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=== release 1.18.3 ===

2021-01-13 21:11:22 +0000  Tim-Philipp Müller <tim@centricular.com>

	* ChangeLog:
	* NEWS:
	* RELEASE:
	* gst-plugins-bad.doap:
	* meson.build:
	  Release 1.18.3

2020-12-21 14:06:53 +0530  Raju Babannavar <raju.babannavar@gmail.com>

	* gst/dvbsuboverlay/dvb-sub.c:
	  dvbsuboverlay: Add support for dynamic resolution update.
	  Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/1487
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1954>

2020-12-16 18:32:25 +0200  Sebastian Dröge <sebastian@centricular.com>

	* sys/decklink/gstdecklinkaudiosrc.cpp:
	  decklinkaudiosrc: Fix duration of the first audio frame after each discont
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1891>

2020-12-30 22:52:01 +0800  Haihua Hu <jared.hu@nxp.com>

	* ext/dash/gstdashsink.c:
	* ext/dash/gstmpdrootnode.c:
	  dashsink: fix critical log when exit dynamic pipeline
	  availability-start-time and publish-time shared the same
	  GstDateTime object, this object will be unref twice and
	  cause reference count issue. Should use g_value_dup_boxed()
	  to copy this object.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1932>

2020-12-29 09:41:05 +0800  Haihao Xiang <haihao.xiang@intel.com>

	* sys/msdk/gstmsdkcontext.c:
	  msdk: needn't close mfx session when failed
	  Otherwise we will get double free issue because mfx session is closed in
	  finalize. See
	  https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1867#note_739346
	  for the double free issue.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1927>

2020-12-10 11:11:04 +0800  Haihao Xiang <haihao.xiang@intel.com>

	* sys/msdk/gstmsdkcontext.c:
	* sys/msdk/msdk.c:
	  msdk: check GstMsdkContext instead of mfxSession instance
	  When creating a GstMsdkContext instance, it also creates a mfxSession
	  instance, so we may check GstMsdkContext instead of mfxSession instance
	  to make sure MSDK is available. In addition, according to MSDK doc [1],
	  MFXVideoCORE_SetHandle function should be executed before any actual
	  usage of library including queries, otherwise the behavior is
	  unexpected, so we should call MFXVideoCORE_QueryPlatform after
	  MFXVideoCORE_SetHandle on Linux
	  [1] https://github.com/Intel-Media-SDK/MediaSDK/blob/master/doc/mediasdk-man.md#working-with-va-api-applications
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1921>

2020-12-28 04:33:11 +0900  Seungha Yang <seungha@centricular.com>

	* sys/d3d11/gstd3d11shader.c:
	  d3d11shader: Fix ID3DBlob object leak
	  Even if HLSL compiler was able to compile our shader code, D3DCompile()
	  might return ID3DBlob object for compile warnings and the object
	  should be released.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1922>

2020-12-29 13:15:10 +0200  Sebastian Dröge <sebastian@centricular.com>

	* ext/assrender/gstassrender.c:
	  assrender: Don't try unlocking unlocked mutex
	  When flushing right at the beginning of the video chain function or
	  when failing negotiation at the top of the function, the assrender mutex
	  would be unlocked without being previously locked.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1923>

2020-12-20 22:12:44 +0900  Seungha Yang <seungha@centricular.com>

	* sys/d3d11/gstd3d11videosink.c:
	  d3d11videosink: Prepare window once streaming started
	  ... instead of READY state. READY state is too early for setting
	  overlay window handle especially playbin/playsink scenario
	  since playsink will set given overlay handle on videosink once
	  READY state change of videosink is ensured.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1905>

2020-12-10 15:37:14 +0800  Lim Siew Hoon <siew.hoon.lim@intel.com>

	* gst/inter/gstintervideosrc.c:
	  intervideosrc: fix negotiation of interlaced caps
	  In 1.0 the field in caps is called "interlace-mode", not "interlaced".
	  Fixes #1480
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1878>

2020-12-07 14:54:28 +0100  Jan Alexander Steffens (heftig) <jan.steffens@ltnglobal.com>

