Need check input caps when map
+	  V4L2_XFER_FUNC_709 back to GST_VIDEO_TRANSFER_BT2020_12 and
+	  GST_VIDEO_TRANSFER_BT2020_10
+	  Fixes: https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/816
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/841>
+
+2020-12-07 10:01:53 +0100  Tobias Ronge <tobiasr@axis.com>
+
+	* gst/rtsp/gstrtspsrc.c:
+	  rtspsrc: Do not wait for response while flushing
+	  Due to the may_cancel flag in GstRTSPConnection, receiving might not get
+	  cancelled when supposed to. In this case, gst_rtsp_src_receive_response
+	  will have to wait until timeout instead but if busy receiving RTP
+	  data, this timeout will never occur.
+	  With this patch, gst_rtsp_src_receive_response returns GST_RTSP_EINTR
+	  if flushing is set to TRUE instead of continuing to receive.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/831>
+
+2021-01-14 19:13:03 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* ext/dv/meson.build:
+	  meson: allow libdv subproject fallback
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/854>
+
+2020-12-21 13:55:58 +0100  Xabier Rodriguez Calvar <calvaris@igalia.com>
+
+	* gst/isomp4/qtdemux.c:
+	  qtdemux: Allow streams with no specified protection system ID
+	  This is necessary in cases like CMAF where there won't be any events
+	  passing thru.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/852>
+
+2021-01-07 16:57:27 +0800  Hou Qi <qi.hou@nxp.com>
+
+	* docs/gst_plugins_cache.json:
+	* sys/v4l2/gstv4l2object.c:
+	  v4l2object: Map correct video format for RGBA
+	  Map V4L2_PIX_FMT_RGBA32 pixel format to GST_VIDEO_FORMAT_RGBA instead of
+	  GST_VIDEO_FORMAT_RGB video format to support RGBA.
+	  Fixes #823
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/848>
+
+2021-01-02 13:06:16 +0530  Sanchayan Maity <sanchayan@asymptotic.io>
+
+	* gst/udp/gstudpsrc.c:
+	  udpsrc: Fix marker links
+	  These should be with a single ':'. The double '::' results in a CI with
+	  build failure message like below.
+	  ERROR: [links]: (mandatory-link-not-found): Mandatory link Link GstSocketTimestamp -> None (GstSocketTimestamp) could not be resolved
+	  ERROR: [check-missing-since-markers]: (missing-since-marker): Missing since marker for udpsrc:socket-timestamp
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/828>
+
+2020-12-17 11:24:07 +0530  Sanchayan Maity <sanchayan@asymptotic.io>
+
+	* docs/gst_plugins_cache.json:
+	* gst/udp/gstudpsrc.c:
+	* gst/udp/gstudpsrc.h:
+	  udpsrc: Allow use of socket control message timestamps for DTS
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/828>
+
+2020-12-09 20:20:18 +1100  Matthew Waters <matthew@centricular.com>
+
+	* docs/gst_plugins_cache.json:
+	* gst/videofilter/gstvideoflip.c:
+	* gst/videofilter/gstvideoflip.h:
+	* tests/check/elements/videoflip.c:
+	  videoflip: fix possible crash when setting the video-direction while running
+	  A classic case of not enough locking.
+	  One interesting thing with this is the interaction between the
+	  rotation value and caps negotiation.  i.e. the width/height of the caps
+	  can be swapped depending on the video-direction property.  We can't lock
+	  the entirety of the caps negotiation for obvious reasons so we need to
+	  do something else.  This takes the approach of trying to use a single
+	  rotation value throughout the entirety of the negotiation and then
+	  subsequent output frame in a kind of latching sequence.
+	  Fixes: https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/792
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/836>
+
+2020-12-09 19:49:47 +1100  Matthew Waters <matthew@centricular.com>
+
+	* tests/check/elements/videoflip.c:
+	* tests/check/meson.build:
+	  tests: add tests for videoflip
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/836>
+
+2020-12-30 13:38:46 +0100  Ignacio Casal Quinteiro <qignacio@amazon.com>
+
+	* gst/deinterlace/meson.build:
+	  deinterlace: force -DPREFIX on macos
+	  This is due to a bug in meson where it will not detect properly
+	  the compiler if the symbols need an undercore.
+	  https://github.com/mesonbuild/meson/issues/5482
+	  Fixes #821
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/845>
+
+2020-12-15 11:36:27 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* docs/gst_plugins_cache.json:
+	* gst/rtsp/gstrtspsrc.c:
+	  rtspsrc: Use proper types instead of G_TYPE_POINTER for the RTSP messages in the "handle-request" signal
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/842>
+
+2020-12-10 14:27:49 +0200  Vivia Nikolaidou <vivia@ahiru.eu>
+
+	* gst/multifile/gstsplitmuxsink.c:
+	* gst/multifile/gstsplitmuxsink.h:
+	  splitmuxsink: Avoid deadlock when releasing a pad from a running muxer
+	  Might not drain correctly
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/838>
+
+2020-12-11 11:24:14 +0800  Hou Qi <qi.hou@nxp.com>
+
+	* sys/v4l2/gstv4l2object.c:
+	  v4l2object: Use active resolution during fallback colorspace probe
+	  For legacy drivers that don't implement ENUM_FRAMESIZE, use active
+	  resolution to probe colorspace. This can improve the accuracy of the
+	  result when the colorspace depends on the resolution. This fixes a
+	  wrong colorspace issue on board with vendor bsp at resolution 2560x1440.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/830>
+
+2020-12-12 04:02:37 +0100  Mathieu Duponchelle <mathieu@centricular.com>
+
+	* gst/rtpmanager/gstrtpst2022-1-fecdec.c:
+	  rtpst2022-1-fecdec: don't xor out of bounds
+	  When reconstituting packets from a stream with variable packet
+	  sizes, don't xor larger packets past the length of the protected
+	  packet
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/839>
+
+2020-12-12 04:00:41 +0100  Mathieu Duponchelle <mathieu@centricular.com>
+
+	* gst/rtpmanager/gstrtpst2022-1-fecenc.c:
+	  rtpst2022-1-fecenc: memset when reallocating xored payload
+	  When protecting packets with a variable payload length, we
+	  reallocate the xored payload when needed. It is a good idea
+	  to memset the extended memory to 0 so that we don't xor
+	  data with garbage!
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/839>
+
+2020-12-12 03:56:11 +0100  Mathieu Duponchelle <mathieu@centricular.com>
+
+	* gst/rtpmanager/gstrtpst2022-1-fecdec.c:
+	* gst/rtpmanager/gstrtpst2022-1-fecenc.c:
+	  rtpst2022-1-fec-*: protect additional RTP header fields
+	  While the standard is a bit vague about whether the padding,
+	  extension and marker bits should be protected:
+	  > The usage, by senders and receivers, of the following bits shall
+	  > be defined by the associated video/audio transport standards:
+	  It is obviously necessary and useful for some formats (eg VP8)
+	  that those indeed be protected.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/839>
+
+2020-12-12 03:28:56 +1100  Jan Schmidt <jan@centricular.com>
+
+	* tests/check/elements/splitmuxsink.c:
+	  splitmuxsink: Unit test - check format/opened/closed sequence
+	  Check the sequence of format-location/fragment-opened/fragment-closed
+	  events is respected. There should be 1 format-location call for each
+	  fragment-opened message, and 1 fragment-closed for each.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/833>
+
+2020-12-09 00:40:52 +1100  Jan Schmidt <jan@centricular.com>
+
+	* gst/multifile/gstsplitmuxsink.c:
+	* gst/multifile/gstsplitmuxsink.h:
+	  splitmuxsink: Fix for 'reference bytes muxed' check.
+	  https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/798
+	  introduced a check in the need-new-fragment logic to avoid starting a
+	  new fragment unless there has been some data on the reference stream,
+	  but the check is done against the number of bytes that have been
+	  received on the input, not the number that were released for output
+	  into the current fragment.
+	  Fix the check to remember and test against bytes that have been sent
+	  for output.
+	  This also fixes a problem where starting a new fragment fails to
+	  request a new filename from the format-location signal.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/833>
+
+2020-09-15 00:27:24 +1000  Jan Schmidt <jan@centricular.com>
+
+	* gst/multifile/gstsplitmuxsink.c:
+	  splitmuxsink: Add debug for fragment opened/closed msgs
+	  When posting fragment-opened and fragment-closed messages,
+	  put a debug statement in the logs
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/833>
+
+2020-08-18 16:06:14 +1000  Jan Schmidt <jan@centricular.com>
+
+	* gst/multifile/gstsplitmuxsink.c:
+	  splitmuxsink: Convert asserts into element errors.
+	  Change some g_assert into element errors so that they can be
+	  caught and the pipeline shut down.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/833>
+
+2020-07-10 15:36:54 +1000  Matthew Waters <matthew@centricular.com>
+
+	* docs/gst_plugins_cache.json:
+	* gst/rtpmanager/gstrtpfunnel.c:
+	* gst/rtpmanager/gstrtphdrext-twcc.c:
+	* gst/rtpmanager/gstrtphdrext-twcc.h:
+	* gst/rtpmanager/gstrtpmanager.c:
+	* gst/rtpmanager/meson.build:
+	  rtpmanager: update for rtp header extensions
+	  Provide an implementation of the transport-wide-cc header extension and
+	  use it in rtpfunnel.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/808>
+
+2020-11-15 11:30:07 +0000  Jose Quaresma <quaresma.jose@gmail.com>
+
+	* sys/rpicamsrc/meson.build:
+	  rpicamsrc: add vchostif library as it is required to build successful
+	  fix: undefined reference to `vc_gencmd'
+	  /usr/src/debug/gstreamer1.0-plugins-good/1.18.1-r0/build/../gst-plugins-good-1.18.1/sys/rpicamsrc/RaspiCamControl.c:1440: undefined reference to `vc_gencmd'
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/818>
+
+2020-11-25 17:51:24 +0100  Marijn Suijten <marijns95@gmail.com>
+
+	* tests/check/elements/rtp-payloading.c:
+	  tests/rtp-payloading: Use new AudioFormatInfo::fill_silence function
+	  The function is renamed to be properly associated with AudioFormatInfo
+	  (its instance) instead of AudioFormat (an unrelated enum), see [1] for
+	  the rename itself.
+	  [1]: https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/940
+
+2020-11-24 22:11:50 +0530  Nirbheek Chauhan <nirbheek@centricular.com>
+
+	* gst/deinterlace/meson.build:
+	* meson.build:
+	  deinterlace: Enable x86 assembly with nasm on MSVC
+	  We need to remove x86inc.asm from the list of compiled assembly files
+	  because it is not supposed to be compiled separately. It is directly
+	  included by yadif.asm, and it exports no symbols.
+	  The object file was getting ignored on all platforms except on msvc
+	  where it was causing a linker hang when building with debugging
+	  enabled because the object file had no debug symbols (or similar).
+	  We've seen this before in FFmpeg too, which uses nasm:
+	  https://gitlab.freedesktop.org/gstreamer/meson-ports/ffmpeg/-/merge_requests/46
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/825>
+
+2020-11-19 17:47:21 +1100  Matthew Waters <matthew@centricular.com>
+
+	* ext/qt/gstqtoverlay.cc:
+	* ext/qt/gstqtsink.cc:
+	  qml: add some docs on display and contexts
+	  Especially considering some dynamic pipeline scenarios.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/822>
+
+2020-11-18 20:09:24 +0100  Tim Schneider <tim.schneider94@t-online.de>
+
+	* sys/rpicamsrc/gstrpicamsrc.c:
+	  rpicamsrc: Added "src->started = FALSE;" to gst_rpi_cam_src_stop
+	  Makes the element reusable multiple times after a state change back to READY.
+	  Fixes #105
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/823>
+
+2020-11-12 09:32:30 +0800  Bing Song <bing.song@nxp.com>
+
+	* docs/gst_plugins_cache.json:
+	* sys/v4l2/gstv4l2object.c:
+	  v4l2: caps negotiate wrong as interlace feature
+	  gst_caps_simplify() will move interlace format before normal video
+	  format. It will cause caps negotiate prefer interlaced caps which
+	  isn't expected. Seperate normal caps and interlaced caps and then
+	  merge it will keep prefer progress video format.
+	  Add ARGB/BGRA for interlaced caps.
+	  Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/802
+	  Part-of <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/813>
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/813>
+
+2020-11-13 21:25:42 +0100  Havard Graff <havard.graff@gmail.com>
+
+	* gst/rtpmanager/rtpsession.c:
+	* tests/check/elements/rtpsession.c:
+	  rtpsession: never send on a non-internal source
+	  This will end up as a "received" packet, due to the code in
+	  source_push_rtp, which will think this is a packet being received.
+	  Instead drop the packet and hope that either:
+	  1. Something upstream responds to the GstRTPCollision event and changes
+	  SSRC used for sending.
+	  2. That the application responds to the "on-ssrc-collision" signal, and
+	  forces the sender (payloader) to change its SSRC.
+	  3. That the BYE sent to the existing user of this SSRC will respond to
+	  the BYE, and that we timeout this source, so we can continue sending
+	  using the chosen SSRC.
+	  The test reproduces a scenario where we previously would have sent
+	  on a non-internal source.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/817>
+
+2020-11-13 12:39:53 +0100  Havard Graff <havard.graff@gmail.com>
+
+	* gst/rtpmanager/rtpsource.c:
+	  rtpsource: rewrite timeout-check to avoid underflow
+	  If current_time is < collision_timeout, we get an uint64 underflow, and
+	  the check will trigger prematurely.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/817>
+
+2020-11-13 14:58:44 +0200  Vivia Nikolaidou <vivia@ahiru.eu>
+
+	* gst/audioparsers/gstaacparse.c:
+	  aacparse: Fix caps change handling
+	  In baseparse we set the fixed caps flag on all src pads, therefore the
+	  source pad caps query in get_allowed_caps will return the current caps.
+	  Current caps won't necessarily intersect with the new caps (e.g. sample
+	  rate change). Replace get_allowed_caps with peer_query_caps.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/816>
+
+2020-11-12 23:39:21 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* tests/check/elements/qtdemux.c:
+	  tests: qtdemux: fix typo in caps field
+	  timesacle -> timescale
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/815>
+
+2020-11-12 23:38:21 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* tests/check/elements/qtdemux.c:
+	  tests: qtdemux: fix crash on 32-bit architectures
+	  Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/803
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/815>
+
+2020-09-14 13:12:50 +0530  Sanchayan Maity <sanchayan@asymptotic.io>
+
+	* docs/gst_plugins_cache.json:
+	* gst/rtp/gstrtp.c:
+	* gst/rtp/gstrtpldacpay.c:
+	* gst/rtp/gstrtpldacpay.h:
+	* gst/rtp/meson.build:
+	  rtp: ldacpay: Add LDAC RTP payloader
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/757>
+
+2020-11-03 15:58:30 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* ext/qt/gstqsgtexture.cc:
+	* ext/qt/gstqsgtexture.h:
+	* ext/qt/qtitem.cc:
+	  qmlglsink: Keep old buffers around a bit longer if they were bound by QML
+	  We don't know exactly when QML will stop using them but it should be
+	  safe to unref them after at least 2 more buffers were bound.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/810>
+
+2020-11-10 18:18:12 +0000  ChrisDuncanAnyvision <chrisd@anyvision.co>
+
+	* gst/rtsp/gstrtspsrc.c:
+	* gst/rtsp/gstrtspsrc.h:
+	  rtspsrc: Ensure same group-id used for both TCP/UDP stream-start events
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/811>
+
+2020-11-10 16:17:23 +0000  ChrisDuncanAnyvision <chrisd@anyvision.co>
+
+	* gst/rtsp/gstrtspsrc.c:
+	  rtspsrc: Use consistent URI hashed stream-id for UDP and TCP/Interleaved streams
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/811>
+
+2020-11-04 18:43:04 +0530  Nirbheek Chauhan <nirbheek@centricular.com>
+
+	* meson.build:
+	  meson: Enable some MSVC warnings for parity with GCC/Clang
+	  This makes it easier to do development with MSVC by making it warn
+	  on common issues that GCC/Clang error out for in our CI configuration.
+	  Continuation from https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/223
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/809>
+
+2020-10-15 21:42:40 -0400  Olivier Crête <olivier.crete@collabora.com>
+
+	* docs/gst_plugins_cache.json:
+	* gst/rtpmanager/rtpsession.c:
+	* gst/rtpmanager/rtpsource.c:
+	* gst/rtpmanager/rtpsource.h:
+	* gst/rtpmanager/rtpstats.h:
+	  rtpsource: Report for which local SSRC is a remote RB reporting on
+	  This is useful in the Bundle case because there may be multiple local
+	  and remote SSRCs in the same session.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/776>
+
+2020-10-29 15:58:38 +0100  Guillaume Desmottes <guillaume.desmottes@collabora.com>
+
+	* docs/gst_plugins_cache.json:
+	* gst/rtp/gstrtpisacdepay.c:
+	* gst/rtp/gstrtpisacpay.c:
+	  docs: update plugins cache
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/530>
+
+2020-03-20 13:15:33 +0100  Guillaume Desmottes <guillaume.desmottes@collabora.com>
+
+	* gst/rtp/gstrtp.c:
+	* gst/rtp/gstrtpisacdepay.c:
+	* gst/rtp/gstrtpisacdepay.h:
+	* gst/rtp/meson.build:
+	  rtp: add rtpisacdepay
+	  Depayload for the iSAC audio codec.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/530>
+
+2020-03-20 13:15:33 +0100  Guillaume Desmottes <guillaume.desmottes@collabora.com>
+
+	* gst/rtp/gstrtp.c:
+	* gst/rtp/gstrtpisacpay.c:
+	* gst/rtp/gstrtpisacpay.h:
+	* gst/rtp/meson.build:
+	  rtp: add rtpisacpay
+	  Payload for the iSAC audio codec.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/530>
+
+2020-11-01 18:36:49 +0000  Dinesh Manajipet <saidinesh5@gmail.com>
+
+	* ext/qt/qtitem.cc:
+	  qmlglsink: Set qtitem's implicit width/height
+	  This can be useful to let the layouts automatically resize qtitem
+	  and also easily query a video's width/height from QML
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/802>
+
+2020-11-01 10:30:27 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/flv/gstflvmux.c:
+	  flvmux: Release pads via GstAggregator
+	  See https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/797
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/801>
+
+2020-10-26 12:40:49 +1100  Matthew Waters <matthew@centricular.com>
+
+	* gst/isomp4/atoms.c:
+	* gst/isomp4/atoms.h:
+	* gst/isomp4/gstqtmux.c:
+	  qtmux: support muxing multiple codec_data for h264/h265
+	  Each codec_data is put into its own SampleTableEntry inside the stsd.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/787>
+
+2020-10-29 14:54:16 +0100  Stéphane Cerveau <scerveau@collabora.com>
+
+	* docs/gst_plugins_cache.json:
+	* gst/debugutils/gstnavseek.c:
+	* gst/debugutils/gstnavseek.h:
+	  navseek: add hold_eos property
