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Hello community,
here is the log from the commit of package gstreamer-rtsp-server for
openSUSE:Factory checked in at 2022-05-14 22:54:50
++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
Comparing /work/SRC/openSUSE:Factory/gstreamer-rtsp-server (Old)
and /work/SRC/openSUSE:Factory/.gstreamer-rtsp-server.new.1538 (New)
++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
Package is "gstreamer-rtsp-server"
Sat May 14 22:54:50 2022 rev:35 rq:976630 version:1.20.2
Changes:
--------
---
/work/SRC/openSUSE:Factory/gstreamer-rtsp-server/gstreamer-rtsp-server.changes
2022-04-06 21:52:14.959048228 +0200
+++
/work/SRC/openSUSE:Factory/.gstreamer-rtsp-server.new.1538/gstreamer-rtsp-server.changes
2022-05-14 22:54:57.555200654 +0200
@@ -1,0 +2,8 @@
+Mon May 9 11:06:24 UTC 2022 - Antonio Larrosa <[email protected]>
+
+- Update to version 1.20.2:
+ + rtspclientsink: fix possible shutdown deadlock in
+ collect_streams()
+ + Minor spelling fixes
+
+-------------------------------------------------------------------
Old:
----
gst-rtsp-server-1.20.1.tar.xz
New:
----
gst-rtsp-server-1.20.2.tar.xz
++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
Other differences:
------------------
++++++ gstreamer-rtsp-server.spec ++++++
--- /var/tmp/diff_new_pack.TSv676/_old 2022-05-14 22:54:57.931201124 +0200
+++ /var/tmp/diff_new_pack.TSv676/_new 2022-05-14 22:54:57.935201129 +0200
@@ -20,7 +20,7 @@
%define _name gst-rtsp-server
Name: gstreamer-rtsp-server
-Version: 1.20.1
+Version: 1.20.2
Release: 0
Summary: GStreamer-based RTSP server library
License: LGPL-2.0-or-later
++++++ gst-rtsp-server-1.20.1.tar.xz -> gst-rtsp-server-1.20.2.tar.xz ++++++
diff -urN '--exclude=CVS' '--exclude=.cvsignore' '--exclude=.svn'
'--exclude=.svnignore' old/gst-rtsp-server-1.20.1/ChangeLog
new/gst-rtsp-server-1.20.2/ChangeLog
--- old/gst-rtsp-server-1.20.1/ChangeLog 2022-03-14 12:33:40.000000000
+0100
+++ new/gst-rtsp-server-1.20.2/ChangeLog 2022-05-03 00:29:29.000000000
+0200
@@ -1,7 +1,52 @@
+=== release 1.20.2 ===
+
+2022-05-02 23:29:25 +0100 Tim-Philipp M??ller <[email protected]>
+
+ * NEWS:
+ * RELEASE:
+ * docs/gst_plugins_cache.json:
+ * gst-rtsp-server.doap:
+ * meson.build:
+ Release 1.20.2
+
+2022-05-02 23:29:19 +0100 Tim-Philipp M??ller <[email protected]>
+
+ * ChangeLog:
+ Update ChangeLogs for 1.20.2
+
+2022-02-15 13:39:43 +0000 Pierre Bourr?? <[email protected]>
+
+ * gst/rtsp-sink/gstrtspclientsink.c:
+ rtspclientsink: fix possible shutdown deadlock collect_streams()
+ Part-of:
<https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2278>
+
+2022-04-13 14:34:57 +0200 Marc Leeman <[email protected]>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ gst-rtsp-server: minor spelling fixes
+ Part-of:
<https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2172>
+
+2022-03-28 21:03:16 +1100 Matthew Waters <[email protected]>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: remove unused variable:
+ Fixes:
+ ../gst/rtsp-server/rtsp-stream.c:2670:9: error: variable 'n_messages'
set but not used [-Werror,-Wunused-but-set-variable]
+ guint n_messages = 0;
+ ^
+ Part-of:
<https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2068>
+
+2022-03-14 14:48:01 +0000 Tim-Philipp M??ller <[email protected]>
+
+ * docs/gst_plugins_cache.json:
+ * meson.build:
+ Back to development
+
=== release 1.20.1 ===
2022-03-14 11:33:33 +0000 Tim-Philipp M??ller <[email protected]>
+ * ChangeLog:
* NEWS:
* RELEASE:
* docs/gst_plugins_cache.json:
diff -urN '--exclude=CVS' '--exclude=.cvsignore' '--exclude=.svn'
'--exclude=.svnignore' old/gst-rtsp-server-1.20.1/NEWS
new/gst-rtsp-server-1.20.2/NEWS
--- old/gst-rtsp-server-1.20.1/NEWS 2022-03-14 12:33:40.000000000 +0100
+++ new/gst-rtsp-server-1.20.2/NEWS 2022-05-03 00:29:29.000000000 +0200
@@ -2,13 +2,13 @@
GStreamer 1.20.0 was originally released on 3 February 2022.
