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 2022-02-09 20:38:56
++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
Comparing /work/SRC/openSUSE:Factory/gstreamer-rtsp-server (Old)
and /work/SRC/openSUSE:Factory/.gstreamer-rtsp-server.new.1898 (New)
++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
Package is "gstreamer-rtsp-server"
Wed Feb 9 20:38:56 2022 rev:32 rq:952077 version:1.18.6
Changes:
--------
---
/work/SRC/openSUSE:Factory/gstreamer-rtsp-server/gstreamer-rtsp-server.changes
2022-01-22 08:19:10.394503849 +0100
+++
/work/SRC/openSUSE:Factory/.gstreamer-rtsp-server.new.1898/gstreamer-rtsp-server.changes
2022-02-09 20:39:45.562456244 +0100
@@ -1,0 +2,15 @@
+Fri Feb 4 19:45:17 UTC 2022 - Bj??rn Lie <bjorn....@gmail.com>
+
+- Update to version 1.18.6:
+ + rtsp-stream: fix get_rates raciness
+ + rtsp-media: Only unprepare a media if it was not already
+ unpreparing anyway
+ + rtsp-media: Unprepare suspended medias too
+ + rtsp-client: make sure sessmedia will not get freed while used
+ + rtsp-media: Also mark receive-only (RECORD) medias as prepared
+ when unsuspending
+ + rtsp-session: Don't unref medias twice if it is removed inside
+ + examples: Fix leak in appsrc2 example
+- Drop service, use source url, upstream changes in git.
+
+-------------------------------------------------------------------
Old:
----
_service
gst-rtsp-server-1.18.5.tar.xz
New:
----
gst-rtsp-server-1.18.6.tar.xz
++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
Other differences:
------------------
++++++ gstreamer-rtsp-server.spec ++++++
--- /var/tmp/diff_new_pack.sugAjJ/_old 2022-02-09 20:39:46.142457631 +0100
+++ /var/tmp/diff_new_pack.sugAjJ/_new 2022-02-09 20:39:46.150457651 +0100
@@ -20,15 +20,13 @@
%define _name gst-rtsp-server
Name: gstreamer-rtsp-server
-Version: 1.18.5
+Version: 1.18.6
Release: 0
Summary: GStreamer-based RTSP server library
License: LGPL-2.0-or-later
Group: Productivity/Multimedia/Other
URL: https://gstreamer.freedesktop.org
-# Disable tarball source url, use _service
-#Source0: %%{url}/src/gst-rtsp-server/%%{_name}-%%{version}.tar.xz
-Source0: %{_name}-%{version}.tar.xz
+Source0: %{url}/src/gst-rtsp-server/%{_name}-%{version}.tar.xz
Source99: gstreamer-rtsp-server-rpmlintrc
BuildRequires: hotdoc
++++++ gst-rtsp-server-1.18.5.tar.xz -> gst-rtsp-server-1.18.6.tar.xz ++++++
diff -urN '--exclude=CVS' '--exclude=.cvsignore' '--exclude=.svn'
'--exclude=.svnignore' old/gst-rtsp-server-1.18.5/ChangeLog
new/gst-rtsp-server-1.18.6/ChangeLog
--- old/gst-rtsp-server-1.18.5/ChangeLog 2021-09-08 21:04:14.000000000
+0200
+++ new/gst-rtsp-server-1.18.6/ChangeLog 2022-02-02 16:08:18.000000000
+0100
@@ -1,3 +1,91 @@
+=== release 1.18.6 ===
+
+2022-02-02 15:08:18 +0000 Tim-Philipp M??ller <t...@centricular.com>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * gst-rtsp-server.doap:
+ * meson.build:
+ Release 1.18.6
+
+2021-10-07 13:00:10 +0300 Sebastian Dr??ge <sebast...@centricular.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: Unprepare suspended medias too
+ Previously suspended medias immediately reached the UNPREPARED state
+ without going through the media's unprepare() vfunc. This didn't allow
+ the media subclass to do any additional cleanup, and for example the
+ shutdown-eos property of GstRTSPMedia was ignored.
+ Part-of:
<https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/221>
+
+2022-01-20 17:13:36 -0600 Michael Gruner <michael.gru...@ridgerun.com>
+
+ * examples/test-appsrc2.c:
+ gst-rtsp-server: Fix leak in appsrc2 example
+ In the need-data appsrc callback, a buffer is pulled from the
+ appsink. This buffer is then copied so that metadata is writable.