	* ext/srt/gstsrtobject.c:
	  srt: Don't take object lock calling gst_srt_object_get_stats
	  This function takes the sock lock. This can result in a deadlock when
	  another thread holding the sock lock is trying to take the object lock.
	  Thread A (Holds object lock, wants sock lock):
	  #2  gst_srt_object_get_stats at gst-plugins-bad/ext/srt/gstsrtobject.c:1753
	  #3  gst_srt_object_get_property_helper at gst-plugins-bad/ext/srt/gstsrtobject.c:409
	  #4  gst_srt_sink_get_property at gst-plugins-bad/ext/srt/gstsrtsink.c:95
	  #5  g_object_get_property from libgobject-2.0.so.0
	  Thread B (Holds sock lock, wants object lock):
	  #2  gst_element_post_message_default at gstreamer/gst/gstelement.c:2069
	  #3  gst_element_post_message at gstreamer/gst/gstelement.c:2123
	  #4  gst_element_message_full_with_details at gstreamer/gst/gstelement.c:2259
	  #5  gst_element_message_full at gstreamer/gst/gstelement.c:2298
	  #6  gst_srt_object_send_headers at gst-plugins-bad/ext/srt/gstsrtobject.c:1407
	  #7  gst_srt_object_send_headers at gst-plugins-bad/ext/srt/gstsrtobject.c:1444
	  #8  gst_srt_object_write_to_callers at gst-plugins-bad/ext/srt/gstsrtobject.c:1444
	  #9  gst_srt_object_write at gst-plugins-bad/ext/srt/gstsrtobject.c:1598
	  #10 gst_srt_sink_render at gst-plugins-bad/ext/srt/gstsrtsink.c:179
	  Fixes d2d00e07acc2b1ab1ae5a728ef5dc33c9dee7869.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1863>

2020-12-04 17:02:00 +1100  Matthew Waters <matthew@centricular.com>

	* gst/rtmp2/rtmp/rtmpclient.c:
	* gst/rtmp2/rtmp/rtmpconnection.c:
	* gst/rtmp2/rtmp/rtmpconnection.h:
	  rtmp2/connection: pass the parent cancellable down to the connection
	  Otherwise, when rtpm2src cancels an inflight operation that has a queued
	  message stored, then the rtmp connection operation is not stopped.
	  If the cancellation occurs during rtmp connection start up, then
	  rtpm2src does not have any way of accessing the connection object as it
	  has not been returned yet.  As a result, rtpm2src will cancel, the
	  connection will still be processing things and the
	  GMainContext/GMainLoop associated with the outstanding operation will be
	  destroyed.  All outstanding operations and the rtmpconnection object will
	  therefore be leaked in this case.
	  Fixes: https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/1425
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1865>

2020-12-06 23:57:01 +0000  Tim-Philipp Müller <tim@centricular.com>

	* meson.build:
	  Back to development

=== release 1.18.2 ===

2020-12-06 13:24:10 +0000  Tim-Philipp Müller <tim@centricular.com>

	* ChangeLog:
	* NEWS:
	* RELEASE:
	* gst-plugins-bad.doap:
	* meson.build:
	  Release 1.18.2

2020-08-25 14:56:50 +0100  Chris Bass <floobleflam@gmail.com>

	* ext/ttml/ttmlparse.c:
	  ttmlparse: Handle whitespace before XML declaration
	  When ttmlparse is in, e.g., an MPEG-DASH pipeline, there may be
	  whitespace between successive TTML documents in ttmlparse's accumulated
	  input. As libxml2 will fail to parse documents that have whitespace
	  before the opening XML declaration, ensure that any preceding whitespace
	  is not passed to libxml2.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1859>

2020-08-25 14:54:31 +0100  Chris Bass <floobleflam@gmail.com>

	* ext/ttml/ttmlparse.c:
	  ttmlparse: Ensure only single TTML doc parsed
	  The parser handles only one TTML file at a time, therefore if there are
	  multiple TTML documets in the input, parse only the first.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1859>

2020-12-03 14:12:06 +0100  Edward Hervey <edward@centricular.com>

	* ext/hls/gsthlsdemux.c:
	  hlsdemux: Use actual object for logging
	  i.e. the pad of the stream
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1858>

