ChangeLog

2020-05-25 20:59:50 +0900  Seungha Yang <seungha@centricular.com>

	* sys/mediafoundation/gstmfcaptureengine.cpp:
	* sys/mediafoundation/gstmfcapturewinrt.cpp:
	* sys/mediafoundation/gstmfsourcereader.cpp:
	* sys/mediafoundation/gstmftransform.cpp:
	* sys/mediafoundation/gstmfutils.cpp:
	* sys/mediafoundation/mediacapturewrapper.cpp:
	  mediafoundation: Use G_BEGIN_DECLS/G_END_DECLS pair everywhere
	  ... instead of extern "c" {} block.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1241>

2020-05-20 23:23:08 +0900  Seungha Yang <seungha@centricular.com>

	* sys/mediafoundation/AsyncOperations.h:
	* sys/mediafoundation/gstmfcapturewinrt.cpp:
	* sys/mediafoundation/gstmfcapturewinrt.h:
	* sys/mediafoundation/gstmfdevice.c:
	* sys/mediafoundation/gstmfsourceobject.c:
	* sys/mediafoundation/gstmfvideosrc.c:
	* sys/mediafoundation/mediacapturewrapper.cpp:
	* sys/mediafoundation/mediacapturewrapper.h:
	* sys/mediafoundation/meson.build:
	* sys/mediafoundation/plugin.c:
	  mediafoundation: Add support video capture on UWP app
	  New video capture implementation using WinRT Media APIs for UWP app.
	  Due to the strict permission policy of UWP, device enumeration and
	  open should be done via new WinRT APIs and to get permission from users,
	  it will invoke permission dialog on UI.
	  Strictly saying, this implementation is not a part of MediaFoundation
	  but structurally it's very similar to MediaFoundation API.
	  So we can avoid some code duplication by adding this implementation
	  into MediaFoundation plugin.
	  This implementation requires UniversalApiContract version >= 6.0
	  which is part of Windows 10 version 1803 (Redstone 4)
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1241>

2020-05-20 23:56:38 +0900  Seungha Yang <seungha@centricular.com>

	* sys/mediafoundation/gstmfsourceobject.c:
	* sys/mediafoundation/gstmfsourceobject.h:
	  mfsourceobject: Move device name, path, and index to public struct
	  ... so that subclass can access each value and update them.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1241>

2020-05-20 22:59:19 +0900  Seungha Yang <seungha@centricular.com>

	* sys/mediafoundation/gstmfcaptureengine.cpp:
	* sys/mediafoundation/gstmfsourceobject.c:
	* sys/mediafoundation/gstmfsourceobject.h:
	* sys/mediafoundation/gstmfsourcereader.cpp:
	  mediafoundation: Fix typo in source object impl.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1241>

2020-05-25 15:36:38 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/audiobuffersplit/gstaudiobuffersplit.c:
	  audiobuffersplit: Unset DISCONT flag if not discontinuous
	  And also set/unset the RESYNC flag accordingly.
	  It can happen that the flag is preserved by GstAdapter from the input
	  buffer. For example if a big input buffer is split into many small ones,
	  each of the small ones would have the flag set.
	  All other buffer flags seem safe to keep here if they were set,
	  including the GAP flag.
	  Also ensure that the buffer is actually writable before changing any
	  flags or metadata on it.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1298>

2020-05-25 19:22:50 +0900  Seungha Yang <seungha@centricular.com>

	* sys/mediafoundation/gstmftransform.cpp:
	  mftransform: Clear unused output IMediaSample
	  If MFT doesn't produce encoded output, need to free allocated
	  output sample and buffer.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1297>

2020-05-25 01:49:00 +1000  Jan Schmidt <jan@centricular.com>

	* gst/mpegtsdemux/tsdemux.c:
	  tsdemux: Handle old streams claiming to be HDMV with Opus
	  GStreamer 1.16 and earlier produced streams with HDMV registration id
	  but with Opus audio streams on the stream ID that AC-4 now uses. Make
	  sure those still play back by special casing the check for AC-4 in HDMV
	  Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/1295
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1296>

2020-05-24 06:22:07 +1000  Jan Schmidt <jan@centricular.com>

	* ext/srt/gstsrtobject.c:
	  srt: Don't leak the connection_poll_id on close()
	  Attempting to reach an inactive SRT peer in caller mode
	  was leaking an fd every few seconds in the gst_srt_object_close()/open()
	  loop.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1293>

2020-05-24 19:12:28 +0900  Seungha Yang <seungha@centricular.com>

	* sys/mediafoundation/gstmfvideoenc.cpp:
	  mfvideoenc: Fix huge memory leak
	  Subclass must unref passed GstVideoCodecFrame on GstVideoEncoder::handle_frame()
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1294>

2020-05-19 10:47:25 -0400  Thibault Saunier <tsaunier@igalia.com>

	* ext/soundtouch/gstpitch.cc:
	  pitch: Remove useless restriction on number of channel
	  It handles any number of channels just fine
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1292>

2020-05-23 02:33:24 +0900  Seungha Yang <seungha@centricular.com>

	* gst-libs/gst/codecs/gsth264decoder.c:
	  h264decoder: Disallow multiple slice group as we don't support FMO
	  Even though it might be supported by accelerator, baseclass is not
	  ready to support it.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1291>