+ The copy is pushed to the appsrc but it doesn't take ownership
+ of the buffer so we need to manually unref it. The original buffer
+ is finally unreffed when the sample is freed.
+ Part-of:
<https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/220>
+
+2021-12-16 21:04:53 +0100 Mathieu Duponchelle <math...@centricular.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: fix get_rates raciness
+ Prior to this patch, we considered that a stream was blocking
+ whenever a pad probe was triggered for either the RTP pad or
+ the RTCP pad.
+ This led to situations where we subsequently unblocked and expected
+ to find a segment on the RTP pad, which was racy.
+ Instead, we now only consider that the stream is blocking when
+ the pad probe for the RTP pad has triggered with a blockable object
+ (buffer, buffer list, gap event).
+ The RTCP pad is simply blocked without affecting the state of the
+ stream otherwise.
+ Fixes #929
+ Part-of:
<https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/218>
+
+2021-10-06 18:19:29 +0300 Sebastian Dr??ge <sebast...@centricular.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: Only unprepare a media if it was not already unpreparing
anyway
+ Part-of:
<https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/216>
+
+2021-10-03 23:25:23 +0200 Ognyan Tonchev <ogn...@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-session.c:
+ * gst/rtsp-server/rtsp-session.h:
+ rtsp-client: make sure sessmedia will not get freed while used
+ handle_*_request() functions were all retrieving the session media
from
+ the session by calling gst_rtsp_session_get_media () which is a
transfer-none
+ call. If a session timeout happens at that time, the session media
may get freed
+ making the pointer invalid..
+ Fixes #757
+ Part-of:
<https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/215>
+
+2021-10-05 19:37:40 +0300 Sebastian Dr??ge <sebast...@centricular.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: Also mark receive-only (RECORD) medias as prepared when
unsuspending
+ Previously the status was only changed for other medias.
+ Part-of:
<https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/214>
+
+2021-10-01 13:51:37 +0300 Sebastian Dr??ge <sebast...@centricular.com>
+
+ * gst/rtsp-server/rtsp-session.c:
+ rtsp-session: Don't unref medias twice if it is removed inside
gst_rtsp_session_filter() while the mutex is shortly released
+ Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/757
+ Part-of:
<https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/213>
+
+2021-09-09 00:13:01 +0100 Tim-Philipp M??ller <t...@centricular.com>
+
+ * docs/gst_plugins_cache.json:
+ * meson.build:
+ Back to development
+
=== release 1.18.5 ===
2021-09-08 20:04:14 +0100 Tim-Philipp M??ller <t...@centricular.com>
@@ -5,6 +93,7 @@
* ChangeLog:
* NEWS:
* RELEASE:
+ * docs/gst_plugins_cache.json:
* gst-rtsp-server.doap:
* meson.build:
Release 1.18.5
diff -urN '--exclude=CVS' '--exclude=.cvsignore' '--exclude=.svn'
'--exclude=.svnignore' old/gst-rtsp-server-1.18.5/NEWS
new/gst-rtsp-server-1.18.6/NEWS
--- old/gst-rtsp-server-1.18.5/NEWS 2021-09-08 21:04:14.000000000 +0200
+++ new/gst-rtsp-server-1.18.6/NEWS 2022-02-02 16:08:18.000000000 +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 1.18.5 and was released
-on 8 September 2021.
+The latest bug-fix release in the 1.18 series is 1.18.6 and was released
+on 2 February 2022.
See https://gstreamer.freedesktop.org/releases/1.18/ for the latest
version of this document.
-Last updated: Wednesday 8 September 2021, 11:00 UTC (log)
+Last updated: Wednesday 2 February 2022, 11:30 UTC (log)
Introduction
@@ -2103,9 +2103,8 @@
Known Issues
- GStreamer 1.18 versions <= 1.18.4 would fail to build on Linux with
- Meson 0.58 due to an issue with the include directories. Either
- apply the patch or build with an older Meson version (<= 0.57) until
- there is a GStreamer 1.18.5 release that includes the fix.
+ Meson 0.58 due to an issue with the include directories.
+ GStreamer >= 1.18.5 includes a fix for this.
Contributors
@@ -3183,16 +3182,180 @@
- List of Merge Requests applied in 1.18.5
- List of Issues fixed in 1.18.5
+1.18.6
+
+The sixth 1.18 bug-fix release (1.18.6) was released on 2 February 2022.