2020-12-03 06:55:00 -0500  Arun Raghavan <arun@asymptotic.io>

	* ext/curl/gstcurlbasesink.c:
	* ext/curl/gstcurlfilesink.c:
	* ext/curl/gstcurlftpsink.c:
	* ext/curl/gstcurlhttpsink.c:
	* ext/curl/gstcurlhttpsrc.c:
	* ext/curl/gstcurlsmtpsink.c:
	* ext/curl/gstcurlsshsink.c:
	* ext/curl/gstcurltlssink.c:
	  curl: Remove incorrect GST_DEBUG_OBJECT() calls
	  klass is not a GstObject, and these debugs print should likely not be
	  around anyway.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1857>

2020-11-10 14:48:28 +0100  Edward Hervey <edward@centricular.com>

	* gst-libs/gst/adaptivedemux/gstadaptivedemux.c:
	  adaptivedemux: Don't calculate bitrate for header/index fragments
	  They are generally substantially smaller than regular fragments, and therefore
	  we end up pushing totally wrong bitrates downstream.
	  Fixes erratic buffering issues with DASH introduced by
	  66f5e874352016e29f555e3ce693b23474e476db
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1819>

2020-11-09 11:41:10 +0100  Edward Hervey <edward@centricular.com>

	* ext/dash/gstdashdemux.c:
	* gst-libs/gst/adaptivedemux/gstadaptivedemux.c:
	* gst-libs/gst/adaptivedemux/gstadaptivedemux.h:
	  adaptivedemux: Store QoS values on the element
	  Storing it per-stream requires taking the manifest lock which can apparenly be
	  hold for aeons. And since the QoS event comes from the video rendering thread
	  we *really* do not want to do that.
	  Storing it as-is in the element is fine, the important part is knowing the
	  earliest time downstream.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1819>

2020-11-05 13:48:27 +0200  Edward Hervey <edward@centricular.com>

	* gst/mpegtsdemux/mpegtspacketizer.c:
	* gst/mpegtsdemux/mpegtspacketizer.h:
	* gst/mpegtsdemux/mpegtsparse.c:
	  tsparse: Forward incoming timestamps
	  Ensure we properly forward the upstream PTS/DTS on the regular and program
	  source pads. All packets being processed will carry over the latest PTS/DTS (as
	  a reconstructed GstBuffer).
	  Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/1419
	  And properly forward PTS/DTS for program pads (which wasn't the case before)
	  Original patch by Vivia Nikolaidou <vivia@ahiru.eu>
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1849>

2020-12-02 09:39:45 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst-libs/gst/adaptivedemux/gstadaptivedemux.c:
	  adaptivedemux: Don't log with non-GObject objects
	  Instead of using the streams, log with the pad of the streams.
	  https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/1457
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1847>

2020-12-02 15:08:44 +0000  Nicolas Dufresne <nicolas@ndufresne.ca>

	* sys/v4l2codecs/gstv4l2decoder.c:
	  Revert "v4l2codecs: decoder: Unmark previously pending request"
	  This reverts commit a3e6d9fc24098fc27fa3fb10c4d189fa61e67500
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1848>

2020-11-05 17:14:22 +0000  Philippe Normand <philn@igalia.com>

	* gst-libs/gst/player/gstplayer.c:
	  player: Fix get_current_subtitle_track annotation
	  As the info returned is a new object, the annotation should be transfer-full,
	  similarly to the get_current_{audio,video}_track() implementations.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1834>

2020-11-11 18:21:25 +0900  Seungha Yang <seungha@centricular.com>

	* gst/mpegdemux/gstmpegdemux.c:
	  mpegdemux: Set duration on seeking query if possible
	  Set duration on seeking query in the same way as duration query handler.
	  Otherwise application might get confused as if the duration is unknown.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1831>