2020-05-23 00:57:23 +0900  Seungha Yang <seungha@centricular.com>

	* sys/nvcodec/gstnvh264dec.c:
	  nvh264sldec: Fix wrong scaling list matrix scan order
	  Quatization matrix of NVDEC should be raster scan order but
	  h264parser stores it in zig-zag scan order. We need to convert
	  the matrix.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1290>

2020-05-21 11:20:39 +0000  Andrey Sazonov <andrey.sazonov@intel.com>

	* gst/asfmux/gstasfmux.c:
	  asfmux: consistent sscanf args usage
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1286>

2020-05-20 07:35:28 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2codecs/gstv4l2codech264dec.c:
	  v4l2codecs: h264: Add missing break
	  There was a missing break for the 4:4:4 case which would break the sizeimage
	  calculation. We don't currently have hardware that supports 4:4:4, so this
	  code wasn't tested. This was detected by Coverity.
	  CID 1463592 1463591
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1283>

2020-05-21 14:28:38 +0000  Andrey Sazonov <andrey.sazonov@intel.com>

	* gst-libs/gst/audio/gstplanaraudioadapter.c:
	  planaraudioadapter: fix possible NULL ptr dereference
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1288>

2020-05-21 11:24:51 +0000  Andrey Sazonov <andrey.sazonov@intel.com>

	* gst/sdp/gstsdpdemux.c:
	  sdpdemux: fix klocwork issues
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1287>

2020-05-19 14:58:35 +1000  Matthew Waters <matthew@centricular.com>

	* sys/androidmedia/gstamcvideodec.c:
	  amc/videodec: only retrieve the stride/slice-height for raw output
	  When outputting to a surface, these values may not exist.
	  As found on a Google Pixel 3.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1284>

2020-05-14 17:13:00 +0200  Stéphane Cerveau <scerveau@collabora.com>

	* ext/openjpeg/meson.build:
	  meson: add libopenjp2 fallback for openjpeg
	  As a wrap is now available in gst-build, the fallback
	  can be used.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1270>

2020-05-13 15:02:41 -0700  Ederson de Souza <ederson.desouza@intel.com>

	* ext/avtp/meson.build:
	  avtp: Add libavtp fallback dependence
	  So that libavtp can be found if added as subproject on gst-build.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1271>

2020-05-01 15:58:09 +0900  Seungha Yang <seungha@centricular.com>

	* sys/mediafoundation/gstmfdevice.c:
	* sys/mediafoundation/gstmfdevice.h:
	* sys/mediafoundation/meson.build:
	* sys/mediafoundation/plugin.c:
	  mediafoundation: Add device provider implementation
	  Only static device probing is supported for now
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1236>

2020-05-01 15:12:43 +0900  Seungha Yang <seungha@centricular.com>

	* sys/mediafoundation/gstmfsourceobject.c:
	  mfsourceobject: Store selected device path, name and index
	  Update path, name and index with selected device so that checked by
	  get_property() after constructed.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1236>

2020-05-20 10:54:21 +0200  Edward Hervey <edward@centricular.com>

	* gst/rtmp2/gstrtmp2src.c:
	  rtmp2src: Answer scheduling query
	  Just like for rtmpsrc, we must inform downstream that we are a
	  sequential (i.e. don't do random access efficiently) and
	  bandwith-limited (i.e. might need buffering downstream) element
	  Fixes buffering issues with playbin3
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1282>

2020-05-06 12:27:56 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2codecs/gstv4l2codech264dec.c:
	* sys/v4l2codecs/gstv4l2codecvp8dec.c:
	* sys/v4l2codecs/gstv4l2decoder.c:
	* sys/v4l2codecs/gstv4l2decoder.h:
	  v4l2slh264dec: Request large enough bitstream buffer
	  The Cedrus driver would otherwise choose 1KB buffer, which is too small.
	  This follows what some drivers do, which is simply to use the size a
	  packed raw image would have. Specifications do not really guaranty any minimum
	  compression ratio.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1268>

2020-05-05 17:55:19 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2codecs/gstv4l2codech264dec.c:
	* sys/v4l2codecs/gstv4l2codecvp8dec.c:
	* sys/v4l2codecs/gstv4l2decoder.c:
	* sys/v4l2codecs/gstv4l2decoder.h:
	  v4l2slh264dec: Add slice based decoder support
	  This adds support for slice based decoder like the Allwinner/Cedrus driver. In
	  order to keep things efficient, we hold the sink buffer until we reach the end
	  of the picture. Note that as we don't know which one is last, we lazy queue the
	  slices. This effectively introduces one slice latency.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1268>

2020-04-30 15:17:05 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2codecs/gstv4l2codech264dec.c:
	* sys/v4l2codecs/gstv4l2codecvp8dec.c:
	  v4l2codecdec: Fix error handling
	  If none of the format the HW produce is supported, the fiter will be NULL,
	  which would lead to assertion when trying to release it.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1268>

2020-04-30 14:18:47 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2codecs/gstv4l2decoder.c:
	* sys/v4l2codecs/gstv4l2format.c:
	  v4l2decoder: Add legacy non-multiplanar support
	  The Cedrus driver uses the lagacy buffer type (non-mplane). This automatically
	  detect and use the right v4l2_buf_type. This also affect code using
	  v4l2_buffer and v4l2_format structures.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1268>