-The latest bug-fix release in the 1.20 series is 1.20.1 and was released
-on 14 March 2022.
+The latest bug-fix release in the 1.20 series is 1.20.2 and was released
+on 2 May 2022.
See https://gstreamer.freedesktop.org/releases/1.20/ for the latest
version of this document.
-Last updated: Monday 14 March 2022, 00:30 UTC (log)
+Last updated: Monday 2 May 2022, 0:30 UTC (log)
Introduction
@@ -2076,6 +2076,221 @@
- List of Merge Requests applied in 1.20.1
- List of Issues fixed in 1.20.1
+1.20.2
+
+The second 1.20 bug-fix release (1.20.2) was released on 2 May 2022.
+
+This release only contains bugfixes and it should be safe to update from
+1.20.x.
+
+Highlighted bugfixes in 1.20.2
+
+- avviddec: Remove vc1/wmv3 override and fix crashes on WMV files with
+ FFMPEG 5.0+
+- macOS: fix plugin discovery for GStreamer installed via brew and fix
+ loading of Rust plugins
+- rtpbasepayload: various header extension handling fixes
+- rtpopusdepay: fix regression in stereo input handling if
+ sprop-stereo is not advertised
+- rtspclientsink: fix possible shutdown deadlock
+- mpegts: gracefully handle ???empty??? program maps and fix AC-4
+ detection
+- mxfdemux: Handle empty VANC packets and fix EOS handling
+- playbin3: various playbin3, uridecodebin3, and playsink fixes
+- ptpclock: fix initial sync-up with certain devices
+- gltransformation: let graphene alloc its structures memory aligned
+- webrtcbin fixes and webrtc sendrecv example improvements
+- video4linux2: various fixes including some fixes for Raspberry Pi
+ users
+- videorate segment handling fixes and other fixes
+- nvh264dec, nvh265dec: Fix broken key-unit trick modes and reverse
+ playback
+- wpe: Reintroduce persistent WebContext
+- cerbero: Make it easier to consume 1.20.1 macOS GStreamer .pkgs
+- build fixes and gobject annotation fixes
+- bug fixes, security fixes, memory leak fixes, and other stability
+ and reliability improvements
+
+gstreamer
+
+- devicemonitor: clean up signal handlers and hidden providers list
+- Leaks tracer: fix pthread_atfork return value check leading to bogus
+ warning in log
+- Rust plugins: Not picked up by the plugin loader on macOS
+- Failed to use plugins of latest GStreamer version 1.20.x installed
+ by brew on macOS
+- ptpclock: Allow at least 100ms delay between Sync/Follow_Up and
+ Delay_Req/Delay_Resp messages. Fixes problems acquiring initial sync
+ with certain devices
+- meson: Add -Wl,-rpath,${libdir} on macOS
+- registry: skip Rust dep builddirs when searching for plugins
+ recursively
+
+gst-plugins-base
+
+- appsrc: Clarify buffer ref semantics in signals documentation
+- appsrc: fix annotations for bindings
+- typefind: Skip extension parsing for data:// URIs, fixing regression
+ with mp4 files serialised to data uris
+- playbin3: various fixes
+- playbin3: fix missing lock when unknown stream type in pad-removed
+ cb
+- decodebin3: fix collection leaks
+- decodebin3: Don???t duplicate stream selections
+- discoverer: chain up to parent finalize methods in all our types to
+ fix memory leaks
+- glmixerbin: slightly better pad/element creation
+- gltransformation: let graphene alloc its structures memory aligned
+- ogg: fix possible buffer overrun
+- rtpbasepayload: Don???t write header extensions if there???s no
+ corresponding???
