Release 1.18.11.18.1

6 files changed, 314 insertions, 22 deletions

diff --git a/ChangeLog b/ChangeLog
index 19b8cce..71fbc00 100644
--- a/ChangeLog
+++ b/ChangeLog
@@ -1,10 +1,86 @@
+=== release 1.18.1 ===
+
+2020-10-26 11:15:28 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* ChangeLog:
+	* NEWS:
+	* RELEASE:
+	* gst-rtsp-server.doap:
+	* meson.build:
+	  Release 1.18.1
+
+2020-10-08 22:17:16 +0200  Mathieu Duponchelle <mathieu@centricular.com>
+
+	* gst/rtsp-server/rtsp-stream.c:
+	  rtsp-stream: make use of blocked_running_time in query_position
+	  When blocking, the sink element will not have received a buffer
+	  yet and the position query will fail. Instead, we make use of
+	  the running time of the buffer we blocked on.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/163>
+
+2020-10-06 00:04:17 +0200  Mathieu Duponchelle <mathieu@centricular.com>
+
+	* gst/rtsp-server/rtsp-stream.c:
+	  rtsp-stream: collect rtp info when blocking
+	  We don't unblock the stream anymore before replying to the
+	  play request (883ddc72bb5bc57c95a9e167814d1ac53fe1b443),
+	  so the sinks don't have a last-sample after potentially flush
+	  seeking. seek_trickmode waits for preroll however, which means
+	  the stream will block and wait for a first buffer. Subsequent
+	  calls to get_rtpinfo() can thus make use of the information.
+	  See https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/issues/115
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/163>
+
+2020-09-05 00:30:42 +0200  Mathieu Duponchelle <mathieu@centricular.com>
+
+	* gst/rtsp-server/rtsp-media.c:
+	* gst/rtsp-server/rtsp-server-internal.h:
+	* gst/rtsp-server/rtsp-stream.c:
+	  rtsp-media: set a 0 storage size for TCP receivers
+	  ulpfec correction is obviously useless when receiving a stream
+	  over TCP, and in TCP modes the rtp storage receives non
+	  timestamped buffers, causing it to queue buffers indefinitely,
+	  until the queue grows so large that sanity checks kick in and
+	  warnings start to get emitted.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/158>
+
+2020-08-21 03:02:40 +0200  Mathieu Duponchelle <mathieu@centricular.com>
+
+	* gst/rtsp-server/rtsp-stream.c:
+	  rtsp-stream: preroll on gap events
+	  This allows negotiating a SDP with all streams present, but only
+	  start sending packets at some later point in time
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/157>
+
+2020-08-25 16:10:36 +0200  Mathieu Duponchelle <mathieu@centricular.com>
+
+	* gst/rtsp-server/rtsp-media.c:
+	  rtsp-media: do not unblock on unsuspend
+	  rtsp_media_unsuspend() is called from handle_play_request()
+	  before sending the play response. Unblocking the streams here
+	  was causing data to be sent out before the client was ready
+	  to handle it, with obvious side effects such as initial packets
+	  getting discarded, causing decoding errors.
+	  Instead we can simply let the media streams be unblocked when
+	  the state of the media is set to PLAYING, which occurs after
+	  sending the play response.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/156>
+
+2020-09-08 17:44:37 +0100  Tim-Philipp Müller <tim@centricular.com>
+
+	* docs/gst_plugins_cache.json:
+	* meson.build:
+	  Back to development
+
 === release 1.18.0 ===
 
 2020-09-08 00:08:29 +0100  Tim-Philipp Müller <tim@centricular.com>
 
+	* .gitlab-ci.yml:
 	* ChangeLog:
 	* NEWS:
 	* RELEASE:
+	* docs/gst_plugins_cache.json:
 	* gst-rtsp-server.doap:
 	* meson.build:
 	  Release 1.18.0
diff --git a/NEWS b/NEWS
index dba9c7c..c427651 100644
--- a/NEWS
+++ b/NEWS
@@ -1,11 +1,14 @@
 GStreamer 1.18 Release Notes
 
-GStreamer 1.18.0 was originally released on 7 September 2020.
+GStreamer 1.18.0 was originally released on 8 September 2020.
+
+The latest bug-fix release in the 1.18 series is 1.18.1 and was released
+on 26 October 2020.
 
 See https://gstreamer.freedesktop.org/releases/1.18/ for the latest
 version of this document.
 
