Script 'mail_helper' called by obssrc
Hello community,
here is the log from the commit of package gstreamer-rtsp-server for
openSUSE:Factory checked in at 2021-01-21 21:54:22
++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
Comparing /work/SRC/openSUSE:Factory/gstreamer-rtsp-server (Old)
and /work/SRC/openSUSE:Factory/.gstreamer-rtsp-server.new.28504 (New)
++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
Package is "gstreamer-rtsp-server"
Thu Jan 21 21:54:22 2021 rev:28 rq:864318 version:1.18.3
Changes:
--------
---
/work/SRC/openSUSE:Factory/gstreamer-rtsp-server/gstreamer-rtsp-server.changes
2020-12-13 17:30:04.384369829 +0100
+++
/work/SRC/openSUSE:Factory/.gstreamer-rtsp-server.new.28504/gstreamer-rtsp-server.changes
2021-01-21 21:54:23.237770982 +0100
@@ -1,0 +2,8 @@
+Sat Jan 16 20:00:07 UTC 2021 - Bj??rn Lie <[email protected]>
+
+- Update to version 1.18.3:
+ + rtsp-media: Only count senders when counting blocked streams
+ + rtsp-client: Only unref client watch context on finalize, to
+ avoid deadlock
+
+-------------------------------------------------------------------
Old:
----
gst-rtsp-server-1.18.2.tar.xz
New:
----
gst-rtsp-server-1.18.3.tar.xz
++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
Other differences:
------------------
++++++ gstreamer-rtsp-server.spec ++++++
--- /var/tmp/diff_new_pack.zMHVIL/_old 2021-01-21 21:54:23.885771227 +0100
+++ /var/tmp/diff_new_pack.zMHVIL/_new 2021-01-21 21:54:23.885771227 +0100
@@ -1,7 +1,7 @@
#
# spec file for package gstreamer-rtsp-server
#
-# Copyright (c) 2020 SUSE LLC
+# Copyright (c) 2021 SUSE LLC
#
# All modifications and additions to the file contributed by third parties
# remain the property of their copyright owners, unless otherwise agreed
@@ -20,7 +20,7 @@
%define _name gst-rtsp-server
Name: gstreamer-rtsp-server
-Version: 1.18.2
+Version: 1.18.3
Release: 0
Summary: GStreamer-based RTSP server library
License: LGPL-2.0-or-later
++++++ _service ++++++
--- /var/tmp/diff_new_pack.zMHVIL/_old 2021-01-21 21:54:23.913771238 +0100
+++ /var/tmp/diff_new_pack.zMHVIL/_new 2021-01-21 21:54:23.913771238 +0100
@@ -9,7 +9,7 @@
<!--
<param name="changesgenerate">enable</param>
-->
- <param name="revision">1.18.2</param>
+ <param name="revision">1.18.3</param>
<param name="scm">git</param>
</service>
<service name="recompress" mode="disabled">
++++++ gst-rtsp-server-1.18.2.tar.xz -> gst-rtsp-server-1.18.3.tar.xz ++++++
diff -urN '--exclude=CVS' '--exclude=.cvsignore' '--exclude=.svn'
'--exclude=.svnignore' old/gst-rtsp-server-1.18.2/ChangeLog
new/gst-rtsp-server-1.18.3/ChangeLog
--- old/gst-rtsp-server-1.18.2/ChangeLog 2020-12-06 14:24:47.000000000
+0100
+++ new/gst-rtsp-server-1.18.3/ChangeLog 2021-01-13 22:12:06.000000000
+0100
@@ -1,3 +1,35 @@
+=== release 1.18.3 ===
+
+2021-01-13 21:12:06 +0000 Tim-Philipp M??ller <[email protected]>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * gst-rtsp-server.doap:
+ * meson.build:
+ Release 1.18.3
+
+2020-12-17 15:27:27 +0100 Tobias Ronge <[email protected]>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: Only count senders when counting blocked streams
+ Only sender streams sends the GstRTSPStreamBlocking message, so only
+ these should be counted before setting media status to prepared.