+	  This property will tell the element to hold
+	  the EOS event and keep it until the next
+	  keystroke.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/792>
+
+2020-10-31 12:52:04 +1100  Jan Schmidt <jan@centricular.com>
+
+	* tests/check/elements/splitmuxsrc.c:
+	  splitmuxsrc: Fix comment in a test
+	  Fix a comment in the splitmuxsrc robust muxing test so it
+	  describes the test properly.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/798>
+
+2020-10-31 12:49:08 +1100  Jan Schmidt <jan@centricular.com>
+
+	* gst/multifile/gstsplitmuxsink.c:
+	* gst/multifile/gstsplitmuxsink.h:
+	  splitmuxsink: Change EOS catching logic.
+	  Add a new state for ending the overall stream, and use it to decide
+	  whether to pass the final EOS message up the bus instead of dropping
+	  it. Fixes a small race that makes the testsuite sometimes not generate
+	  the last fragment(s) sometimes because the wrong EOS gets
+	  allowed through too early.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/798>
+
+2020-10-31 02:19:07 +1100  Jan Schmidt <jan@centricular.com>
+
+	* gst/multifile/gstsplitmuxsink.c:
+	* gst/multifile/gstsplitmuxsink.h:
+	  splitmuxsink: Don't use the element state lock
+	  Using the element state lock to avoid splitmuxsink shutting
+	  down while doing element manipulations can lead to a deadlock on
+	  shutdown if a fragment switch happens at exactly the wrong moment.
+	  Use a private mutex and a shutdown boolean instead.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/798>
+
+2020-10-30 03:38:15 +1100  Jan Schmidt <jan@centricular.com>
+
+	* gst/multifile/gstsplitmuxsink.c:
+	  splitmuxsink: Don't busy loop on a non-ready pad.
+	  If a pad gets into the check_completed_gop method and then
+	  the underlying conditions change on the reference context,
+	  things could get stuck in a busy loop when the context should
+	  instead jump back out and wait for more data.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/798>
+
+2020-10-30 03:36:51 +1100  Jan Schmidt <jan@centricular.com>
+
+	* gst/multifile/gstsplitmuxsrc.c:
+	  splitmuxsrc: Mark running=false on shutdown.
+	  Make sure that any late gst_element_call_async() callbacks
+	  know that the elements is shutting down and bail out instead
+	  of operating on the element we're trying to stop.
+	  Fixes a spurious test failure in elements_splitmuxsrc
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/798>
+
+2020-10-29 02:36:35 +1100  Jan Schmidt <jan@centricular.com>
+
+	* gst/multifile/gstsplitmuxsink.c:
+	  splitmuxsink: Forward EOS messages from async fragments.
+	  Re-enable forwarding EOS messages from fragments that are completing
+	  asynchronously, so that splitmuxsink itself won't go EOS until they
+	  are complete. This was disabled to work around a bug in core that
+	  is fixed in
+	  https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/683
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/798>
+
+2020-09-17 22:56:01 +1000  Jan Schmidt <jan@centricular.com>
+
+	* gst/multifile/gstsplitmuxsink.c:
+	* gst/multifile/gstsplitmuxsink.h:
+	  splitmuxsink: Never start a new fragment with no reference buffers
+	  If there has been no bytes from the reference stream muxed into
+	  the current fragment, then time can't have advanced, there's no
+	  GOP... this fragment would be broken or empty, so wait for some
+	  data on the reference buffer.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/798>
+
+2020-10-29 02:38:16 +1100  Jan Schmidt <jan@centricular.com>
+
+	* gst/isomp4/gstqtmux.c:
+	  qtmux: Chain up when releasing pad, and fix some locking.
+	  Release pads by calling up into aggregator so it can do the right
+	  things. Don't clean up the pad until after that.
+	  Add some missing locks around some accesses to shared pad state.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/797>
+
+2018-08-13 15:35:11 +0200  Stian Selnes <stian@pexip.com>
+
+	* gst/rtp/gstrtpvp9depay.c:
+	* gst/rtp/gstrtpvp9depay.h:
+	* tests/check/elements/rtpvp9.c:
+	  rtpvp9depay: Improve SVC parsing, aggregate all layers
+	  - Fix start and end of picture to support multiple layers. Start of
+	  picture is the first packet of the base layer, while end of picture
+	  is when the marker bit is set (last packet of the enhancement
+	  layers).
+	  - All "layers" (aka "frames") of a picture are pushed downstream in a
+	  single buffer when picture is complete.
+	  - Forgive SID=0 for enhancement layers (invalid, but Chrome and
+	  Firefox sends it)
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/773>
+
+2020-10-30 03:09:48 +0100  Stian Selnes <stian@pexip.com>
+
+	* gst/rtp/gstrtpvp8depay.c:
+	* gst/rtp/gstrtpvp8depay.h:
+	* tests/check/elements/rtpvp8.c:
+	  rtpvp8depay: Send lost events when marker bit is missing
+	  This means the previous frame was incomplete.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/796>
+
+2020-10-14 23:17:53 +0200  Knut Saastad <Knut@bitflow.io>
+
+	* gst/rtp/gstrtpvp8depay.c:
+	* gst/rtp/gstrtpvp8pay.c:
+	  rtpvp9depay: detect incomplete frames and bail out
+	  If a packet with the B bit set arrives but we haven't received
+	  a packet with the marker or E bits set to end the previous frame,
+	  we know the current frame was incomplete.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/795>
+
+2020-10-14 23:17:53 +0200  Knut Saastad <Knut@bitflow.io>
+
+	* gst/rtp/gstrtpvp9depay.c:
+	  rtpvp9depay: detect incomplete frames and bail out
+	  If a packet with the B bit set arrives but we haven't received
+	  a packet with the marker or E bits set to end the previous frame,
+	  we know the current frame was incomplete.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/769>
+
+2020-10-14 01:28:50 +0200  Mikhail Fludkov <misha@pexip.com>
+
+	* gst/rtp/gstrtpvp8depay.c:
+	* gst/rtp/gstrtpvp8depay.h:
+	* gst/rtp/gstrtpvp9depay.c:
+	* gst/rtp/gstrtpvp9depay.h:
+	* tests/check/elements/rtpvp8.c:
+	* tests/check/elements/rtpvp9.c:
+	  rtpvp*depay: possibly forward might-have-been-fec PacketLost events
+	  This is ad adaptation of a Pexip patch for dealing with spurious
+	  GstRTPPacketLost events caused by lost ulpfec packets: as FEC packets
+	  under that scheme are spliced in the same sequence domain as the media
+	  packets, it is not generally possible to determine whether a lost packet
+	  was a FEC packet or a media packet.
+	  When upstreaming pexip's ulpfec patches, we decided to drop all lost
+	  events at the base depayloader level, and where the original patch
+	  from pexip was making use of picture ids and marker bits to determine
+	  whether a packet should be forwarded, this patch makes use of those
+	  to determine whether they should be dropped instead (by removing their
+	  might-have-been-fec field).
+	  Spurious lost events coming out of the depayloader can cause the
+	  decoder to stop decoding until the next keyframe and / or request a new
+	  keyframe, and while this is not desirable it makes sense to forward
+	  that information when we have other means to determine whether a lost
+	  packet was indeed a FEC packet, as is the case with VP8 / VP9 payloads
+	  when they carry a picture id.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/769>
+
+2020-10-20 23:22:36 +1100  Jan Schmidt <jan@centricular.com>
+
+	* gst/rtp/gstrtph264depay.c:
+	  rtph264depay: Preserve SPS/PPS arrival order.
+	  Even if SPS/PPS haven't changed, make sure to move them to the
+	  end of the tracking array if needed, so we always know what the
+	  most recent entries are, in case we need to discard the oldest
+	  when generating codec_data.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/775>
+
+2020-10-17 00:05:15 +1100  Jan Schmidt <jan@centricular.com>
+
+	* gst/rtp/gstrtph264depay.c:
+	  rtph264depay: Warn when max SPS/PPS are collected in AVC mode.
+	  The AVC codec_data has a flaw that it can only accomodate
+	  31 SPS headers, even though H.264 can have 32, and 255 PPS,
+	  when there can be 256 in H.264. When streaming RTP some
+	  clients like to cycle through SPS/PPS ids when changing
+	  configuration and can eventually accumulate a full set.
+	  In that case, we have no choice but to discard one (oldest)
+	  entry, or else the count written into the codec_data is wrong
+	  and downstream decoding failures ensue.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/775>
+
+2020-10-28 00:29:05 +0100  Havard Graff <havard.graff@gmail.com>
+
+	* gst/rtpmanager/gstrtpjitterbuffer.c:
+	* gst/rtpmanager/rtptimerqueue.c:
+	* gst/rtpmanager/rtptimerqueue.h:
+	* tests/check/elements/rtpjitterbuffer.c:
+	* tests/check/elements/rtptimerqueue.c:
+	  rtpjitterbuffer: don't send multiple instant RTX for the same packet
+	  Due to us not properly acknowleding the time when the last RTX was sent
+	  when scheduling a new one, it can easily happen that due to the packet
+	  you are requesting have a PTS that is slightly old (but not too old when
+	  adding the latency of the jitterbuffer), both its calculated second and
+	  third (etc.) timeout could already have passed. This would lead to a burst
+	  of RTX requests, which acts completely against its purpose, potentially
+	  spending a lot more bandwidth than needed.
+	  This has been properly reproduced in the test:
+	  test_rtx_not_bursting_requests
+	  The good news is that slightly re-thinking the logic concerning
+	  re-requesting RTX, made it a lot simpler to understand, and allows us
+	  to remove two members of the RtpTimer which no longer serves any purpose
+	  due to the refactoring. If desirable the whole "delay" concept can actually
+	  be removed completely from the timers, and simply just added to the timeout
+	  by the caller of the API. But that can be a change for a another time.
+	  The only external change (other than the improved behavior around bursting
+	  RTX) is that the "delay" field now stricly represents the delay between
+	  the PTS of the RTX-requested packet and the time it is requested on,
+	  whereas before this calculation was more about the theoretical calculated
+	  delay. This is visible in three other RTX-tests where the delay had
+	  to be adjusted slightly. I am confident however that this change is
+	  correct.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/789>
+
+2020-10-27 23:43:49 +1100  Jan Schmidt <jan@centricular.com>
+
+	* gst/matroska/matroska-mux.c:
+	  matroska-mux: Fix sparse stream crash
+	  https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/656
+	  introduced an invalid memory access when debug is enabled, by casting
+	  the wrong pointer to a GstCollectPad. Fixing that showed the original
+	  change was incorrect and leads to an infinite loop in the
+	  testsuite. This patch fixes both problems.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/788>
+
+2020-10-22 15:29:01 -0300  Thibault Saunier <tsaunier@igalia.com>
+
+	* ext/vpx/gstvpxenc.c:
+	  vpx: Fix the check to unfixed/unknown framerate to set bitrate
+	  0/1 means unknown framerate not X/0 (which is illegal).
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/782>
+
+2020-10-22 09:17:26 -0400  Arun Raghavan <arun@asymptotic.io>
+
+	* gst/rtp/gstrtputils.c:
+	  rtputils: Count metas with an empty tag list for copying/keeping
+	  The GstMetaInfos registered in core do not set their tags to NULL, but
+	  instead use an empty list (non-NULL list with a single NULL value).
+	  Let's check explicitly for that so as to not miss some metas.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/779>
+
+2020-10-16 16:05:45 -0700  Bastien Reboulet <bastien.reboulet@gmail.com>
+
+	* ext/qt/qtitem.cc:
+	  qmlglsink: fix crash when created/destroyed in quick succession
+	  The crash is caused by a race condition where the render thread
+	  calls a method on the QtGLVideoItem instance that was
+	  previously destroyed by the main thread.
+	  Also, less frequently, QtGLVideoItem::onSceneGraphInitialized
+	  is called when QQuickItem::window is null, also causing a crash.
+	  Fixes #798
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/777>
+
+2020-10-19 18:23:25 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* sys/v4l2/gstv4l2videodec.c:
+	* sys/v4l2/gstv4l2videoenc.c:
+	  v4l2codec: Garbage collect old frames if they accumulate because of codec bugs
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/778>
+
+2020-10-19 17:56:04 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* sys/v4l2/gstv4l2bufferpool.c:
+	* sys/v4l2/gstv4l2bufferpool.h:
+	* sys/v4l2/gstv4l2sink.c:
+	* sys/v4l2/gstv4l2src.c:
+	* sys/v4l2/gstv4l2transform.c:
+	* sys/v4l2/gstv4l2videodec.c:
+	* sys/v4l2/gstv4l2videoenc.c:
+	  v4l2codec: Pass system frame number as timestamp and use it to retrieve back frames reliably
+	  System frame numbers are supposed to be unique and correct drivers are
+	  passing through timestamps without modification from the output/sink to the
+	  capture/src side.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/778>
+
+2020-09-24 13:13:00 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+	* docs/gst_plugins_cache.json:
+	* gst/rtpmanager/gstrtpbin.c:
+	* gst/rtpmanager/gstrtpbin.h:
+	  rtpbin: Add clear-ssrc action
+	  This action signal will delegate to clear-ssrc onto the rtpssrcdemux element
+	  associated with the session. This allow rtpbin users to clear pads and
+	  elements for a specific ssrc that is known to no longer be in use. This
+	  happens when a pad is reused in rtpsrc or ristsrc.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/736>
+
+2017-09-08 20:02:13 +0100  John-Mark Bell <jmb@pexip.com>
+
+	* gst/rtp/gstrtpvp8pay.c:
+	* gst/rtp/gstrtpvp8pay.h:
+	* tests/check/elements/rtpvp8.c:
+	* tests/check/meson.build:
+	  rtpvp8pay: payload temporally scaled bitstreams.
+	  Co-Authored-By: Vincent Sanders <vince@pexip.com>
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/728>
+
+2017-11-17 15:11:41 +0100  Stian Selnes <stian@pexip.com>
+
+	* docs/gst_plugins_cache.json:
+	* gst/rtp/gstrtpvp8pay.c:
+	* gst/rtp/gstrtpvp8pay.h:
+	  rtpvp8pay: Add picture-id-offset property
+	  Add property to set the initial value for picture-id. RFC7741 says
+	  that picture-id MAY be initialized to a random value, thus it's also
+	  valid to simply set it to a fixed initial value. A fixed value is very
+	  useful for testing.
+	  Default behavior is not changed.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/728>
+
+2017-03-16 15:23:28 +0100  Mikhail Fludkov <misha@pexip.com>
+
+	* gst/rtp/gstrtpvp8pay.c:
+	  rtpvp8pay: move duplicate code to separate functions
+	  Two new functions to modify picture id:
+	  gst_rtp_vp8_pay_picture_id_reset - picks random picture id of
+	  appropriate bitsize
+	  gst_rtp_vp8_pay_picture_id_increment - increments picture id taking
+	  care of wrapping
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/728>
+
+2017-09-08 08:13:05 +0100  John-Mark Bell <jmb@pexip.com>
+
+	* docs/gst_plugins_cache.json:
+	* ext/vpx/gstvpxenc.c:
+	  vp8enc: expect bps for temporal-scalability-target-bitrate.
+	  Consistency with target-bitrate is less surprising and with
+	  modern libvpx additional configuration is required to make
+	  temporal scaling work.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/728>
+
+2017-09-08 08:19:20 +0100  John-Mark Bell <jmb@pexip.com>
+
+	  vp8enc: finish support for temporally scaled encoding
+	  - introduce two new properties:
+	  * temporal-scalability-layer-flags:
+	  Provide fine-grained control of layer encoding to the
+	  outside world. The flags sequence should be a multiple of
+	  the periodicity and is indexed by a running count of encoded
+	  frames modulo the sequence length.
+	  * temporal-scalability-layer-sync-flags:
+	  Specify the pattern of inter-layer synchronisation (i.e.
+	  which of the frames generated by the layer encoding
+	  specification represent an inter-layer synchronisation).
+	  There must be one entry per entry in
+	  temporal-scalability-layer-flags.
+	  - apply temporal scalability settings and expose as buffer
+	  metadata.
+	  This allows the codec to allocate a given frame to the correct
+	  internal bitrate allocator. Additionally, all the
+	  non-bitstream metadata needed to payload a temporally scaled
+	  stream is now attached to each output buffer as a
+	  GstVideoVP8Meta.
+	  - add unit test for temporally scaled encoding.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/728>
+
+2020-10-15 18:21:54 +0200  Stéphane Cerveau <scerveau@collabora.com>
+
+	* gst/isomp4/qtdemux.c:
+	* gst/rtpmanager/gstrtpjitterbuffer.c:
+	* gst/udp/gstudp.c:
+	* meson.build:
+	  meson: update glib minimum version to 2.56
+	  In order to support the symbol g_enum_to_string in various
+	  project using GStreamer ( gst-validate etc.), the glib minimum
+	  version should be 2.56.0.
+	  Remove compat code as glib requirement
+	  is now > 2.56
+	  Version used by Ubuntu 18.04 LTS
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/774>
+
+2020-10-14 14:30:34 +0200  Mathieu Duponchelle <mathieu@centricular.com>
+
+	* gst/rtpmanager/gstrtpst2022-1-fecenc.c:
+	  rtpst2022-1-fecenc: fix input seqnum check
+	  We need to cast the incremented last seqnum to guint16 for
+	  consistent checks on wraparound
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/770>
+
+2020-09-12 09:02:30 +0200  Jan Alexander Steffens (heftig) <jan.steffens@ltnglobal.com>
+
+	* gst/flv/gstflvmux.c:
+	* gst/flv/gstflvmux.h:
+	  flvmux: Correct time types
+	  - last_dts is in milliseconds, not nanoseconds as expected for
+	  GstClockTime. Make it a generic guint64.
+	  - Use GstClockTime for the fields that actually contain nanoseconds.
+	  None of them should become negative.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/766>
+
+2020-10-09 09:31:27 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/rtpmanager/gstrtpst2022-1-fecenc.c:
+	  rtpst2022-1-fecenc: Don't unconditionally use GLib 2.60 APIs
+	  g_queue_clear_full() in this case.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/768>
+
+2020-10-08 18:54:55 +0200  Mathieu Duponchelle <mathieu@centricular.com>
+
+	* gst/rtp/rtpulpfeccommon.c:
+	  rtpulpfec: fix potential alignment issue in xor function
+	  https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/753#note_646453
+	  for context
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/753>
+
+2020-10-06 03:03:13 +0200  Mathieu Duponchelle <mathieu@centricular.com>
+
+	* docs/gst_plugins_cache.json:
+	* gst/rtpmanager/gstrtpbin.c:
+	* gst/rtpmanager/gstrtpbin.h:
+	* gst/rtpmanager/gstrtpmanager.c:
+	* gst/rtpmanager/gstrtpst2022-1-fecenc.c:
+	* gst/rtpmanager/gstrtpst2022-1-fecenc.h:
+	* gst/rtpmanager/meson.build:
+	* tests/check/elements/rtpst2022-1-fecenc.c:
+	* tests/check/meson.build:
+	  rtpmanager: implement SMPTE 2022-1 FEC encoder
+	  + improve integration of FEC encoders in rtpbin
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/753>
+
+2020-10-06 03:13:30 +0200  Mathieu Duponchelle <mathieu@centricular.com>
+
+	* docs/gst_plugins_cache.json:
+	* gst/rtpmanager/gstrtpbin.c:
+	* gst/rtpmanager/gstrtpbin.h:
+	* gst/rtpmanager/gstrtpmanager.c:
+	* gst/rtpmanager/gstrtpst2022-1-fecdec.c:
+	* gst/rtpmanager/gstrtpst2022-1-fecdec.h:
+	* gst/rtpmanager/meson.build:
+	* tests/check/elements/rtpst2022-1-fecdec.c:
+	* tests/check/meson.build:
+	  rtpmanager: implement SMPTE 2022-1 FEC decoder
+	  + improve integration of FEC decoders in rtpbin
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/753>
+
+2020-07-08 17:28:31 -0400  Olivier Crête <olivier.crete@collabora.com>
+
+	* gst/rtpmanager/gstrtpfunnel.c:
+	* tests/check/elements/rtpfunnel.c:
+	  rtpfunnel: Also forward custom sticky event
+	  This is useful to track metadata about each group of packets
+	  Also include a unit test
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/666>
+
+2020-09-29 09:44:54 -0300  Thibault Saunier <tsaunier@igalia.com>
+
+	* docs/gst_plugins_cache.json:
+	* gst/isomp4/gstqtmux-doc.c:
+	* gst/isomp4/gstqtmux.c:
+	* gst/isomp4/gstqtmuxmap.c:
+	  isomp4: Rename GstQTMux to GstBaseQTMux to avoid breaking API
+	  Since 52b63de19ada283c1180c8fc00cacb1465fdf10f the qtmux GType was
+	  renamed GstQTMuxElement which breaks presets, revert that change.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/755>
+
+2020-09-28 18:25:21 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/rtp/gstrtpdvpay.c:
+	* gst/rtp/gstrtph261pay.c:
+	* gst/rtp/gstrtph263pay.c:
+	* gst/rtp/gstrtph263ppay.c:
+	* gst/rtp/gstrtph264pay.c:
+	* gst/rtp/gstrtph265pay.c:
+	* gst/rtp/gstrtpj2kpay.c:
+	* gst/rtp/gstrtpjpegpay.c:
+	* gst/rtp/gstrtpklvpay.c:
+	* gst/rtp/gstrtpmp4vpay.c:
+	* gst/rtp/gstrtpmpvpay.c:
+	* gst/rtp/gstrtptheorapay.c:
+	* gst/rtp/gstrtpvp8pay.c:
+	* gst/rtp/gstrtpvp9pay.c:
+	  rtp: Fix allocations to support source-info property
+	  Use gst_rtp_base_payload_allocate_output_buffer() instead of
+	  gst_rtp_buffer_new_allocate() in order to allocate RTP buffer with
+	  correct number of CSRCs according to the meta.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/612>
+
+2015-10-23 11:08:56 +0200  Stian Selnes <stian@pexip.com>
+
+	* gst/rtp/gstrtpvp8pay.c:
+	  rtpvp8pay: Fix allocation to support source-info property
+	  Use gst_rtp_base_payload_allocate_output_buffer() in order to allocate
+	  RTP buffer with correct number of CSRCs according to the meta.
+	  Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/314
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/612>
+
+2020-09-28 15:36:00 +1000  Matthew Waters <matthew@centricular.com>
+
+	* gst/isomp4/gstqtmux.c:
+	  qtmux: output the correct limits in error messages
+	  Having the current bytes being less than the limit was confusing!
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/732>
+
+2020-07-31 16:47:37 +1000  Matthew Waters <matthew@centricular.com>
+
+	* gst/isomp4/gstqtmux.c:
+	* gst/isomp4/gstqtmux.h:
+	* tests/check/elements/qtmux.c:
+	  qtmux: properly support initial caps nego failure
+	  Scenario:
+	  - gap event causes h264parse to push made up caps that may fail checks
+	  inside qtmux (e.g missing codec_data).
+	  - the caps event has already been marked as received and is sticky on
+	  the sink pad
+	  - gst_qt_mux_pad_can_renegotiate() will retrieve the failed caps event
+	  using gst_pad_get_current_caps() and reject the correct updated caps
+	  with codec_data.
+	  - Failure!
+	  Keep track of the configured caps ourselves instead of relying on the
+	  sticky event on the pad.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/732>
+
+2020-07-22 15:34:44 +1000  Matthew Waters <matthew@centricular.com>
+
+	* gst/isomp4/gstqtmux.c:
+	* gst/isomp4/gstqtmux.h:
+	  qtmux: support non-seekable downstream mode
+	  Write an mdat per buffer in that case.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/732>
+
+2020-09-23 15:25:36 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+	* gst/rtpmanager/gstrtpbin.c:
+	  rtpbin: Remove the rtpjitterbuffer with the stream
+	  Since !348, the jitterbuffer was only removed with the session. This restores
+	  the original behaviour and removes the jitterbuffer when the stream is
+	  removed. This avoid accumulating jitterbuffer objects into the bin when a
+	  session is reused.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/735>
+
+2020-09-23 13:26:51 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+	* gst/rtpmanager/gstrtpbin.c:
+	  rtpbin: Cleanup dead code
+	  The rtpjitterbuffer is now part of the session elements, we no longer need
+	  to do the ref_sink dance when signalling it. It is already owned by the bin
+	  when signalled. Also, the code that handles generic session elements already
+	  handle the ref_sink() calls since:
+	  03dc22951bacb6fdc3868c8f801e6a52c33a745f
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/735>
+
+2020-09-18 16:09:20 +1000  Matthew Waters <matthew@centricular.com>
+
+	* gst/rtp/gstrtph264depay.c:
+	* gst/rtp/gstrtph264depay.h:
+	* gst/rtp/gstrtph265depay.c:
+	* gst/rtp/gstrtph265depay.h:
+	* tests/check/elements/rtph264.c:
+	  rtph26*depay: drop FU's without a corresponding start bit
+	  If we have not received a FU with a start bit set, any subsequent FU
+	  data is not useful at all and would result in an invalid stream.
+	  This case is constructed from multiple requirements in
+	  RFC 3984 Section 5.8 and RFC 7798 Section 4.4.3.  Following are excerpts
+	  from RFC 3984 but RFC 7798 contains similar language.
+	  The FU in a single FU case is forbidden:
+	  A fragmented NAL unit MUST NOT be transmitted in one FU; i.e., the
+	  Start bit and End bit MUST NOT both be set to one in the same FU
+	  header.
+	  and dropping is possible:
+	  If a fragmentation unit is lost, the receiver SHOULD discard all
+	  following fragmentation units in transmission order corresponding to
+	  the same fragmented NAL unit.
+	  The jump in seqnum case is supported by this from the specification
+	  instead of implementing the forbidden_zero_bit mangling:
+	  If a fragmentation unit is lost, the receiver SHOULD discard all
+	  following fragmentation units in transmission order corresponding to
+	  the same fragmented NAL unit.
+	  A receiver in an endpoint or in a MANE MAY aggregate the first n-1
+	  fragments of a NAL unit to an (incomplete) NAL unit, even if fragment
+	  n of that NAL unit is not received.  In this case, the
+	  forbidden_zero_bit of the NAL unit MUST be set to one to indicate a
+	  syntax violation.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/730>
+
+2020-09-20 21:06:19 +0900  Seungha Yang <seungha@centricular.com>
+
+	* gst/imagefreeze/gstimagefreeze.c:
+	  imagefreeze: Response caps query from srcpad
+	  ... and chain up to default query handler for unhandled query types.
+	  Unhandled query shouldn't be returned with FALSE if there's no special needs.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/731>
+
+2020-09-16 12:15:09 +1000  Matthew Waters <matthew@centricular.com>
+
+	* docs/gst_plugins_cache.json:
+	* gst/isomp4/gstqtmux-doc.c:
+	* gst/isomp4/gstqtmux-doc.h:
+	* gst/isomp4/gstqtmux.c:
+	* gst/isomp4/gstqtmux.h:
+	  qtmux: make documentation happy
+	  introduce a base qtmux class that we can install documentation snippets
+	  on instead of duplicating across alll the isomp4 elements
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/643>
+
+2020-05-28 19:40:24 +1000  Matthew Waters <matthew@centricular.com>
+
+	* docs/gst_plugins_cache.json:
+	* gst/isomp4/atoms.c:
+	* gst/isomp4/atoms.h:
+	* gst/isomp4/gstqtmux.c:
+	* gst/isomp4/gstqtmux.h:
+	* gst/isomp4/gstqtmuxmap.c:
+	* tests/check/elements/qtmux.c:
+	  isomp4/mux: add a fragment mode for initial moov with data
+	  Used by some proprietary software for their fragmented files.
+	  Adds some support for multi-stream fragmented files
+	  Flow is as follows.
+	  1. The first 'fragment' is written as a self-contained fragmented
+	  mdat+moov complete with an edit list and durations, tags, etc.
+	  2. Subsequent fragments are written with a mdat+moof and each stream is
+	  interleaved as data arrives (currently ignoring the interleave-*
+	  properties).  data-offsets in both the traf and the trun ensure
+	  data is read from the correct place on demuxing.  Data/chunk offsets
+	  are also kept for writing out the final moov.
+	  3. On finalisation, the initial moov is invalidated to a hoov and the
+	  size of the first mdat is extended to cover the entire file contents.
+	  Then a moov is written as regularly would in moov-at-end mode (the
+	  default).
+	  This results in a file that is playable throughout while leaving a
+	  finalised file on completion for players that do not understand
+	  fragmented mp4.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/643>
+
+2020-06-25 16:37:56 +1000  Matthew Waters <matthew@centricular.com>
+
+	* gst/isomp4/qtdemux.c:
+	  qtdemux: increase some logging on streams and sample parsing
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/643>
+
+2020-06-25 16:35:45 +1000  Matthew Waters <matthew@centricular.com>
+
+	* gst/isomp4/qtdemux.c:
+	  qtdemux: bail out when encountering an atom with a size of 0
+	  A size 0 atom means the atom extends to the end of the file.  No further
+	  valid atoms will ever follow.  Avoids a subsequent scan for an atom from
+	  one byte earlier after encountering a size 0 atom.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/643>
+
+2020-06-25 16:33:04 +1000  Matthew Waters <matthew@centricular.com>
+
+	* gst/isomp4/qtdemux.c:
+	  qtdemux: fix subsequent moof parsing after moov with valid samples
+	  reset the moof_offset back to its original value like is done in the
+	  error case just before.
+	  Fixes subsequent parsing of a moof following a moov that contains valid
+	  samples in a non-streaming fragmented mp4.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/643>
+
+2020-06-25 16:30:28 +1000  Matthew Waters <matthew@centricular.com>
+
+	* gst/isomp4/qtdemux.c:
+	  qtdemux: extend edit list when fragmented
+	  When we are fragmented, the edit list may only refer to the portion of
+	  the media that is in the moov.  Extend the edit list stop time when we
+	  if there is only one qt segment and we are reading a fragmented file.
+	  Fixes playback of some fragmented mp4 files generated by proprietary
+	  programs.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/643>
+
+2020-09-15 14:22:13 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+	* meson_options.txt:
+	  meson: Allow overriding qt5 feature
+	  This will allow controlling that feature from gst-build
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/729>
+
+2015-11-17 19:14:01 -0500  Olivier Crête <olivier.crete@collabora.com>
+
+	* gst/multifile/gstsplitmuxsrc.c:
+	  splitmuxsrc: Implement segment query
+	  Fixes #239
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/713>
+
+2020-09-14 10:15:35 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* docs/gst_plugins_cache.json:
+	* gst/rtp/gstrtpmp4gdepay.c:
+	  rtpmp4gdepay: Allow lower-case "aac-hbr" instead of correct "AAC-hbr"
+	  Various live555 based products are using the wrong "mode" string or
+	  seem to assume case-insensitive matching, which is wrong.
+	  Examples for this are the Yuan SC6C0N1 mini and the Kiloview E2.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/727>
+
+2020-05-02 02:21:00 +0200  Stefan Brüns <stefan.bruens@rwth-aachen.de>
+
+	* gst/isomp4/qtdemux.c:
+	  qtdemux: Add support for AAX encrypted audio streams
+	  This is modelled after the DASH Common Encryption scheme, but is somewhat
+	  simpler as more parts are fixed, i.e. just one encryption scheme.
+	  The output caps are fixed to 'application/x-aavd'. All information
+	  required for decryption are part of the 'adrm' atom, which is passed
+	  on as a property. The property is attached to the buffer.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/577>
+
+2020-05-02 02:20:44 +0200  Stefan Brüns <stefan.bruens@rwth-aachen.de>
+
+	* gst/isomp4/fourcc.h:
+	* gst/isomp4/qtdemux.c:
+	* gst/isomp4/qtdemux_types.c:
+	  qtdemux: Add 'aavd' and related fourcc codes for AAX encrypted audio
+	  The 'aavd' box is contained in the 'stsd' sample description. The 'aavd'
+	  box follows the layout of an 'mp4a' entry, i.e. it contains a single
+	  standard 'esds' extension box, and the two proprietary 'adrm' and 'aabd'
+	  extension boxes.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/577>
+
+2014-06-23 08:46:37 +0200  Haakon Sporsheim <haakon@pexip.com>
+
+	* ext/vpx/gstvp8dec.c:
+	* ext/vpx/gstvp9dec.c:
+	* ext/vpx/gstvpxdec.c:
+	* ext/vpx/gstvpxdec.h:
+	  vpxdec: request a sync point on decoder errors
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/725>
+
+2020-09-13 18:31:57 +0200  Camilo Celis Guzman <camilo@pexip.com>
+
+	* gst/rtp/gstrtpvrawpay.c:
+	  rtp/vrawpay: use alloc_output_buffer from base class
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/726>
+
+2020-09-07 23:20:58 +0800  Ricky Tang <ricky@deepsentinel.com>
+
+	* docs/gst_plugins_cache.json:
+	* gst/rtsp/gstrtspsrc.c:
+	  rtspsrc: Fix push-backchannel-buffer parameter mismatch
+	  When using python, signal parameter must match with function.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/724>
+
+2020-09-10 11:24:32 +0200  Jérôme Laheurte <jlaheurte@quividi.net>
+
+	* ext/jpeg/gstjpegdec.c:
+	  jpegdec: check buffer size before dereferencing. Fixes #541
+	  Some cameras (Panacast) have buggy drivers/firmware which send
+	  invalid JPEG frames, containing no data, which makes jpegdec
+	  crash because it assumes the frame is at least 2 bytes long.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/723>
+
+2020-09-10 11:11:00 +0200  Jan Alexander Steffens (heftig) <jan.steffens@ltnglobal.com>
+
+	* gst/flv/gstflvmux.c:
+	  flvmux: Improve logging of gst_flv_mux_buffer_to_tag_internal
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/722>
+
+2020-09-09 15:12:53 +0200  Jan Alexander Steffens (heftig) <jan.steffens@ltnglobal.com>
+
+	* gst/flv/gstflvmux.c:
+	  flvmux: Move stream skipping to GstAggregatorPadClass.skip_buffer
+	  Besides looking like the correct place to put this, it allows us to drop
+	  the entire aggregator queue. The old implementation only dropped at most
+	  one buffer for each call of aggregate.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/722>
+
+2020-09-08 17:35:50 +0200  Havard Graff <havard@pexip.com>
+
+	* sys/v4l2/gstv4l2object.c:
+	  v4l2object: plug memory-leak
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/718>
+
+2020-08-28 18:09:15 +0200  Mathieu Duponchelle <mathieu@centricular.com>
+
+	* docs/gst_plugins_cache.json:
+	* ext/vpx/gstvp9enc.c:
+	* ext/vpx/gstvp9enc.h:
+	* ext/vpx/meson.build:
+	  vp9enc: expose row-mt property
+	  With recent libvpx versions, multithreading can be enabled on
+	  a per-tile basis, instead of on a per tile-column basis.
+	  In combination with the new tile-rows property, this allows the
+	  encoder to make much better use of the available CPU power.
+	  Bump minimum libvpx version to 1.7.0
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/707>
+
+2020-08-28 17:45:48 +0200  Mathieu Duponchelle <mathieu@centricular.com>
+
+	* docs/gst_plugins_cache.json:
+	* ext/vpx/gstvpxenc.c:
+	  vpxenc: change default for deadline to good quality
+	  Having the deadline set to best quality causes the encoder
+	  to be absurdly slow, most real-life users will want the good
+	  quality tradeoff instead.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/707>
+
+2020-08-28 17:39:47 +0200  Mathieu Duponchelle <mathieu@centricular.com>
+
+	* docs/gst_plugins_cache.json:
+	* ext/vpx/gstvp9enc.c:
+	* ext/vpx/gstvp9enc.h:
+	  vp9enc: expose tile-columns and tile-rows properties
+	  Based on patch by Stian Selnes <stian@pexip.com>.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/707>
+
+2020-08-28 17:35:26 +0200  Mathieu Duponchelle <mathieu@centricular.com>
+
+	* ext/vpx/gstvpxenc.c:
+	* ext/vpx/gstvpxenc.h:
+	  vpxenc: add configure_encoder virtual method
+	  For subclasses to expose format-specific properties
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/707>
+
+2020-09-08 20:57:33 +0200  Mathieu Duponchelle <mathieu@centricular.com>
+
+	* gst/multifile/gstsplitmuxsink.c:
+	  splitmuxsink: fix sink pad release while PLAYING
+	  - Release the split mux lock while removing the probes
+	  - Flush the sinkpad to unblock other pads
+	  - Turn check_completed_gop into a do while statement, when
+	  waking up we want to recheck whether the current GOP is
+	  ready for sending
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/719>
+
+2017-10-31 09:40:33 +0000  John-Mark Bell <jmb@pexip.com>
+
+	* tests/check/elements/vp8enc.c:
+	  vp8enc: improve unit tests
+	  - make test_encode_simple cope with libvpx built with
+	  CONFIG_REALTIME_ONLY. Sadly, there's no way to detect this at
+	  runtime beyond trying to set lag-in-frames to >0, pushing a
+	  buffer and catching the GST_FLOW_NOT_NEGOTIATED return.
+	  - fix bitrot in test_encode_simple_when_bitrate_set_to_zero.
+	  - port test_encode_simple to GstHarness and introduce a separate
+	  test for the lag-in-frames property.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/708>
+
+2020-08-21 16:03:09 +0200  Jakub Adam <jakub.adam@collabora.com>
+
+	* docs/gst_plugins_cache.json:
+	  docs: Update plugin cache
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/705>
+
+2020-03-24 19:35:07 +0100  Jakub Adam <jakub.adam@collabora.com>
+
+	* ext/vpx/gstvp9dec.c:
+	* ext/vpx/gstvp9enc.c:
+	* ext/vpx/gstvpxenc.c:
+	  vpx: Support GST_VIDEO_FORMAT_I422_10LE
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/705>
+
+2020-03-24 17:16:59 +0100  Jakub Adam <jakub.adam@collabora.com>
+
+	* ext/vpx/gstvp9dec.c:
+	* ext/vpx/gstvp9enc.c:
+	* ext/vpx/gstvpxenc.c:
+	  vpx: Support GST_VIDEO_FORMAT_I420_10LE
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/705>
+
+2020-03-23 21:44:30 +0100  Jakub Adam <jakub.adam@collabora.com>
+
+	* ext/vpx/gstvp9enc.c:
+	* ext/vpx/gstvpxenc.c:
+	  vp9enc: support GST_VIDEO_FORMAT_Y444
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/705>
+
+2020-09-08 17:30:35 +0100  Tim-Philipp Müller <tim@centricular.com>
+
+	* .gitlab-ci.yml:
+	  ci: include template from gst-ci master branch again
+
+2020-09-08 16:58:37 +0100  Tim-Philipp Müller <tim@centricular.com>
+
+	* docs/gst_plugins_cache.json:
+	* meson.build:
+	  Back to development
+
 === release 1.18.0 ===
 