+- rtpbasepayload: always store input buffer meta before negotiation
+- rtpbasepayload: fix transfer annotation for push and push_list
+- subparse: don???t try to index string with -1
+- riff-media: fix memory leak after usage for g_strjoin()
+- playbin/playbin3: Allow setting a NULL URI
+- playsink: Complete reconfiguration on pad release.
+- parsebin: Expose streams of unknown type
+- pbutils: Fix wmv screen description detection
+- subparse: don???t deref a potentially NULL variable
+- rawvideoparse: set format from caps in
+ gst_raw_video_parse_set_config_from_caps
+- videodecoder: release stream lock after handling gap events
+- videorate: fix assertion when pushing last and only buffer without
+ duration
+- videorate: Revert ???don???t reset on segment update??? to fix segment
+ handling regressions
+- gst-play-1.0, gst-launch-1.0: Enable win32 high-resolution timer
+ also for MinGW build
+
+gst-plugins-good
+
+- deinterlace: silence unused-but-set werror from imported code
+- qtdemux: fix leak of channel_mapping
+- rtpopusdepay: missing sprop-stereo should not assume mono
+- rtpjitterbuffer: Fix invalid memory access in
+ rtp_jitter_buffer_pop()
+- rtpptdemux: fix leak of caps when ignoring a pt
+- rtpredenc: quieten warning about ignoring header extensions
+- soup: Fix pre-processor macros in souploader for libsoup-3.0
+- twcc: Note that twcc-stats packet loss counts reordering as loss +
+ add some logging
+- video4linux2: Manual backports for RPi users
+- wavparse: handle URI query in any parse state, fixing audio track
+ selection issue in GES
+- wavparse: Unset DISCONT buffer flag for divided into multiple
+ buffers in push mode
+
+gst-plugins-bad
+
+- av1parse: Fix several issues about the colorimetry.
+- av1parse: fix up various possible logic errors
+- dashsink: fix missing mutex unlock in error code path when failing
+ to get content
+- d3d11videosink: Fix for unhandled mouse double click events
+- interlace: Also handle a missing ???interlace-mode??? field as
+ progressive
+- msdk: fix build with MSVC
+- mxfdemux: Fix issues at EOS
+- mxfdemux: Handle empty VANC packets
+- nvh264dec, nvh265dec: Fix broken key-unit trick and reverse playback
+- nvvp9sldec: Increase DPB size to cover render delay
+- rvsg: fix cairo include
+- tsdemux: Fix AC-4 detection in MPEG-TS
+- tsdemux: Handle ???empty??? PMT gracefully
+- va: pool: don???t advertise the GST_BUFFER_POOL_OPTION_VIDEO_ALIGNMENT
+ option any more
+- v4l2codecs: Fix memory leak
+- v4l2videodec: set frame duration according to framerate
+- webrtcbin: Update documentation of ???get-stats??? action signal
+- webrtcbin: Check data channel transport for notifying
+ ???ice-gathering-state???