+ Part-of:
<https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/182>
+
+2020-12-14 14:12:38 +1300 Lawrence Troup <[email protected]>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: Only unref client watch context on finalize, to avoid
deadlock
+ Fixes
https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/issues/127
+ Part-of:
<https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/179>
+
+2020-12-06 23:57:09 +0000 Tim-Philipp M??ller <[email protected]>
+
+ * docs/gst_plugins_cache.json:
+ * meson.build:
+ Back to development
+
=== release 1.18.2 ===
2020-12-06 13:24:47 +0000 Tim-Philipp M??ller <[email protected]>
@@ -5,6 +37,7 @@
* ChangeLog:
* NEWS:
* RELEASE:
+ * docs/gst_plugins_cache.json:
* gst-rtsp-server.doap:
* meson.build:
Release 1.18.2
diff -urN '--exclude=CVS' '--exclude=.cvsignore' '--exclude=.svn'
'--exclude=.svnignore' old/gst-rtsp-server-1.18.2/NEWS
new/gst-rtsp-server-1.18.3/NEWS
--- old/gst-rtsp-server-1.18.2/NEWS 2020-12-06 14:24:47.000000000 +0100
+++ new/gst-rtsp-server-1.18.3/NEWS 2021-01-13 22:12:06.000000000 +0100
@@ -2,13 +2,13 @@
GStreamer 1.18.0 was originally released on 8 September 2020.
-The latest bug-fix release in the 1.18 series is 1.18.2 and was released
-on 6 December 2020.
+The latest bug-fix release in the 1.18 series is 1.18.3 and was released
+on 13 January 2021.
See https://gstreamer.freedesktop.org/releases/1.18/ for the latest
version of this document.
-Last updated: Monday 26 October 2020, 11:00 UTC (log)
+Last updated: Wednesday 13 January 2021, 20:00 UTC (log)
Introduction
@@ -2576,6 +2576,147 @@
- List of Merge Requests applied in 1.18.2
- List of Issues fixed in 1.18.2
+1.18.3
+
+The third 1.18 bug-fix release (1.18.3) was released on 13 January 2021.
+
+This release only contains bugfixes and it should be safe to update from
+1.18.x.
+
+Highlighted bugfixes in 1.18.3
+
+- fix ogg playback regression for ogg files that also have ID3 or APE
+ tags
+- compositor: fix artefacts and invalid memory access when blending
+ subsampled formats
+- exported mini object ref/unref/copy functions for use in bindings
+ such as gstreamer-sharp
+- Add support for Apple silicon (M1) to cerbero package builder
+- Ship RIST plugin in binary packages
+- various stability, performance and reliability improvements
+- memory leak fixes
+- build fixes
+
+gstreamer
+
+- gst: Add non-inline ref/unref/copy/replace methods for various mini
+ objects (buffer, bufferlist, caps, context, event, memory, message,
+ promise, query, sample, taglist, uri) for use in bindings such as
+ gstreamer-sharp
+- harness: don???t use GST_DEBUG_OBJECT with GstHarness which is not a
+ GObject
+
+gst-plugins-base
+
+- audiorate: Make buffer writable before changing its metadata
+- compositor: fix blending of subsampled components
+- decodebin3: When reconfiguring a slot make sure that the ghostpad is
+ unlinked
+- decodebin3: Release selection lock when pushing EOS
+- encodebasebin: Ensure that parsers are compatible with selected
+ encoders
+- tagdemux: resize and trim buffer in place to fix interaction with
+ oggdemux
+- videoaggregator: Pop out old buffers on timeout
+- video-blend: fix blending 8-bit and 16-bit frames together
+- appsrc: fix signal documentation
+- gl: document some GL caps specifics