+
+This release only contains bugfixes and security fixes and it should be
+safe to update from 1.18.x.
+
+Highlighted bugfixes in 1.18.6
+
+- tagdemux: Fix crash when presented with malformed files (security
+ fix)
+- video-converter: Fix broken gamma remap with high bitdepth YUV
+ output
+- shout2send: Fix issues with libshout >= 2.4.2
+- mxfdemux: fix regression with VANC tracks that only contains packet
+ types we don???t handle
+- Better plugin loading error reporting on Windows
+- Fixes for deprecations in Python 3.10
+- build fixes, memory leak fixes, reliability fixes
+- security fixes
+
+gstreamer
+
+- gstplugin: Fix for UWP build
+- gstplugin: Better warnings on plugin load failure on Windows
+- gst-ptp-helper: Do not disable multicast loopback
+- concat: fix qos event handling
+- pluginfeature: Fix object leak
+- baseparse: fix invalid avg_bitrate after reset
+- multiqueue: Fix query unref race on flush
+- gst: Initialize optional event/message fields when parsing
+- bitwriter: Fix the trailing bits lost when getting its data.
+- multiqueue: never consider a queue that is not waiting
+- input-selector: Use proper segments when cleaning cached buffers
+
+gst-plugins-base
+
+- tagdemux: Fix crash when presented with malformed files (security
+ fix)
+- videoencoder: make sure the buffer is writable before modifying
+ metadata
+- video-converter: Fix for broken gamma remap with high bitdepth YUV
+ output
+- sdpmessage: fix mapping single char fmtp params
+- oggdemux: fix a race in push mode when performing the duration seek
+- uridecodebin: Fix critical warnings
+- audio-converter: Fix resampling when there???s nothing to output
+- tcp: fix build on Solaris
+- uridecodebin3: Nullify current item after all play items are freed.
+- audio-resampler: Fix segfault when we can???t output any frames
+- urisourcebin: Handle sources with dynamic pads and pads already
+ present
+- playbin2/3: autoplug/caps: don???t expand caps to ANY
+- uridecodebin3/urisourcebin: Reusability fixes
+- rtspconnection: Only reset timeout when socket is unused
+- gstvideoaggregator.c: fix build with gcc 4.8
+
+gst-plugins-good
+
+- rtspsrc: Fix critical while serializing timeout element message
+- multifilesrc: fix caps leak
+- shout2: Add compatibility for libshout >= 2.4.2 shout_open return
+ values
+- v4l2: Update fmt if padded height is greater than fmt height
+- v4l2bufferpool: set video alignment of video meta
+- qtmux: fix deadlock in gst_qt_mux_prepare_moov_recovery
+- matroska: Add support for muxing/demuxing ffv1
+- qtdemux: Try to build AAC codec-data whenever it???s possible
+
+gst-plugins-bad
+
+- interlace: Fix a double-unref on shutdown
+- webrtcbin: Chain up to parent constructed method
+- webrtc: fix log error message in function
+ gst_webrtc_bin_set_local_description
+- mxfdemux: don???t error out if VANC track only contains packets we
+ don???t handle
+- av1parser: Fix data type of film grain param
+- assrender: Support RFC8081 mime types
+- pitch: Specify layout as required for negotiation
+- magicleap: update lumin_rt libraries names to the latest official
+ version
+- codecs: h265decoder: Fix per-slice leak
+- mpeg4videoparse: fix criticals trying to insert configs that don???t
+ exist yet
+- webrtcbin: Always set SINK/SRC flags
+- mpegtspacketizer: memcmp potentially seen_before data
+- zxing: update to support version 1.1.1
+
+gst-plugins-ugly
+
+- No changes
+
+gst-libav
+
+- avcodecmap: Add support for GBRA_10LE/BE
+
+gst-rtsp-server
+
+- rtsp-stream: fix get_rates raciness
+- rtsp-media: Only unprepare a media if it was not already unpreparing
+ anyway
+- rtsp-media: Unprepare suspended medias too
+- rtsp-client: make sure sessmedia will not get freed while used
+- rtsp-media: Also mark receive-only (RECORD) medias as prepared when
+ unsuspending
+- rtsp-session: Don???t unref medias twice if it is removed inside???