2020-05-05 17:50:22 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2codecs/linux/h264-ctrls.h:
	* sys/v4l2codecs/linux/types-compat.h:
	* sys/v4l2codecs/linux/v4l2-common.h:
	* sys/v4l2codecs/linux/v4l2-controls.h:
	* sys/v4l2codecs/linux/videodev2.h:
	  v4l2codecs: Update kernel headers
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1268>

2020-05-16 21:52:59 +0900  Seungha Yang <seungha@centricular.com>

	* sys/d3d11/gstd3d11colorconvert.c:
	* sys/d3d11/gstd3d11colorconvert.h:
	  d3d11convert: Fix fallback texture setup when resolution is not even number
	  When texture format is semi-planar, resolution should be even number,
	  and add missing P016 format handling
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1275>

2020-05-16 21:45:02 +0900  Seungha Yang <seungha@centricular.com>

	* sys/d3d11/gstd3d11colorconvert.c:
	  d3d11convert: Fix fallback texture copy
	  Fix texture copy when input texture has non-zero subresource index
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1275>

2020-05-16 20:45:23 +0900  Seungha Yang <seungha@centricular.com>

	* sys/d3d11/gstd3d11colorconvert.c:
	* sys/d3d11/plugin.c:
	  d3d11: Add support for video rescale and rename element to d3d11convert
	  GstD3D11ColorConverter implementation is able to rescale video as well.
	  By doing colorspace conversion and rescale at once, we can save
	  one cycle of shader pipeline which will can save GPU resource.
	  Since this element can support color space conversion and rescale,
	  it's renamed as d3d11convert
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1275>

2020-05-16 20:12:33 +0900  Seungha Yang <seungha@centricular.com>

	* sys/d3d11/gstd3d11colorconvert.c:
	* sys/d3d11/gstd3d11utils.c:
	* sys/d3d11/gstd3d11utils.h:
	  d3d11: Move scoring util method for colorspace conversion to colorconvert element
	  It's used only by colorconvert element.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1275>

2020-05-16 11:14:58 +0200  Víctor Manuel Jáquez Leal <vjaquez@igalia.com>

	* gst-libs/gst/codecs/gsth264decoder.c:
	  codecs: h264decoder: chain finalize vmethod
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1238>

2020-05-13 17:23:12 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* gst-libs/gst/codecparsers/gsth264parser.c:
	  codecparsers: h264: Only set relevant default weight values
	  This is minor optimization to avoid setting values we don't need. It also
	  makes debugging easier since only relevant values a non-zero now.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1238>

2020-05-13 15:32:44 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* gst-libs/gst/codecparsers/gsth264parser.c:
	  codecparsers: h264: Fix default ref list size
	  The default in PPS was not applied properly. The default does not apply for
	  I-Slice and l1 default only applies for B-Slice.  This fixes the slice values
	  for num_ref_idx_l0_active_minus1 and num_ref_idx_l1_active_minus1.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1238>

2020-05-12 12:23:15 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* gst-libs/gst/codecs/gsth264decoder.c:
	  codecs: h264decoder: Use calculated values for max_pic_num/frame_num
	  The parser pre-calculate these already, just use them.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1238>

2020-05-03 17:30:34 +0200  Víctor Manuel Jáquez Leal <vjaquez@igalia.com>

	* gst-libs/gst/codecs/gsth264decoder.c:
	* gst-libs/gst/codecs/gsth264decoder.h:
	* sys/d3d11/gstd3d11h264dec.c:
	* sys/nvcodec/gstnvh264dec.c:
	* sys/v4l2codecs/gstv4l2codech264dec.c:
	  codecs: h264decoder: ref pic lists as decode_slice parameters
	  Pass reference picture lists to decode_slice() vmethods
	  Change gstv4l2codech264dec and gstnvh264dec accordingly.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1238>

2020-04-27 16:53:45 +0200  Víctor Manuel Jáquez Leal <vjaquez@igalia.com>

	* gst-libs/gst/codecs/gsth264decoder.c:
	* gst-libs/gst/codecs/gsth264decoder.h:
	  codecs: h264decoder: handle reference picture lists
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1238>

2020-05-15 14:56:27 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* gst-libs/gst/codecs/gsth264decoder.c:
	* gst-libs/gst/codecs/gsth264picture.c:
	* gst-libs/gst/codecs/gsth264picture.h:
	  codecs: h264decoder: Port from GList to GArray
	  Using glist requires a lot of small allocation at runtime and also
	  it comes with a slow sort algorithm. As we play with that for very
	  frame and slices, use GArray instead. Note that we cache some arrays
	  in the instance as there is no support for stack allocated arrays
	  in GArray.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1238>

2020-05-08 17:56:48 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* gst-libs/gst/codecs/gsth264decoder.c:
	* gst-libs/gst/codecs/gsth264picture.c:
	  codecs: h264decoder: Make get_long_ref_by_pic_num() transfer none
	  We don't use the extra reference, so let's just avoid the extra
	  ref/unref.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1238>