2020-11-09 18:27:14 +0100  Edward Hervey <edward@centricular.com>

	* gst/mpegtsdemux/mpegtspacketizer.c:
	  mpegtspacketizer: Handle PCR issues with adaptive streams
	  A lot of content producers out there targetting "adaptive streaming" are riddled
	  with non-compliant PCR streams (essentially all the players out there just use
	  PTS/DTS and don't care about the PCR).
	  In order to gracefully cope with these, we detect them appropriately and any
	  small (< 15s) PCR resets get gracefully ignored.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1787>

2020-11-13 17:50:03 +0100  Edward Hervey <edward@centricular.com>

	* gst/mpegtsdemux/mpegtspacketizer.c:
	  mpegtsdemux: Fix off by one error
	  Turns out timestamps of zero are valid :) Fixes issues with streams where the
	  PTS/DTS would be equal to the first PCR.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1820>

2020-11-10 14:48:28 +0100  Edward Hervey <edward@centricular.com>

	* gst-libs/gst/adaptivedemux/gstadaptivedemux.c:
	  adaptivedemux: Don't calculate bitrate for header/index fragments
	  They are generally substantially smaller than regular fragments, and therefore
	  we end up pushing totally wrong bitrates downstream.
	  Fixes erratic buffering issues with DASH introduced by
	  66f5e874352016e29f555e3ce693b23474e476db
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1818>

2020-11-11 18:07:57 +0100  Edward Hervey <edward@centricular.com>

	* ext/hls/gsthlsdemux.c:
	  hlsdemux: Don't double-free variant streams on errors
	  If an error happened switching to a new variant, we switch back to the previous
	  one ... except it will be unreffed when settin git.
	  In order to avoid such issues, keep a reference to the old variant until we're
	  sure we don't need it anymore
	  Fixes cases of double-free on variants and its contents
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1817>

2020-10-30 14:07:02 +0000  Julian Bouzas <julian.bouzas@collabora.com>

	* sys/nvcodec/gstcudautils.c:
	  nvcodec: leave g_once_init when all quarks are initialized
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1784>

2020-11-05 13:30:49 +0000  Jason Pereira <mindriot88@users.noreply.github.com>

	* docs/plugins/gst_plugins_cache.json:
	* sys/decklink/gstdecklink.cpp:
	  decklink: correct framerate 2KDCI 23.98
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1774>

2020-11-05 09:01:47 +0100  Rafostar <40623528+Rafostar@users.noreply.github.com>

	* gst-libs/gst/player/gstplayer.c:
	  player: call ref_sink on pipeline
	  Otherwise `gst_player_get_pipeline()` will return a floating reference which may confuse bindings and lead to crash.
	  Fixes #1450
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1770>

2020-11-03 02:22:23 +1100  Jan Schmidt <jan@centricular.com>

	* ext/vulkan/vkdeviceprovider.c:
	  vkdeviceprovider: Avoid deadlock on physical device
	  Don't hold the object lock on the vk physical device while
	  constructing a GstVulkanDevice around it, as
	  GstVulkanDevice can make calls on the physical device that
	  require the object lock.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1759>

2020-11-02 08:46:25 +0000  Randy Li <ayaka@soulik.info>

	* ext/wayland/wlvideoformat.c:
	  wlvideoformat: fix DMA format convertor
	  In the most of case, this typo would work. But for
	  ARGB8888 and XRGB8888, which shm format is not based on fourcc,
	  which would never appear in format enumeration.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1758>

2020-10-31 01:23:36 +1100  Jan Schmidt <jan@centricular.com>

	* tests/check/elements/dtls.c:
	  dtls: Catch bus errors and fail instead of hanging.
	  If the DTLS elements fail, they post a bus error and don't signal any
	  key negotiation. Catch the bus error and fail the test early instead
	  of letting it hang and time out.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1746>

2020-10-30 22:52:18 +1100  Jan Schmidt <jan@centricular.com>

	* ext/sctp/gstsctpdec.c:
	* ext/sctp/gstsctpenc.c:
	  sctp: Do downward state change logic after chaining up.
	  Call the parent state_change function first when changing state
	  downward, to make sure that the element has stopped before cleaning
	  it up.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1746>

