Release 1.18.41.18.4

6 files changed, 266 insertions, 9 deletions

diff --git a/ChangeLog b/ChangeLog
index b0630df..10a3a7c 100644
--- a/ChangeLog
+++ b/ChangeLog
@@ -1,3 +1,87 @@
+=== release 1.18.4 ===
+
+2021-03-15 17:49:53 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* ChangeLog:
+	* NEWS:
+	* RELEASE:
+	* gst-rtsp-server.doap:
+	* meson.build:
+	  Release 1.18.4
+
+2021-02-15 12:07:15 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* tests/check/gst/rtspclientsink.c:
+	  tests: rtspclientsink: fix some leaks
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/192>
+
+2021-02-15 12:26:30 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* gst/rtsp-sink/gstrtspclientsink.c:
+	  rtspclientsink: mark cached caps as maybe-leaked to make leaks tracer happy
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/192>
+
+2021-02-15 12:07:45 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* tests/check/gst/rtspclientsink.c:
+	  rtspclientsink: add unit test for potential shutdown deadlock
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/191>
+
+2021-02-15 12:01:34 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* gst/rtsp-sink/gstrtspclientsink.c:
+	  rtspclientsink: fix deadlock on shutdown before preroll
+	  Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/issues/130
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/191>
+
+2021-02-01 12:16:46 +0100  Branko Subasic <branko@axis.com>
+
+	* gst/rtsp-server/rtsp-stream.c:
+	  rtsp-stream: avoid deadlock in send_func
+	  Currently the send_func() runs in a thread of its own which is started
+	  the first time we enter handle_new_sample(). It runs in an outer loop
+	  until priv->continue_sending is FALSE, which happens when a TEARDOWN
+	  request is received. We use a local variable, cont, which is initialized
+	  to TRUE, meaning that we will always enter the outer loop, and at the
+	  end of the outer loop we assign it the value of priv->continue_sending.
+	  Within the outer loop there is an inner loop, where we wait to be
+	  signaled when there is more data to send. The inner loop is exited when
+	  priv->send_cookie has changed value, which it does when more data is
+	  available or when a TEARDOWN has been received.
+	  But if we get a TEARDOWN before send_func() is entered we will get stuck
+	  in the inner loop because no one will increase priv->session_cookie
+	  anymore.
+	  By not entering the outer loop in send_func() if priv->continue_sending
+	  is FALSE we make sure that we do not get stuck in send_func()'s inner
+	  loop should we receive a TEARDOWN before the send thread has started.
+	  Change-Id: I7338a0ea60ea435bb685f875965f5165839afa20
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/188>
+
+2021-01-22 08:58:23 +0100  Branko Subasic <branko@axis.com>
+
+	* gst/rtsp-server/rtsp-client.c:
+	  rtsp-client: cleanup transports during TEARDOWN
+	  When tunneling RTP over RTSP the stream transports are stored in a hash
+	  table in the GstRTSPClientPrivate struct. They are used for, among other
+	  things, mapping channel id to stream transports when receiving data from
+	  the client. The stream tranports are created and added to the hash table
+	  in handle_setup_request(), but unfortuately they are not removed in
+	  handle_teardown_request(). This means that if the client sends data on
+	  the RTSP connection after it has sent the TEARDOWN, which is often the
+	  case when audio backchannel is enabled, handle_data() will still be able
+	  to map the channel to a session transport and pass the data along to it.
+	  Which eventually leads to a failing assert in gst_rtsp_stream_recv_rtp()
+	  because the stream is no longer joined to a bin.
+	  We avoid this by removing the stream transports from the hash table when
+	  we handle the TEARDOWN request.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/186>
+
+2021-01-14 02:17:42 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* docs/gst_plugins_cache.json:
+	* meson.build:
+	  Back to development
+
 === release 1.18.3 ===
 
 2021-01-13 21:12:06 +0000  Tim-Philipp Müller <tim@centricular.com>
@@ -5,6 +89,7 @@
 	* ChangeLog:
 	* NEWS:
 	* RELEASE:
+	* docs/gst_plugins_cache.json:
 	* gst-rtsp-server.doap:
 	* meson.build:
 	  Release 1.18.3
diff --git a/NEWS b/NEWS
index 1ac82bd..3ada886 100644
--- a/NEWS
+++ b/NEWS
@@ -2,13 +2,13 @@ GStreamer 1.18 Release Notes
 
 GStreamer 1.18.0 was originally released on 8 September 2020.
 
-The latest bug-fix release in the 1.18 series is 1.18.3 and was released
-on 13 January 2021.
+The latest bug-fix release in the 1.18 series is 1.18.4 and was released
+on 15 March 2021.
 