 2020-09-08 00:05:14 +0100  Tim-Philipp Müller <tim@centricular.com>
 
+	* .gitlab-ci.yml:
 	* ChangeLog:
 	* NEWS:
 	* README:
 	* RELEASE:
+	* docs/gst_plugins_cache.json:
 	* gst-plugins-good.doap:
 	* meson.build:
 	  Release 1.18.0
diff --git a/NEWS b/NEWS
index dba9c7c47..cc6c3b4a8 100644
--- a/NEWS
+++ b/NEWS
@@ -1,11 +1,23 @@
-GStreamer 1.18 Release Notes
+GStreamer 1.20 Release Notes
 
-GStreamer 1.18.0 was originally released on 7 September 2020.
+GStreamer 1.20 has not been released yet. It is scheduled for release
+around July 2021.
 
-See https://gstreamer.freedesktop.org/releases/1.18/ for the latest
+1.19.x is the unstable development version that is being developed in
+the git master branch and which will eventually result in 1.20, and
+1.19.1 is the current development release in that series
+
+It is expected that feature freeze will be around June/July 2021,
+followed by several 1.19 pre-releases and the new 1.20 stable release
+around July 2021.
+
+1.20 will be backwards-compatible to the stable 1.18, 1.16, 1.14, 1.12,
+1.10, 1.8, 1.6,, 1.4, 1.2 and 1.0 release series.
+
+See https://gstreamer.freedesktop.org/releases/1.20/ for the latest
 version of this document.
 
-Last updated: Monday 7 September 2020, 10:30 UTC (log)
+Last updated: Sunday 30 May 2021, 16:00 UTC (log)
 
 Introduction
 
@@ -18,1639 +30,87 @@ fixes and other improvements.
 
 Highlights
 
--   GstTranscoder: new high level API for applications to transcode
-    media files from one format to another
-
--   High Dynamic Range (HDR) video information representation and
-    signalling enhancements
-
--   Instant playback rate change support
-
--   Active Format Description (AFD) and Bar Data support
-
--   ONVIF trick modes support in both GStreamer RTSP server and client
-
--   Hardware-accelerated video decoding on Windows via DXVA2 /
-    Direct3D11
-
--   Microsoft Media Foundation plugin for video capture and
-    hardware-accelerated video encoding on Windows
-
--   qmlgloverlay: New overlay element that renders a QtQuick scene over
-    the top of an input video stream
-
--   New imagesequencesrc element to easily create a video stream from a
-    sequence of jpeg or png images
-
--   dashsink: Add new sink to produce DASH content
-
--   dvbsubenc: DVB Subtitle encoder element
-
--   TV broadcast compliant MPEG-TS muxing with constant bitrate muxing
-    and SCTE-35 support
-
--   rtmp2: new RTMP client source and sink element implementation
-
--   svthevcenc: new SVT-HEVC-based H.265 video encoder
-
--   vaapioverlay compositor element using VA-API
-
--   rtpmanager support for Google’s Transport-Wide Congestion Control
-    (twcc) RTP extension
-
--   splitmuxsink and splitmuxsrc gained support for auxiliary video
-    streams
-
--   webrtcbin now contains some initial support for renegotiation
-    involving stream addition and removal
-
--   New RTP source and sink elements to easily set up RTP streaming via
-    rtp:// URIs
-
--   New Audio Video Transport Protocol (AVTP) plugin for Time-Sensitive
-    Applications
-
--   Support for the Video Services Forum’s Reliable Internet Stream
-    Transport (RIST) TR-06-1 Simple Profile
-
--   Universal Windows Platform (UWP) support
-
--   rpicamsrc element for capturing from the Raspberry Pi camera
-
--   RTSP Server TCP interleaved backpressure handling improvements as
-    well as support for Scale/Speed headers
-
--   GStreamer Editing Services gained support for nested timelines,
-    per-clip speed rate control and the OpenTimelineIO format.
-
--   Autotools build system has been removed in favour of Meson
+-   this section will be completed in due course
 
 Major new features and changes
 
 Noteworthy new features and API
 
-Instant playback rate changes
-
-Changing the playback rate as quickly as possible so far always required
-a flushing seek. This generally works, but has the disadvantage of
-flushing all data from the playback pipeline and requiring the demuxer
-or parser to do a full-blown seek including resetting its internal state
-and resetting the position of the data source. It might also require
-considerable decoding effort to get to the right position to resume
-playback from at the higher rate.
-
-This release adds a new mechanism to achieve quasi-instant rate changes
-in certain playback pipelines without interrupting the flow of data in
-the pipeline. This is activated by sending a seek with the
-GST_SEEK_FLAG_INSTANT_RATE_CHANGE flag and start_type = stop_type =
-GST_SEEK_TYPE_NONE. This flag does not work for all pipelines, in which
-case it is necessary to fall back to sending a full flushing seek to
-change the playback rate. When using this flag, the seek event is only
-allowed to change the current rate and can modify the trickmode flags
-(e.g. keyframe only or not), but it is not possible to change the
-current playback position, playback direction or do a flush.
-
-This is particularly useful for streaming use cases like HLS or DASH
-where the streaming download should not be interrupted when changing
-rate.
-
-Instant rate changing is handled in the pipeline in a specific sequence
-which is detailed in the seeking design docs. Most elements don’t need
-to worry about this, only elements that sync to the clock need some
-special handling which is implemented in the GstBaseSink base class, so
-should be taken care of automatically in most normal playback pipelines
-and sink elements.
-
-See Jan’s GStreamer Conference 2019 talk “Changing Playback Rate
-Instantly” for more information.
-
-You can try this feature by passing the -i command line option to
-gst-play-1.0. It is supported at least by qtdemux, tsdemux, hlsdemux,
-and dashdemux.
-
-Google Transport-Wide Congestion Control
-
-rtpmanager now supports the parsing and generating of RTCP messages for
-the Google Transport-Wide Congestion Control RTP Extension, as described
-in:
-https://tools.ietf.org/html/draft-holmer-rmcat-transport-wide-cc-extensions-01.
-
-This “just” provides the required plumbing/infrastructure, it does not
-actually make effect any actual congestion control on the sender side,
-but rather provides information for applications to use to make such
-decisions.
-
-See Håvard’s “Google Transport-Wide Congestion Control” talk for more
-information about this feature.
-
-GstTranscoder: a new high-level transcoding API for applications
-
-The new GstTranscoder library, along with transcodebin and
-uritranscodebin elements, provides high level API for applications to
-transcode media files from one format to another. Watch Thibault’s talk
-“GstTranscoder: A High Level API to Quickly Implement Transcoding
-Capabilities in your Applications” for more information.
-
-This also comes with a gst-transcoder-1.0 command line utility to
-transcode one URI into another URI based on the specified encoding
-profile.
-
-Active Format Description (AFD) and Bar Data support
-
-The GstVideo Ancillary Data API has gained support for Active Format
-Description (AFD) and Bar data.
-
-This includes various two new buffer metas: GstVideoAFDMeta and
-GstVideoBarMeta.
-
-GStreamer now also parses and extracts AFD/Bar data in the h264/h265
-video parsers, and supports both capturing them and outputting them in
-the decklink elements. See Aaron’s lightning talk at the GStreamer
-Conference for more background.
-
-ONVIF trick modes support in both GStreamer RTSP server and client
-
--   Support for the various trick modes described in section 6 of the
-    ONVIF streaming spec has been implemented in both gst-rtsp-server
-    and rtspsrc.
--   Various new properties in rtspsrc must be set to take advantage of
-    the ONVIF support
--   Examples are available here: test-onvif-server.c and
-    test-onvif-client.c
--   Watch Mathieu Duponchelle’s talk “Implementing a Trickmode Player
-    with ONVIF, RTSP and GStreamer” for more information and a live
-    demo.
-
-GStreamer Codecs library with decoder base classes
-
-This introduces a new library in gst-plugins-bad which contains a set of
-base classes that handle bitstream parsing and state tracking for the
-purpose of decoding different codecs. Currently H264, H265, VP8 and VP9
-are supported. These bases classes are meant primarily for internal use
-in GStreamer and are used in various decoder elements in connection with
-low level decoding APIs like DXVA, NVDEC, VAAPI and V4L2 State Less
-decoders. The new library is named gstreamer-codecs-1.0 /
-libgstcodecs-1.0 and is not yet guaranteed to be API stable across major
-versions.
-
-MPEG-TS muxing improvements
-
-The GStreamer MPEG-TS muxer has seen major improvements on various
-fronts in this cycle:
-
--   It has been ported to the GstAggregator base class which means it
-    can work in defined-latency mode with live input sources and
-    continue streaming if one of the inputs stops producing data.
-
--   atscmux, a new ATSC-specific tsmux subclass
-
--   Constant Bit Rate (CBR) muxing support via the new bitrate property
-    which allows setting the target bitrate in bps. If this is set the
-    muxer will insert null packets as padding to achieve the desired
-    multiplex-wide constant bitrate.
-
--   compliance fixes for TV broadcasting use cases (esp. ATSC). See
-    Jan’s talk “TV Broadcast compliant MPEG-TS” for details.
-
--   Streams can now be added and removed at runtime: Until now, any
-    streams in tsmux had to be present when the element started
-    outputting its first buffer. Now they can appear at any point during
-    the stream, or even disappear and reappear later using the same PID.
-
--   new pcr-interval property allows applications to configure the
-    desired interval instead of hardcoding it
-
--   basic SCTE-35 support. This is enabled by setting the scte-35-pid
-    property on the muxer. Sending SCTE-35 commands is then done by
-    creating the appropriate SCTE-35 GstMpegtsSection and sending them
-    on the muxer.
-
--   MPEG-2 AAC handling improvements
+-   this section will be filled in in due course
 
 New elements
 
--   New qmlgloverlay element for rendering a QtQuick scene over the top
-    of a video stream. qmlgloverlay requires that Qt support adopting an
-    external OpenGL context and is known to work on X11 and Windows.
-    Wayland is known not to work due to limitations within Qt. Check out
-    the example to see how it works.
-
--   The clocksync element is a generic element that can be placed in a
-    pipeline to synchronise passing buffers to the clock at that point.
-    This is similar to identity sync=true, but because it isn’t
-    GstBaseTransform-based, it can process GstBufferLists without
-    breaking them into separate GstBuffers. It is also more discoverable
-    than the identity option. Note that you do not need to insert this
-    element into your pipeline to make GStreamer sync to the pipeline
-    clock, this is usually handled automatically by the elements in the
-    pipeline (sources and sinks mostly). This element is useful to feed
-    non-live input such as local files into elements that expect live
-    input such as webrtcbin.`
-
--   New imagesequencesrc element to easily create a video stream from a
-    sequence of JPEG or PNG images (or any other encoding where the type
-    can be detected), basically a multifilesrc made specifically for
-    image sequences.
-
--   rpicamsrc element for capturing raw or encoded video (H.264, MJPEG)
-    from the Raspberry Pi camera. This works much like the popular
-    raspivid command line utility but outputs data nicely timestamped
-    and formatted in order to integrate nicely with other GStreamer
-    elements. Also comes with a device provider so applications can
-    discover the camera if available.
-
--   aatv and cacatv video filters that transform video ASCII art style
-
--   avtp: new Audio Video Transport Protocol (AVTP) plugin for Linux.
-    See Andre Guedes’ talk “Audio/Video Bridging (AVB) support in
-    GStreamer” for more details.
-
--   clockselect: a pipeline element that enables clock selection/forcing
-    via gst-launch pipeline syntax.
-
--   dashsink: Add new sink to produce DASH content. See Stéphane’s talk
-    or blog post for details.
-
--   dvbsubenc: a DVB subtitle encoder element
-
--   microdns: a libmicrodns-based mdns device provider to discover RTSP
-    cameras on the local network
-
--   mlaudiosink: new audio sink element for the Magic Leap platform,
-    accompanied by an MLSDK implementation in the amc plugin
-
--   msdkvp9enc: VP9 encoder element for the Intel MediaSDK
-
--   rist: new plugin implementing support for the Video Services Forum’s
-    Reliable Internet Stream Transport (RIST) TR-06-1 Simple Profile.
-    See Nicolas’ blog post “GStreamer support for the RIST
-    Specification” for more details.
-
--   rtmp2: new RTMP client source and sink elements with fully
-    asynchronous network operations, better robustness and additional
-    features such as handling ping and stats messages, and adobe-style
-    authentication. The new rtmp2src and rtmp2sink elements should be
-    API-compatible with the old rtmpsrc / rtmpsink elements and should
-    work as drop-in replacements.
-
--   new RTP source and sink elements to easily set up RTP streaming via
-    rtp:// URIs: The rtpsink and rtpsrc elements add an URI interface so
-    that streams can be decoded with decodebin using rtp:// URIs. These
-    can be used as follows: ``` gst-launch-1.0 videotestsrc ! x264enc !
-    rtph264pay config-interval=3 ! rtpsink uri=rtp://239.1.1.1:1234
-
-    gst-launch-1.0 videotestsrc ! x264enc ! rtph264pay config-interval=1
-    ! rtpsink uri=rtp://239.1.2.3:5000 gst-launch-1.0 rtpsrc
-    uri=rtp://239.1.2.3:5000?encoding-name=H264 ! rtph264depay !
-    avdec_h264 ! videoconvert ! xvimagesink
-
-    gst-launch-1.0 videotestsrc ! avenc_mpeg4 ! rtpmp4vpay
-    config-interval=1 ! rtpsink uri=rtp://239.1.2.3:5000 gst-launch-1.0
-    rtpsrc uri=rtp://239.1.2.3:5000?encoding-name=MP4V-ES ! rtpmp4vdepay
-    ! avdec_mpeg4 ! videoconvert ! xvimagesink ```
-
--   svthevcenc: new SVT-HEVC-based H.265 video encoder
-
--   switchbin: new helper element which chooses between a set of
-    processing chains (paths) based on input caps, and changes the
-    active chain if new caps arrive. Paths are child objects, which are
-    accessed by the GstChildProxy interface. See the switchbin
-    documentation for a usage example.
-
--   vah264dec: new experimental va plugin with an element for H.264
-    decoding with VA-API using GStreamer’s new stateless decoder
-    infrastructure (see Linux section below).
-
--   v4l2codecs: introduce an V4L2 CODECs Accelerator supporting the new
-    CODECs uAPI in the Linux kernel (see Linux section below)
-
--   zxing new plugin to detect QR codes and barcodes, based on libzxing
-
--   also see the Rust plugins section below which contains plenty of new
-    exciting plugins written in Rust!
+-   this section will be filled in in due course
 
