Release 1.18.11.18.1

5 files changed, 766 insertions, 21 deletions

diff --git a/ChangeLog b/ChangeLog
index 86cb24720..3e7246dc1 100644
--- a/ChangeLog
+++ b/ChangeLog
@@ -1,7 +1,536 @@
+=== release 1.18.1 ===
+
+2020-10-26 11:14:43 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* ChangeLog:
+	* NEWS:
+	* RELEASE:
+	* gst-plugins-bad.doap:
+	* meson.build:
+	  Release 1.18.1
+
+2020-10-16 12:29:02 +0100  Andrew Wesie <andrew@theori.io>
+
+	* gst-libs/gst/codecparsers/gsth264parser.c:
+	  codecparsers: h264parser: guard against ref_pic_markings overflow
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1704>
+
+2020-10-15 12:08:19 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+	* sys/v4l2codecs/gstv4l2codecvp8dec.c:
+	  v4l2codecs: vp8: Allow baseclass from skipping frames
+	  In preparation for !1670, this will allow the base class from skipping frames
+	  that should not be displayed. Previously it would complain about unordered
+	  decoding taking place in the driver.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1701>
+
+2020-10-15 12:05:45 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+	* sys/v4l2codecs/gstv4l2decoder.c:
+	  v4l2codecs: decoder: Unmark previously pending request
+	  requests are executed in order, so while dequeuing sink buffers for previous
+	  request, also mark these request as no longer pending. This will allow reusing
+	  the request later.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1701>
+
+2020-10-15 11:35:04 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+	* sys/v4l2codecs/gstv4l2decoder.c:
+	  v4l2codecs: decoder: Properly remove pending requests
+	  Pass the pointer instead of NULL in order to find and remove properly any
+	  pending request from the queue. This coding error was leading to use after
+	  free in error and early exit cases.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1701>
+
+2020-10-04 02:02:16 +0900  Seungha Yang <seungha@centricular.com>
+
+	* gst/videoparsers/gsth265parse.c:
+	  h265parse: Don't enable passthrough by default
+	  SEI messages contain various information which wouldn't be conveyed
+	  by using upstream CAPS (HDR, timecode for example).
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1694>
+
+2020-10-12 14:15:49 +0200  Jan Alexander Steffens (heftig) <jan.steffens@ltnglobal.com>
+
+	* ext/srt/gstsrtsrc.c:
+	  srtsrc: Prevent `delay` from being negative
+	  `delay` should be a GstClockTimeDiff since SRT time is int64_t.
+	  All values are in local time so we should never see a srctime that's in
+	  the future. If we do, clamp the delay to 0 and warn about it.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1690>
+
+2020-10-12 14:12:24 +0200  Jan Alexander Steffens (heftig) <jan.steffens@ltnglobal.com>
+
+	* ext/srt/gstsrtsrc.c:
+	  srtsrc: Don't calculate a delay if the srctime is 0
+	  A zero srctime is a missing srctime. Apparently this can happen when
+	  ["the connection is not between SRT peers or if Timestamp-Based Packet
+	  Delivery mode (TSBPDMODE) is not enabled"][1] so it may not apply to us,
+	  but it's best to be defensive.
+	  [1]: https://github.com/Haivision/srt/blob/v1.4.2/docs/API.md#sending-and-receiving
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1690>
+
+2020-10-12 14:09:28 +0200  Jan Alexander Steffens (heftig) <jan.steffens@ltnglobal.com>
+
+	* ext/srt/gstsrtsrc.c:
+	  srtsrc: Defend against missing clock
+	  If we don't have a clock, stop the source instead of asserting in
+	  gst_clock_get_time. This can happen when the element is removed from the
+	  pipeline while it's playing.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1690>
+
+2020-10-12 11:50:59 +0100  Tim-Philipp Müller <tim@centricular.com>
+
+	* docs/plugins/gst_plugins_cache.json:
+	* gst/rtmp2/gstrtmp2.c:
+	* gst/rtmp2/gstrtmp2sink.c:
+	  rtmp2sink: don't expose stop-commands property in backported patches
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1628>
+
+2020-08-19 14:51:17 +0300  Nazar Mokrynskyi <nazar@mokrynskyi.com>
+
+	* docs/meson.build:
+	* gst/rtmp2/gstrtmp2sink.c:
+	* gst/rtmp2/rtmp/rtmpclient.h:
+	  rtmp2sink: add docs section with since marker on new stop-commands property
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1628>
+
+2020-08-18 19:16:40 +0300  Nazar Mokrynskyi <nazar@mokrynskyi.com>
+
+	* docs/plugins/gst_plugins_cache.json:
+	* gst/rtmp2/gstrtmp2sink.c:
+	  rtmp2: fix code style, update documentation cache
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1628>
+
+2020-08-18 14:05:26 +0200  Jan Alexander Steffens (heftig) <jan.steffens@ltnglobal.com>
+
+	* gst/rtmp2/gstrtmp2.c:
+	* gst/rtmp2/gstrtmp2sink.c:
+	* gst/rtmp2/rtmp/rtmpclient.c:
+	* gst/rtmp2/rtmp/rtmpclient.h:
+	  rtmp2: Clean up (improve) GstRtmpStopCommands type
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1628>
+
+2020-05-02 04:49:42 +0300  Nazar Mokrynskyi <nazar@mokrynskyi.com>
+
+	* gst/rtmp2/gstrtmp2sink.c:
+	* gst/rtmp2/rtmp/rtmpclient.c:
+	* gst/rtmp2/rtmp/rtmpclient.h:
+	  rtmp2sink: handle EOS event and close stream
+	  https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/1285
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1628>
+
+2020-10-05 11:07:25 +0200  Guillaume Desmottes <guillaume.desmottes@collabora.com>
+
+	* gst/videoparsers/gsth265parse.c:
+	  h265parse: set interlace-mode=interleaved on interlaced content
+	  interlace-mode=alternate is a special case of interlace-mode=interleaved
+	  where the fields are split using two different buffers.
+	  We should use the latter instead of the former to no break compat with
+	  elements supporting only 'interleaved'.
+	  Decoders producing alternate, such as OMX on the Zynq, should change the
+	  interlace-mode on their output caps.
+	  Fix https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/issues/825
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1683>
+
+2020-10-06 13:39:23 +0200  Jan Alexander Steffens (heftig) <jan.steffens@ltnglobal.com>
+
+	* ext/srt/gstsrtobject.c:
+	  srt: Remove unused sa_family tracking
+	  Now that SRT no longer needs the family when creating the socket, this
+	  code has become useless.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1685>
+
+2020-10-02 03:39:40 +0200  Niklas Hambüchen <mail@nh2.me>
+
+	* ext/srt/gstsrtobject.c:
+	* ext/srt/meson.build:
+	  srt: Move off deprecated `srt_socket()`.
+	  See https://github.com/Haivision/srt/blob/73ee1e1a3e3adc2702a9a5057d101ef80447b38c/docs/API-functions.md#srt_socket
+	  `srt_create_socket()` was added in
+	  https://github.com/Haivision/srt/commit/4b897ba92d34f1829a1c6e419eeab17f0763a0fc
+	  and srt `v1.3.0` is the first release that has it.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1685>
+
+2020-10-01 17:31:13 +0200  Jan Alexander Steffens (heftig) <jan.steffens@ltnglobal.com>
+
+	* ext/srt/gstsrt.c:
+	  srt: Register a log handler
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1685>
+
+2020-09-25 19:17:35 +0200  Jan Alexander Steffens (heftig) <jan.steffens@ltnglobal.com>
+
+	* ext/srt/gstsrtobject.c:
+	  srt: Avoid removing invalid sockets from the polls
+	  This would provoke error messages from SRT.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1685>
+
+2020-09-25 19:08:17 +0200  Jan Alexander Steffens (heftig) <jan.steffens@ltnglobal.com>
+
+	* ext/srt/gstsrtobject.c:
+	  srt: Fix use of srt_startup
+	  `srt_startup` can also return 1 if it was successful. Avoid warning in
+	  this case.
+	  Avoid a race when checking whether we need to call it at all.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1685>
+
+2020-10-06 12:35:12 +0200  Jan Alexander Steffens (heftig) <jan.steffens@ltnglobal.com>
+
+	* ext/srt/gstsrtobject.c:
+	  srt: Fix parameter types used for socket options
+	  The [SRT documentation][1] specifies exact types for the socket options.
+	  Make sure we match these.
+	  This reverts the linger workaround in commit 84f8dbd932029220ee86154dd
+	  and extends srt_constant_params to support other types than int.
+	  [1]: https://github.com/Haivision/srt/blob/master/docs/APISocketOptions.md
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1685>
+
+2020-10-07 11:04:30 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* pkgconfig/gstreamer-webrtc-uninstalled.pc.in:
+	* pkgconfig/gstreamer-webrtc.pc.in:
+	  webrtc: Require gstreamer-sdp in the pkg-config file
+	  Some headers include API from it.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1678>
+
+2020-10-06 11:45:36 +0200  Edward Hervey <edward@centricular.com>
+
+	* ext/srt/gstsrtobject.c:
+	* ext/srt/gstsrtobject.h:
+	* ext/srt/gstsrtsrc.c:
+	* ext/srt/gstsrtsrc.h:
+	  srtsrc: Fix timestamping
+	  SRT provides the original timestamp of a packet (with drift/skew corrected for
+	  local clock), which is what should be used for timestamping the outgoing
+	  buffers. This ensures that we output the packets with the same timestamp (and by
+	  extension rate) as the original feed.
+	  Also detect if packets were dropped (by checking the sequence number) and
+	  properly set DISCONT flag on the outgoing buffer.
+	  Finally answer the latency queries
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1677>
+
+2020-10-08 17:52:05 +0200  Jan Alexander Steffens (heftig) <jan.steffens@ltnglobal.com>
+
+	* ext/srt/gstsrtsink.c:
+	  srt: Consume the error from gst_srt_object_write