 See https://gstreamer.freedesktop.org/releases/1.18/ for the latest
 version of this document.
 
-Last updated: Wednesday 13 January 2021, 20:00 UTC (log)
+Last updated: Monday 15 March 2021, 13:00 UTC (log)
 
 Introduction
 
@@ -2717,6 +2717,168 @@ List of merge requests and issues fixed in 1.18.3
 -   List of Merge Requests applied in 1.18.3
 -   List of Issues fixed in 1.18.3
 
+1.18.4
+
+The fourth 1.18 bug-fix release (1.18.4) was released on 15 March 2021.
+
+This release only contains bugfixes and security fixes and it should be
+safe to update from 1.18.x.
+
+Highlighted bugfixes in 1.18.4
+
+-   important security fixes for ID3 tag reading, matroska and realmedia
+    parsing, and gst-libav audio decoding
+-   audiomixer, audioaggregator: input buffer handling fixes
+-   decodebin3: improve stream-selection message handling
+-   uridecodebin3: make “caps” property work
+-   wavenc: fix writing of INFO chunks in some cases
+-   v4l2: bt601 colorimetry, allow encoder resolution changes, fix
+    decoder frame rate negotiation
+-   decklinkvideosink: fix auto format detection, and fixes for 29.97fps
+    framerate output
+-   mpeg-2 video handling fixes when seeking
+-   avviddec: fix bufferpool negotiation and possible memory corruption
+    when changing resolution
+-   various stability, performance and reliability improvements
+-   memory leak fixes
+-   build fixes: rpicamsrc, qt overlay example, d3d11videosink on UWP
+
+gstreamer
+
+-   info: Don’t leak log function user_data if the debug system is
+    compiled out
+-   task: Use SetThreadDescription() Win32 API for setting thread names,
+    which preserves thread names in dump files.
+-   buffer, memory: Mark info in map functions as caller-allocates and
+    pass allocation params as const pointers where possible
+-   clock: define AUTO_CLEANUP_FREE_FUNC for GstClockID
+
+gst-plugins-base
+
+-   tag: id3v2: fix frame size check and potential invalid reads
+-   audio: Fix gst_audio_buffer_truncate() meta handling for
+    non-interleaved audio
+-   audioresample: respect buffer layout when draining
+-   audioaggregator: fix input_buffer ownership
+-   decodebin3: change stream selection message owner, so that the app
+    sends the stream-selection event to the right element
+-   rtspconnection: correct data_size when tunneled mode
+-   uridecodebin3: make caps property work
+-   video-converter: Don’t upsample invalid lines
+-   videodecoder: Fix racy critical when pool negotiation occurs during
+    flush
+-   video: Convert gst_video_info_to_caps() to take self as const ptr
+-   examples: added qt core dependency for qt overlay example
+
+gst-plugins-good
+
+-   matroskademux: header parsing fixes
+-   rpicamsrc: depend on posix threads and vchiq_arm to fix build on
+    raspios again
+-   wavenc: Fixed INFO chunk corruption, caused by odd sized data not
+    being padded
+-   wavpackdec: Add floating point format support to fix distortions in
+    some cases
+-   v4l2: recognize V4L2 bt601 colorimetry again
+-   v4l2videoenc: support resolution change stream encode
+-   v4l2h265codec: fix HEVC profile string issue
+-   v4l2object: Need keep same transfer as input caps
+-   v4l2videodec: Fix vp8 and vp9 streams can’t play on board with
+    vendor bsp
+-   v4l2videodec: fix src side frame rate negotiation
+
+gst-plugins-bad
+
+-   avwait: Don’t post messages with the mutex locked
+-   d3d11h264dec: Reconfigure decoder object on DPB size change and keep
+    track of actually configured DPB size
+-   dashsink: fix double unref of sinkpad caps
+-   decklinkvideosink: Use correct numerator for 29.97fps
+-   decklinkvideosink: fix auto format detection
+-   decklinksrc: Use a more accurate capture time
+-   d3d11videosink: Fix build error on UWP
+-   interlace: negotiation and buffer leak fixes
+-   mpegvideoparse: do not clip, so decoder receives data from keyframe
+    even if it’s before the segment start
+-   mpegtsparse: Fix switched DTS/PTS when set-timestamps=false
+-   nvh264sldec: Reopen decoder object if larger DPB size is required
+-   sdpsrc: fix double free if sdp is provided as string via the
+    property
+-   vulkan: Fix elements long name.
+
+gst-plugins-ugly
+
+-   rmdemux: Make sure we have enough data available when parsing
+    audio/video packets
+
+gst-libav
+
+-   avviddec: take the maximum of the height/coded_height
+-   viddec: don’t configure an incorrect buffer pool when receiving a
+    gap event
+-   audiodec: fix stack overflow in gst_ffmpeg_channel_layout_to_gst()
+
+gst-rtsp-server
+
+-   rtspclientsink: fix deadlock on shutdown if no data has been
+    received yet
+-   rtspclientsink: fix leaks in unit tests
+-   rtsp-stream: avoid deadlock in send_func
+-   rtsp-client: cleanup transports during TEARDOWN
+
+gstreamer-vaapi
+
+-   h264 encoder: append encoder exposure to aud
+-   postproc: Fix a problem of propose_allocation when passthrough
+-   glx: Iterate over FBConfig and select 8 bit color size
+
+gstreamer-sharp
+
+-   no changes
+
+gst-omx
+
+-   no changes
+
+gst-python
+
+-   no changes
+
+gst-editing-services
+
+-   group: Use proper group constructor
+
+gst-integration-testsuites
+
+-   no changes
+
+gst-build
+
+-   no changes
+
+Cerbero build tool and packaging changes in 1.18.4
+
+-   macOS: more BigSur fixes
+-   glib: Backport patch to set thread names on Windows 10
+
+Contributors to 1.18.4
+
+Alicia Boya García, Ashley Brighthope, Bing Song, Branko Subasic, Edward
+Hervey, Guillaume Desmottes, Haihua Hu, He Junyan, Hou Qi, Jan Alexander
+Steffens (heftig), Jeongki Kim, Jordan Petridis, Knobe, Kristofer
+Björkström, Marijn Suijten, Matthew Waters, Paul Goulpié, Philipp Zabel,
+Rafał Dzięgiel, Sebastian Dröge, Seungha Yang, Staz M, Stéphane Cerveau,
+Thibault Saunier, Tim-Philipp Müller, Víctor Manuel Jáquez Leal, Vivia
+Nikolaidou, Vladimir Menshakov,
+
+… 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 1.18.4
+
+-   List of Merge Requests applied in 1.18.4
+-   List of Issues fixed in 1.18.4
+
 Schedule for 1.20
 