2020-05-06 12:23:34 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* gst-libs/gst/codecs/gsth264decoder.c:
	* gst-libs/gst/codecs/gsth264picture.c:
	  codecs: h264decoder: Make get_short_ref_by_pic_num() transfer none
	  We don't use the extra reference, so let's just avoid the extra
	  ref/unref.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1238>

2020-05-19 15:39:50 +0200  Stéphane Cerveau <scerveau@collabora.com>

	* tests/check/meson.build:
	  tests: fix nalutils file name
	  The filename was too long causing issues with ccache
	  Fix https://gitlab.freedesktop.org/gstreamer/gst-build/-/issues/97
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1281>

2020-05-18 14:19:04 +0200  Jan Alexander Steffens (heftig) <jan.steffens@ltnglobal.com>

	* tests/check/elements/mpegtsdemux.c:
	* tests/check/meson.build:
	  mpegtsdemux: tests: Add simple tests for tsparse and tsdemux
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1274>

2020-05-15 17:05:59 +0200  Jan Alexander Steffens (heftig) <jan.steffens@ltnglobal.com>

	* gst/mpegtsdemux/mpegtsbase.c:
	* gst/mpegtsdemux/mpegtsbase.h:
	* gst/mpegtsdemux/mpegtsparse.c:
	  mpegtsdemux: Close a buffer leak and simplify input_done
	  tsparse leaked input buffers quite badly:
	  GST_TRACERS=leaks GST_DEBUG=GST_TRACER:9 gst-launch-1.0 audiotestsrc num-buffers=3 ! avenc_aac ! mpegtsmux ! tsparse ! fakesink
	  The input_done vfunc was passed the input buffer, which it had to
	  consume. For this reason, the base class takes a reference on the buffer
	  if and only if input_done is not NULL.
	  Before 34af8ed66a7c63048ce0bdf59bbe61011d7c6292, input_done was used in
	  tsparse to pass on the input buffer on the "src" pad. That commit
	  changed the code to packetize for that pad as well and removed the use
	  of input_done.
	  Afterwards, 0d2e9085236ed94622c327f73489e558cc95d05f set input_done
	  again in order to handle automatic alignment of the output buffers to
	  the input buffers. However, it ignored the provided buffer and did not
	  even unref it, causing a leak.
	  Since no code makes use of the buffer provided with input_done, just
	  remove the argument in order to simplify things a bit.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1274>

2020-05-17 10:27:03 +0200  Mats Lindestam <matslm@axis.com>

	* ext/curl/gstcurlhttpsink.c:
	  gstcurlhttpsink: Set 'Expect: 100-continue'-header
	  In the upgrade of libcurl from 7.64.1 to 7.69.1 the
	  EXPECT_100_THRESHOLD has been increased from 1 Kb to 1 Mb
	  (see https://curl.haxx.se/mail/lib-2020-01/0050.html).
	  This caused the gstcurlhttpsink to not being able to rewind
	  and resend in the case, e.g. response '401 Unauthorized'.
	  Now the 'Expect: 100-continue'-header is explicitly set in
	  the gstcurlhttpsink.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1276>

2020-04-29 16:43:06 -0400  Arun Raghavan <arun@asymptotic.io>

	* sys/opensles/openslessink.c:
	* sys/opensles/openslessrc.c:
	  opensles: Remove hard-coded buffer-/latency-time values
	  These were originally required in early Android versions, but are no
	  longer needed.

2020-05-14 20:47:06 +0900  Seungha Yang <seungha@centricular.com>

	* sys/mediafoundation/gstmfcaptureengine.cpp:
	* sys/mediafoundation/gstmfsourceobject.c:
	* sys/mediafoundation/gstmfsourceobject.h:
	* sys/mediafoundation/gstmfsourcereader.cpp:
	  mediafoundation: Refactor GstMFSourceObject implementation
	  * Move CoInitializeEx/CoUninitialize pair into thread function in order to
	  ensure MTA COM thread
	  * Move common code to baseclass
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1269>

2020-05-14 20:17:33 +0900  Seungha Yang <seungha@centricular.com>

	* sys/mediafoundation/gstmfh264enc.cpp:
	* sys/mediafoundation/gstmfh265enc.cpp:
	* sys/mediafoundation/gstmftransform.cpp:
	* sys/mediafoundation/gstmftransform.h:
	* sys/mediafoundation/plugin.c:
	  mediafoundation: Remove COM thread constraints from GstMFTransform object
	  Move CoInitializeEx/CoUninitialize pair into our dedicated thread so that
	  we can ensure COM thread is MTA. This will remove thread constraints
	  around plugin init.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1269>

2020-05-15 13:52:06 +1000  Matthew Waters <matthew@centricular.com>

	* sys/androidmedia/gstamcvideodec.c:
	  amcvideodec: fix sync meta copying not taking a reference
	  Fixup for
	  9b9e39be248389370e80b429da5a528418733483: amc: Fix crash when a sync_meta survives its sink
	  https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/603
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1272>

2020-04-13 18:09:55 +0900  J. Kim <jeongseok.kim@sk.com>

	* ext/srt/gstsrtobject.c:
	  srtobject: add streamid property
	  The stream id starts with '#!::' according to SRT Access Control[1],
	  but GstURI requires URI encoded string.This commit introduces additional
	  property to set the id by normal string.
	  [1] https://github.com/Haivision/srt/blob/master/docs/AccessControl.md