+- libvisual: workaround clang compiler warning
+
+gst-plugins-good
+
+- deinterlace: fix build of assembly optimisations on macOS
+- splitmuxsink: Avoid deadlock when releasing a pad from a running
+ muxer
+- splitmuxsink: fix bogus fragment split
+- v4l2object: Map correct video format for RGBA
+- videoflip: fix possible crash when changing video-direction/method
+ while running
+
+gst-plugins-bad
+
+- assrender: fix mutex handling in certain flushing/error situations
+- dvbsuboverlay: Add support for dynamic resolution update
+- dashsink: fix critical log of dynamic pipeline
+- d3d11shader: Fix ID3DBlob object leak
+- d3d11videosink: Prepare window once streaming started
+- decklinkaudiosrc: Fix duration of the first audio frame after each
+ discont
+- intervideosrc: fix negotiation of interlaced caps
+- msdk: needn???t close mfx session when failed, fixes double free /
+ potential crash
+- msdk: check GstMsdkContext instead of mfxSession instance
+- srt: fix locking when retrieving stats
+- rtmp2src: fix leaks when connection is cancelled during startup or
+ connection fails
+
+gst-plugins-ugly
+
+- no changes
+
+gst-libav
+
+- avauddec: Drain decoder on decoding failure, fixes timestamps after
+ decoding errors
+
+gst-rtsp-server
+
+- rtsp-media: Only count senders when counting blocked streams
+- rtsp-client: Only unref client watch context on finalize, to avoid
+ deadlock
+
+gstreamer-vaapi
+
+- no changes
+
+gstreamer-sharp
+
+- no changes
+
+gst-omx
+
+- no changes
+
+gst-python
+
+- no changes
+
+gst-editing-services
+
+- launch: Ensure to add required ref to profiles from project
+- tests: fix meson test env setup to make sure we use the right
+ gst-plugin-scanner
+
+gst-integration-testsuites
+
+- no changes
+
+gst-build
+
+- meson: Update zlib.wrap to use wrapdb instead of github fork
+
+Cerbero build tool and packaging changes in 1.18.3
+
+- Add support for Apple silicon
+- Build and ship RIST plugin
+
+Contributors to 1.18.3
+
+Andoni Morales Alastruey, Edward Hervey, Haihao Xiang, Haihua Hu, Hou
+Qi, Ignacio Casal Quinteiro, Jakub Adam, Jan Alexander Steffens
+(heftig), Jan Schmidt, Jordan Petridis, Lawrence Troup, Lim Siew Hoon,
+Mathieu Duponchelle, Matthew Waters, Nicolas Dufresne, Raju Babannavar,
+Sebastian Dr??ge, Seungha Yang, Thibault Saunier, Tim-Philipp M??ller,
+Tobias Ronge, Vivia Nikolaidou,
+
+??? 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 1.18.3
+
+- List of Merge Requests applied in 1.18.3
+- List of Issues fixed in 1.18.3
+
Schedule for 1.20
Our next major feature release will be 1.20, and 1.19 will be the
@@ -2583,8 +2724,9 @@
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
-2021, with the first 1.20 stable release around February/March 2021.
+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.
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 -urN '--exclude=CVS' '--exclude=.cvsignore' '--exclude=.svn'
'--exclude=.svnignore' old/gst-rtsp-server-1.18.2/RELEASE
new/gst-rtsp-server-1.18.3/RELEASE
--- old/gst-rtsp-server-1.18.2/RELEASE 2020-12-06 14:24:47.000000000 +0100
+++ new/gst-rtsp-server-1.18.3/RELEASE 2021-01-13 22:12:06.000000000 +0100
@@ -1,4 +1,4 @@
-This is GStreamer gst-rtsp-server 1.18.2.
+This is GStreamer gst-rtsp-server 1.18.3.