+- webrtcbin: Avoid access of freed memory
+- wpe: Reintroduce persistent WebContext
+- Build: use CMake to find some openssl and exr deps
+- Fix multiple ???unused-but-set variable??? compiler warnings
+
+gst-plugins-ugly
+
+- x264enc: Don???t try to fixate ANY allowed caps
+
+gst-libav
+
+- video decoders: fix frame leak on negotiation error
+- Fix build on systems without C++ compiler
+- avviddec: Remove vc1/wmv3 override (fixing crash with FFmpeg 5
+- Segfaults on ASF/WMV files with FFMPEG 5.0+
+
+gst-rtsp-server
+
+- rtspclientsink: fix possible shutdown deadlock in collect_streams()
+- Minor spelling fixes
+
+gstreamer-vaapi
+
+- No changes
+
+gstreamer-sharp
+
+- No changes
+
+gst-omx
+
+- No changes
+
+gst-python
+
+- Fix build on systems without C++ compiler
+
+gst-editing-services
+
+- License clarification: GES is released under the LGPL2+ license
+
+gst-examples:
+
+- Fix build on macOS with gtk+-quartz-3.0
+- player android: add missing dummy.cpp
+- player android: update for android changes
+- webrtc_sendrecv.py: Link pads instead of elements
+- webrtc_sendrecv.py: Implement all negotiation modes + bugfixes
+
+Development build environment + gst-full build
+
+- meson: provide gobject-cast-checks, glib-checks and glib-asserts
+ options at top level as well
+
+Cerbero build tool and packaging changes in 1.20.2
+
+- macOS: Make it easier to consume 1.20.1 GStreamer .pkgs
+- Android: fix text relocation regression on Android (x86/ x86_64
+ platforms)
+
+Bindings
+
+- appsrc: fix annotations for bindings
+- bindings: The out args for gst_rtp_buffer_get_extension_data*() are
+ optional
+- rtpbasepayload: fix transfer annotation for push and push_list
+
+Contributors to 1.20.2
+
+Bastian Krause, Benjamin Gaignard, Camilo Celis Guzman, Chun-wei Fan,
+Corentin Damman, Daniel Stone, Dongil Park, Edward Hervey, Fabrice
+Fontaine, Guillaume Desmottes, Havard Graff, He Junyan, Hoonhee Lee, Hou
+Qi, Jan Schmidt, Marc Leeman, Mathieu Duponchelle, Matthew Waters,
+Nicolas Dufresne, Nirbheek Chauhan, Philippe Normand, Pierre Bourr??,
+Sangchul Lee, Sebastian Dr??ge, Seungha Yang, St??phane Cerveau, Thibault
+Saunier, Tim-Philipp M??ller, Tong Wu, Tristan Matthews, Tulio Beloqui,
+Wonchul Lee, Zhao Zhili,
+
+??? 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 1.20.2
+
+- List of Merge Requests applied in 1.20.2
+- List of Issues fixed in 1.20.2
+
Schedule for 1.22
Our next major feature release will be 1.22, and 1.21 will be the
diff -urN '--exclude=CVS' '--exclude=.cvsignore' '--exclude=.svn'
'--exclude=.svnignore' old/gst-rtsp-server-1.20.1/RELEASE
new/gst-rtsp-server-1.20.2/RELEASE
--- old/gst-rtsp-server-1.20.1/RELEASE 2022-03-14 12:33:40.000000000 +0100
+++ new/gst-rtsp-server-1.20.2/RELEASE 2022-05-03 00:29:29.000000000 +0200
@@ -1,4 +1,4 @@
-This is GStreamer gst-rtsp-server 1.20.1.
+This is GStreamer gst-rtsp-server 1.20.2.