+- examples: Fix leak in appsrc2 example
+
+gstreamer-vaapi
+
+- libs: video-format: Check if formats map is not NULL
+- vaapidecode: Autogenerate caps template
+- vaapipostproc: copy over metadata also when using system allocated
+ buffer
+
+gst-python
+
+- Avoid treating float as int (fix for Python 3.10)
+
+gst-editing-services
+
+- meson: Remove duplicate definition of ???examples??? option
+
+gst-devtools
+
+- No changes
+
+gst-integration-testsuites
+
+- No changes
+
+gst-build
+
+- env: Fix deprecations from python 3.10
+- Various fixes for macOS
+- update FFmpeg wrap to 4.3.3
+
+Cerbero build tool and packaging changes in 1.18.6
+
+- Some fixes for Fedora 34
+- cerbero: Backport fix for removed loop param of PriorityQueue()
+- cerbero: Fix support for Fedora 35
+- Add support for Visual Studio 2022
+- openssl.recipe: Fix crash on iOS TestFlight
+- UnixBootstrapper: remove sudo as root user
+- bzip2.recipe: bump version to 1.0.8
+- openssl.recipe: upgrade to version 1.1.1l
+
+Contributors to 1.18.6
+
+Antonio Ospite, C??lestin Marot, Dave Pich??, Erlend Eriksen, Fabrice
+Fontaine, Guillaume Desmottes, Haihua Hu, He Junyan, Jakub Adam, Jan
+Alexander Steffens (heftig), Jan Schmidt, Jeremy Cline, Jordan Petridis,
+Mathieu Duponchelle, Matthew Waters, Mengkejiergeli Ba, Michael Gruner,
+Nirbheek Chauhan, Ognyan Tonchev, Pascal Hache, Rafa?? Dzi??giel,
+Sebastian Dr??ge, Seungha Yang, St??phane Cerveau, Teng En Ung,Thibault
+Saunier, Thomas Klausner, Tim-Philipp M??ller, Tobias Reineke, Tobias
+Ronge, Tomasz Andrzejak, Trung Do, V??ctor Manuel J??quez Leal, 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 1.18.6
+
+- List of Merge Requests applied in 1.18.6
+- List of Issues fixed in 1.18.6
+
Schedule for 1.20
-Our next major feature release will be 1.20, and 1.19 will be the
-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
-September/October 2021, with the first 1.20 stable release hopefully
-towards the end of October 2021.
+Our next major feature release will be 1.20, and will be released in
+early February 2022. You can track its progress on the 1.20 Release
+Notes page.
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-1.18.5/RELEASE
new/gst-rtsp-server-1.18.6/RELEASE
--- old/gst-rtsp-server-1.18.5/RELEASE 2021-09-08 21:04:14.000000000 +0200
+++ new/gst-rtsp-server-1.18.6/RELEASE 2022-02-02 16:08:18.000000000 +0100
@@ -1,4 +1,4 @@
-This is GStreamer gst-rtsp-server 1.18.5.
+This is GStreamer gst-rtsp-server 1.18.6.
The GStreamer team is thrilled to announce a new major feature release
of your favourite cross-platform multimedia framework!
@@ -82,7 +82,7 @@
For help and support, please subscribe to and send questions to the
gstreamer-devel mailing list (see below for details).
-There is also a #gstreamer IRC channel on the Freenode IRC network.
+There is also a #gstreamer IRC channel on the OFTC IRC network.