2020-05-12 05:00:36 +0530  Nirbheek Chauhan <nirbheek@centricular.com>

	* ext/modplug/meson.build:
	* ext/openni2/meson.build:
	* meson.build:
	  meson: Pass native: false to add_languages()
	  This is needed for cross-compiling without a build machine compiler
	  available. The option was added in 0.54, but we only need this in
	  Cerbero and it doesn't affect older versions so it should be ok.
	  Will only cause a spurious warning.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1266>

2020-05-12 10:55:45 -0400  Alex Hoenig <alexander.hoenig@progeny.net>

	* gst/mpegtsmux/gstbasetsmux.c:
	  mpegtsmux: detect and ignore gap buffers
	  Fixes #1291.  Without this, when a stream has gaps and then resumes, the next buffer PTS that is written to the TS is given the PTS of the first gap.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1263>

2020-05-12 16:05:01 +1000  Matthew Waters <matthew@centricular.com>

	* ext/closedcaption/gstccconverter.c:
	  ccconverter: check fraction multiply for overflow
	  It should not happen and if it does, something went very wrong earlier
	  CID 1463350
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1262>

2020-05-12 16:01:42 +1000  Matthew Waters <matthew@centricular.com>

	* ext/closedcaption/gstccconverter.c:
	  ccconverter: tighten up a couple of NULL checks
	  CID 1463347
	  CID 1463346
	  CID 1463345
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1262>

2020-05-12 16:00:58 +1000  Matthew Waters <matthew@centricular.com>

	* ext/closedcaption/gstccconverter.c:
	  ccconverter: fix unintialized read of mapped output info in error case
	  We only need to gst_buffer_unmap() if we have gst_buffer_map()ed.  In
	  most cases we can shorten the lenght of time we need to map the output
	  buffer.  Fix similar occurences elsewhere.
	  CID 1463349
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1262>

2020-05-12 15:24:32 +1000  Matthew Waters <matthew@centricular.com>

	* ext/closedcaption/gstccconverter.c:
	  ccconverter: fix uninitialized read in error case
	  CID 1463351
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1262>

2020-05-10 17:38:11 +0800  Ting-Wei Lan <lantw@src.gnome.org>

	* sys/v4l2codecs/gstv4l2codecdevice.c:
	* sys/v4l2codecs/linux/media.h:
	* sys/v4l2codecs/linux/types-compat.h:
	* sys/v4l2codecs/meson.build:
	  v4l2codecs: Fix compilation error on FreeBSD
	  This commit does the following things to fix compilation on FreeBSD:
	  1. Add required typedefs to linux/types-compat.h.
	  2. Remove unnecessary include linux/ioctl.h and replace linux/types.h
	  with linux/types-compat.h. Both files do not exist on FreeBSD.
	  3. Check the header including makedev macro. FreeBSD does not have
	  sys/sysmacros.h, and including it unconditionally causes error.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1259>

2020-05-11 17:14:09 +1000  Matthew Waters <matthew@centricular.com>

	* ext/closedcaption/gstccconverter.c:
	  ccconverter: implement discont handling
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1116>

2020-05-07 23:59:30 +1000  Matthew Waters <matthew@centricular.com>

	* ext/closedcaption/gstccconverter.c:
	* tests/check/elements/ccconverter.c:
	  ccconverter: use a better padding byte sequence for writing cdp
	  0xf8 can be interpreted as cea608 data at the beginning of a cdp packet
	  as the cc_valid bit is not checked when cc_valid in (0b00 or 0b01).
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1116>

2020-03-19 17:42:13 +1100  Matthew Waters <matthew@centricular.com>

	* ext/closedcaption/gstccconverter.c:
	* ext/closedcaption/gstccconverter.h:
	* tests/check/elements/ccconverter.c:
	  ccconverter: split temporary storage into 3
	  Instead of storing the raw cc_data, store the 2 cea608 fields individually
	  as well as the ccp data.
	  Simply copying the input cc_data to the output cc_data violates a number of
	  requirements in the cea708 specification.  The most prominent being, that
	  cea608 triples must be placed at the beginning of each cdp.
	  We also need to comply with the framerate-dpendent limits for both the
	  cea608 and the ccp data which may involve splitting or merging some
	  cea608 data but not ccp data or vice versa.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1116>

2020-03-17 17:23:44 +1100  Matthew Waters <matthew@centricular.com>

	* ext/closedcaption/gstccconverter.c:
	* tests/check/elements/ccconverter.c:
	  ccconvert: compact input cc_data where possible
	  Skip over padding cc_data triples.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1116>

2020-03-13 10:54:02 +1100  Matthew Waters <matthew@centricular.com>

	* ext/closedcaption/gstccconverter.c:
	* ext/closedcaption/gstccconverter.h:
	* tests/check/elements/ccconverter.c:
	  ccconverter: implement support for CDP framerate conversions
	  - Any format involving CDP is supported.
	  - Time codes (if present) are scaled as well.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1116>

2020-03-12 16:08:54 +1100  Matthew Waters <matthew@centricular.com>

	* tests/check/elements/ccconverter.c:
	* tests/check/meson.build:
	  tests/ccconverter: test the time codes are successfully passed through
	  Where time codes are not stored in the caption data themselves
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1116>

