From 22d32acd45c74127a23ac52ecf80e3f2067e4c4b Mon Sep 17 00:00:00 2001 From: Tim-Philipp Müller Date: Mon, 15 Mar 2021 17:49:53 +0000 Subject: Release 1.18.4 --- ChangeLog | 85 ++++++++++++++++++++++ NEWS | 174 ++++++++++++++++++++++++++++++++++++++++++-- RELEASE | 2 +- docs/gst_plugins_cache.json | 2 +- gst-rtsp-server.doap | 10 +++ meson.build | 2 +- 6 files changed, 266 insertions(+), 9 deletions(-) diff --git a/ChangeLog b/ChangeLog index b0630df298..10a3a7c3ad 100644 --- a/ChangeLog +++ b/ChangeLog @@ -1,3 +1,87 @@ +=== release 1.18.4 === + +2021-03-15 17:49:53 +0000 Tim-Philipp Müller + + * ChangeLog: + * NEWS: + * RELEASE: + * gst-rtsp-server.doap: + * meson.build: + Release 1.18.4 + +2021-02-15 12:07:15 +0000 Tim-Philipp Müller + + * tests/check/gst/rtspclientsink.c: + tests: rtspclientsink: fix some leaks + Part-of: + +2021-02-15 12:26:30 +0000 Tim-Philipp Müller + + * gst/rtsp-sink/gstrtspclientsink.c: + rtspclientsink: mark cached caps as maybe-leaked to make leaks tracer happy + Part-of: + +2021-02-15 12:07:45 +0000 Tim-Philipp Müller + + * tests/check/gst/rtspclientsink.c: + rtspclientsink: add unit test for potential shutdown deadlock + Part-of: + +2021-02-15 12:01:34 +0000 Tim-Philipp Müller + + * 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: + +2021-02-01 12:16:46 +0100 Branko Subasic + + * 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: + +2021-01-22 08:58:23 +0100 Branko Subasic + + * 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: + +2021-01-14 02:17:42 +0000 Tim-Philipp Müller + + * docs/gst_plugins_cache.json: + * meson.build: + Back to development + === release 1.18.3 === 2021-01-13 21:12:06 +0000 Tim-Philipp Müller @@ -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 1ac82bdf4a..3ada8869cd 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 a17c568eb1..467b5efbce 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 91bfc903fd..c29f8d2bf5 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 cbb6437190..a906095d88 100644 --- a/gst-rtsp-server.doap +++ b/gst-rtsp-server.doap @@ -30,6 +30,16 @@ RTSP server library based on GStreamer + + + 1.18.4 + 1.18 + + 2021-03-15 + + + + 1.18.3 diff --git a/meson.build b/meson.build index 747f690acc..91891e56c5 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']) -- cgit v1.2.3