==== Developers ====
diff -urN '--exclude=CVS' '--exclude=.cvsignore' '--exclude=.svn'
'--exclude=.svnignore' old/gst-rtsp-server-1.18.5/docs/gst_plugins_cache.json
new/gst-rtsp-server-1.18.6/docs/gst_plugins_cache.json
--- old/gst-rtsp-server-1.18.5/docs/gst_plugins_cache.json 2021-09-08
21:04:14.000000000 +0200
+++ new/gst-rtsp-server-1.18.6/docs/gst_plugins_cache.json 2022-02-02
16:08:18.000000000 +0100
@@ -321,7 +321,7 @@
"construct": false,
"construct-only": false,
"controllable": false,
- "default": "GStreamer/1.18.5",
+ "default": "GStreamer/1.18.6",
"mutable": "null",
"readable": true,
"type": "gchararray",
diff -urN '--exclude=CVS' '--exclude=.cvsignore' '--exclude=.svn'
'--exclude=.svnignore' old/gst-rtsp-server-1.18.5/examples/test-appsrc2.c
new/gst-rtsp-server-1.18.6/examples/test-appsrc2.c
--- old/gst-rtsp-server-1.18.5/examples/test-appsrc2.c 2021-09-08
21:04:14.000000000 +0200
+++ new/gst-rtsp-server-1.18.6/examples/test-appsrc2.c 2022-02-02
16:08:18.000000000 +0100
@@ -63,6 +63,7 @@
GST_BUFFER_PTS (buffer) = pts;
GST_BUFFER_DTS (buffer) = dts;
g_signal_emit_by_name (appsrc, "push-buffer", buffer, &ret);
+ gst_buffer_unref (buffer);
}
/* we don't need the appsink sample anymore */
diff -urN '--exclude=CVS' '--exclude=.cvsignore' '--exclude=.svn'
'--exclude=.svnignore' old/gst-rtsp-server-1.18.5/gst/rtsp-server/rtsp-client.c
new/gst-rtsp-server-1.18.6/gst/rtsp-server/rtsp-client.c
--- old/gst-rtsp-server-1.18.5/gst/rtsp-server/rtsp-client.c 2021-09-08
21:04:14.000000000 +0200
+++ new/gst-rtsp-server-1.18.6/gst/rtsp-server/rtsp-client.c 2022-02-02
16:08:18.000000000 +0100
@@ -1445,7 +1445,7 @@
path = klass->make_path_from_uri (client, ctx->uri);
/* get a handle to the configuration of the media in the session */
- sessmedia = gst_rtsp_session_get_media (session, path, &matched);
+ sessmedia = gst_rtsp_session_dup_media (session, path, &matched);
if (!sessmedia)
goto not_found;
@@ -1480,6 +1480,7 @@
/* unmanage the media in the session, returns false if all media session
* are torn down. */
keep_session = gst_rtsp_session_release_media (session, sessmedia);
+ g_object_unref (sessmedia);
/* construct the response now */
code = GST_RTSP_STS_OK;
@@ -1530,6 +1531,7 @@
send_generic_response (client,
GST_RTSP_STS_ONLY_AGGREGATE_OPERATION_ALLOWED, ctx);
g_free (path);
+ g_object_unref (sessmedia);
return FALSE;
}
sig_failed:
@@ -1539,6 +1541,7 @@
send_generic_response (client, sig_result, ctx);
gst_rtsp_media_unlock (media);
g_object_unref (media);
+ g_object_unref (sessmedia);
return FALSE;
}
}
@@ -1697,7 +1700,7 @@
path = klass->make_path_from_uri (client, ctx->uri);
/* get a handle to the configuration of the media in the session */
- sessmedia = gst_rtsp_session_get_media (session, path, &matched);
+ sessmedia = gst_rtsp_session_dup_media (session, path, &matched);
if (!sessmedia)
goto not_found;
@@ -1744,6 +1747,7 @@
/* the state is now READY */
gst_rtsp_session_media_set_rtsp_state (sessmedia, GST_RTSP_STATE_READY);
+ g_object_unref (sessmedia);
g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_PAUSE_REQUEST], 0,
ctx);
@@ -1777,6 +1781,7 @@
GST_ERROR ("client %p: no aggregate path %s", client, path);
send_generic_response (client,
GST_RTSP_STS_ONLY_AGGREGATE_OPERATION_ALLOWED, ctx);
+ g_object_unref (sessmedia);
g_free (path);
return FALSE;
}
@@ -1786,6 +1791,7 @@
gst_rtsp_status_as_text (sig_result));
send_generic_response (client, sig_result, ctx);