+	  Instead of leaking it.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1681>
+
+2020-10-05 16:40:55 -0400  Olivier Crête <olivier.crete@collabora.com>
+
+	* ext/dtls/gstdtlsconnection.c:
+	  dtlsconnection: Ignore OpenSSL system call errors
+	  OpenSSL shouldn't be making real system calls, so we can safely
+	  ignore syscall errors. System interactions should happen through
+	  our BIO. So especially don't look at the system's errno, as it
+	  should be meaningless.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1684>
+
+2020-10-09 10:24:50 +0200  Jacek Tomaszewski <lord.jacold@gmail.com>
+
+	* COPYING:
+	* COPYING.LIB:
+	  Replace LGPL v2 with LGPL v2.1 in COPYING and remove COPYING.LIB
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1682>
+
+2020-10-02 11:42:07 +0200  Jacek Tomaszewski <lord.jacold@gmail.com>
+
+	* COPYING:
+	  Replace GPL v2 with LGPL v2 in COPYING file
+	  Fixes #1422
+	  https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/1422
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1682>
+
+2020-10-08 17:48:20 +0200  Jan Alexander Steffens (heftig) <jan.steffens@ltnglobal.com>
+
+	* ext/srt/gstsrtobject.c:
+	  srt: Check socket state before retrieving payload size
+	  The connection might be broken, which we should detect instead of just
+	  aborting the write.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1680>
+
+2020-10-08 18:25:59 +0200  Jakub Adam <jakub.adam@collabora.com>
+
+	* ext/x265/gstx265enc.c:
+	  x265enc: fix deadlock on reconfig
+	  Don't attempt to obtain encoder lock that is already held by
+	  gst_x265_enc_encode_frame().
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1679>
+
+2020-10-07 05:05:25 +0900  Seungha Yang <seungha@centricular.com>
+
+	* sys/mediafoundation/gstmfsourcereader.cpp:
+	  mfvideosrc: Use only the first video stream per device
+	  Non-first video stream might not be working with current
+	  implementation. It could be non-video (e.g., photo source) and then
+	  ReadSample() might be blocked forever.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1676>
+
+2020-09-25 22:00:26 +0530  raghavendra <raghavendra.rao@collabora.com>
+
+	* ext/srt/gstsrtobject.c:
+	  srtobject: typecast SRTO_LINGER to linger
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1675>
+
+2020-09-29 15:52:21 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* docs/plugins/gst_plugins_cache.json:
+	* sys/decklink/gstdecklink.cpp:
+	  decklink: Correctly order the different dependent mode tables
+	  One was forgotten in 309f6187fef890c7ffa49305f38e89beac3b1423.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1652>
+
+2020-09-19 00:26:35 +0900  Seungha Yang <seungha@centricular.com>
+
+	* sys/wasapi/gstwasapisrc.c:
+	* sys/wasapi/gstwasapisrc.h:
+	  wasapisrc: Make sure that wasapisrc produces data in loopback mode
+	  An oddness of wasapi loopback feature is that capture client will not
+	  produce any data if there's no outputting sound to corresponding
+	  render client. In other words, if there's no sound to render,
+	  capture task will stall. As an option to solve such issue, we can
+	  add timeout to wake up from capture thread if there's no incoming data
+	  within given time interval. But it seems to be glitch prone.
+	  Another approach is that we can keep pushing silence data into
+	  render client so that capture client can keep capturing data
+	  (even if it's just silence).
+	  This patch will choose the latter one because it's more straightforward
+	  way and it's likely produce glitchless sound than former approach.
+	  A bonus point of this approach is that loopback capture on Windows7/8
+	  will work with this patch. Note that there's an OS bug prior to Windows10
+	  when loopback capture client is running with event-driven mode.
+	  To work around the bug, event signalling should be handled manually
+	  for read thread to wake up.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1651>
+
+2020-09-29 23:46:00 +1000  Matthew Waters <matthew@centricular.com>
+
+	* ext/wpe/gstwpesrc.cpp:
+	  wpesrc: fix some caps leaks using the non-GL output
+	  Always chain up to the parent _stop() implementation as it unrefs some
+	  caps (among other things).
+	  Fixes: https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/1409
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1650>
+
+2020-06-16 11:16:37 +0900  Hosang Lee <hosang10.lee@lge.com>
+
+	* ext/smoothstreaming/gstmssmanifest.c:
+	* ext/smoothstreaming/gstmssmanifest.h:
+	  smoothstreaming: clear live adapter on seek
+	  In live streaming, buffers sent by souphttpsrc are pushed to the live
+	  adapter. The buffers in the adapter are sent out of mssdemux when it