2020-03-12 15:06:46 +1100  Matthew Waters <matthew@centricular.com>

	* ext/closedcaption/gstccconverter.c:
	* ext/closedcaption/gstccconverter.h:
	  ccconverter: introduce define for max cdp packet length
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1116>

2020-03-12 15:01:02 +1100  Matthew Waters <matthew@centricular.com>

	* ext/closedcaption/gstccconverter.c:
	  ccconverter: don't rely on external state in *_internal()
	  This allows using the _internal() variants for simply converting some
	  caption data without relying on any external state.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1116>

2020-03-12 14:06:49 +1100  Matthew Waters <matthew@centricular.com>

	* ext/closedcaption/gstccconverter.c:
	* ext/closedcaption/gstccconverter.h:
	* tests/check/elements/ccconverter.c:
	  ccconverter: cc_count limits are per framerate
	  Enforce this and add a test for cdp input being too large.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1116>

2020-03-12 12:54:41 +1100  Matthew Waters <matthew@centricular.com>

	* ext/closedcaption/gstccconverter.c:
	  ccconverter: refactor cdp id, fps, max_cc_count into a table
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1116>

2020-03-12 09:55:40 +1100  Matthew Waters <matthew@centricular.com>

	* ext/closedcaption/gstccconverter.c:
	  ccconverter: pivot to implementing generate_output
	  Will make a n-n buffer element much easier to implement.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1116>

2020-05-09 19:59:46 +0200  Jan Alexander Steffens (heftig) <jan.steffens@ltnglobal.com>

	* gst-libs/gst/vulkan/gstvkerror.c:
	  vulkan: Drop use of VK_RESULT_BEGIN_RANGE
	  This was removed in Vulkan 1.2.140.
	  > Shortly after 2020-04-24, we will be removing the automatically
	  > generated `VK_*_BEGIN_RANGE`, `VK_*_END_RANGE`, and `VK_*_RANGE_SIZE`
	  > tokens from the Vulkan headers. These tokens are currently defined for
	  > some enumerated types, but are explicitly not part of the Vulkan API.
	  > They existed only to support some Vulkan implementation internals,
	  > which no longer require them. We will be accepting comments on this
	  > topic in [#1230], but we strongly suggest any external projects using
	  > these tokens immediately migrate away from them.
	  [#1230]: https://github.com/KhronosGroup/Vulkan-Docs/issues/1230
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1255>

2020-05-08 22:36:01 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/audiobuffersplit/gstaudiobuffersplit.c:
	* gst/audiobuffersplit/gstaudiobuffersplit.h:
	  audiobuffersplit: Perform discont tracking on running time
	  Otherwise we would have to drain on every segment event. Like this we
	  can handle segment events that don't cause a discontinuity in running
	  time to be handled without draining.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1254>

2020-05-08 21:36:44 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/audiobuffersplit/gstaudiobuffersplit.c:
	* gst/audiobuffersplit/gstaudiobuffersplit.h:
	  audiobuffersplit: Keep incoming and outgoing segments separate
	  We might have to drain already queued input based on the old segment
	  before forwarding the new segment event. The new segment is only
	  forwarded after a discont as otherwise we might cause unnecessary
	  timestamp jumps as we output buffers timestamped based on sample counts.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1254>

2020-04-30 16:01:13 +0000  Chris Ayoup <ayochris@amazon.com>

	* ext/webrtc/gstwebrtcbin.c:
	* ext/webrtc/gstwebrtcice.c:
	* ext/webrtc/gstwebrtcice.h:
	  webrtc: move filtering properties to webrtcice
	  We want webrtcbin to only expose properties that are defined in JSEP, so
	  these additional properties should be moved out.  In order to access
	  them, the webrtcice instance is exposed from webrtcbin.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1223>

2020-04-17 18:00:59 +0000  Chris Ayoup <ayochris@amazon.com>

	* ext/webrtc/gstwebrtcbin.c:
	* ext/webrtc/gstwebrtcice.c:
	* ext/webrtc/gstwebrtcice.h:
	  webrtc: allow setting local IP addresses
	  If a local IP address is specified, ICE gathering can be much faster
	  in environments where there are multiple IP addreses but only some are
	  usable (for example, if you are running docker on the machine).
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1223>

2020-04-16 22:37:35 +0000  Chris Ayoup <ayochris@amazon.com>

	* ext/webrtc/gstwebrtcbin.c:
	* ext/webrtc/gstwebrtcice.c:
	  webrtc: Allow toggling TCP and UDP candidates
	  Add some properties to allow TCP and UDP candidates to be toggled.  This
	  is useful in cases where someone is using this element in an environment
	  where it is known in advance whether a given transport will work or not
	  and will prevent wasting time generating and checking candidate pairs
	  that will not succeed.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1223>

2020-04-02 10:44:31 +0800  Haihao Xiang <haihao.xiang@intel.com>

	* sys/msdk/gstmsdkvpp.c:
	  msdkvpp: clear the parameters after closing the session
	  Otherwise the stale values are used for the new process.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1159>