 New element features and additions
 
-GStreamer core
-
--   filesink: Add a new “full” buffer mode. Previously the default and
-    full modes were the same. Now the default mode is like before: it
-    accumulates all buffers in a buffer list until the threshold is
-    reached and then writes them all out, potentially in multiple
-    writes. The new full mode works by always copying memory to a single
-    memory area and writing everything out with a single write once the
-    threshold is reached.
-
--   multiqueue: Add stats property and
-    current-level-{buffers, bytes, time} pad properties to query the
-    current levels of the corresponding internal queue.
-
-Plugins Base
-
--   alsa: implement a device provider
-
--   alsasrc: added use-driver-timestamp property to force use of
-    pipeline timestamps (and disable driver timestamps) if so desired
-
--   audioconvert: fix changing the mix-matrix property at runtime
-
--   appsrc: added support for segment forwarding or custom GstSegments
-    via GstSample, enabled via the handle-segment-change property. This
-    only works for segments in TIME format for now.
-
--   compositor: various performance optimisations, checkerboard drawing
-    fixes, and support for VUYA format
-
--   encodebin: Fix and refactor smart encoding; ensure that a single
-    segment is pushed into encoders; improve force-key-unit event
-    handling.
-
--   opusenc: Add low delay option (audio-type=restricted-lowdelay) to
-    disable the SILK layer and achieve only 5ms delay.
-
--   opusdec: add stats property to retrieve various decoder statistics.
-
--   uridecodebin3: Let decodebin3 do its stream selection if no one
-    answers
-
--   decodebin3: Avoid overriding explicit user selection of streams
-
--   playbin: add flag to force use of software decoders over any
-    hardware decoders that might also be available
-
--   playbin3, playbin: propagate sink context
-
--   rawvideoparse: Fix tiling support, allow setting colorimetry
-
--   subparse: output plain utf8 text instead of pango-markup formatted
-    text if downstream requires it, useful for interop with elements
-    that only accept utf8-formatted subtitles such as muxers or closed
-    caption converters.
-
--   tcpserversrc, tcpclientsrc: add stats property with TCP connection
-    stats (some are only available on Linux though)
-
--   timeoverlay: add show-times-as-dates, datetime-format and
-    datetime-epoch properties to display times with dates
-
--   videorate: Fix changing rate property during playback; reverse
-    playback fixes; update QoS events taking into account our rate
-
--   videoscale: pass through and transform size sensitive metas instead
-    of just dropping them
-
-Plugins Good
-
--   avidemux can handle H.265 video now. Our advice remains to
-    immediately cease all contact and communication with anyone who
-    hands you H.265 video in an AVI container, however.
-
--   avimux: Add support for S24LE and S32LE raw audio and v210 raw video
-    formats; support more than 2 channels of raw audio.
-
--   souphttpsrc: disable session sharing and cookie jar when the cookies
-    property is set; correctly handle seeks past the end of the content
-
--   deinterlace: new YADIF deinterlace method which should provide
-    better quality than the existing methods and is LGPL licensed;
-    alternate fields are supported as input to the deinterlacer as well
-    now, and there were also fixes for switching the deinterlace mode on
-    the fly.
-
--   flvmux: in streamable mode allow adding new pads even if the initial
-    header has already been written. Old clients will only process the
-    initial stream, new clients will get a header with the new streams.
-    The skip-backwards-streams property can be used to force flvmux to
-    skip and drop a few buffers rather than produce timestamps that go
-    backward and confuse librtmp-based clients. There’s also better
-    handling for timestamp rollover when streaming for a long time.
-
--   imagefreeze: Add live mode, which can be enabled via the new is-live
-    property. In this mode frames will only be output in PLAYING state
-    according to the negotiated framerate, skipping frames if the output
-    can’t keep up (e.g. because it’s blocked downstream). This makes it
-    possible to actually use imagefreeze in live pipelines without
-    having to manually ensure somehow that it starts outputting at the
-    current running time and without still risking to fall behind
-    without recovery.
-
--   matroskademux, qtdemux: Provide audio lead-in for some lossy formats
-    when doing accurate seeks, to make sure we can actually decode
-    samples at the desired position. This is especially important for
-    non-linear audio/video editing use-cases.
-
--   matroskademux, matroskamux: Handle interlaced field order (tff, bff)
-
--   matroskamux:
-
-    -   new offset-to-zero property to offset all streams to start at
-        zero. This takes the timestamp of the earliest stream and
-        offsets it so that it starts at 0. Some software (VLC,
-        ffmpeg-based) does not properly handle Matroska files that start
-        at timestamps much bigger than zero, which could happen with
-        live streams.
-    -   added a creation-time property to explicitly set the creation
-        time to write into the file headers. Useful when remuxing, for
-        example, but also for live feeds where the DateUTC header can be
-        set a UTC timestamp corresponding to the beginning of the file.
-    -   the muxer now also always waits for caps on sparse streams, and
-        warns if caps arrive after the header has already been sent,
-        otherwise the subtitle track might be silently absent in the
-        final file. This might affect applications that send sparse data
-        into matroskamux via an appsrc element, which will usually not
-        send out the initial caps before it sends out the first buffer.
-
--   pulseaudio: device provider improvements: fix discovery of
-    newly-added devices and hide the alsa device provider if we provide
-    alsa devices
-
--   qtdemux: raw audio handling improvements, support for AC4 audio, and
-    key-units trickmode interval support
-
--   qtmux:
-
-    -   was ported to the GstAggregator base class which allows for
-        better handling of live inputs, but might entail minor
-        behavioural changes for sparse inputs if inputs are not live.
-    -   has also gained a force-create-timecode-trak property to create
-        a timecode trak in non-mov flavors, which may not be supported
-        by Apple but is supported by other software such as Final Cut
-        Pro X
-    -   also a force-chunks property to force the creation of chunks
-        even in single-stream files, which is required for Apple ProRes
-        certification.
-    -   also supports 8k resolutions in prefill mode with ProRes.
-
--   rtpbin gained a request-jitterbuffer signal which allows
-    applications to plug in their own jitterbuffer implementation such
-    as the threadsharing jitterbuffer from the Rust plugins, for
-    example.
-
--   rtprtxsend: add clock-rate-map property to allow generic RTP input
-    caps without a clock-rate whilst still supporting the max-size-time
-    property for bundled streams.
-
--   rtpssrcdemux: introduce max-streams property to guard against
-    attacks where the sender changes SSRC for every RTP packet.
-
--   rtph264pay, rtph264pay: implement STAP-A and various aggregation
-    modes controled by the new aggegrate-mode property: none to not
-    aggregate NAL units (as before), zero-latency to aggregate NAL units
-    until a VCL or suffix unit is included, or max to aggregate all NAL
-    units with the same timestamp (which adds one frame of latency). The
-    default has been kept at none for backwards compatibility reasons
-    and because various RTP/RTSP implementions don’t handle aggregation
-    well. For WebRTC use cases this should be set to zero-latency,
-    however.
-
--   rtpmp4vpay: add support for config-interval=-1 to resend headers
-    with each IDR keyframe, like other video payloaders.
-
--   rtpvp8depay: Add wait-for-keyframe property for waiting until the
-    next keyframe after packet loss. Useful if the video stream was not
-    encoded with error resilience enabled, in which case packet loss
-    tends to cause very bad artefacts when decoding, and waiting for the
-    next keyframe instead improves user experience considerably.
-
--   splitmuxsink and splitmuxsrc can now handle auxiliary video streams
-    in addition to the primary video stream. The primary video stream is
-    still used to select fragment cut points at keyframe boundaries.
-    Auxilliary video streams may be broken up at any packet - so
-    fragments may not start with a keyframe for those streams.
-
--   splitmuxsink:
-
-    -   new muxer-preset and sink-preset properties for setting
-        muxer/sink presets
-    -   a new start-index property to set the initial fragment id
-    -   and a new muxer-pad-map property which explicitly maps
-        splitmuxsink pads to the muxer pads they should connect to,
-        overriding the implicit logic that tries to match pads but
-        yields arbitrary names.
-    -   Also includes the actual sink element in the fragment-opened and
-        fragment-closed element messages now, which is especially useful
-        for sinks without a location property or when finalisation of
-        the fragments is done asynchronously.
-
--   videocrop: add support for Y444, Y41B and Y42B pixel formats
-
--   vp8enc, vp9enc: change default value of VP8E_SET_STATIC_THRESHOLD
-    from 0 to 1 which matches what Google WebRTC does and results in
-    lower CPU usage; also added a new bit-per-pixel property to select a
-    better default bitrate
-
--   v4l2: add support for ABGR, xBGR, RGBA, and RGBx formats and for
-    handling interlaced video in alternate fields interlace mode (one
-    field per buffer instead of one frame per picture with both fields
-    interleaved)
-
--   v4l2: Profile and level probing support for H264, H265, MPEG-4,
-    MPEG-2, VP8, and VP9 video encoders and decoders
-
-Plugins Ugly
-
--   asfdemux: extract more metadata: disc number and disc count
-
--   x264enc:
-
-    -   respect YouTube bitrate recommendation when user sets the
-        YouTube profile preset
-    -   separate high-10 video formats from 8-bit formats to improve
-        depth negotiation and only advertise suitable input raw formats
-        for the desired output depth
-    -   forward downstream colorimetry and chroma-site restrictions to
-        upstream elements
-    -   support more color primaries/mappings
-
-Plugins Bad
-
--   av1enc: add threads, row-mt and tile-{columns,rows} properties for
-    this AOMedia AV1 encoder
-
--   ccconverter: implement support for CDP framerate conversions
-
--   ccextractor: Add remove-caption-meta property to remove caption
-    metas from the outgoing video buffers
-
--   decklink: add support for 2K DCI video modes, widescreen NTSC/PAL,
-    and for parsing/outputting AFD/Bar data. Also implement a simple
-    device provider for Decklink devices.
-
--   dtlsrtpenc: add rtp-sync property which synchronises RTP streams to
-    the pipeline clock before passing them to funnel for merging with
-    RTCP.
-
--   fdkaac: also decode MPEG-2 AAC; encoder now supports more
-    multichannel/surround sound layouts
-
--   hlssink2: add action signals for custom playlist/fragment handling:
-    Instead of always going through the file system API we allow the
-    application to modify the behaviour. For the playlist itself and
-    fragments, the application can provide a GOutputStream. In addition
-    the sink notifies the application whenever a fragment can be
-    deleted.
-
--   interlace: can now output data in alternate fields mode; added field
-    switching mode for 2:2 field pattern
-
--   iqa: Add a mode property to enable strict mode that checks that all
-    the input streams have the exact same number of frames; also
-    implement the child proxy interface
-
--   mpeg2enc: add disable-encode-retries property for lower CPU usage
-
--   mpeg4videoparse: allow re-sending codec config at IDR via
-    config-interval=-1
-
--   mpegtsparse: new alignment property to determine number of TS
-    packets per output buffer, useful for feeding an MPEG-TS stream for
-    sending via udpsink. This can be used in combination with the
-    split-on-rai property that makes sure to start a new output buffer
-    for any TS packet with the Random Access Indicator set. Also set
-    delta unit buffer flag on non-random-access buffers.
-
--   mpegdemux: add an ignore-scr property to ignore the SCR in
-    non-compliant MPEG-PS streams with a broken SCR, which will work as
-    long as PTS/DTS in the PES header is consistently increasing.
-
--   tsdemux:
-
-    -   add an ignore-pcr property to ignore MPEG-TS streams with broken
-        PCR streams on which we can’t reliably recover correct
-        timestamps.
-    -   new latency property to allow applications to lower the
-        advertised worst-case latency of 700ms if they know their
-        streams support this (must have timestamps in higher frequency
-        than required by the spec)
-    -   support for AC4 audio
-
--   msdk - Intel Media SDK plugin for hardware-accelerated video
-    decoding and encoding on Windows and Linux:
-
-    -   mappings for more video formats: Y210, Y410, P012_LE, Y212_LE
-    -   encoders now support bitrate changes and input format changes in
-        playing state
-    -   msdkh264enc, msdkh265enc: add support for CEA708 closed caption
-        insertion
-    -   msdkh264enc, msdkh265enc: set Region of Interest (ROI) region
-        from ROI metas
-    -   msdkh264enc, msdkh265enc: new tune property to enable low-power
-        mode
-    -   msdkh265enc: add support 12-bit 4:2:0 encoding and 8-bit 4:2:2
-        encoding and VUYA, Y210, and Y410 as input formats
-    -   msdkh265enc: add support for screen content coding extension
-    -   msdkh265dec: add support for main-12/main-12-intra,
-        main-422-10/main-422-10-intra 10bit,
-        main-422-10/main-422-10-intra 8bit,
-        main-422-12/main-422-12-intra, main-444-10/main-444-10-intra,
-        main-444-12/main-444-12-intra, and main-444 profiles
-    -   msdkvp9dec: add support for 12-bit 4:4:4
-    -   msdkvpp: add support for Y410 and Y210 formats, cropping via
-        properties, and a new video-direction property.
-
--   mxf: Add support for CEA-708 CDP from S436 essence tracks. mxfdemux
-    can now handle Apple ProRes
-
--   nvdec: add H264 + H265 stateless codec implementation nvh264sldec
-    and nvh265sldec with fewer features but improved latency. You can
-    set the environment variable GST_USE_NV_STATELESS_CODEC=h264 to use
-    the stateless decoder variant as nvh264dec instead of the “normal”
-    NVDEC decoder implementation.
-
--   nvdec: add support for 12-bit 4:4:4/4:2:0 and 10-bit 4:2:0 decoding
-
--   nvenc:
-
-    -   add more rate-control options, support for B-frame encoding (if
-        device supports it), an aud property to toggle Access Unit
-        Delimiter insertion, and qp-{min,max,const}-{i,p,b} properties.
-    -   the weighted-pred property enables weighted prediction.
-    -   support for more input formats, namely 8-bit and 10-bit RGB
-        formats (BGRA, RGBA, RGB10A2, BGR10A2) and YV12 and VUYA.
-    -   on-the-fly resolution changes are now supported as well.
-    -   in case there are multiple GPUs on the system, there are also
-        per-GPU elements registered now, since different devices will
-        have different capabilities.
-    -   nvh265enc can now support 10-bit YUV 4:4:4 encoding and 8-bit
-        4:4:4 / 10-bit 4:2:0 formats up to 8K resolution (with some
-        devices). In case of HDR content HDR related SEI nals will be
-        inserted automatically.
-
--   openjpeg: enable multi-threaded decoding and add support for
-    sub-frame encoding (for lower latency)
-
--   rtponviftimestamp: add opt-out “drop-out-of-segment” property
-
--   spanplc: new stats property
-
--   srt: add support for IPv6 and for using hostnames instead of IP
-    addresses; add streamid property, but also allow passing the id via
-    the stream URI; add wait-for-connection property to srtsink
-
--   timecodestamper: this element was rewritten with an updated API
-    (properties); it has gained many new properties, seeking support and
-    support for linear timecode (LTC) from an audio stream.
-
--   uvch264src now comes with a device provider to advertise available
-    camera sources that support this interface (mostly Logitech C920s)
-
--   wpe: Add software rendering support and support for mouse scroll
-    events
-
--   x265enc: support more 8/10/12 bits 4:2:0, 4:2:2 and 4:4:4 profiles;
-    add support for mastering display info and content light level
-    encoding SEIs
-
-gst-libav
-
--   Add mapping for SpeedHQ video codec used by NDI
-
--   Add mapping for aptX and aptX-HD
-
--   avivf_mux: support VP9 and AV1
-
--   avvidenc: shift output buffer timestamps and output segment by 1h
-    just like x264enc does, to allow for negative DTS.
-
--   avviddec: Limit default number of decoder threads on systems with
-    more than 16 cores, as the number of threads used in avdec has a
-    direct impact on the latency of the decoder, which is of as many
-    frames as threads, so a large numbers of threads can make for
-    latency levels that can be problematic in some applications.
-
--   avviddec: Add thread-type property that allows applications to
-    specify the preferred multithreading method (auto, frame, slice).
-    Note that thread-type=frame may introduce additional latency
-    especially in live pipelines, since it introduces a decoding delay
-    of number of thread frames.
+-   this section will be filled in in due course
 
 Plugin and library moves
 
--   There were no plugin moves or library moves in this cycle.
+-   this section will be filled in in due course
 
--   The rpicamsrc element was moved into -good from an external
-    repository on github.
+-   There were no plugin moves or library moves in this cycle.
 
 Plugin removals
 
 The following elements or plugins have been removed:
 
--   The yadif video deinterlacing plugin from gst-plugins-bad, which was
-    one of the few GPL licensed plugins, has been removed in favour of
-    deinterlace method=yadif.
-
--   The avdec_cdgraphics CD Graphics video decoder element from
-    gst-libav was never usable in GStreamer and we now have a cdgdec
-    element written in Rust in gst-plugins-rs to replace it.
-
--   The VDPAU plugin has been unmaintained and unsupported for a very
-    long time and does not have the feature set we expect from
-    hardware-accelerated video decoders. It’s been superseded by the
-    nvcodec plugin leveraging NVIDIA’s NVDEC API.
+-   this section will be filled in in due course
 
 Miscellaneous API additions
 
-GStreamer core
-
--   gst_task_resume(): This new API allows resuming a task if it was
-    paused, while leaving it in stopped state if it was stopped or not
-    started yet. This can be useful for callback-based driver workflows,
-    where you basically want to pause and resume the task when buffers
-    are notified while avoiding the race with a gst_task_stop() coming
-    from another thread.
-
--   info: add printf extensions GST_TIMEP_FORMAT and GST_STIMEP_FORMAT
-    for printing GstClockTime/GstClockTimeDiff pointers, which is much
-    more convenient to use in debug log statements than the usual
-    GST_TIME_FORMAT-followed-by-GST_TIME_ARGS dance. Also add an
-    explicit GST_STACK_TRACE_SHOW_NONE enum value.
-
--   gst_element_get_current_clock_time() and
-    gst_element_get_current_running_time(): new helper functions for
-    getting an element clock’s time, and the clock time minus base time,
-    respectively. Useful when adding additional input branches to
-    elements such as compositor, audiomixer, flvmux, interleave or
-    input-selector to determine initial pad offsets and such.
-
--   seeking: Add GST_SEEK_FLAG_TRICKMODE_FORWARD_PREDICTED to just skip
-    B-frames during trick mode, showing both keyframes + P-frame, and
-    add support for it in h264parse and h265parse.
-
--   elementfactory: add GST_ELEMENT_FACTORY_TYPE_HARDWARE to allow
-    elements to advertise that they are hardware-based or interact with
-    hardware. This has multiple applications:
-
-    -   it makes it possible to easily differentiate hardware and
-        software based element implementations such as audio or video
-        encoders and decoders. This is useful in order to force the use
-        of software decoders for specific use cases, or to check if a
-        selected decoder is actually hardware-accelerated or not.
-    -   elements interacting with hardware and their respective drivers
-        typically don’t know the actually supported capabilities until
-        the element is set into at least READY state and can open a
-        device handle and probe the hardware.
-
--   gst_uri_from_string_escaped(): identical to gst_uri_from_string()
-    except that the userinfo and fragment components of the URI will not
-    be unescaped while parsing. This is needed for correctly parsing
-    usernames or passwords with : in them .
-
--   paramspecs: new GstParamSpec flag GST_PARAM_CONDITIONALLY_AVAILABLE
-    to indicate that a property might not always exist.
-
--   gst_bin_iterate_all_by_element_factory_name() finds elements in a
-    bin by factory name
-
--   pad: gst_pad_get_single_internal_link() is a new convenience
-    function to return the single internal link of a pad, which is
-    useful e.g. to retrieve the output pad of a new multiqueue request
-    pad.
-
--   datetime: Add constructors to create datetimes with timestamps in
-    microseconds, gst_date_time_new_from_unix_epoch_local_time_usecs()
-    and gst_date_time_new_from_unix_epoch_utc_usecs().
-
--   gst_debug_log_get_lines() gets debug log lines formatted in the same
-    way the default log handler would print them
-
--   GstSystemClock: Add GST_CLOCK_TYPE_TAI as GStreamer abstraction for
-    CLOCK_TAI, to support transmission offloading features where network
-    packets are timestamped with the time they are deemed to be actually
-    transmitted. Useful in combination with the new AVTP plugin.
-
--   miscellaneous utility functions: gst_clear_uri(),
-    gst_structure_take().
-
--   harness: Added gst_harness_pull_until_eos()
-
--   GstBaseSrc:
-
-    -   gst_base_src_new_segment() allows subclasses to update the
-        segment to be used at runtime from the ::create() function. This
-        deprecates gst_base_src_new_seamless_segment()
-    -   gst_base_src_negotiate() allows subclasses to trigger format
-        renegotiation at runtime from inside the ::create() or ::alloc()
-        function
-
--   GstBaseSink: new stats property and gst_base_sink_get_stats() method
-    to retrieve various statistics such as average frame rate and
-    dropped/rendered buffers.
-
--   GstBaseTransform: gst_base_transform_reconfigure() is now public
-    API, useful for subclasses that need to completely re-implement the
-    ::submit_input_buffer() virtual method
-
--   GstAggregator:
-
-    -   gst_aggregator_update_segment() allows subclasses to update the
-        output segment at runtime. Subclasses should use this function
-        rather than push a segment event onto the source pad directly.
-    -   new sample selection API:
-        -   subclasses should now call gst_aggregator_selected_samples()
-            from their ::aggregate() implementation to signal that they
-            have selected the next samples they will aggregate
-        -   GstAggregator will then emit the samples-selected signal
-            where handlers can then look up samples per pad via
-            gst_aggregator_peek_next_sample().
-        -   This is useful for example to atomically update input pad
-            properties in mixer subclasses such as compositor.
-            Applications can now update properties with precise control
-            of when these changes will take effect, and for which input
-            buffer(s).
-    -   gst_aggregator_finish_buffer_list() allows subclasses to push
-        out a buffer list, improving efficiency in some cases.
-    -   a ::negotiate() virtual method was added, for consistency with
-        other base classes and to allow subclasses to completely
-        override the negotiation behaviour.
-    -   the new ::sink_event_pre_queue() and ::sink_query_pre_queue()
-        virtual methods allow subclasses to intercept or handle
-        serialized events and queries before they’re queued up
-        internally.
-
-GStreamer Plugins Base Libraries
-
-Audio library
-
--   audioaggregator, audiomixer: new output-buffer-duration-fraction
-    property which allows use cases such as keeping the buffers output
-    by compositor on one branch and audiomixer on another perfectly
-    aligned, by requiring the compositor to output a n/d frame rate, and
-    setting output-buffer-duration-fraction to d/n on the audiomixer.
-
--   GstAudioDecoder: new max-errors property so applications can
-    configure at what point the decoder should error out, or tell it to
-    just keep going
-
--   gst_audio_make_raw_caps() and gst_audio_formats_raw() are
-    bindings-friendly versions of the GST_AUDIO_CAPS_MAKE() C macro.
-
--   gst_audio_info_from_caps() now handles encoded audio formats as well
-
-PbUtils library
-
--   GstEncodingProfile:
-    -   Do not restrict number of similar profiles in a container
-    -   add GstValue serialization function
--   codec utils now support more H.264/H.265 profiles/levels and have
-    improved extension handling
-
-RTP library
-
--   rtpbasepayloader: Add scale-rtptime property for scaling RTP
-    timestamp according to the segment rate (equivalent to RTSP speed
-    parameter). This is useful for ONVIF trickmodes via RTSP.
-
--   rtpbasepayload: add experimental property for embedding twcc
-    sequencenumbers for Transport-Wide Congestion Control (gated behind
-    the GST_RTP_ENABLE_EXPERIMENTAL_TWCC_PROPERTY environment
-    variable) - more generic API for enabling this is expected to land
-    in the next development cycle.
-
--   rtcpbuffer: add RTPFB_TYPE_TWCC for Transport-Wide Congestion
-    Control
-
--   rtpbuffer: add
-    gst_rtp_buffer_get_extension_onebyte_header_from_bytes()``, so that one can parse theGBytes`
-    returned by gst_rtp_buffer_get_extension_bytes()
-
--   rtpbasedepayload: Add max-reorder property to make the
-    previously-hardcoded value when to consider a sender to have
-    restarted configurable. In some scenarios it’s particularly useful
-    to set max-reorder=0 to disable the behaviour that the depayloader
-    will drop packets: when max-reorder is set to 0 all
-    reordered/duplicate packets are considered coming from a restarted
-    sender.
-
-RTSP library
-
--   add gst_rtsp_url_get_request_uri_with_control() to create request
-    uri combined with control url
-
--   GstRTSPConnection: add the possibility to limit the Content-Length
-    for RTSP messages via
-    gst_rtsp_connection_set_content_length_limit(). The same
-    functionality is also exposed in gst-rtsp-server.
-
-SDP library
-
--   add support for parsing the extmap attribute from caps and storing
-    inside caps The extmap attribute allows mapping RTP extension header
-    IDs to well-known RTP extension header specifications. See RFC8285
-    for details.
-
-Tags library
-
--   update to latest iso-code and support more languages
-
--   add tags for acoustid id & acoustid fingerprint, plus MusicBrainz ID
-    handling fixes
-
-Video library
-
--   High Dynamic Range (HDR) video information representation and
-    signalling enhancements:
-
-    -   New APIs for HDR video information representation and
-        signalling:
-        -   GstVideoMasteringDisplayInfo: display color volume info as
-            per SMPTE ST 2086
-        -   GstVideoContentLightLevel: content light level specified in
-            CEA-861.3, Appendix A.
-        -   plus functions to serialise/deserialise and add them to or
-            parse them from caps
-        -   gst_video_color_{matrix,primaries,transfer}_{to,from}_iso():
-            new utilility functions for conversion from/to ISO/IEC
-            23001-8
-        -   add ARIB STD-B67 transfer chracteristic function
-        -   add SMPTE ST 2084 support and BT 2100 colorimetry
-        -   define bt2020-10 transfer characteristics for clarity:
-            bt707, bt2020-10, and bt2020-12 transfer characteristics are
-            functionally identical but have their own unique values in
-            the specification.
-    -   h264parse, h265parse: Parse mastering display info and content
-        light level from SEIs.
-    -   matroskademux: parse HDR metadata
-    -   matroskamux: Write MasteringMetadata and Max{CLL,FALL}. Enable
-        muxing with HDR meta data if upstream provided it
-    -   avviddec: Extract HDR information if any and map bt2020-10, PQ
-        and HLG transfer functions
-
--   added bt601 transfer function (for completeness)
-
--   support for more pixel formats:
-
-    -   Y412 (packed 12 bits 4:4:4:4)
-    -   Y212 (packed 12 bits 4:2:2)
-    -   P012 (semi-planar 4:2:0)
-    -   P016_{LE,BE} (semi-planar 16 bits 4:2:0)
-    -   Y444_16{LE,BE} (planar 16 bits 4:4:4)
-    -   RGB10A2_LE (packed 10-bit RGB with 2-bit alpha channel)
-    -   NV12_32L32 (NV12 with 32x32 tiles in linear order)
-    -   NV12_4L4 (NV12 with 4x4 tiles in linear order)
-
--   GstVideoDecoder:
-
-    -   new max-errors property so applications can configure at what
-        point the decoder should error out, or tell it to just keep
-        going
-
-    -   new qos property to disable dropping frames because of QoS, and
-        post QoS messages on the bus when dropping frames. This is
-        useful for example in a scenario where the decoded video is
-        tee-ed off to go into a live sink that syncs to the clock in one
-        branch, and an encoding and save to file pipeline in the other
-        branch. In that case one wouldn’t want QoS events from the video
-        sink make the decoder drop frames because that would also leave
-        gaps in the encoding branch then.
-
--   GstVideoEncoder:
-
-    -   gst_video_encoder_finish_subframe() is new API to push out
-        subframes (e.g. slices), so encoders can split the encoding into
-        subframes, which can be useful to reduce the overall end-to-end
-        latency as we no longer need to wait for the full frame to be
-        encoded to start decoding or sending out the data.
-    -   new min-force-key-unit-interval property allows configuring the
-        minimum interval between force-key-unit requests and prevents a
-        big bitrate increase if a lot of key-units are requested in a
-        short period of time (as might happen in live streaming RTP
-        pipelines when packet loss is detected).
-    -   various force-key-unit event handling fixes
-
--   GstVideoAggregator, compositor, glvideomixer: expose
-    max-last-buffer-repeat property on pads. This can be used to have a
-    compositor display either the background or a stream on a lower
-    zorder after a live input stream freezes for a certain amount of
-    time, for example because of network issues.
-
--   gst_video_format_info_component() is new API to find out which
-    components are packed into a given plane, which is useful to prevent
-    us from assuming a 1-1 mapping between planes and components.
-
--   gst_video_make_raw_caps() and gst_video_formats_raw() are
-    bindings-friendly versions of the GST_VIDEO_CAPS_MAKE() C macro.
-
--   video-blend: Add support for blending on top of 16 bit per component
-    formats, which makes sure we can support every currently supported
-    raw video format for blending subtitles or logos on top of video.
-
--   GST_VIDEO_BUFFER_IS_TOP_FIELD() and
-    GST_VIDEO_BUFFER_IS_BOTTOM_FIELD() convenience macros to check
-    whether the video buffer contains only the top field or bottom field
-    of an interlaced picture.
-
--   GstVideoMeta now includes an alignment field with the
-    GstVideoAlignment so buffer producers can explicitly specify the
-    exact geometry of the planes, allowing users to easily know the
-    padded size and height of each plane. Default values will be used if
-    this is not set.
-
-    Use gst_video_meta_set_alignment() to set the alignment and
-    gst_video_meta_get_plane_size() or gst_video_meta_get_plane_height()
-    to compute the plane sizes or plane heights based on the information
-    in the video meta.
-
--   gst_video_info_align_full() works like gst_video_info_align() but
-    also retrieves the plane sizes.
-
-MPEG-TS library
-
--   support for SCTE-35 sections
-
--   extend support for ATSC tables:
-
-    -   System Time Table (STT)
-    -   Master Guide Table (MGT)
-    -   Rating Region Table (RRT)
+-   this section will be filled in in due course
 