-Last updated: Monday 7 September 2020, 10:30 UTC (log)
+Last updated: Monday 26 October 2020, 11:00 UTC (log)
 
 Introduction
 
@@ -28,7 +31,8 @@ Highlights
 
 -   Active Format Description (AFD) and Bar Data support
 
--   ONVIF trick modes support in both GStreamer RTSP server and client
+-   RTSP server and client implementations gained ONVIF trick modes
+    support
 
 -   Hardware-accelerated video decoding on Windows via DXVA2 /
     Direct3D11
@@ -39,24 +43,25 @@ Highlights
 -   qmlgloverlay: New overlay element that renders a QtQuick scene over
     the top of an input video stream
 
--   New imagesequencesrc element to easily create a video stream from a
+-   imagesequencesrc: New element to easily create a video stream from a
     sequence of jpeg or png images
 
--   dashsink: Add new sink to produce DASH content
+-   dashsink: New sink to produce DASH content
 
--   dvbsubenc: DVB Subtitle encoder element
+-   dvbsubenc: New DVB Subtitle encoder element
 
--   TV broadcast compliant MPEG-TS muxing with constant bitrate muxing
-    and SCTE-35 support
+-   MPEG-TS muxing now also supports TV broadcast compliant muxing with
+    constant bitrate muxing and SCTE-35 support
 
--   rtmp2: new RTMP client source and sink element implementation
+-   rtmp2: New RTMP client source and sink element from-scratch
+    implementation
 
--   svthevcenc: new SVT-HEVC-based H.265 video encoder
+-   svthevcenc: New SVT-HEVC-based H.265 video encoder
 
--   vaapioverlay compositor element using VA-API
+-   vaapioverlay: New compositor element using VA-API
 
--   rtpmanager support for Google’s Transport-Wide Congestion Control
-    (twcc) RTP extension
+-   rtpmanager gained support for Google’s Transport-Wide Congestion
+    Control (twcc) RTP extension
 
 -   splitmuxsink and splitmuxsrc gained support for auxiliary video
     streams
@@ -64,18 +69,18 @@ Highlights
 -   webrtcbin now contains some initial support for renegotiation
     involving stream addition and removal
 
--   New RTP source and sink elements to easily set up RTP streaming via
-    rtp:// URIs
+-   RTP support was enhanced with new RTP source and sink elements to
+    easily set up RTP streaming via rtp:// URIs
 
--   New Audio Video Transport Protocol (AVTP) plugin for Time-Sensitive
-    Applications
+-   avtp: New Audio Video Transport Protocol (AVTP) plugin for
+    Time-Sensitive Applications
 
 -   Support for the Video Services Forum’s Reliable Internet Stream
     Transport (RIST) TR-06-1 Simple Profile
 
 -   Universal Windows Platform (UWP) support
 
--   rpicamsrc element for capturing from the Raspberry Pi camera
+-   rpicamsrc: New element for capturing from the Raspberry Pi camera
 
 -   RTSP Server TCP interleaved backpressure handling improvements as
     well as support for Scale/Speed headers
@@ -2179,7 +2184,208 @@ the git 1.18 branch, which will be a stable branch.
 