gst_rtsp_media_unlock (media);
+ g_object_unref (sessmedia);
g_object_unref (media);
return FALSE;
}
@@ -1795,6 +1801,7 @@
send_generic_response (client, GST_RTSP_STS_METHOD_NOT_VALID_IN_THIS_STATE,
ctx);
gst_rtsp_media_unlock (media);
+ g_object_unref (sessmedia);
g_object_unref (media);
return FALSE;
}
@@ -1803,6 +1810,7 @@
GST_ERROR ("client %p: pausing not supported", client);
send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
gst_rtsp_media_unlock (media);
+ g_object_unref (sessmedia);
g_object_unref (media);
return FALSE;
}
@@ -2066,7 +2074,7 @@
path = klass->make_path_from_uri (client, uri);
/* get a handle to the configuration of the media in the session */
- sessmedia = gst_rtsp_session_get_media (session, path, &matched);
+ sessmedia = gst_rtsp_session_dup_media (session, path, &matched);
if (!sessmedia)
goto not_found;
@@ -2161,6 +2169,7 @@
gst_rtsp_session_media_set_state (sessmedia, GST_STATE_PLAYING);
gst_rtsp_session_media_set_rtsp_state (sessmedia, GST_RTSP_STATE_PLAYING);
+ g_object_unref (sessmedia);
g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_PLAY_REQUEST], 0, ctx);
@@ -2193,6 +2202,7 @@
GST_ERROR ("client %p: no aggregate path %s", client, path);
send_generic_response (client,
GST_RTSP_STS_ONLY_AGGREGATE_OPERATION_ALLOWED, ctx);
+ g_object_unref (sessmedia);
g_free (path);
return FALSE;
}
@@ -2203,6 +2213,7 @@
send_generic_response (client, sig_result, ctx);
gst_rtsp_media_unlock (media);
g_object_unref (media);
+ g_object_unref (sessmedia);
return FALSE;
}
invalid_state:
@@ -2212,6 +2223,7 @@
ctx);
gst_rtsp_media_unlock (media);
g_object_unref (media);
+ g_object_unref (sessmedia);
return FALSE;
}
pipeline_error:
@@ -2221,6 +2233,7 @@
ctx);
gst_rtsp_media_unlock (media);
g_object_unref (media);
+ g_object_unref (sessmedia);
return FALSE;
}
unsuspend_failed:
@@ -2229,6 +2242,7 @@
send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, ctx);
gst_rtsp_media_unlock (media);
g_object_unref (media);
+ g_object_unref (sessmedia);
return FALSE;
}
invalid_mode:
@@ -2237,6 +2251,7 @@
send_generic_response (client, code, ctx);
gst_rtsp_media_unlock (media);
g_object_unref (media);
+ g_object_unref (sessmedia);
return FALSE;
}
unsupported_mode:
@@ -2245,6 +2260,7 @@
send_generic_response (client, GST_RTSP_STS_METHOD_NOT_ALLOWED, ctx);
gst_rtsp_media_unlock (media);
g_object_unref (media);
+ g_object_unref (sessmedia);
return FALSE;
}
get_rates_error:
@@ -2253,6 +2269,7 @@
send_generic_response (client, GST_RTSP_STS_INTERNAL_SERVER_ERROR, ctx);
gst_rtsp_media_unlock (media);
g_object_unref (media);
+ g_object_unref (sessmedia);
return FALSE;
}
adjust_play_response_failed:
@@ -2261,6 +2278,7 @@
send_generic_response (client, code, ctx);
gst_rtsp_media_unlock (media);
g_object_unref (media);
+ g_object_unref (sessmedia);
return FALSE;
}
rtp_info_error:
@@ -2269,6 +2287,7 @@
send_generic_response (client, GST_RTSP_STS_INTERNAL_SERVER_ERROR, ctx);
gst_rtsp_media_unlock (media);
g_object_unref (media);
+ g_object_unref (sessmedia);
return FALSE;
}
}
diff -urN '--exclude=CVS' '--exclude=.cvsignore' '--exclude=.svn'
'--exclude=.svnignore' old/gst-rtsp-server-1.18.5/gst/rtsp-server/rtsp-media.c
new/gst-rtsp-server-1.18.6/gst/rtsp-server/rtsp-media.c
--- old/gst-rtsp-server-1.18.5/gst/rtsp-server/rtsp-media.c 2021-09-08
21:04:14.000000000 +0200
+++ new/gst-rtsp-server-1.18.6/gst/rtsp-server/rtsp-media.c 2022-02-02
16:08:18.000000000 +0100
@@ -4083,12 +4083,15 @@