2020-05-10 11:23:02 +0300  Sebastian Dröge <sebastian@centricular.com>

	* ext/spandsp/gstspanplc.c:
	  spanplc: Don't segfault when retrieving the stats property without a spanplc context
	  For example when trying to get the property value in NULL state.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1258>

2020-05-10 11:16:44 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/onvif/gstrtponviftimestamp.c:
	  onviftimestamp: Add missing `break` in set_property()
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1257>

2020-05-07 14:05:16 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* tests/check/elements/h265parse.c:
	  test: h265parse: Test parsing buffer the ends with half a NAL header
	  This test cover the case where we are parsing, but our current buffers ends
	  with half the NAL header (which is 2 bytes in HEVC). Previously we would
	  throw an error message on the bus.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1251>

2020-05-07 13:59:33 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* gst-libs/gst/codecparsers/gsth265parser.c:
	  h265parse: Ensure parsing ends on start-code + full header
	  The parser is used all over the place assuming that after calling
	  gst_h265_parser_identify_nalu(), the start-code found is can also be
	  identified. In H264 this works, because scan_for_start_code rely on
	  gst_byte_reader_masked_scan_uint32() that ensures that 1 byte passed the 3
	  bytes start code is found. But for HEVC, we need two bytes to identify the
	  following NAL.
	  This patch will return NO_NAL_END, even if a start code is found in the case
	  there was not enough bytes. This solution was chosen to maintain backward
	  compatibility, and reduce complexicity.
	  Fixes #1287
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1251>

2020-05-07 11:09:23 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* tests/check/elements/h264parse.c:
	* tests/check/elements/h265parse.c:
	  test: h264/h265: Add test for four bytes start code initial skip
	  This test detects if the parser have skipped too much and dropped meaninful
	  NALs.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1251>

2020-05-07 12:02:40 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* tests/check/elements/h264parse.c:
	* tests/check/elements/h265parse.c:
	* tests/check/elements/parser.c:
	* tests/check/elements/parser.h:
	  test: h264/h265: Constify all test buffers
	  This ensure that no test modify other tests data.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1251>

2020-05-07 11:06:45 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* gst/videoparsers/gsth264parse.c:
	* gst/videoparsers/gsth265parse.c:
	  h264/h265parse: Fix initial skip
	  Account for start codes possibly be 4 bytes. For HEVC, also take into
	  account that we might be missing only one of the two identification
	  bytes.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1251>

2020-05-07 08:29:28 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* gst/videoparsers/gsth265parse.c:
	  h265parse: Ensure correct timestamps
	  If the input has a miss-placed filler zero byte (e.g. a filler without a 4
	  bytes start code on the next NAL), we would endup using the same timestamp
	  twice. Ask the base class to read the timestamp from the buffer were the NAL
	  actually starts.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1251>

2020-05-07 07:43:30 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* gst-libs/gst/codecparsers/gsth264parser.c:
	  h264parser: Removed impossible error case
	  Same as done for H264, this error was trying to catch the case where we had
	  a start code without any bytes afterward. This will never happen since the
	  start code scanner only returns a match if there is one byte after start
	  code (pattern 0x00000100 / mask 0xffffff00). In H264, once byte is sufficient
	  to identify the NALU.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1251>

2020-05-06 22:28:34 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* gst-libs/gst/codecparsers/gsth264parser.c:
	* gst/videoparsers/gsth264parse.c:
	* tests/check/elements/h264parse.c:
	  h264parse: Properly handle 4 bytes start code
	  This will stop stripping four bytes start code. This was fixed and broken
	  again as it was causing the a timestamp shift. We now call
	  gst_base_parse_set_ts_at_offset() with the offset of the first NAL to ensure
	  that fixing a moderatly broken input stream won't affect the timestamps. We
	  also fixes the unit test, removing a comment about the stripping behaviour not
	  being correct.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1251>

2020-05-06 22:18:12 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* gst-libs/gst/codecparsers/gsth265parser.c:
	  h265parser: Fix NAL size check for identification
	  Unlike H264, H265 requires 2 bytes after the start code to allow NAL
	  identification. This would otherwise report a broken NAL and skip
	  important data.
	  Fixes #1287
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1251>

2020-05-06 22:13:45 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* gst-libs/gst/codecparsers/gsth265parser.c:
	  h265parser: Removed impossible error case
	  This error was trying to catch the case where we had a start code without any
	  bytes afterward. This will never happen since the start code scanner only returns
	  a match if there is one byte adter start code (pattern 0x00000100 / mask
	  0xffffff00).
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1251>

2020-04-29 16:19:08 +0800  Xu Guangxin <guangxin.xu@intel.com>

	* sys/msdk/gstmsdkbufferpool.c:
	  msdk: bufferpool: set alignment to video meta
	  else gst_video_meta_validate_alignment will report error like
	  "videometa gstvideometa.c:416:gst_video_meta_validate_alignment: Stride of plane 0 defined in meta (384) is different from the one computed from the alignment (320)"
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1224>

2020-05-06 20:04:17 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/timecode/gsttimecodestamper.c:
	  timecodestamper: Unref latency query after usage
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1249>