 1.18.0
 
-1.18.0 was released on 7 September 2020.
+1.18.0 was released on 8 September 2020.
+
+1.18.1
+
+The first 1.18 bug-fix release (1.18.1) was released on 26 October 2020.
+
+This release only contains bugfixes and it should be safe to update from
+1.18.0.
+
+Highlighted bugfixes in 1.18.1
+
+-   important security fixes
+-   bug fixes and memory leak fixes
+-   various stability and reliability improvements
+
+gstreamer
+
+-   aggregator: make peek() has() pop() drop() buffer API threadsafe
+-   gstvalue: don’t write to const char *
+-   meson: Disallow DbgHelp for UWP build
+-   info: Fix build on Windows ARM64 device
+-   build: use cpu_family for arch checks
+-   basetransform: Fix in/outbuf confusion of _default_transform_meta
+-   Fix documentation
+-   info: Load DbgHelp.dll using g_module_open()
+-   padtemplate: mark documentation caps as may be leaked
+-   gstmeta: intern registered impl string
+-   aggregator: Hold SRC_LOCK while unblocking via SRC_BROADCAST()
+-   ptp_helper_post_install.sh: deal with none
+-   skip elements/leak.c if tracer is not available
+-   aggregator: Wake up source pad in PAUSED<->PLAYING transitions
+-   input-selector: Wake up blocking pads when releasing them
+-   ptp: Also handle gnu/kfreebsd
+
+gst-plugins-base
+
+-   theoradec: Set telemetry options only if they are nonzero
+-   glslstage: delete shader on finalize of stage
+-   urisourcebin: Fix crash caused by use after free
+-   decodebin3: Store stream-start event on output pad before exposing
+    it
+-   Add some missing nullable annotations
+-   typefind/xdgmime: Validate mimetypes to be valid GstStructure names
+    before using them
+-   uridecodebin3: Forward upstream events to decodebin3 directly
+-   video-converter: Add fast paths from v210 to I420/YV12, Y42B, UYVY
+    and YUY2
+-   videoaggregator: Limit accepted caps by template caps
+-   gstrtpbuffer: fix header extension length validation
+-   decodebin3: only force streams-selected seqnum after a
+    select-streams
+-   videodecoder: don’t copy interlace-mode from reference state
+-   enable abi checks
+-   multihandlesink: Don’t pass NULL caps to gst_caps_is_equal
+-   audio: video: Fix in/outbuf confusion of transform_meta
+-   meson: Always wrap “prefix” option with join_paths() to make Windows
+    happy
+-   videoaggregator: ensure peek_next_sample() uses the correct caps
+-   meson: Actually build gstgl without implicit include dirs
+-   videoaggregator: Don’t require any pads to be configured for
+    negotiating source pad caps
+-   gst-libs: gl: Fix documentation typo and clarify
+    gl_memory_texsubimage
+-   audioaggregator: Reset offset if the output rate is renegotiated
+-   video-anc: Implement transform functions for AFD/Bar metas
+-   appsrc: Wake up the create() function on caps changes
+-   rtpbasepayload: do not forget delayed segment when forwarding gaps
+
+gst-plugins-good
+
+-   v4l2object: Only offer inactive pools and if needed
+-   vpx: Fix the check to unfixed/unknown framerate to set bitrate
+-   qmlglsink: fix crash when created/destroyed in quick succession
+-   rtputils: Count metas with an empty tag list for copying/keeping
+-   rtpbin: Remove the rtpjitterbuffer with the stream
+-   rtph26*depay: drop FU’s without a corresponding start bit
+-   imagefreeze: Response caps query from srcpad
+-   rtpmp4gdepay: Allow lower-case “aac-hbr” instead of correct
+    “AAC-hbr”
+-   rtspsrc: Fix push-backchannel-buffer parameter mismatch
+-   jpegdec: check buffer size before dereferencing
+-   flvmux: Move stream skipping to GstAggregatorPadClass.skip_buffer
+-   v4l2object: plug memory leak
+-   splitmuxsink: fix sink pad release while PLAYING
+
+gst-plugins-bad
+
+-   codecparsers: h264parser: guard against ref_pic_markings overflow
+-   v4l2codecs: Various fixes
+-   h265parse: Don’t enable passthrough by default
+-   srt: Fix “Fix timestamping”
+-   srt: Fixes for 1.4.2
+-   dtlsconnection: Ignore OpenSSL system call errors
+-   h265parse: set interlace-mode=interleaved on interlaced content
+-   Replace GPL v2 with LGPL v2 in COPYING file
+-   srt: Consume the error from gst_srt_object_write
+-   srt: Check socket state before retrieving payload size
+-   x265enc: fix deadlock on reconfig
+-   webrtc: Require gstreamer-sdp in the pkg-config file
+-   srtsrc: Fix timestamping
+-   mfvideosrc: Use only the first video stream per device
+-   srtobject: typecast SRTO_LINGER to linger
+-   decklink: Correctly order the different dependent mode tables
+-   wasapisrc: Make sure that wasapisrc produces data in loopback mode
+-   wpesrc: fix some caps leaks using the non-GL output
+-   smoothstreaming: clear live adapter on seek
+-   vtdec/vulkan: use Shared storage mode for IOSurface textures
+-   wpe: Move webview load waiting to WPEView
+-   wpe: Use proper callback for TLS errors signal handling
+-   kmssink: Do not source using padded width/height
+-   avtp: avtpaafdepay: fix crash when building caps
+-   opencv: set opencv_dep when option is disabled to fix the build
+-   line21encoder: miscellaneous enhancements
+-   Hls youtube issues with urisourcebin/queue2
+-   rtmp2: Replace stats queue with stats lock
+-   rtmp2sink: support EOS event for graceful connection shutdown
+-   mpegtsmux: Make handling of sinkpads thread-safe
+-   hlssink2: Actually release splitmuxsink’s pads
+-   mpegtsmux: Don’t create streams with reserved PID
+
+gst-plugins-ugly
+
+-   no changes
+
+gst-libav
+
+-   avaudenc/avvidenc: Reopen encoding session if it’s required
+-   avauddec/audenc/videnc: Don’t return GST_FLOW_EOS when draining
+-   avauddec/avviddec: Avoid dropping non-OK flow return
+-   avcodecmap: Enable 24 bit WMA Lossless decoding
+
+gst-rtsp-server
+
+-   rtsp-stream: collect rtp info when blocking
+-   rtsp-media: set a 0 storage size for TCP receivers
+-   rtsp-stream: preroll on gap events
+-   rtsp-media: do not unblock on unsuspend
+
+gstreamer-vaapi
+
+-   decoder: don’t reply src caps query with allowed if pad is fixed
+-   plugins: decode: fix a DMA caps typo in ensure_allowed_srcpad_caps
+
+gstreamer-sharp
+
+-   Add bindings for some missing 1.18 API
+
+gst-omx
+
+-   omxvideodec: support interlace-mode=interleaved input
+
+gst-python
+
+-   no changes
+
+gst-editing-services
+
+-   ges: Do not recreate auto-transitions when changing clip assets
+-   ges: Fix a copy/paste mistake in meson file
+
+gst-integration-testsuites
+
+-   medias: Update for h265parse passthrough behavior change
+-   update validate.test.h265parse.alternate test
+
+gst-build
+
+-   windows: Detect Strawberry Perl and error out early
+-   {pygobject,pycairo}.wrap: point to stable refs
+
+Cerbero build tool and packaging changes in 1.18.1
+
+-   Add macOS Big Sur support
+-   gst-plugins-bad: Ship rtpmanagerbad plugin
+-   gstreamer-1.0: Don’t enable DbgHelp for UWP build
+-   pango: fix font corruption on windows
+-   cairo: use thread local storage to grant one windows HDC per thread
+-   small fixes for Xcode 12
+-   cerbero: Re-add alsa-devel to bootstrap on Linux
+-   FreeType: update to 2.10.4 to fix security vulnerability
+
+Contributors to 1.18.1
+
+Aaron Boxer, Adam Williamson, Andrew Wesie, Arun Raghavan, Bastien
+Reboulet, Brent Gardner, Edward Hervey, François Laignel, Guillaume
+Desmottes, Havard Graff, He Junyan, Hosang Lee, Jacek Tomaszewski, Jakub
+Adam, Jan Alexander Steffens (heftig), Jan Schmidt, Jérôme Laheurte,
+Jordan Petridis, Marc Leeman, Marian Cichy, Marijn Suijten, Mathieu
+Duponchelle, Matthew Waters, Michael Tretter, Nazar Mokrynskyi, Nicolas
+Dufresne, Niklas Hambüchen, Nirbheek Chauhan, Olivier Crête, Philippe
+Normand, raghavendra, Ricky Tang, Sebastian Dröge, Seungha Yang,
+sohwan.park, Stéphane Cerveau, Thibault Saunier, Tim-Philipp Müller, Tom
+Schoonjans, Víctor Manuel Jáquez Leal, Will Miller, Xavier Claessens, Xℹ
+Ruoyao, Zebediah Figura,
+
+… and many others who have contributed bug reports, translations, sent
+suggestions or helped testing. Thank you all!
+
+List of merge requests and issues fixed in 1.18.1
+
+-   List of Merge Requests applied in 1.18.1
+-   List of Issues fixed in 1.18.1
 