The GStreamer team is thrilled to announce a new major feature release
of your favourite cross-platform multimedia framework!
diff -urN '--exclude=CVS' '--exclude=.cvsignore' '--exclude=.svn'
'--exclude=.svnignore' old/gst-rtsp-server-1.20.1/docs/gst_plugins_cache.json
new/gst-rtsp-server-1.20.2/docs/gst_plugins_cache.json
--- old/gst-rtsp-server-1.20.1/docs/gst_plugins_cache.json 2022-03-14
12:33:40.000000000 +0100
+++ new/gst-rtsp-server-1.20.2/docs/gst_plugins_cache.json 2022-05-03
00:29:29.000000000 +0200
@@ -321,7 +321,7 @@
"construct": false,
"construct-only": false,
"controllable": false,
- "default": "GStreamer/1.20.1",
+ "default": "GStreamer/1.20.2",
"mutable": "null",
"readable": true,
"type": "gchararray",
diff -urN '--exclude=CVS' '--exclude=.cvsignore' '--exclude=.svn'
'--exclude=.svnignore' old/gst-rtsp-server-1.20.1/gst/rtsp-server/rtsp-stream.c
new/gst-rtsp-server-1.20.2/gst/rtsp-server/rtsp-stream.c
--- old/gst-rtsp-server-1.20.1/gst/rtsp-server/rtsp-stream.c 2022-03-14
12:33:40.000000000 +0100
+++ new/gst-rtsp-server-1.20.2/gst/rtsp-server/rtsp-stream.c 2022-05-03
00:29:29.000000000 +0200
@@ -2667,7 +2667,6 @@
GstSample *sample;
GstBuffer *buffer;
GstBufferList *buffer_list;
- guint n_messages = 0;
gboolean is_rtp;
GPtrArray *transports;
@@ -2699,10 +2698,6 @@
/* We will get one message-sent notification per buffer or
* complete buffer-list. We handle each buffer-list as a unit */
- if (buffer)
- n_messages += 1;
- if (buffer_list)
- n_messages += 1;
transports = priv->tr_cache;
if (transports)
@@ -5589,7 +5584,7 @@
pad = gst_object_ref (priv->send_src[0]);
} else {
g_mutex_unlock (&priv->lock);
- GST_WARNING_OBJECT (stream, "Couldn't obtain postion: erroneous pipeline");
+ GST_WARNING_OBJECT (stream, "Couldn't obtain position: erroneous
pipeline");
return FALSE;
}
g_mutex_unlock (&priv->lock);
@@ -5597,7 +5592,7 @@
if (sink) {
if (!gst_element_query_position (sink, GST_FORMAT_TIME, position)) {
GST_WARNING_OBJECT (stream,
- "Couldn't obtain postion: position query failed");
+ "Couldn't obtain position: position query failed");
gst_object_unref (sink);
return FALSE;
}
@@ -5608,7 +5603,7 @@
event = gst_pad_get_sticky_event (pad, GST_EVENT_SEGMENT, 0);
if (!event) {
- GST_WARNING_OBJECT (stream, "Couldn't obtain postion: no segment event");
+ GST_WARNING_OBJECT (stream, "Couldn't obtain position: no segment
event");
gst_object_unref (pad);
return FALSE;
}
@@ -5784,7 +5779,7 @@
* Add a receiver and sender part to the pipeline based on the transport from
* SETUP.
*
- * Returns: %TRUE if the stream has been sucessfully updated.
+ * Returns: %TRUE if the stream has been successfully updated.