2020-05-06 11:47:56 +0300  Sebastian Dröge <sebastian@centricular.com>

	* ext/musepack/gstmusepackdec.c:
	  musepackdec: Don't fail all queries if no sample rate is known yet
	  The sample rate is only needed for the POSITION/DURATION queries and we
	  would otherwise fail important queries like the CAPS query.
	  Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/498
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1248>

2020-05-01 07:46:56 +0200  Luka Blaskovic <lblasc@znode.net>

	* ext/opencv/meson.build:
	  opencv: allow compilation against 4.3.x
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1235>

2020-05-06 15:36:19 +1000  Matthew Waters <matthew@centricular.com>

	* ext/webrtc/webrtcdatachannel.c:
	* tests/check/elements/webrtcbin.c:
	  webrtc: fix an off-by-one calculating low-threshold
	  We were not signalling low-threshold when the previous amount was at
	  exactly the low-threshold mark.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1247>

2020-05-06 15:35:26 +1000  Matthew Waters <matthew@centricular.com>

	* tests/check/elements/webrtcbin.c:
	  webrtc: fix a slightly racy test
	  There is no guarantee that the peer data channel has transitioned to
	  open when we do.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1247>

2020-05-06 15:49:05 +1000  Matthew Waters <matthew@centricular.com>

	* ext/webrtc/gstwebrtcbin.c:
	  webrtc: remove debugging leftover
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1247>

2020-05-06 00:30:34 +1000  Matthew Waters <matthew@centricular.com>

	* ext/webrtc/gstwebrtcbin.c:
	* ext/webrtc/gstwebrtcbin.h:
	* ext/webrtc/sctptransport.c:
	* ext/webrtc/utils.h:
	* ext/webrtc/webrtcdatachannel.c:
	  webrtc: always reply to a promise
	  Otherwise, we defeat the purpose of a promise.
	  We were not replying when the state was closed.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1240>

2020-05-06 00:25:45 +1000  Matthew Waters <matthew@centricular.com>

	* ext/webrtc/gstwebrtcbin.c:
	* ext/webrtc/gstwebrtcice.c:
	* ext/webrtc/gstwebrtcice.h:
	  webrtc: name threads based on the element name
	  Makes debugging a busy loop possibly easier
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1240>

2020-05-05 17:15:51 +1000  Matthew Waters <matthew@centricular.com>

	* tests/check/elements/webrtcbin.c:
	  tests/webrtc: fix a data channel leak in a test
	  test_data_channel_create_after_negotiate overrides the data_channel_data
	  without ever freeing it.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1240>

2020-05-05 17:14:46 +1000  Matthew Waters <matthew@centricular.com>

	* ext/webrtc/gstwebrtcbin.c:
	  webrtc: correctly use the pad template
	  GstHarness uses this for releasing request pads correctly. Fixes
	  numerous leaks in the webrtc unit tests.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1240>

2020-05-05 14:35:10 +1000  Matthew Waters <matthew@centricular.com>

	* ext/webrtc/gstwebrtcbin.c:
	  webrtc: Fix a couple of renegotiation races
	  When negotiating the SDP we should only connect the streams that are
	  actually mentioned in the SDP.  All other streams are not relevant at
	  this time and would likely be part of a future SDP update.  Fixes a
	  couple of the renegotiation webrtc unit tests.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1240>

2020-05-05 14:33:34 +1000  Matthew Waters <matthew@centricular.com>

	* tests/check/elements/webrtcbin.c:
	  tests/webrtc: move bus thread creation earlier
	  Fixes a small deadlock race where the bus watch GSource could execute before
	  the unlock mutex GSource.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1240>

2020-05-04 14:39:45 +1000  Matthew Waters <matthew@centricular.com>

	* tests/check/meson.build:
	  tests: add libnice to the plugin loading whitelist
	  Allows webrtcbin to actually unit test some negotiation scenarios.
	  Without this, only two of the possible 33 tests are being executed.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1240>

2020-05-05 12:01:21 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2codecs/gstv4l2codecvp8dec.c:
	  v4l2slvp8dec: Flip the meaning of segment_feature_mode
	  In section 9.3.4 a), segment_feature_mode have 0 for absolute and 1 for delta,
	  while in 19.2, it says the opposite. But the reference code, which usually
	  rules over the text state that 1 means absolute:
	  if (hdr->update_data)
	  {
	  hdr->abs = bool_get_bit(bool);
	  And uses it with that meaning to decide weither to override the existing value
	  or just add the detla. This fixes multiple decoding issues.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1245>

2020-05-04 15:33:39 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2codecs/gstv4l2codecvp8dec.c:
	  v4l2slvp8dec: Copy header version
	  This field was not copied.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1245>

2020-05-04 14:54:23 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2codecs/gstv4l2codecvp8dec.c:
	  v4l2slvp8dec: Add debugging for reference frames
	  This simply trace the frame number of the references used for decoding.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1245>

2020-05-04 14:52:45 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2codecs/gstv4l2codecvp8dec.c:
	  v4l2slvp8dec: Ensure width/height is always set
	  Our parser strictly read the bitstream. As it's known from DXVA that always
	  having a valid width/height might be needed, use the cached width/height
	  instead of the value from the parser. This didn't impact Hantro driver.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1245>

2020-05-04 14:52:02 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2codecs/gstv4l2codecvp8dec.c:
	  v4l2slvp8dec: Fix debug category name
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1245>