 Miscellaneous performance, latency and memory optimisations
 
-As always there have been many performance and memory usage improvements
-across all components and modules. Some of them have already been
-mentioned elsewhere so won’t be repeated here.
-
-The following list is only a small snapshot of some of the more
-interesting optimisations that haven’t been mentioned in other contexts
-yet:
-
--   caps negotiation, structure and GValue performance optimizations
-
--   systemclock: clock waiting performance improvements (moved from
-    GstPoll to GCond for waiting), especially on Windows.
-
--   rtpsession: add support for buffer lists on the recv path for better
-    performance with higher packet rate streams.
-
--   rtpjitterbuffer: internal timer handling has been rewritten for
-    better performance, see Nicolas’ talk “Revisiting RTP Jitter Buffer
-    Timers” for more details.
-
--   H.264/H.265 parsers and RTP payloaders/depayloaders have been
-    optimised for latency to make sure data is processed and pushed out
-    as quickly as possible
-
--   video-scaler: correctness and performance improvements, esp. for
-    interlaced formats and GBRA
-
--   GstVideoEncoder has gained new API to push out subframes
-    (e.g. slices), so encoders can split the encoding into subframes,
-    which can be useful to reduce the overall end-to-end latency as we
-    no longer need to wait for the full frame to be encoded to start
-    decoding or sending out the data.
-
-    This is complemented by the new GST_VIDEO_BUFFER_FLAG_MARKER which
-    is a video-specific buffer flag to mark the end of a video frame, so
-    elements can know that they have received all data for a frame
-    without waiting for the beginning of the next frame. This is similar
-    to how the RTP marker flag is used in many RTP video mappings.
-
-    The video encoder base class now also releases the internal stream
-    lock before pushing out data, so as to not block the input side of
-    things from processing more data in the meantime.
+-   this section will be filled in in due course
 
 Miscellaneous other changes and enhancements
 
--   it is now possible to modify the initial rank of plugin features
-    without modifying the source code or writing code to do so
-    programmatically via the GST_PLUGIN_FEATURE_RANK environment
-    variable. Users can adjust the rank of plugin(s) by passing a
-    comma-separated list of feature:rank pairs where rank can be a
-    numerical value or one of NONE, MARGINAL, SECONDARY, PRIMARY, and
-    MAX. Example: GST_PLUGIN_FEATURE_RANK=myh264dec:MAX,avdec_h264:NONE
-    sets the rank of the myh264dec element feature to the maximum and
-    that of avdec_h264 to 0 (none), thus ensuring that myh264dec is
-    prefered as H264 decoder in an autoplugging context.
-
--   GstDeviceProvider now does a static probe on start as fallback for
-    providers that don’t support dynamic probing to make things easier
-    for users
-
-WebRTC
-
--   webrtcbin now contains initial support for renegotiation involving
-    stream addition and removal. There are a number of caveats to this
-    initial renegotiation support and many complex scenarios are known
-    to require some work.
-
--   webrtcbin now exposes the internal ICE object for advanced
-    configuration options. Using the internal ICE object, it is possible
-    to toggle UDP or TCP connection usage as well as provide local
-    network addresses.
-
--   Fix a number of call flows within webrtcbin’s GstPromise handling
-    where a promise was never replied to. This has been fixed and now a
-    promise will always receive a reply.
-
--   webrtcbin now exposes a latency property for configuring the
-    internal rtpjitterbuffer latency and buffering when receiving
-    streams.
-
--   webrtcbin now only synchronises the RTP part of a stream, allowing
-    RTCP messages to skip synchronisation entirely.
-
--   Fixed most of the webrtcbin state properties (connection-state,
-    ice-connection-state, signaling-state, but not ice-gathering-state
-    as that requires newer API in libnice and will be fixed in the next
-    release series) to advance through the state values correctly. Also
-    implemented DTLS connection states in the DTLS elements so that
-    peer-connection-state is not always new.
-
--   webrtcbin now accounts for the a=ice-lite attribute in a remote SDP
-    offer and will configure the internal ICE implementation
-    accordingly.
-
--   webrtcbin will now resolve .local candidate addresses using the
-    system DNS resolver. .local candidate addresses are now produced by
-    web browsers to help protect the privacy of users.
-
--   webrtcbin will now add candidates found in the SDP to the internal
-    ICE agent. This was previously unsupported and required using the
-    add-ice-candidate signal manually from the application.
-
--   webrtcbin will now correctly parse a TURN URI that contains a
-    username or password with a : in it.
-
--   The GStreamer WebRTC library gained a GstWebRTCDataChannel object
-    roughly matching the interface exposed by the WebRTC specification
-    to allow for easier binding generation and use of data channels.
-
-OpenGL integration
-
-GStreamer OpenGL bindings/build related changes
-
--   The GStreamer OpenGL library (libgstgl) now ships pkg-config files
-    for platform-specific API where libgstgl provides a public
-    integration interface and a pkg-config file for a dependency on the
-    detected OpenGL headers. The new list of pkg-config files in
-    addition to the original gstreamer-gl-1.0 are gstreamer-gl-x11-1.0,
-    gstreamer-gl-wayland-1.0, gstreamer-gl-egl-1.0, and
-    gstreamer-gl-prototypes-1.0 (for OpenGL headers when including
-    gst/gl/gstglfuncs.h).
-
--   GStreamer OpenGL now ships some platform-specific introspection data
-    for platforms that have a public interface. This should allow for
-    easier integration with bindings involving platform specific
-    functionality. The new introspection data files are named
-    GstGLX11-1.0, GstGLWayland-1.0, and GstGLEGL-1.0.
-
-GStreamer OpenGL Features
-
--   The iOS implementation no longer accesses UIKit objects off the main
-    thread fixing a loud warning message when used in iOS applications.
-
--   Support for mouse and keyboard handling using the GstNavigation
-    interface was added for the wayland implementation complementing the
-    already existing support for the X11 and Windows implementations.
-
--   A new helper base class for source elements, GstGLBaseSrc is
-    provided to ease writing source elements producing OpenGL video
-    frames.
-
--   Support for some more 12-bit and 16-bit video formats (Y412_LE,
-    Y412_BE, Y212_LE, Y212_BE, P012_LE, P012_BE, P016, NV16, NV61) was
-    added to glcolorconvert.
-
--   glupload can now import dma-buf’s into external-oes textures.
-
--   A new display type for EGLDevice-based systems was added. It is
-    currently opt-in by using either the GST_GL_PLATFORM=egl-device
-    environment variable or manual construction
-    (gst_gl_display_egl_device_new*()) due to compatibility issues with
-    some platforms.
-
--   Support was added for WinRT/UWP using the ANGLE project for running
-    OpenGL-based pipelines within a UWP application.
-
--   Various elements now support changing the GstGLDisplay to be used at
-    runtime in simple cases. This is primarily helpful for changing or
-    adding an OpenGL-based video sink that must share an OpenGL context
-    with an external source to an already running pipeline.
-
-GStreamer Vulkan integration
-
--   There is now a GStreamer Vulkan library to provide integration
-    points and helpers with applications and external GStreamer Vulkan
-    based elements. The structure of the library is modelled similarly
-    to the already existing GStreamer OpenGL library. Please note that
-    the API is still unstable and may change in future releases,
-    particularly around memory handling. The GStreamer Vulkan library
-    contains objects for sharing the vkInstance, vkDevice, vkQueue,
-    vkImage, VkMemory, etc with other elements and/or the application as
-    well as some helper objects for using Vulkan in an application or
-    element.
-
--   Added support for building and running on/for the Android and
-    Windows systems to complement the existing XCB, Wayland, MacOS, and
-    iOS implementations.
-
--   XCB gained support for mouse/keyboard events using the GstNavigation
-    API.
-
--   New vulkancolorconvert element for converting between color formats.
-    vulkancolorconvert can currently convert to/from all 8-bit RGBA
-    formats as well as 8-bit RGBA formats to/from the YUV formats AYUV,
-    NV12, and YUY2.
-
--   New vulkanviewconvert element for converting between stereo view
-    layouts. vulkanviewconvert can currently convert between all of the
-    single memory formats (side-by-side, top-bottom, column-interleaved,
-    row-interleaved, checkerboard, left, right, mono).
-
--   New vulkanimageidentity element for a blit from the input vulkan
-    image/s to a new vulkan image/s.
-
--   The vulkansink element can now scale the input image to the output
-    window/surface size where that information is available.
-
--   The vulkanupload element can now configure a transfer from system
-    memory to VulkanImage-based memory. Previously, this required two
-    vulkanupload elements.
+-   this section will be filled in in due course
 
 Tracing framework and debugging improvements
 
--   gst_tracing_get_active_tracers() returns a list of active tracer
-    objects. This can be used to interact with tracers at runtime using
-    GObject API such as action signals. This has been implemented in the
-    leaks tracer for snapshotting and retrieving leaked/active objects
-    at runtime.
-
--   The leaks tracer can now be interacted with programmatically at
-    runtime via GObject action signals:
-
-    -   get-live-object returns a list of live (allocated) traced
-        objects
-    -   log-live-objects logs a list of live objects into the debug log.
-        This is the same as sending the SIGUSR1 signal on unix systems,
-        but works on all operating systems including Windows.
-    -   activity-start-tracking, activity-get-checkpoint,
-        activity-log-checkpoint, activity-stop-tracking: add support for
-        tracking and checkpointing objects, similar to what was
-        previously available via SIGUSR2 on unix systems, but works on
-        all operating systems including Windows.
-
--   various GStreamer gdb debug helper improvements:
-
-    -   new ‘gst-pipeline-tree’ command
-    -   more gdb helper functions: gst_element_pad(), gst_pipeline() and
-        gst_bin_get()
-    -   support for queries and buffers
-    -   print more info for segment events, print event seqnums, object
-        pointers and structures
-    -   improve gst-print command to show more pad and element
-        information
+-   this section will be filled in in due course
 
 Tools
 
-gst-launch-1.0
-
--   now prints the pipeline position and duration if available when the
-    pipeline is advancing. This is hopefully more user-friendly and
-    gives visual feedback on the terminal that the pipeline is actually
-    up and running. This can be disabled with the --no-position command
-    line option.
-
--   the parse-launch pipeline syntax now has support for presets:
-    use@preset=<preset-name>" after an element to load a preset.
-
-gst-inspect-1.0
-
--   new --color command line option to force coloured output even if not
-    connected to a tty
-
-gst-tester-1.0 (new)
-
--   gst-tester-1.0 is a new tool for plugin developers to launch
-    .validatetest files with TAP compatible output, meaning it can
-    easily and cleanly be integrated with the meson test harness. It
-    allows you to use gst-validate (from the gst-devtools module) to
-    write integration tests in any GStreamer repository whilst keeping
-    the tests as close as possible to the code. The tool transparently
-    handles gst-validate being installed or not: if it is not installed
-    those integration tests will simply be skipped.
-
-gst-play-1.0
-
--   interactive keyboard controls now also work on Windows
-
-gst-transcoder-1.0 (new)
-
--   gst-transcoder-1.0 is a new command line tool to transcode one URI
-    into another URI based on the specified encoding profile using the
-    new GstTranscoder API (see above).
+-   this section will be filled in in due course
 
 GStreamer RTSP server
 
--   Fix issue where the first few packets (i.e. keyframes) could
-    sometimes be dropped if the rtsp media pipeline had a live input.
-    This was a regression from GStreamer 1.14. There are more fixes
-    pending for that which will hopefully land in 1.18.1.
-
--   Fix backpressure handling when sending data in TCP interleave mode
-    where RTSP requests and responses and RTP/RTCP packets flow over the
-    same RTSP TCP connection: The previous implementation would at some
-    point stop sending data to other clients when a single client
-    stopped consuming data or did not consume data fast enough. This
-    obviously created problems for shared media, where the same stream
-    from a single producer pipeline is sent to multiple clients. Instead
-    we now manage a backlog in the server’s stream-transport component
-    and remove slow clients once this backlog exceeds a maximum duration
-    (which is currently hardcoded).
-
--   Onvif Streaming Specification trick modes support (see section at
-    the beginning)
-
--   Scale/Speed header support: Speed will deliver the data at the
-    requested speed, which means increasing the data bandwidth for
-    speeds > 1.0. Scale will attempt to do the same without affecting
-    the overall bandwidth requirement vis-a-vis normal playback speed
-    (e.g. it might drop data for fast-forward playback).
-
--   rtspclientsink: send buffer lists in one go for better performance
+-   this section will be filled in in due course
 