*
* Since: 1.14
*/
@@ -5816,7 +5811,7 @@
priv->is_complete = TRUE;
g_mutex_unlock (&priv->lock);
- GST_DEBUG_OBJECT (stream, "pipeline sucsessfully updated");
+ GST_DEBUG_OBJECT (stream, "pipeline successfully updated");
return TRUE;
create_receiver_error:
diff -urN '--exclude=CVS' '--exclude=.cvsignore' '--exclude=.svn'
'--exclude=.svnignore'
old/gst-rtsp-server-1.20.1/gst/rtsp-sink/gstrtspclientsink.c
new/gst-rtsp-server-1.20.2/gst/rtsp-sink/gstrtspclientsink.c
--- old/gst-rtsp-server-1.20.1/gst/rtsp-sink/gstrtspclientsink.c
2022-03-14 12:33:40.000000000 +0100
+++ new/gst-rtsp-server-1.20.2/gst/rtsp-sink/gstrtspclientsink.c
2022-05-03 00:29:29.000000000 +0200
@@ -3602,6 +3602,18 @@
}
static gboolean
+gst_rtsp_client_sink_is_stopping (GstRTSPClientSink * sink)
+{
+ gboolean is_stopping;
+
+ GST_OBJECT_LOCK (sink);
+ is_stopping = sink->task == NULL;
+ GST_OBJECT_UNLOCK (sink);
+
+ return is_stopping;
+}
+
+static gboolean
gst_rtsp_client_sink_collect_streams (GstRTSPClientSink * sink)
{
GstRTSPStreamContext *context;
@@ -3640,7 +3652,8 @@
continue;
g_mutex_lock (&sink->preroll_lock);
- while (!context->prerolled && !sink->conninfo.flushing) {
+ while (!context->prerolled && !sink->conninfo.flushing
+ && !gst_rtsp_client_sink_is_stopping (sink)) {
GST_DEBUG_OBJECT (sink, "Waiting for caps on stream %d", context->index);
g_cond_wait (&sink->preroll_cond, &sink->preroll_lock);
}
@@ -4374,18 +4387,6 @@
return res;
}
-static gboolean
-gst_rtsp_client_sink_is_stopping (GstRTSPClientSink * sink)
-{
- gboolean is_stopping;
-
- GST_OBJECT_LOCK (sink);
- is_stopping = sink->task == NULL;
- GST_OBJECT_UNLOCK (sink);
-
- return is_stopping;
-}
-
static GstRTSPResult
gst_rtsp_client_sink_record (GstRTSPClientSink * sink, gboolean async)
{
@@ -4989,6 +4990,10 @@
g_cond_broadcast (&sink->block_streams_cond);
g_mutex_unlock (&sink->block_streams_lock);
+ g_mutex_lock (&sink->preroll_lock);
+ g_cond_broadcast (&sink->preroll_cond);
+ g_mutex_unlock (&sink->preroll_lock);
+
/* make sure it is not running */
GST_RTSP_STREAM_LOCK (sink);
GST_RTSP_STREAM_UNLOCK (sink);
diff -urN '--exclude=CVS' '--exclude=.cvsignore' '--exclude=.svn'
'--exclude=.svnignore' old/gst-rtsp-server-1.20.1/gst-rtsp-server.doap
new/gst-rtsp-server-1.20.2/gst-rtsp-server.doap
--- old/gst-rtsp-server-1.20.1/gst-rtsp-server.doap 2022-03-14
12:33:40.000000000 +0100
+++ new/gst-rtsp-server-1.20.2/gst-rtsp-server.doap 2022-05-03
00:29:29.000000000 +0200
@@ -32,6 +32,16 @@
<release>
<Version>
+ <revision>1.20.2</revision>
+ <branch>1.20</branch>
+ <name></name>
+ <created>2022-05-02</created>
+ <file-release
rdf:resource="https://gstreamer.freedesktop.org/src/gst-rtsp-server/gst-rtsp-server-1.20.2.tar.xz";
/>
+ </Version>
+ </release>
+
+ <release>
+ <Version>
<revision>1.20.1</revision>
<branch>1.20</branch>
<name></name>
diff -urN '--exclude=CVS' '--exclude=.cvsignore' '--exclude=.svn'
'--exclude=.svnignore' old/gst-rtsp-server-1.20.1/meson.build
new/gst-rtsp-server-1.20.2/meson.build
--- old/gst-rtsp-server-1.20.1/meson.build 2022-03-14 12:33:40.000000000
+0100
+++ new/gst-rtsp-server-1.20.2/meson.build 2022-05-03 00:29:29.000000000
+0200
@@ -1,5 +1,5 @@
project('gst-rtsp-server', 'c',
- version : '1.20.1',
+ version : '1.20.2',
meson_version : '>= 0.59',
default_options : ['warning_level=1', 'buildtype=debugoptimized'])