 Our next major feature release will be 1.20, and 1.19 will be the
@@ -2724,9 +2886,9 @@ unstable development version leading up to the stable 1.20 release. The
 development of 1.19/1.20 will happen in the git master branch.
 
 The plan for the 1.20 development cycle is yet to be confirmed, but it
-is now expected that feature freeze will take place some time in
-January/February 2021, with the first 1.20 stable release hopefully
-around February/March 2021.
+is now expected that feature freeze will take place some time in April
+2021, with the first 1.20 stable release hopefully around April/May
+2021.
 
 1.20 will be backwards-compatible to the stable 1.18, 1.16, 1.14, 1.12,
 1.10, 1.8, 1.6, 1.4, 1.2 and 1.0 release series.
diff --git a/RELEASE b/RELEASE
index a17c568..467b5ef 100644
--- a/RELEASE
+++ b/RELEASE
@@ -1,4 +1,4 @@
-This is GStreamer gst-rtsp-server 1.18.3.
+This is GStreamer gst-rtsp-server 1.18.4.
 
 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 91bfc90..c29f8d2 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/1.18.3.1",
+                        "default": "GStreamer/1.18.4",
                         "mutable": "null",
                         "readable": true,
                         "type": "gchararray",
diff --git a/gst-rtsp-server.doap b/gst-rtsp-server.doap
index cbb6437..a906095 100644
--- a/gst-rtsp-server.doap
+++ b/gst-rtsp-server.doap
@@ -32,6 +32,16 @@ RTSP server library based on GStreamer
 
  <release>
   <Version>
+   <revision>1.18.4</revision>
+   <branch>1.18</branch>
+   <name></name>
+   <created>2021-03-15</created>
+   <file-release rdf:resource="https://gstreamer.freedesktop.org/src/gst-rtsp-server/gst-rtsp-server-1.18.4.tar.xz" />
+  </Version>
+ </release>
+
+ <release>
+  <Version>
    <revision>1.18.3</revision>
    <branch>1.18</branch>
    <name></name>
diff --git a/meson.build b/meson.build
index 747f690..91891e5 100644
--- a/meson.build
+++ b/meson.build
@@ -1,5 +1,5 @@
 project('gst-rtsp-server', 'c',
-  version : '1.18.3.1',
+  version : '1.18.4',
   meson_version : '>= 0.48',
   default_options : ['warning_level=1', 'buildtype=debugoptimized'])