 GStreamer VAAPI
 
--   A lot of work was done adding support for media-driver (iHD), the
-    new VAAPI driver for Intel, mostly for Gen9 onwards.
-
--   Available color formats and frame sizes are now detected at run-time
-    according to the context configuration.
-
--   Gallium drivers have been re-enabled in the allowed drivers list
-
--   Improved the mapping between VA formats and GStreamer formats by
-    generating a mapping table at run-time since even among different
-    drivers the mapping might be different, particularly for RGB with
-    little endianness.
-
--   The experimental Flexible Encoding Infrastructure (FEI) elements
-    have been removed since they were not really actively maintained or
-    tested.
-
--   Enhanced the juggling of DMABuf buffers and VASurface metas
-
--   New vaapioverlay element: a compositor element using VA VPP blend
-    capabilities to accelerate overlaying and compositing. Example
-    pipeline:
-
-          gst-launch-1.0 -vf videotestsrc ! vaapipostproc ! tee name=testsrc ! queue \
-          ! vaapioverlay sink_1::xpos=300 sink_1::alpha=0.75 name=overlay ! vaapisink \
-          testsrc. ! queue ! overlay.
-
-vaapipostproc
-
--   added video-orientation support, supporting frame mirroring and
-    rotation
-
--   added cropping support, either via properties (crop-left,
-    crop-right, crop-bottom and crop-top) or buffer meta.
-
--   new skin-tone-enhancenment-level property which is the iHD
-    replacement of the i965 driver’s sink-tone-level. Both are
-    incompatible with each other, so both were kept.
-
--   handle video colorimetry
-
--   support HDR10 tone mapping
-
-vaapisink
-
--   resurrected wayland backend for non-weston compositors by extracting
-    the DMABuf from the VASurface and rendering it.
-
--   merged the video overlay API for wayland. Now applications can
-    define the “window” to render on.
-
--   demoted the vaapisink element to secondary rank since libva
-    considers rendering as a second-class feature.
-
-VAAPI Encoders
-
--   new common target-percentage property which is the desired target
-    percentage of bitrate for variable rate control.
-
--   encoders now extract their caps from the driver at registration
-    time.
-
--   vaapivp9enc: added support for low power mode and support for
-    profile 2 (profile 0 by default)
-
--   vaapih264enc: new max-qp property that sets the maximum quantization
-    value. Support for ICQ and QBVR bitrate control mode, adding a
-    quality-factor property for these modes. Support baseline profile as
-    constrained-baseline
-
--   vaapih265enc:
-
-    -   support for main-444 and main-12 encoding profiles.
-    -   new max-qp property that sets the maximum quantization value.
-    -   support for ICQ and QBVR bitrate control mode, adding a
-        quality-factor property for these modes.
-    -   handle SCC profiles.
-    -   num-tile-cols and num-tile-row properties to specify the number
-        of tiles to use.
-    -   the low-delay-b property was deprecated and is now determined
-        automatically.
-    -   improved profile selection through caps.
-
-VAAPI Decoders
-
--   Decoder surfaces are not bound to their context any longer and can
-    thus be created and used dynamically, removing the deadlock
-    headache.
-
--   Reverse playback is now fluid
-
--   Forward Region-of-Interest (ROI) metas downstream
-
--   GLTextureUploadMeta uses DMABuf when GEM is not available. Now
-    Gallium drivers can use this meta for rendering with EGL.
-
--   vaapivp9dec: support for 4:2:2 and 4:4:4 chroma type streams
-
--   vaapih265dec: skip all pictures prior to the first I-frame. Enable
-    passing range extension flags to the driver. Handle SCC profiles.
-
--   vaapijpegdec: support for 4:0:0, 4:1:1, 4:2:2 and 4:4:4 chroma types
-    pictures
-
--   vaapih264dec: handle baseline streams as constrained-baseline if
-    possible and make it more tolerant when encountering unknown NALs
+-   this section will be filled in in due course
 
 GStreamer OMX
 
--   omxvideoenc: use new video encoder subframe API to push out slices
-    as soon as they’re ready
-
--   omxh264enc, omxh265enc: negotiate subframe mode via caps. To enable
-    it, force downstream caps to video/x-h264,alignment=nal or
-    video/x-h265,alignment=nal.
-
--   omxh264enc: Add ref-frames property
-
--   Zynq ultrascale+ specific video encoder/decoder improvements:
-
-    -   GRAY8 format support
-    -   support for alternate fields interlacing mode
-    -   video encoder: look-ahead, long-term-ref, and long-term-freq
-        properties
+-   this section will be filled in in due course
 
 GStreamer Editing Services and NLE
 
--   Added nested timelines and subproject support so that GES projects
-    can be used as clips, potentially serializing nested projects in the
-    main file or referencing external project files.
-
--   Implemented an OpenTimelineIO GES formatter. This means GES and
-    GStreamer can now load and save projects in all the formats
-    supported by otio.
-
--   Implemented a GESMarkerList object which allow setting timed
-    metadata on any GES object.
-
--   Fixed audio rendering issues during clip transition by ensuring that
-    a single segment is pushed into encoders.
-
--   The GESUriClipAsset API is now MT safe.
-
--   Added ges_meta_container_register_static_meta() to allow fixing a
-    type for a specific metadata without actually setting a value.
-
--   The framepositioner element now handles resizing the project and
-    keeps the same positioning when the aspect ratio is not changed .
-
--   Reworked the documentation, making it more comprehensive and much
-    more detailed.
-
--   Added APIs to retrieve natural size and framerate of a clip (for
-    example in the case of URIClip it is the framerate/size of the
-    underlying file).
-
--   ges_container_edit() is now deprecated and GESTimelineElement gained
-    the ges_timeline_element_edit() method so the editing API is now
-    usable from any element in the timeline.
-
--   GESProject::loading was added so applications can be notified about
-    when a new timeline starts loading.
-
--   Implemented the GstStream API in GESTimeline.
-
--   Added a way to add a timeoverlay inside the test source (potentially
-    with timecodes).
-
--   Added APIs to convert times to frame numbers and vice versa:
-
-    -   ges_timeline_get_frame_time()
-
-    -   ges_timeline_get_frame_at()
-
-    -   ges_clip_asset_get_frame_time()
-
-    -   ges_clip_get_timeline_time_from_source_frame()
-
-        Quite a few validate tests have been implemented to check the
-        behavior for various demuxer/codec formats
-
--   Added ges_layer_set_active_for_tracks() which allows muting layers
-    for the specified tracks
-
--   Deprecated GESImageSource and GESMultiFileSource now that we have
-    imagesequencesrc which handles the imagesequence “protocol”
-
--   Stopped exposing ‘deinterlacing’ children properties for clip types
-    where they do not make sense.
-
--   Added support for simple time remapping effects
+-   this section will be filled in in due course
 
 GStreamer validate
 
--   Introduced the concept of “Test files” allowing to implement “all
-    included” test cases, meaning that inside the file the following can
-    be defined:
-
-    -   The application arguments
-    -   The validate configurations
-    -   The validate scenario
-
-    This replaces the previous big dictionary file in
-    gst-validate-launcher to implement specific test cases.
-
-    We set several variables inside the files (as well as inside
-    scenarios and config files) to make them relocatable.
-
-    The file format has been enhanced so it is easier to read and write,
-    for example line ending with a coma or (curly) brackets can now be
-    used as continuation marker so you do not need to add \ at the end
-    of lines to write a structure on several lines.
-
--   Support the imagesequence “protocol” and added integration tests for
-    it.
-
--   Added action types to allow the scenario to run the Test Clock for
-    better reproducibility of tests.
-
--   Support generating tests to check that seeking is frame accurate
-    (base on ssim).
-
--   Added ways to record buffers checksum (in different ways) in the
-    validateflow module.
-
--   Added vp9 encoding tests.
-
--   Enhanced seeking action types implementation to allow support for
-    segment seeks.
-
--   Output improvements:
-
-    -   Logs are now in markdown formats (and bat is used to dump them
-        if available).
-    -   File format issues in scenarios/configs/tests files are nicely
-        reported with the line numbers now.
+-   this section will be filled in in due course
 
 GStreamer Python Bindings
 
--   Python 2.x is no longer supported
-
--   Support mapping buffers without any memcpy:
-
-    -   Added a ContextManager to make the API more pythonic
-
-            with buf.map(Gst.MapFlags.READ | Gst.MapFlags.WRITE) as info:
-                info.data[42] = 0
-
--   Added high-level helper API for constructing pipelines:
-
-    -   Gst.Bin.make_and_add(factory_name, instance_name=None)
-    -   Gst.Element.link_many(element, ...)
+-   this section will be filled in in due course
 
 GStreamer C# Bindings
 
--   Bind gst_buffer_new_wrapped() manually to fix memory handling.
-
--   Fix gst_promise_new_with_change_func() where bindgen didn’t properly
-    detect the func as a closure.
-
--   Declare GstVideoOverlayComposition and GstVideoOverlayRectangle as
-    opaque type and subclasses of Gst.MiniObject. This changes the API
-    but without this all usage will cause memory corruption or simply
-    not work.
-
--   on Windows, look for gstreamer, glib and gobject DLLs using the MSVC
-    naming convention (i.e. gstvideo-1.0-0.dll instead of
-    libgstvideo-1.0-0.dll).
-
-    The names of these DLLs have to be hardcoded in the bindings, and
-    most C# users will probably be using the Microsoft toolchain anyway.
-
-    This means that the MSVC compiler is now required to build the
-    bindings, MingW will no longer work out of the box.
+-   this section will be filled in in due course
 
 GStreamer Rust Bindings and Rust Plugins
 
 The GStreamer Rust bindings are released separately with a different
 release cadence that’s tied to gtk-rs, but the latest release has
-already been updated for the new GStreamer 1.18 API, so there’s
-absolutely no excuse why your next GStreamer application can’t be
-written in Rust anymore.
+already been updated for the upcoming new GStreamer 1.20 API.
 
 gst-plugins-rs, the module containing GStreamer plugins written in Rust,
 has also seen lots of activity with many new elements and plugins.
@@ -1659,6 +119,8 @@ What follows is a list of elements and plugins available in
 gst-plugins-rs, so people don’t miss out on all those potentially useful
 elements that have no C equivalent.
 
+-   FIXME: add new elements
+
 Rust audio plugins
 
 -   audiornnoise: New element for audio denoising which implements the
@@ -1724,73 +186,11 @@ Generic Rust plugins
 
 Build and Dependencies
 
--   The Autotools build system has finally been removed in favour of the
-    Meson build system. Developers who currently use gst-uninstalled
-    should move to gst-build.
-
--   API and plugin documentation are no longer built with gtk_doc. The
-    gtk_doc documentation has been removed in favour of a new unified
-    documentation module built with hotdoc (also see “Documentation
-    improvements” section below). Distributors should use the
-    documentation release tarball instead of trying to package hotdoc
-    and building the documentation from scratch.
-
--   gst-plugins-bad now includes an internal copy of libusrsctp, as
-    there are problems in usrsctp with global shared state, lack of API
-    stability guarantees, and the absence of any kind of release
-    process. We also can’t rely on distros shipping a version with the
-    fixes we need. Both firefox and Chrome bundle their own copies too.
-    It is still possible to build against an external copy of usrsctp if
-    so desired.
-
--   nvcodec no longer needs the NVIDIA NVDEC/NVENC SDKs available at
-    build time, only at runtime. This allows distributions to ship this
-    plugin by default and it will just start to work when the required
-    run-time SDK libraries are installed by the user, without users
-    needing to build and install the plugin from source.
-
--   the gst-editing-services tarball is now named gst-editing-services
-    for consistency (used to be gstreamer-editing-services).
-
--   the gst-validate tarball has been superseded by the gst-devtools
-    tarball for consistency with the git module name.
+-   this section will be filled in in due course
 
 gst-build
 
-gst-build is a meta-module and serves primarily as our uninstalled
-development environment. It makes it easy to build most of GStreamer,
-but unlike Cerbero it only comes with a limited number of external
-dependencies that can be built as subprojects if they are not found on
-the system.
-
-gst-build is based on Meson and replaces the old autotools
-gst-uninstalled script.
-
--   The ‘uninstalled’ target has been renamed to ‘devenv’
-
--   Experimental gstreamer-full library containing all built plugins and
-    their deps when building with -Ddefault_library=static. A monolithic
-    library is easier to distribute, and may be required in some
-    environments. GStreamer core, GLib and GObject are always included,
-    but external dependencies are still dynamically linked. The
-    gst-full-libraries meson option allows adding other GStreamer
-    libraries to the gstreamer-full build. This is an experiment for now
-    and its behaviour or API may still change in future releases.
-
--   Add glib-networking as a subproject when glib is a subproject and
-    load gio modules in the devenv, tls option control whether to use
-    openssl or gnutls.
-
--   git-worktree: Allow multiple worktrees for subproject branches
-
--   Guard against meson being run from inside the uninstalled devenv, as
-    this might have unexpected consequences.
-
--   our ffmpeg and x264 meson ports have been updated to the latest
-    stable version (you might need to update the subprojects checkout
-    manually though, or just remove the checkouts so meson checks out
-    the latest version again; improvements for this are pending in
-    meson, but not merged yet).
+-   this section will be filled in in due course
 
 Cerbero
 
@@ -1800,405 +200,97 @@ Windows, Android, iOS and macOS.
 
 General improvements
 
--   Recipe build steps are done in parallel wherever possible. This
-    leads to massive improvements in overall build time.
--   Several recipes were ported to Meson, which improved build times
--   Moved from using both GnuTLS and OpenSSL to only OpenSSL
--   Moved from yasm to nasm for all assembly compilation
--   Support zsh when running the cerbero shell command
--   Numerous version upgrades for dependencies
--   Default to xz for tarball binary packages. bz2 can be selected with
-    the --compress-method option to package.
--   Added boolean variant for controlling the optimization level:
-    -v optimization
--   Ship .pc pkgconfig files for all plugins in the binary packages
--   CMake and nasm will only be built by Cerbero if the system versions
-    are unusable
--   The nvcodec variant was removed and the nvcodec plugin is built by
-    default now (as it no longer requires the SDK to be installed at
-    build time, only at runtime)
+-   this section will be filled in in due course
 
 macOS / iOS
 
--   Minimum iOS SDK version bumped to 11.0
--   Minimum macOS SDK version bumped to 10.11
--   No longer need to manually add support for newer iOS SDK versions
--   Added Vulkan elements via MoltenVK
--   Build times were improved by code-signing all build tools
--   macOS framework ships all gstreamer libraries instead of an outdated
-    subset
--   Ship pkg-config in the macOS framework package
--   fontconfig: Fix EXC_BAD_ACCESS crash on iOS ARM64
--   Improved App Store compatibility by setting LC_VERSION_MIN_MACOSX,
-    fixing relocations, and improved bitcode support
+-   this section will be filled in in due course
 
 Windows
 
--   MinGW-GCC toolchain was updated to 8.2. It uses the Universal CRT
-    instead of MSVCRT which eliminates cross-CRT issues in the Visual
-    Studio build.
--   Require Windows 7 or newer for running binaries produced by Cerbero
--   Require Windows x86_64 for running Cerbero to build binary packages
--   Cerbero no longer uses C:/gstreamer/1.0 as a prefix when building.
-    That prefix is reserved for use by the MSI installers.
--   Several recipes can now be buit with Visual Studio instead of MinGW.
-    Ported to meson: opus, libsrtp, harfbuzz, cairo, openh264, libsoup,
-    libusrsctp. Existing build system: libvpx, openssl.
--   Support building using Visual Studio for 32-bit x86. Previously we
-    only supported building for 32-bit x86 using the MinGW toolchain.
--   Fixed annoying msgmerge popups in the middle of cerbero builds
--   Added configuration options vs_install_path and vs_install_version
-    for specifying custom search locations for older Visual Studio
-    versions that do not support vswhere. You can set these in
-    ~/.cerbero/cerbero.cbc where ~ is the MSYS homedir, not your Windows
-    homedir.
--   New Windows-specific plugins: d3d11, mediafoundation, wasapi2
--   Numerous compatibility and reliability fixes when running Cerbero on
-    Windows, especially non-English locales
--   proxy-libintl now exports the same symbols as gettext, which makes
-    it a drop-in replacement
--   New mapping variant for selecting the Visual Studio CRT to use:
-    -v vscrt=<value>. Valid values are md, mdd, and auto (default). A
-    separate prefix is used when building with either md (release) or
-    mdd (debug), and the outputted package will have +debug in the
-    filename. This variant is also used for selecting the correct Qt
-    libraries (debug vs release) to use when building with -v qt5 on
-    Windows.
--   Support cross-compile on Windows to Windows ARM64 and ARMv7
--   Support cross-compile on Windows to the Universal Windows Platform
-    (UWP). Only the subset of plugins that can be built entirely with
-    Visual Studio will be selected in this case. To do so, use the
-    config/cross-uwp-universal.cbc configuration, which will build
-    ARM64, x86, and x86_64 binaries linked to the release CRT, with
-    optimizations enabled, and debugging turned on. You can combine this
-    with -v vscrt=mdd to produce binaries linked to the debug CRT. You
-    can turn off optimizations with the -v nooptimization variant.
+-   this section will be filled in in due course
 
 Windows MSI installer
 
--   Require Windows 7 or newer for running GStreamer
--   Fixed some issues with shipping of pkg-config in the Windows
-    installers
--   Plugin PDB debug files are now shipped in the development package,
-    not the runtime package
--   Ship installers for 32-bit binaries built with Visual Studio
--   Ship debug and release “universal” (ARM64, X86, and X86_64) tarballs
-    built for the Universal Windows Platform
--   Windows MSI installers now install into separate prefixes when
-    building with MSVC and MinGW. Previously both would be installed
-    into C:/gstreamer/1.0/x86 or C:/gstreamer/1.0/x86_64. Now, the
-    installation prefixes are:
-
-  ----------------------------------------------------------------------------------------------------------------
-  Target                      Path                                 Build options
-  --------------------------- ------------------------------------ -----------------------------------------------
-  MinGW 32-bit                C:/gstreamer/1.0/mingw_x86           -c config/win32.cbc
-
-  MinGW 64-bit                C:/gstreamer/1.0/mingw_x86_64        -c config/win64.cbc
-
-  MSVC 32-bit                 C:/gstreamer/1.0/msvc_x86            -c config/win32.cbc -v visualstudio
-
-  MSVC 64-bit                 C:/gstreamer/1.0/msvc_x86_64         -c config/win64.cbc -v visualstudio
-
-  MSVC 32-bit (debug)         C:/gstreamer/1.0/msvc-debug_x86      -c config/win32.cbc -v visualstudio,vscrt=mdd
-
-  MSVC 64-bit (debug)         C:/gstreamer/1.0/msvc-debug_x86_64   -c config/win64.cbc -v visualstudio,vscrt=mdd
-  ----------------------------------------------------------------------------------------------------------------
-
-Note: UWP binary packages are tarballs, not MSI installers.
+-   this section will be filled in in due course
 
 Linux
 
--   Support creating MSI installers using WiX when cross-compiling to
-    Windows
--   Support running cross-windows binaries with Wine when using the
-    shell and runit cerbero commands
--   Added bash-completion support inside the cerbero shell on Linux
--   Require a system-wide installation of openssl on Linux
--   Added variant -v vaapi to build gstreamer-vaapi and the new gstva
-    plugin
--   Debian packaging was disabled because it does not work. Help in
-    fixing this is appreciated.
--   Trimmed the list of packages needed for bootstrap on Linux
+-   this section will be filled in in due course
 
 Android
 
--   Updated to NDK r21
--   Support Vulkan
--   Support Qt 5.14+ binary package layout
+-   this section will be filled in in due course
 
 Platform-specific changes and improvements
 
 Android
 
--   opensles: Remove hard-coded buffer-/latency-time values and allow
-    openslessink to handle 48kHz streams.
-
--   photography interface and camera source: Add additional settings
-    relevant to Android such as: Exposure mode property, extra colour
-    tone values (aqua, emboss, sketch, neon), extra scene modes
-    (backlight, flowers, AR, HDR), and missing virtual methods for
-    exposure mode, analog gain, lens focus, colour temperature, min &
-    max exposure time. Add new effects and scene modes to Camera
-    parameters.
+-   this section will be filled in in due course
 
 macOS and iOS
 
--   vtdec can now output to Vulkan-backed memory for zerocopy support
-    with the Vulkan elements.
+-   this section will be filled in in due course
 