+	  is greater than 4096 bytes.
+	  Occasionally, when seeking in live streams, if seek occurs just
+	  after the last data chunk was received, and if this data chunk is
+	  smaller than 4096 bytes, it will be kept in the live adapter.
+	  This remaining data in the live adapter will be erroneously prepended
+	  to the new data that is downloaded after seek and pushed out.
+	  When qtdemux receives this data, since it does not start with
+	  a moof box, it is impossible to demux the fragment, and bogus
+	  size error will occur.
+	  Clear out the live adapter on seek so that no unnecessary remaining
+	  data is pushed out together with the new fragment after seeking.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1649>
+
+2020-09-23 20:08:46 +1000  Matthew Waters <matthew@centricular.com>
+
+	* sys/applemedia/videotexturecache-vulkan.mm:
+	  vtdec/vulkan: use Shared storage mode for IOSurface textures
+	  Fixes a debug assertion with i(Pad)OS 14:
+	  'IOSurface textures must use MTLStorageModeShared'
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1648>
+
+2020-09-14 09:48:48 +0100  Philippe Normand <philn@igalia.com>
+
+	* ext/wpe/gstwpesrc.cpp:
+	  wpe: Plug event leak
+	  Handled events don't go through the default pad event handler, so they need to
+	  be unreffed in this case.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1647>
+
+2020-08-25 01:57:55 +1000  Jan Schmidt <jan@centricular.com>
+
+	* ext/wpe/gstwpesrc.cpp:
+	  wpesrc: Don't crash if WPE doesn't generate a buffer.
+	  On creating a 2nd wpesrc in a new pipeline in an app that already
+	  has a runnig wpesrc, WPE sometimes doesn't return a buffer on request,
+	  leading to a crash. This commit fixes the crash, but not the underlying
+	  failure - a 2nd wpesrc can still error out instead.
+	  Partially fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/1386
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1647>
+
+2020-09-11 12:51:56 +0100  Philippe Normand <philn@igalia.com>
+
+	* ext/wpe/WPEThreadedView.cpp:
+	  wpe: Plug SHM buffer leaks
+	  Fixes #1409
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1647>
+
+2020-09-10 14:39:58 +0100  Philippe Normand <philn@igalia.com>
+
+	* ext/wpe/WPEThreadedView.cpp:
+	* ext/wpe/WPEThreadedView.h:
+	  wpe: Move webview load waiting to WPEView
+	  As waiting for the load to be finished is specific to the WebView, it should be
+	  done from our WPEView, not from the WPEContextThread. This fixes issues where
+	  multiple wpesrc elements are created in sequence. Without this patch the first
+	  view might receive erroneous buffer notifications.
+	  Fixes #1386
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1647>
+
+2020-09-10 13:05:23 +0100  Philippe Normand <philn@igalia.com>
+
+	* ext/wpe/WPEThreadedView.cpp:
+	  wpe: Use proper callback for TLS errors signal handling
+	  The load-failed and load-failed-with-tls-errors signals expect distinct callback
+	  signatures.
+	  Fixes #1388
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1646>
+
+2020-09-17 17:39:25 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+	* sys/kms/gstkmssink.c:
+	  kmssink: Do not source using padded width/height
+	  The width/height from the video meta can be padded width, height. But when
+	  sourcing from padded buffer, we only want to use the valid pixels. This
+	  rectangle is from the crop meta, orther it is deduces from the caps. The width
+	  and height from the caps is save in the parent class, use these instead of the
+	  GstVideoInfo when settting the src rectangle.
+	  This fixes an issue with 1080p video displaying repeated or green at the
+	  padded bottom 8 lines (seen with v4l2codecs).
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1642>
+
+2020-09-15 17:09:57 +0200  Marian Cichy <m.cichy@pengutronix.de>
+
+	* ext/avtp/gstavtpaafdepay.c:
+	  avtp: avtpaafdepay: fix crash when building caps
+	  gst_caps_new_simple gets wrong types for rate and channel which
+	  may lead to a crash.
+	  As 64-bit values for rate, depth, format, channels does not
+	  make much sense and since any other functionality in gstreamer
+	  expects G_TYPE_INT for channels and rate, we should stick to that
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1641>
+
+2020-09-10 10:35:11 -0700  Adam Williamson <awilliam@redhat.com>
+
+	* ext/opencv/meson.build:
+	  opencv: set opencv_dep when option is disabled (#1406)
+	  The examples build file checks opencv_dep, so it still needs to
+	  be set even if the option is disabled.
+	  Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/1406
+	  Signed-off-by: Adam Williamson <awilliam@redhat.com>