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-1.18.2/docs/gst_plugins_cache.json
new/gst-rtsp-server-1.18.3/docs/gst_plugins_cache.json
--- old/gst-rtsp-server-1.18.2/docs/gst_plugins_cache.json 2020-12-06
14:24:47.000000000 +0100
+++ new/gst-rtsp-server-1.18.3/docs/gst_plugins_cache.json 2021-01-13
22:12:06.000000000 +0100
@@ -321,7 +321,7 @@
"construct": false,
"construct-only": false,
"controllable": false,
- "default": "GStreamer/1.18.2",
+ "default": "GStreamer/1.18.3",
"mutable": "null",
"readable": true,
"type": "gchararray",
diff -urN '--exclude=CVS' '--exclude=.cvsignore' '--exclude=.svn'
'--exclude=.svnignore' old/gst-rtsp-server-1.18.2/gst/rtsp-server/rtsp-client.c
new/gst-rtsp-server-1.18.3/gst/rtsp-server/rtsp-client.c
--- old/gst-rtsp-server-1.18.2/gst/rtsp-server/rtsp-client.c 2020-12-06
14:24:47.000000000 +0100
+++ new/gst-rtsp-server-1.18.3/gst/rtsp-server/rtsp-client.c 2021-01-13
22:12:06.000000000 +0100
@@ -765,9 +765,13 @@
/* the watch and related state should be cleared before finalize
* as the watch actually holds a strong reference to the client */
g_assert (priv->watch == NULL);
- g_assert (priv->watch_context == NULL);
g_assert (priv->rtsp_ctrl_timeout == NULL);
+ if (priv->watch_context) {
+ g_main_context_unref (priv->watch_context);
+ priv->watch_context = NULL;
+ }
+
gst_rtsp_client_set_send_func (client, NULL, NULL, NULL);
gst_rtsp_client_set_send_messages_func (client, NULL, NULL, NULL);
@@ -1308,11 +1312,6 @@
rtsp_ctrl_timeout_remove (client);
}
- if (priv->watch_context) {
- g_main_context_unref (priv->watch_context);
- priv->watch_context = NULL;
- }
-
g_mutex_unlock (&priv->watch_lock);
}
@@ -5071,11 +5070,6 @@
rtsp_ctrl_timeout_remove (client);
}
- if (priv->watch_context) {
- g_main_context_unref (priv->watch_context);
- priv->watch_context = NULL;
- }
-
return GST_RTSP_STS_OK;
/* ERRORS */
@@ -5174,10 +5168,6 @@
gst_rtsp_client_set_send_messages_func (client, NULL, NULL, NULL);
rtsp_ctrl_timeout_remove (client);
gst_rtsp_client_session_filter (client, cleanup_session, &closed);
- if (priv->watch_context) {
- g_main_context_unref (priv->watch_context);
- priv->watch_context = NULL;
- }
if (closed)
g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_CLOSED], 0, NULL);
@@ -5210,6 +5200,7 @@
priv = client->priv;
g_return_val_if_fail (priv->connection != NULL, 0);
g_return_val_if_fail (priv->watch == NULL, 0);
+ g_return_val_if_fail (priv->watch_context == NULL, 0);
/* make sure noone will free the context before the watch is destroyed */
priv->watch_context = g_main_context_ref (context);
diff -urN '--exclude=CVS' '--exclude=.cvsignore' '--exclude=.svn'
'--exclude=.svnignore' old/gst-rtsp-server-1.18.2/gst/rtsp-server/rtsp-media.c
new/gst-rtsp-server-1.18.3/gst/rtsp-server/rtsp-media.c
--- old/gst-rtsp-server-1.18.2/gst/rtsp-server/rtsp-media.c 2020-12-06
14:24:47.000000000 +0100
+++ new/gst-rtsp-server-1.18.3/gst/rtsp-server/rtsp-media.c 2021-01-13
22:12:06.000000000 +0100
@@ -3146,19 +3146,20 @@
}
static void