 Windows
 
--   d3d11videosink: new Direct3D11-based video sink with support for
-    HDR10 rendering if supported.
-
--   Hardware-accelerated video decoding on Windows via DXVA2 /
-    Direct3D11 using native Windows APIs rather than per-vendor SDKs
-    (like MSDK for Intel or NVCODEC for NVidia). Plus modern Direct3D11
-    integration rather than the almost 20-year old Direct3D9 from
-    Windows XP times used in d3dvideosink. Formats supported for
-    decoding are H.264, H.265, VP8, and VP9, and zero-copy operation
-    should be supported in combination with the new d3d11videosink. See
-    Seungha’s blog post “Windows DXVA2 (via Direct3D 11) Support in
-    GStreamer 1.17” for more details.
-
--   Microsoft Media Foundation plugin for hardware-accelerated video
-    encoding on Windows using native Windows APIs rather than per-vendor
-    SDKs. Formats supported for encoding are H.264, H.265 and VP9. Also
-    includes audio encoders for AAC and MP3. See Seungha’s blog post
-    “Bringing Microsoft Media Foundation to GStreamer” for some more
-    details about this.
-
--   new mfvideosrc video capture source element using the latest Windows
-    APIs rather than ancient APIs used by ksvideosrc/winks. ksvideosrc
-    should be considered deprecated going forward.
-
--   d3d11: add d3d11convert, a color space conversion and rescaling
-    element using shaders, and introduce d3d11upload and d3d11download
-    elements that work just like glupload and gldownload but for D3D11.
-
--   Universal Windows Platform (UWP) support, including official
-    GStreamer binary packages for it. Check out Nirbheek’s latest blog
-    post “GStreamer 1.18 supports the Universal Windows Platform” for
-    more details.
-
--   systemclock correctness and reliability fixes, and also don’t start
-    the system clock at 0 any longer (which shouldn’t make any
-    difference to anyone, as absolute clock time values are supposed to
-    be meaningless in themselves, only the rate of increase matters).
-
--   toolchain specific plugin registry: the registry cache is now named
-    differently for MSVC and MinGW toolchains/packages, which should
-    avoid problems when switching between binaries built with a
-    different toolchain.
-
--   new wasapi2 plugin mainly to support UWP applications. The core
-    logic of this plugin is almost identical to existing wasapi plugin,
-    but the main target is Windows 10 and UWP. This plugin uses WinRT
-    APIs, so will likely not work on Windows 8 or older. Unlike the
-    existing wasapi plugin, this plugin supports automatic stream
-    routing (auto fallback when device was removed) and device level
-    mute/volume control. Exclusive streaming mode is not supported,
-    however, and loopback features are not implemented yet. It is also
-    only possible to build this plugin with MSVC and the Windows 10 SDK,
-    it can’t be cross-compiled with the MingW toolchain.
-
--   new dxgiscreencapsrc element which uses the Desktop Duplication API
-    to capture the desktop screen at high speed. This is only supported
-    on Windows 8 or later. Compared to the existing elements
-    dxgiscreencapsrc offers much better performance, works in High DPI
-    environments and draws an accurate mouse cursor.
-
--   d3dvideosink was downgraded to secondary rank, d3d11videosink is
-    preferred now. Support OverlayComposition for GPU overlay
-    compositing of subtitles and logos.
-
--   debug log output fixes, esp. with a non-UTF8 locale/codepage
-
--   speex, jack: fixed crashes on Windows caused by cross-CRT issues
-
--   gst-play-1.0 interactive keyboard controls now also work on Windows
+-   this section will be filled in in due course
 
 Linux
 
--   kmssink: Add support for P010 and P016 formats
-
--   vah264dec: new experimental va plugin with an element for H.264
-    decoding with VA-API. This novel approach, different from
-    gstreamer-vaapi, uses the gstcodecs library for decoder state
-    handling, which it is hoped will make for cleaner code because it
-    uses VA-API without further layers or wrappers. Check out Víctor’s
-    blog post “New VA-API H.264 decoder in gst-plugins-bad” for the full
-    lowdown and the limitations of this new plugin, and how to give it a
-    spin.
-
--   v4l2codecs: introduce a V4L2 CODECs Accelerator. This plugin will
-    support the new CODECs uAPI in the Linux kernel, which consists of
-    an accelerator interface similar to DXVA, NVDEC, VDPAU and VAAPI. So
-    far H.264 and VP8 are supported. This is used on certain embedded
-    systems such as i.mx8m, rk3288, rk3399, Allwinner H-series SoCs.
+-   this section will be filled in in due course
 
 Documentation improvements
 
--   unified documentation containing tutorials, API docs, plugin docs,
-    etc. all under one roof, shipped in form of a documentation release
-    tarball containing both devhelp and html documentation.
-
--   all documentation is now generated using hotdoc, gtk-doc is no
-    longer used. Distributors should use the above-mentioned
-    documentation release tarball instead of trying to package hotdoc
-    and building the documentation from scratch.
-
--   there is now documentation for wrapper plugins like gst-libav and
-    frei0r, as well as tracer plugins.
-
--   for more info, check out Thibault’s “GStreamer Documentation”
-    lightning talk from the 2019 GStreamer Conference.
-
--   new API for plugins to support the documentation system:
-
-    -   new GParamSpecFlag GST_PARAM_DOC_SHOW_DEFAULT to make
-        gst-inspect-1.0 (and the documentation) show the paramspec’s
-        default value rather than the actually set value as default
-    -   GstPadTemplate getter and setter for “documentation caps”,
-        gst_pad_template_set_documentation_caps() and
-        gst_pad_template_get_documentation_caps(): This can be used in
-        elements where the caps of pad templates are dynamically
-        generated and/or dependent on the environment, to override the
-        caps shown in the documentation (usually to advertise the full
-        set of possible caps).
-    -   gst_type_mark_as_plugin_api() for marking types as plugin API,
-        used for plugin-internal types like enums, flags, pad
-        subclasses, boxed types, and such.
+-   this section will be filled in in due course
 
 Possibly Breaking Changes
 
--   GstVideo: the canonical list of raw video formats (for use in caps)
-    has been reordered, so video elements such as videotestsrc or
-    videoconvert might negotiate to a different format now than before.
-    The new format might be a higher-quality format or require more
-    processing overhead, which might affect pipeline performance.
-
--   mpegtsdemux used to wrongly advertise H.264 and H.265 video
-    elementary streams as alignment=nal. This has now been fixed and
-    changed to alignment=none, which means an h264parse or h265parse
-    element is now required after tsdemux for some pipelines where there
-    wasn’t one before, e.g. in transmuxing scenarios (tsdemux ! tsmux).
-    Pipelines without such a parser may now fail to link or error out at
-    runtime. As parsers after demuxers and before muxers have been
-    generally required for a long time now it is hoped that this will
-    only affect a small number of applications or pipelines.
-
--   The Android opensles audio source and sink used to have hard-coded
-    buffer-/latency-time values of 20ms. This is no longer needed with
-    newer Android versions and has now been removed. This means a higher
-    or lower value might now be negotiated by default, which can affect
-    pipeline performance and latency.
+-   this section will be filled in in due course
 
 Known Issues
 
--   None in particular
+-   this section will be filled in in due course
+
+-   There are a couple of known WebRTC-related regressions/blockers:
+
+    -   webrtc: DTLS setup with Chrome is broken
+    -   webrtcbin: First keyframe is usually lost
 
 Contributors
 
-Aaron Boxer, Adam Duskett, Adam x Nilsson, Adrian Negreanu, Akinobu
-Mita, Alban Browaeys, Alcaro, Alexander Lapajne, Alexandru Băluț, Alex
-Ashley, Alex Hoenig, Alicia Boya García, Alistair Buxton, Ali Yousuf,
-Ambareesh “Amby” Balaji, Amr Mahdi, Andoni Morales Alastruey, Andreas
-Frisch, Andre Guedes, Andrew Branson, Andrey Sazonov, Antonio Ospite,
-aogun, Arun Raghavan, Askar Safin, AsociTon, A. Wilcox, Axel Mårtensson,
-Ayush Mittal, Bastian Bouchardon, Benjamin Otte, Bilal Elmoussaoui,
-Brady J. Garvin, Branko Subasic, Camilo Celis Guzman, Carlos Rafael
-Giani, Charlie Turner, Cheng-Chang Wu, Chris Ayoup, Chris Lord,
-Christoph Reiter, cketti, Damian Hobson-Garcia, Daniel Klamt, Daniel
-Molkentin, Danny Smith, David Bender, David Gunzinger, David Ing, David
-Svensson Fors, David Trussel, Debarshi Ray, Derek Lesho, Devarsh
-Thakkar, dhilshad, Dimitrios Katsaros, Dmitriy Purgin, Dmitry Shusharin,
-Dominique Leuenberger, Dong Il Park, Doug Nazar, dudengke, Dylan McCall,
-Dylan Yip, Ederson de Souza, Edward Hervey, Eero Nurkkala, Eike Hein,
-ekwange, Eric Marks, Fabian Greffrath, Fabian Orccon, Fabio D’Urso,
-Fabrice Bellet, Fabrice Fontaine, Fanchao L, Felix Yan, Fernando
-Herrrera, Francisco Javier Velázquez-García, Freyr, Fuwei Tang, Gaurav
-Kalra, George Kiagiadakis, Georgii Staroselskii, Georg Lippitsch, Georg
-Ottinger, gla, Göran Jönsson, Gordon Hart, Gregor Boirie, Guillaume
-Desmottes, Guillermo Rodríguez, Haakon Sporsheim, Haihao Xiang, Haihua
-Hu, Havard Graff, Håvard Graff, Heinrich Kruger, He Junyan, Henry
-Wilkes, Hosang Lee, Hou Qi, Hu Qian, Hyunjun Ko, ibauer, Ignacio Casal
-Quinteiro, Ilya Smelykh, Jake Barnes, Jakub Adam, James Cowgill, James
-Westman, Jan Alexander Steffens, Jan Schmidt, Jan Tojnar, Javier Celaya,
-Jeffy Chen, Jennifer Berringer, Jens Göpfert, Jérôme Laheurte, Jim
-Mason, Jimmy Ohn, J. Kim, Joakim Johansson, Jochen Henneberg, Johan
-Bjäreholt, Johan Sternerup, John Bassett, Jonas Holmberg, Jonas Larsson,
-Jonathan Matthew, Jordan Petridis, Jose Antonio Santos Cadenas, Josep
-Torra, Jose Quaresma, Josh Matthews, Joshua M. Doe, Juan Navarro,
-Juergen Werner, Julian Bouzas, Julien Isorce, Jun-ichi OKADA, Justin
-Chadwell, Justin Kim, Keri Henare, Kevin JOLY, Kevin King, Kevin Song,
-Knut Andre Tidemann, Kristofer Björkström, krivoguzovVlad, Kyrylo
-Polezhaiev, Lenny Jorissen, Linus Svensson, Loïc Le Page, Loïc Minier,
-Lucas Stach, Ludvig Rappe, Luka Blaskovic, luke.lin, Luke Yelavich,
-Marcin Kolny, Marc Leeman, Marco Felsch, Marcos Kintschner, Marek
-Olejnik, Mark Nauwelaerts, Markus Ebner, Martin Liska, Martin Theriault,
-Mart Raudsepp, Matej Knopp, Mathieu Duponchelle, Mats Lindestam, Matthew
-Read, Matthew Waters, Matus Gajdos, Maxim Paymushkin, Maxim P.
-Dementiev, Michael Bunk, Michael Gruner, Michael Olbrich, Miguel París
-Díaz, Mikhail Fludkov, Milian Wolff, Millan Castro, Muhammet Ilendemli,
-Nacho García, Nayana Topolsky, Nian Yan, Nicola Murino, Nicolas
-Dufresne, Nicolas Pernas Maradei, Niels De Graef, Nikita Bobkov, Niklas
-Hambüchen, Nirbheek Chauhan, Ognyan Tonchev, okuoku, Oleksandr
-Kvl,Olivier Crête, Ondřej Hruška, Pablo Marcos Oltra, Patricia Muscalu,
-Peter Seiderer, Peter Workman, Philippe Normand, Philippe Renon, Philipp
-Zabel, Pieter Willem Jordaan, Piotr Drąg, Ralf Sippl, Randy Li, Rasmus
-Thomsen, Ratchanan Srirattanamet, Raul Tambre, Ray Tiley, Richard
-Kreckel, Rico Tzschichholz, R Kh, Robert Rosengren, Robert Tiemann,
-Roman Shpuntov, Roman Sivriver, Ruben Gonzalez, Rubén Gonzalez,
-rubenrua, Ryan Huang, Sam Gigliotti, Santiago Carot-Nemesio, Saunier
-Thibault, Scott Kanowitz, Sebastian Dröge, Sebastiano Barrera, Seppo
-Yli-Olli, Sergey Nazaryev, Seungha Yang, Shinya Saito, Silvio
-Lazzeretti, Simon Arnling Bååth, Siwon Kang, sohwan.park, Song Bing,
-Soohyun Lee, Srimanta Panda, Stefano Buora, Stefan Sauer, Stéphane
-Cerveau, Stian Selnes, Sumaid Syed, Swayamjeet, Thiago Santos, Thibault
-Saunier, Thomas Bluemel, Thomas Coldrick, Thor Andreassen, Tim-Philipp
-Müller, Ting-Wei Lan, Tobias Ronge, trilene, Tulio Beloqui, U. Artie
-Eoff, VaL Doroshchuk, Varunkumar Allagadapa, Vedang Patel, Veerabadhran
-G, Víctor Manuel Jáquez Leal, Vivek R, Vivia Nikolaidou, Wangfei, Wang
-Zhanjun, Wim Taymans, Wonchul Lee, Xabier Rodriguez Calvar, Xavier
-Claessens, Xidorn Quan, Xu Guangxin, Yan Wang, Yatin Maan, Yeongjin
-Jeong, yychao, Zebediah Figura, Zeeshan Ali, Zeid Bekli, Zhiyuan Sraf,
-Zoltán Imets,
+-   this section will be filled in in due course
 
 … and many others who have contributed bug reports, translations, sent
 suggestions or helped testing.
 
-Stable 1.18 branch
+Stable 1.20 branch
 
-After the 1.18.0 release there will be several 1.18.x bug-fix releases
+After the 1.20.0 release there will be several 1.20.x bug-fix releases
 which will contain bug fixes which have been deemed suitable for a
 stable branch, but no new features or intrusive changes will be added to
-a bug-fix release usually. The 1.18.x bug-fix releases will be made from
-the git 1.18 branch, which will be a stable branch.
+a bug-fix release usually. The 1.20.x bug-fix releases will be made from
+the git 1.20 branch, which will be a stable branch.
 
-1.18.0
+1.20.0
 
-1.18.0 was released on 7 September 2020.
+1.20.0 is scheduled to be released around July 2021.
 
-Schedule for 1.20
+Schedule for 1.22
 
-Our next major feature release will be 1.20, and 1.19 will be the
-unstable development version leading up to the stable 1.20 release. The
-development of 1.19/1.20 will happen in the git master branch.
+Our next major feature release will be 1.22, and 1.21 will be the
+unstable development version leading up to the stable 1.22 release. The
+development of 1.21/1.22 will happen in the git master branch.
 
-The plan for the 1.20 development cycle is yet to be confirmed, but it
-is now expected that feature freeze will take place some time in January
-2021, with the first 1.20 stable release around February/March 2021.
+The plan for the 1.22 development cycle is yet to be confirmed, but it
+is hoped that feature freeze will take place some time in December 2021.
 
-1.20 will be backwards-compatible to the stable 1.18, 1.16, 1.14, 1.12,
-1.10, 1.8, 1.6, 1.4, 1.2 and 1.0 release series.
+1.22 will be backwards-compatible to the stable 1.20, 1.18, 1.16, 1.14,
+1.12, 1.10, 1.8, 1.6, 1.4, 1.2 and 1.0 release series.
 
 ------------------------------------------------------------------------
 
 These release notes have been prepared by Tim-Philipp Müller with
-contributions from Mathieu Duponchelle, Matthew Waters, Nirbheek
-Chauhan, Sebastian Dröge, Thibault Saunier, and Víctor Manuel Jáquez
-Leal.
+contributions from …
 
 License: CC BY-SA 4.0
diff --git a/README b/README
index 300b39868..c95e53e28 100644
--- a/README
+++ b/README
@@ -1,4 +1,4 @@
-GStreamer 1.18.x stable series
+GStreamer 1.19.x development series
 
 WHAT IT IS
 ----------
diff --git a/RELEASE b/RELEASE
index 852a7102d..43016107f 100644
--- a/RELEASE
+++ b/RELEASE
@@ -1,18 +1,15 @@
-This is GStreamer gst-plugins-good 1.18.0.
+This is GStreamer gst-plugins-good 1.19.1.
 
-The GStreamer team is thrilled to announce a new major feature release
-of your favourite cross-platform multimedia framework!
+GStreamer 1.19 is the development branch leading up to the next major
+stable version which will be 1.20.
 
-As always, this release is again packed with new features, bug fixes and
-other improvements.
-
-The 1.18 release series adds new features on top of the 1.16 series and is
+The 1.19 development series adds new features on top of the 1.18 series and is
 part of the API and ABI-stable 1.x release series of the GStreamer multimedia
 framework.
 
-Full release notes can be found at:
+Full release notes will one day be found at:
 
-  https://gstreamer.freedesktop.org/releases/1.18/
+  https://gstreamer.freedesktop.org/releases/1.20/
 
 Binaries for Android, iOS, Mac OS X and Windows will usually be provided
 shortly after the release.
diff --git a/docs/gst_plugins_cache.json b/docs/gst_plugins_cache.json
index 7d79b3500..005a16f2c 100644
--- a/docs/gst_plugins_cache.json
+++ b/docs/gst_plugins_cache.json
@@ -6607,7 +6607,7 @@
                         "construct": false,
                         "construct-only": false,
                         "controllable": false,
-                        "default": "GStreamer 1.19.0.1 FLV muxer",
+                        "default": "GStreamer 1.19.1 FLV muxer",
                         "mutable": "null",
                         "readable": true,
                         "type": "gchararray",
@@ -6619,7 +6619,7 @@
                         "construct": false,
                         "construct-only": false,
                         "controllable": false,
-                        "default": "GStreamer 1.19.0.1 FLV muxer",
+                        "default": "GStreamer 1.19.1 FLV muxer",
                         "mutable": "null",
                         "readable": true,
                         "type": "gchararray",
@@ -20112,7 +20112,7 @@
                         "construct": false,
                         "construct-only": false,
                         "controllable": false,
-                        "default": "GStreamer/1.19.0.1",
+                        "default": "GStreamer/1.19.1",
                         "mutable": "null",
                         "readable": true,
                         "type": "gchararray",
@@ -22069,7 +22069,7 @@
                         "construct": false,
                         "construct-only": false,
                         "controllable": false,
-                        "default": "GStreamer souphttpsrc 1.19.0.1 ",
+                        "default": "GStreamer souphttpsrc 1.19.1 ",
                         "mutable": "null",
                         "readable": true,
                         "type": "gchararray",
diff --git a/gst-plugins-good.doap b/gst-plugins-good.doap
index 00966a235..ba8918ce9 100644
--- a/gst-plugins-good.doap
+++ b/gst-plugins-good.doap
@@ -34,6 +34,16 @@ the plug-in code, LGPL or LGPL-compatible for the supporting library).
 
  <release>
   <Version>
+   <revision>1.19.1</revision>
+   <branch>master</branch>
+   <name></name>
+   <created>2021-06-01</created>
+   <file-release rdf:resource="https://gstreamer.freedesktop.org/src/gst-plugins-good/gst-plugins-good-1.19.1.tar.xz" />
+  </Version>
+ </release>
+
+ <release>
+  <Version>
    <revision>1.18.0</revision>
    <branch>master</branch>
    <name></name>
diff --git a/meson.build b/meson.build
index 97f76a4d8..e04c71d79 100644
--- a/meson.build
+++ b/meson.build
@@ -1,5 +1,5 @@
 project('gst-plugins-good', 'c',
-  version : '1.19.0.1',
+  version : '1.19.1',
   meson_version : '>= 0.54',
   default_options : [ 'warning_level=1',
                       'buildtype=debugoptimized' ])