 Schedule for 1.20
 
diff --git a/RELEASE b/RELEASE
index b68f1e7..27e090d 100644
--- a/RELEASE
+++ b/RELEASE
@@ -1,4 +1,4 @@
-This is GStreamer gst-rtsp-server 1.18.0.
+This is GStreamer gst-rtsp-server 1.18.1.
 
 The GStreamer team is thrilled to announce a new major feature release
 of your favourite cross-platform multimedia framework!
diff --git a/docs/gst_plugins_cache.json b/docs/gst_plugins_cache.json
index 8fadb77..1a2bd7d 100644
--- a/docs/gst_plugins_cache.json
+++ b/docs/gst_plugins_cache.json
@@ -321,7 +321,7 @@
                         "construct": false,
                         "construct-only": false,
                         "controllable": false,
-                        "default": "GStreamer/1.18.0.1",
+                        "default": "GStreamer/1.18.1",
                         "mutable": "null",
                         "readable": true,
                         "type": "gchararray",
diff --git a/gst-rtsp-server.doap b/gst-rtsp-server.doap
index cd72528..8fd62ec 100644
--- a/gst-rtsp-server.doap
+++ b/gst-rtsp-server.doap
@@ -32,6 +32,16 @@ RTSP server library based on GStreamer
 
  <release>
   <Version>
+   <revision>1.18.1</revision>
+   <branch>1.18</branch>
+   <name></name>
+   <created>2020-10-26</created>
+   <file-release rdf:resource="https://gstreamer.freedesktop.org/src/gst-rtsp-server/gst-rtsp-server-1.18.1.tar.xz" />
+  </Version>
+ </release>
+
+ <release>
+  <Version>
    <revision>1.18.0</revision>
    <branch>master</branch>
    <name></name>
diff --git a/meson.build b/meson.build
index b8a236b..6628069 100644
--- a/meson.build
+++ b/meson.build
@@ -1,5 +1,5 @@
 project('gst-rtsp-server', 'c',
-  version : '1.18.0.1',
+  version : '1.18.1',
   meson_version : '>= 0.48',
   default_options : ['warning_level=1', 'buildtype=debugoptimized'])