-stream_collect_active (GstRTSPStream * stream, guint * active_streams)
+stream_collect_active_sender (GstRTSPStream * stream, guint * active_streams)
{
- if (gst_rtsp_stream_is_complete (stream))
+ if (gst_rtsp_stream_is_complete (stream)
+ && gst_rtsp_stream_is_sender (stream))
(*active_streams)++;
}
static guint
-nbr_active_streams (GstRTSPMedia * media)
+nbr_active_sender_streams (GstRTSPMedia * media)
{
guint ret = 0;
- g_ptr_array_foreach (media->priv->streams, (GFunc) stream_collect_active,
- &ret);
+ g_ptr_array_foreach (media->priv->streams,
+ (GFunc) stream_collect_active_sender, &ret);
return ret;
}
@@ -3271,7 +3272,7 @@
s = gst_message_get_structure (message);
if (gst_structure_has_name (s, "GstRTSPStreamBlocking")) {
gboolean is_complete = FALSE;
- guint n_active_streams;
+ guint n_active_sender_streams;
guint expected_nbr_blocking_msg;
/* to prevent problems when some streams are complete, some are not,
@@ -3279,16 +3280,16 @@
* streams (during DESCRIBE), we will listen to all streams. */
gst_structure_get_boolean (s, "is_complete", &is_complete);
- n_active_streams = nbr_active_streams (media);
- expected_nbr_blocking_msg = n_active_streams;
+ n_active_sender_streams = nbr_active_sender_streams (media);
+ expected_nbr_blocking_msg = n_active_sender_streams;
GST_DEBUG_OBJECT (media, "media received blocking message,"
- " n_active_streams = %d, is_complete = %d",
- n_active_streams, is_complete);
+ " n_active_sender_streams = %d, is_complete = %d",
+ n_active_sender_streams, is_complete);
- if (n_active_streams == 0 || is_complete)
+ if (n_active_sender_streams == 0 || is_complete)
priv->blocking_msg_received++;
- if (n_active_streams == 0)
+ if (n_active_sender_streams == 0)
expected_nbr_blocking_msg = priv->streams->len;
if (priv->blocked && media_streams_blocking (media) &&
diff -urN '--exclude=CVS' '--exclude=.cvsignore' '--exclude=.svn'
'--exclude=.svnignore' old/gst-rtsp-server-1.18.2/gst-rtsp-server.doap
new/gst-rtsp-server-1.18.3/gst-rtsp-server.doap
--- old/gst-rtsp-server-1.18.2/gst-rtsp-server.doap 2020-12-06
14:24:47.000000000 +0100
+++ new/gst-rtsp-server-1.18.3/gst-rtsp-server.doap 2021-01-13
22:12:06.000000000 +0100
@@ -32,6 +32,16 @@
<release>
<Version>
+ <revision>1.18.3</revision>
+ <branch>1.18</branch>
+ <name></name>
+ <created>2021-01-13</created>
+ <file-release
rdf:resource="https://gstreamer.freedesktop.org/src/gst-rtsp-server/gst-rtsp-server-1.18.3.tar.xz";
/>
+ </Version>
+ </release>
+
+ <release>
+ <Version>
<revision>1.18.2</revision>
<branch>1.18</branch>
<name></name>
diff -urN '--exclude=CVS' '--exclude=.cvsignore' '--exclude=.svn'
'--exclude=.svnignore' old/gst-rtsp-server-1.18.2/meson.build
new/gst-rtsp-server-1.18.3/meson.build
--- old/gst-rtsp-server-1.18.2/meson.build 2020-12-06 14:24:47.000000000
+0100
+++ new/gst-rtsp-server-1.18.3/meson.build 2021-01-13 22:12:06.000000000
+0100
@@ -1,5 +1,5 @@
project('gst-rtsp-server', 'c',
- version : '1.18.2',
+ version : '1.18.3',
meson_version : '>= 0.48',
default_options : ['warning_level=1', 'buildtype=debugoptimized'])