2020-10-30 22:49:22 +1100  Jan Schmidt <jan@centricular.com>

	* ext/dtls/gstdtlsconnection.c:
	  dtls: Avoid bio_buffer assertion on shutdown.
	  On shutdown, a previous iteration of dtsl_connection_process()
	  might be incomplete and leave a partial bio_buffer behind.
	  If the DTLS connection is already marked closed, drop out
	  of dtls_connection_process early without asserting.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1746>

2020-10-30 16:31:18 +1100  Jan Schmidt <jan@centricular.com>

	* ext/webrtc/gstwebrtcbin.c:
	  webrtc: Fix a race on shutdown.
	  The main context can disappear in gst_webrtc_bin_enqueue_task()
	  between checking the is_closed flag and enqueueing a source on the
	  main context. Protect the main context with the object lock instead
	  of the PC lock, and hold a ref briefly to make sure it stays alive.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1746>

2020-10-31 21:46:16 +1100  Jan Schmidt <jan@centricular.com>

	* tests/check/elements/dtls.c:
	  tests: Don't set dtlsenc state before linking.
	  Link the dtlsenc in the testsuite before setting it to paused, as it
	  starts a pad task that can generate a not-linked error otherwise.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1745>

2020-10-30 10:02:32 +0200  Sebastian Dröge <sebastian@centricular.com>

	* sys/decklink/gstdecklink.cpp:
	  decklink: Update enum value bounds check in gst_decklink_get_mode()
	  The widescreen modes moved after GST_DECKLINK_MODE_2160p60 and using
	  them now would cause an assertion. This is a regression from
	  309f6187fef890c7ffa49305f38e89beac3b1423.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1740>

2020-10-29 13:43:16 -0400  Xavier Claessens <xavier.claessens@collabora.com>

	* sys/androidmedia/gstamc.c:
	  amc: Fix crash when encoding AVC
	  gstamcvideoenc.c calls gst_amc_avc_profile_to_string() with alternatives
	  set to NULL which causes a crash.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1739>

2020-10-18 17:59:44 +0200  Nicola Murino <nicola.murino@gmail.com>

	* ext/opencv/meson.build:
	  opencv: allow compilation against 4.5.x
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1735>

2020-10-28 00:47:49 +0900  Seungha Yang <seungha@centricular.com>

	* sys/d3d11/gstd3d11videoprocessor.c:
	  d3d11videoprocessor: Fix wrong input/output supportability check
	  The flag argument of ID3D11VideoProcessorEnumerator::CheckVideoProcessorFormat
	  method is output value, not input.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1736>

2020-09-12 02:48:43 +0200  Jan Alexander Steffens (heftig) <heftig@archlinux.org>

	* tests/check/elements/svthevcenc.c:
	  tests: svthevcenc: Fix test_encode_simple
	  Pick the same I420 format the other test use. Without this, the source
	  picks AYUV64, which fails.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1734>

2020-09-23 17:04:55 +0200  Jan Alexander Steffens (heftig) <jan.steffens@ltnglobal.com>

	* gst/mpegtsmux/gstbasetsmux.c:
	  mpegtsmux: Restore intervals when creating TsMux
	  Otherwise the settings from the properties would be overwritten with
	  the defaults.
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1733>

2020-10-27 12:34:33 +0000  Tim-Philipp Müller <tim@centricular.com>

	* meson.build:
	  Back to development

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=== 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:
	* RELEASE:
	* gst-plugins-bad.doap:
	* meson.build:
	  Release 1.18.0

2020-09-07 22:48:27 +0100  Tim-Philipp Müller <tim@centricular.com>

	* meson.build:
	* scripts/dist-translations.py:
	* scripts/meson.build:
	  meson: dist pot file in tarball
	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1558>

2020-09-07 12:14:47 +0300  Sebastian Dröge <sebastian@centricular.com>

	* ext/x265/gstx265enc.c:
	* gst/videoparsers/gsth264parse.c:
	* gst/videoparsers/gsth265parse.c:
	* sys/nvcodec/gstnvdec.c:
	* sys/nvcodec/gstnvh264enc.c:
	* sys/nvcodec/gstnvh265enc.c:
	  gst: Update for gst_video_transfer_function_*() function renaming