priv->prepare_count--;
if (priv->prepare_count > 0)
goto is_busy;
+ if (priv->status == GST_RTSP_MEDIA_STATUS_UNPREPARING)
+ goto is_unpreparing;
GST_INFO ("unprepare media %p", media);
set_target_state (media, GST_STATE_NULL, FALSE);
success = TRUE;
- if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARED) {
+ if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARED
+ || priv->status == GST_RTSP_MEDIA_STATUS_SUSPENDED) {
GstRTSPMediaClass *klass;
klass = GST_RTSP_MEDIA_GET_CLASS (media);
@@ -4108,6 +4111,12 @@
GST_INFO ("media %p was already unprepared", media);
return TRUE;
}
+is_unpreparing:
+ {
+ g_rec_mutex_unlock (&priv->state_lock);
+ GST_INFO ("media %p is already unpreparing", media);
+ return TRUE;
+ }
is_busy:
{
GST_INFO ("media %p still prepared %d times", media, priv->prepare_count);
@@ -4614,9 +4623,8 @@
switch (priv->suspend_mode) {
case GST_RTSP_SUSPEND_MODE_NONE:
- if (gst_rtsp_media_is_receive_only (media))
- break;
- if (media_streams_blocking (media)) {
+ if (!gst_rtsp_media_is_receive_only (media)
+ && media_streams_blocking (media)) {
g_rec_mutex_unlock (&priv->state_lock);
if (gst_rtsp_media_get_status (media) == GST_RTSP_MEDIA_STATUS_ERROR) {
g_rec_mutex_lock (&priv->state_lock);
diff -urN '--exclude=CVS' '--exclude=.cvsignore' '--exclude=.svn'
'--exclude=.svnignore'
old/gst-rtsp-server-1.18.5/gst/rtsp-server/rtsp-session.c
new/gst-rtsp-server-1.18.6/gst/rtsp-server/rtsp-session.c
--- old/gst-rtsp-server-1.18.5/gst/rtsp-server/rtsp-session.c 2021-09-08
21:04:14.000000000 +0200
+++ new/gst-rtsp-server-1.18.6/gst/rtsp-server/rtsp-session.c 2022-02-02
16:08:18.000000000 +0100
@@ -343,20 +343,9 @@
return more;
}
-/**
- * gst_rtsp_session_get_media:
- * @sess: a #GstRTSPSession
- * @path: the path for the media
- * @matched: (out): the amount of matched characters
- *
- * Get the session media for @path. @matched will contain the number of matched
- * characters of @path.
- *
- * Returns: (transfer none) (nullable): the configuration for @path in @sess.
- */
-GstRTSPSessionMedia *
-gst_rtsp_session_get_media (GstRTSPSession * sess, const gchar * path,
- gint * matched)
+static GstRTSPSessionMedia *
+_gst_rtsp_session_get_media (GstRTSPSession * sess, const gchar * path,
+ gint * matched, gboolean dup)
{
GstRTSPSessionPrivate *priv;
GstRTSPSessionMedia *result;
@@ -384,6 +373,9 @@
}
}
}
+
+ if (result && dup)
+ result = g_object_ref (result);
g_mutex_unlock (&priv->lock);
*matched = best;
@@ -392,6 +384,45 @@
}
/**
+ * gst_rtsp_session_get_media:
+ * @sess: a #GstRTSPSession
+ * @path: the path for the media
+ * @matched: (out): the amount of matched characters
+ *
+ * Gets the session media for @path. @matched will contain the number of
matched
+ * characters of @path.
+ *
+ * Returns: (transfer none) (nullable): the configuration for @path in @sess.
+ */
+GstRTSPSessionMedia *
+gst_rtsp_session_get_media (GstRTSPSession * sess, const gchar * path,
+ gint * matched)
+{
+ return _gst_rtsp_session_get_media (sess, path, matched, FALSE);
+}
+
+/**
+ * gst_rtsp_session_dup_media:
+ * @sess: a #GstRTSPSession
+ * @path: the path for the media
+ * @matched: (out): the amount of matched characters
+ *
+ * Gets the session media for @path, increasing its reference count. @matched
+ * will contain the number of matched characters of @path.
+ *
+ * Returns: (transfer full) (nullable): the configuration for @path in @sess,
+ * should be unreferenced when no longer needed.