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1638>
+
+2020-09-04 16:53:03 +0200  Mathieu Duponchelle <mathieu@centricular.com>
+
+	* docs/plugins/gst_plugins_cache.json:
+	* ext/closedcaption/gstline21enc.c:
+	* ext/closedcaption/gstline21enc.h:
+	* tests/check/elements/line21.c:
+	  line21enc: add remove-caption-meta property
+	  Similar to #GstCCExtractor:remove-caption-meta
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1637>
+
+2020-09-04 02:38:58 +0200  Mathieu Duponchelle <mathieu@centricular.com>
+
+	* docs/plugins/gst_plugins_cache.json:
+	* ext/closedcaption/gstline21enc.c:
+	* tests/check/elements/line21.c:
+	  line21enc: heavily constrain video height
+	  We can only determine a correct placement for the CC line
+	  with:
+	  * height == 525 (standard NTSC, line 21 / 22)
+	  * height == 486 (NTSC usable lines + 6 lines for VBI, line 1 / 2)
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1637>
+
+2020-09-04 02:33:52 +0200  Mathieu Duponchelle <mathieu@centricular.com>
+
+	* ext/closedcaption/gstline21enc.c:
+	  line21enc: add support for CDP closed caption meta
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1637>
+
+2020-08-27 11:23:01 +0200  Edward Hervey <edward@centricular.com>
+
+	* gst-libs/gst/adaptivedemux/gstadaptivedemux.c:
+	  adaptivedemux: Add another nominal bitrate fallback calculation
+	  Some HTTP servers don't provide fragment sizes (with the Content-Length HTTP
+	  header). In order to still figure out a nominal bitrate (for usage by queue2),
+	  calculate on when we're done downloading a fragment.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1635>
+
+2020-08-27 11:18:56 +0200  Edward Hervey <edward@centricular.com>
+
+	* gst-libs/gst/adaptivedemux/gstadaptivedemux.c:
+	  adaptivedemux: Handle invalid HTTP duration
+	  The default BYTE DURATION basesrc query handler will return `-1` and TRUE. In
+	  order to properly handle cases where upstream http servers didn't return a valid
+	  Content-Length we also need to check whether it was valid when calculating
+	  bitrates.
+	  Avoids returning completely bogus bitrates with gogol's video streaming services
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1635>
+
+2020-09-01 13:28:44 +0200  Jan Alexander Steffens (heftig) <jan.steffens@ltnglobal.com>
+
+	* gst/rtmp2/gstrtmp2sink.c:
+	* gst/rtmp2/gstrtmp2src.c:
+	* gst/rtmp2/rtmp/rtmpconnection.c:
+	  rtmp2: Replace stats queue with stats lock
+	  Making the thread receiving the stats wait on the loop to respond was
+	  not a good idea, as the latter can get blocked on the streaming thread.
+	  Have get_stats read the values directly, adding a lock to ensure we
+	  don't read garbage.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1629>
+
+2020-09-02 15:29:49 +0200  Jan Alexander Steffens (heftig) <jan.steffens@ltnglobal.com>
+
+	* gst/mpegtsmux/gstbasetsmux.c:
+	* gst/mpegtsmux/gstbasetsmux.h:
+	  mpegtsmux: Make handling of sinkpads thread-safe
+	  Ensure we take the object lock while accessing `GstElement.sinkpads`.
+	  Use an iterator when the code isn't simple to avoid deadlock.
+	  When we find the best pad, take a reference so a concurrent pad
+	  release doesn't destroy the pad before we're done with it.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1625>
+
+2020-09-01 13:29:30 +0200  Jan Alexander Steffens (heftig) <jan.steffens@ltnglobal.com>
+
+	* ext/hls/gsthlssink2.c:
+	  hlssink2: Actually release splitmuxsink's pads
+	  It was looking at the "outer" peer of the ghost pad, not the "inner"
+	  peer (the target).
+	  It provided the wrong pad to gst_element_release_request_pad.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1623>
+
+2020-09-08 10:48:56 +0200  Edward Hervey <edward@centricular.com>
+
+	* gst/mpegtsmux/gstbasetsmux.c:
+	  mpegtsmux: Don't create streams with reserved PID
+	  There are quite a few reserved PID in the various MPEG-TS (and derivate)
+	  specifications which we should definitely not use. Those PID have a certain
+	  meaning and purpose.
+	  Furthermore, a lot of the code in the muxer implementation also makes assumption
+	  on the purpose of streams based on their PID.
+	  Therefore, when requesting a pad with a specific PID, make sure it is not a
+	  restricted PID.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1620>
+
+2020-09-14 14:42:36 +0300  Jordan Petridis <jordan@centricular.com>
+
+	* tests/check/gst-plugins-bad.supp:
+	  validate: plug leak in gssdp
+	  These are triggered by the webrtcbin tests
+	  https://gitlab.gnome.org/GNOME/gssdp/-/issues/10
+
+2020-09-08 17:44:29 +0100  Tim-Philipp Müller <tim@centricular.com>
+
+	* meson.build:
+	  Back to development
+
 === release 1.18.0 ===
 
 2020-09-08 00:07:45 +0100  Tim-Philipp Müller <tim@centricular.com>
 
+	* .gitlab-ci.yml:
 	* ChangeLog:
 	* NEWS:
 	* README:
diff --git a/NEWS b/NEWS
index dba9c7c47..c4276518e 100644
--- a/NEWS
+++ b/NEWS
@@ -1,11 +1,14 @@
 GStreamer 1.18 Release Notes
 
-GStreamer 1.18.0 was originally released on 7 September 2020.
+GStreamer 1.18.0 was originally released on 8 September 2020.
+
+The latest bug-fix release in the 1.18 series is 1.18.1 and was released
+on 26 October 2020.
 
 See https://gstreamer.freedesktop.org/releases/1.18/ for the latest
 version of this document.
 
-Last updated: Monday 7 September 2020, 10:30 UTC (log)
+Last updated: Monday 26 October 2020, 11:00 UTC (log)
 
 Introduction
 
@@ -28,7 +31,8 @@ Highlights
 
 -   Active Format Description (AFD) and Bar Data support
 
--   ONVIF trick modes support in both GStreamer RTSP server and client
+-   RTSP server and client implementations gained ONVIF trick modes
+    support
 
 -   Hardware-accelerated video decoding on Windows via DXVA2 /
     Direct3D11
@@ -39,24 +43,25 @@ Highlights
 -   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
+-   imagesequencesrc: New element to easily create a video stream from a
     sequence of jpeg or png images
 
--   dashsink: Add new sink to produce DASH content
+-   dashsink: New sink to produce DASH content
 
--   dvbsubenc: DVB Subtitle encoder element
+-   dvbsubenc: New DVB Subtitle encoder element
 
--   TV broadcast compliant MPEG-TS muxing with constant bitrate muxing
-    and SCTE-35 support
+-   MPEG-TS muxing now also supports TV broadcast compliant muxing with
+    constant bitrate muxing and SCTE-35 support
 
--   rtmp2: new RTMP client source and sink element implementation
+-   rtmp2: New RTMP client source and sink element from-scratch
+    implementation
 
--   svthevcenc: new SVT-HEVC-based H.265 video encoder
+-   svthevcenc: New SVT-HEVC-based H.265 video encoder
 
--   vaapioverlay compositor element using VA-API
+-   vaapioverlay: New compositor element using VA-API
 
--   rtpmanager support for Google’s Transport-Wide Congestion Control
-    (twcc) RTP extension
+-   rtpmanager gained support for Google’s Transport-Wide Congestion
+    Control (twcc) RTP extension
 
 -   splitmuxsink and splitmuxsrc gained support for auxiliary video
     streams
@@ -64,18 +69,18 @@ Highlights
 -   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
+-   RTP support was enhanced with 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
+-   avtp: 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
+-   rpicamsrc: New element for capturing from the Raspberry Pi camera
 
 -   RTSP Server TCP interleaved backpressure handling improvements as
     well as support for Scale/Speed headers
@@ -2179,7 +2184,208 @@ the git 1.18 branch, which will be a stable branch.
 
 1.18.0
 
-1.18.0 was released on 7 September 2020.
+1.18.0 was released on 8 September 2020.
+
+1.18.1
+
+The first 1.18 bug-fix release (1.18.1) was released on 26 October 2020.
+
+This release only contains bugfixes and it should be safe to update from
+1.18.0.
+
+Highlighted bugfixes in 1.18.1
+
+-   important security fixes
+-   bug fixes and memory leak fixes
+-   various stability and reliability improvements
+
+gstreamer
+
+-   aggregator: make peek() has() pop() drop() buffer API threadsafe
+-   gstvalue: don’t write to const char *
+-   meson: Disallow DbgHelp for UWP build
+-   info: Fix build on Windows ARM64 device
+-   build: use cpu_family for arch checks
+-   basetransform: Fix in/outbuf confusion of _default_transform_meta
+-   Fix documentation
+-   info: Load DbgHelp.dll using g_module_open()
+-   padtemplate: mark documentation caps as may be leaked
+-   gstmeta: intern registered impl string
+-   aggregator: Hold SRC_LOCK while unblocking via SRC_BROADCAST()
+-   ptp_helper_post_install.sh: deal with none
+-   skip elements/leak.c if tracer is not available
+-   aggregator: Wake up source pad in PAUSED<->PLAYING transitions
+-   input-selector: Wake up blocking pads when releasing them
+-   ptp: Also handle gnu/kfreebsd
+
+gst-plugins-base
+
+-   theoradec: Set telemetry options only if they are nonzero
+-   glslstage: delete shader on finalize of stage
+-   urisourcebin: Fix crash caused by use after free
+-   decodebin3: Store stream-start event on output pad before exposing
+    it
+-   Add some missing nullable annotations
+-   typefind/xdgmime: Validate mimetypes to be valid GstStructure names
+    before using them
+-   uridecodebin3: Forward upstream events to decodebin3 directly
+-   video-converter: Add fast paths from v210 to I420/YV12, Y42B, UYVY
+    and YUY2
+-   videoaggregator: Limit accepted caps by template caps
+-   gstrtpbuffer: fix header extension length validation
+-   decodebin3: only force streams-selected seqnum after a
+    select-streams
+-   videodecoder: don’t copy interlace-mode from reference state
+-   enable abi checks
+-   multihandlesink: Don’t pass NULL caps to gst_caps_is_equal
+-   audio: video: Fix in/outbuf confusion of transform_meta
+-   meson: Always wrap “prefix” option with join_paths() to make Windows
+    happy
+-   videoaggregator: ensure peek_next_sample() uses the correct caps
+-   meson: Actually build gstgl without implicit include dirs
+-   videoaggregator: Don’t require any pads to be configured for
+    negotiating source pad caps
+-   gst-libs: gl: Fix documentation typo and clarify
+    gl_memory_texsubimage
+-   audioaggregator: Reset offset if the output rate is renegotiated
+-   video-anc: Implement transform functions for AFD/Bar metas
+-   appsrc: Wake up the create() function on caps changes
+-   rtpbasepayload: do not forget delayed segment when forwarding gaps
+
+gst-plugins-good
+
+-   v4l2object: Only offer inactive pools and if needed
+-   vpx: Fix the check to unfixed/unknown framerate to set bitrate
+-   qmlglsink: fix crash when created/destroyed in quick succession
+-   rtputils: Count metas with an empty tag list for copying/keeping
+-   rtpbin: Remove the rtpjitterbuffer with the stream
+-   rtph26*depay: drop FU’s without a corresponding start bit
+-   imagefreeze: Response caps query from srcpad
+-   rtpmp4gdepay: Allow lower-case “aac-hbr” instead of correct
+    “AAC-hbr”
+-   rtspsrc: Fix push-backchannel-buffer parameter mismatch
+-   jpegdec: check buffer size before dereferencing
+-   flvmux: Move stream skipping to GstAggregatorPadClass.skip_buffer
+-   v4l2object: plug memory leak
+-   splitmuxsink: fix sink pad release while PLAYING
+
+gst-plugins-bad
+
+-   codecparsers: h264parser: guard against ref_pic_markings overflow
+-   v4l2codecs: Various fixes
+-   h265parse: Don’t enable passthrough by default
+-   srt: Fix “Fix timestamping”
+-   srt: Fixes for 1.4.2
+-   dtlsconnection: Ignore OpenSSL system call errors
+-   h265parse: set interlace-mode=interleaved on interlaced content
+-   Replace GPL v2 with LGPL v2 in COPYING file
+-   srt: Consume the error from gst_srt_object_write
+-   srt: Check socket state before retrieving payload size
+-   x265enc: fix deadlock on reconfig
+-   webrtc: Require gstreamer-sdp in the pkg-config file
+-   srtsrc: Fix timestamping
+-   mfvideosrc: Use only the first video stream per device
+-   srtobject: typecast SRTO_LINGER to linger
+-   decklink: Correctly order the different dependent mode tables
+-   wasapisrc: Make sure that wasapisrc produces data in loopback mode
+-   wpesrc: fix some caps leaks using the non-GL output
+-   smoothstreaming: clear live adapter on seek
+-   vtdec/vulkan: use Shared storage mode for IOSurface textures
+-   wpe: Move webview load waiting to WPEView
+-   wpe: Use proper callback for TLS errors signal handling
+-   kmssink: Do not source using padded width/height
+-   avtp: avtpaafdepay: fix crash when building caps
+-   opencv: set opencv_dep when option is disabled to fix the build
+-   line21encoder: miscellaneous enhancements
+-   Hls youtube issues with urisourcebin/queue2
+-   rtmp2: Replace stats queue with stats lock
+-   rtmp2sink: support EOS event for graceful connection shutdown
+-   mpegtsmux: Make handling of sinkpads thread-safe
+-   hlssink2: Actually release splitmuxsink’s pads
+-   mpegtsmux: Don’t create streams with reserved PID
+
+gst-plugins-ugly
+
+-   no changes
+
+gst-libav
+
+-   avaudenc/avvidenc: Reopen encoding session if it’s required
+-   avauddec/audenc/videnc: Don’t return GST_FLOW_EOS when draining
+-   avauddec/avviddec: Avoid dropping non-OK flow return
+-   avcodecmap: Enable 24 bit WMA Lossless decoding
+
+gst-rtsp-server
+
+-   rtsp-stream: collect rtp info when blocking
+-   rtsp-media: set a 0 storage size for TCP receivers
+-   rtsp-stream: preroll on gap events
+-   rtsp-media: do not unblock on unsuspend
+
+gstreamer-vaapi
+
+-   decoder: don’t reply src caps query with allowed if pad is fixed
+-   plugins: decode: fix a DMA caps typo in ensure_allowed_srcpad_caps
+
+gstreamer-sharp
+
+-   Add bindings for some missing 1.18 API
+
+gst-omx
+
+-   omxvideodec: support interlace-mode=interleaved input
+
+gst-python
+
+-   no changes
+
+gst-editing-services
+
+-   ges: Do not recreate auto-transitions when changing clip assets
+-   ges: Fix a copy/paste mistake in meson file
+
+gst-integration-testsuites
+
+-   medias: Update for h265parse passthrough behavior change
+-   update validate.test.h265parse.alternate test
+
+gst-build
+
+-   windows: Detect Strawberry Perl and error out early
+-   {pygobject,pycairo}.wrap: point to stable refs
+
+Cerbero build tool and packaging changes in 1.18.1
+
+-   Add macOS Big Sur support
+-   gst-plugins-bad: Ship rtpmanagerbad plugin
+-   gstreamer-1.0: Don’t enable DbgHelp for UWP build
+-   pango: fix font corruption on windows
+-   cairo: use thread local storage to grant one windows HDC per thread
+-   small fixes for Xcode 12
+-   cerbero: Re-add alsa-devel to bootstrap on Linux
+-   FreeType: update to 2.10.4 to fix security vulnerability
+
+Contributors to 1.18.1
+
+Aaron Boxer, Adam Williamson, Andrew Wesie, Arun Raghavan, Bastien
+Reboulet, Brent Gardner, Edward Hervey, François Laignel, Guillaume
+Desmottes, Havard Graff, He Junyan, Hosang Lee, Jacek Tomaszewski, Jakub
+Adam, Jan Alexander Steffens (heftig), Jan Schmidt, Jérôme Laheurte,
+Jordan Petridis, Marc Leeman, Marian Cichy, Marijn Suijten, Mathieu
+Duponchelle, Matthew Waters, Michael Tretter, Nazar Mokrynskyi, Nicolas
+Dufresne, Niklas Hambüchen, Nirbheek Chauhan, Olivier Crête, Philippe
+Normand, raghavendra, Ricky Tang, Sebastian Dröge, Seungha Yang,
+sohwan.park, Stéphane Cerveau, Thibault Saunier, Tim-Philipp Müller, Tom
+Schoonjans, Víctor Manuel Jáquez Leal, Will Miller, Xavier Claessens, Xℹ
+Ruoyao, Zebediah Figura,
+
+… and many others who have contributed bug reports, translations, sent
+suggestions or helped testing. Thank you all!
+
+List of merge requests and issues fixed in 1.18.1
+
+-   List of Merge Requests applied in 1.18.1
+-   List of Issues fixed in 1.18.1
 
 Schedule for 1.20
 
diff --git a/RELEASE b/RELEASE
index 9229fa19b..68dd88122 100644
--- a/RELEASE
+++ b/RELEASE
@@ -1,4 +1,4 @@
-This is GStreamer gst-plugins-bad 1.18.0.
+This is GStreamer gst-plugins-bad 1.18.1.
 
 The GStreamer team is thrilled to announce a new major feature release
 of your favourite cross-platform multimedia framework!
diff --git a/gst-plugins-bad.doap b/gst-plugins-bad.doap
index 57c427405..91af32ecc 100644
--- a/gst-plugins-bad.doap
+++ b/gst-plugins-bad.doap
@@ -35,6 +35,16 @@ real live maintainer, or some actual wide use.
 
  <release>
   <Version>
+   <revision>1.18.1</revision>
+   <branch>1.18</branch>
+   <name></name>
+   <created>2020-10-26</created>
+   <file-release rdf:resource="https://gstreamer.freedesktop.org/src/gst-plugins-bad/gst-plugins-bad-1.18.1.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 63b37523b..5d52ea9d0 100644
--- a/meson.build
+++ b/meson.build
@@ -1,5 +1,5 @@
 project('gst-plugins-bad', 'c', 'cpp',
-  version : '1.18.0.1',
+  version : '1.18.1',
   meson_version : '>= 0.49',
   default_options : [ 'warning_level=1',
                       'buildtype=debugoptimized' ])