+ *
+ * Since: 1.20
+ */
+GstRTSPSessionMedia *
+gst_rtsp_session_dup_media (GstRTSPSession * sess, const gchar * path,
+ gint * matched)
+{
+ return _gst_rtsp_session_get_media (sess, path, matched, TRUE);
+}
+
+/**
* gst_rtsp_session_filter:
* @sess: a #GstRTSPSession
* @func: (scope call) (allow-none): a callback
@@ -454,19 +485,38 @@
res = func (sess, media, user_data);
g_mutex_lock (&priv->lock);
- } else
+ } else {
res = GST_RTSP_FILTER_REF;
+ }
changed = (cookie != priv->medias_cookie);
switch (res) {
case GST_RTSP_FILTER_REMOVE:
- if (changed)
- priv->medias = g_list_remove (priv->medias, media);
- else
+ if (changed) {
+ GList *l;
+
+ walk = NULL;
+
+ for (l = priv->medias; l; l = l->next) {
+ if (l->data == media) {
+ walk = l;
+ break;
+ }
+ }
+ }
+
+ /* The media might have been removed from the list while the mutex was
+ * unlocked above. In that case there's nothing else to do here as the
+ * only reference to the media owned by this function is in the
+ * visited hash table and that is released in the end
+ */
+ if (walk) {
priv->medias = g_list_delete_link (priv->medias, walk);
+ g_object_unref (media);
+ }
+
cookie = ++priv->medias_cookie;
- g_object_unref (media);
break;
case GST_RTSP_FILTER_REF:
result = g_list_prepend (result, g_object_ref (media));
diff -urN '--exclude=CVS' '--exclude=.cvsignore' '--exclude=.svn'
'--exclude=.svnignore'
old/gst-rtsp-server-1.18.5/gst/rtsp-server/rtsp-session.h
new/gst-rtsp-server-1.18.6/gst/rtsp-server/rtsp-session.h
--- old/gst-rtsp-server-1.18.5/gst/rtsp-server/rtsp-session.h 2021-09-08
21:04:14.000000000 +0200
+++ new/gst-rtsp-server-1.18.6/gst/rtsp-server/rtsp-session.h 2022-02-02
16:08:18.000000000 +0100
@@ -141,7 +141,12 @@
GstRTSPSessionMedia * gst_rtsp_session_get_media (GstRTSPSession
*sess,
const gchar
*path,
gint * matched);
+/* get media in a session, increasing its reference count */
+GST_RTSP_SERVER_API
+GstRTSPSessionMedia * gst_rtsp_session_dup_media (GstRTSPSession
*sess,
+ const gchar
*path,
+ gint * matched);
/**
* GstRTSPSessionFilterFunc:
* @sess: a #GstRTSPSession object
diff -urN '--exclude=CVS' '--exclude=.cvsignore' '--exclude=.svn'
'--exclude=.svnignore' old/gst-rtsp-server-1.18.5/gst/rtsp-server/rtsp-stream.c
new/gst-rtsp-server-1.18.6/gst/rtsp-server/rtsp-stream.c
--- old/gst-rtsp-server-1.18.5/gst/rtsp-server/rtsp-stream.c 2021-09-08
21:04:14.000000000 +0200
+++ new/gst-rtsp-server-1.18.6/gst/rtsp-server/rtsp-stream.c 2022-02-02
16:08:18.000000000 +0100
@@ -5219,7 +5219,7 @@
}
static GstPadProbeReturn
-pad_blocking (GstPad * pad, GstPadProbeInfo * info, gpointer user_data)
+rtp_pad_blocking (GstPad * pad, GstPadProbeInfo * info, gpointer user_data)
{
GstRTSPStreamPrivate *priv;
GstRTSPStream *stream;
@@ -5306,6 +5306,41 @@
return ret;
}
+static GstPadProbeReturn
+rtcp_pad_blocking (GstPad * pad, GstPadProbeInfo * info, gpointer user_data)
+{
+ GstRTSPStreamPrivate *priv;
+ GstRTSPStream *stream;
+ GstPadProbeReturn ret = GST_PAD_PROBE_OK;
+
+ stream = user_data;
+ priv = stream->priv;
+
+ g_mutex_lock (&priv->lock);
+
+ if ((info->type & GST_PAD_PROBE_TYPE_BUFFER) ||
+ (info->type & GST_PAD_PROBE_TYPE_BUFFER_LIST)) {
+ GST_DEBUG_OBJECT (pad, "Now blocking on buffer");
+ } else if ((info->type & GST_PAD_PROBE_TYPE_EVENT_DOWNSTREAM)) {
+ if (GST_EVENT_TYPE (info->data) == GST_EVENT_GAP) {
+ GST_DEBUG_OBJECT (pad, "Now blocking on gap event");
+ ret = GST_PAD_PROBE_OK;
+ } else {
+ ret = GST_PAD_PROBE_PASS;
+ g_mutex_unlock (&priv->lock);
+ goto done;
+ }
+ } else {
+ g_assert_not_reached ();
+ }
+
+ g_mutex_unlock (&priv->lock);
+
+done:
+ return ret;
+}
+
+
static void
set_blocked (GstRTSPStream * stream, gboolean blocked)
{
@@ -5330,11 +5365,20 @@
priv->blocked_buffer = FALSE;
priv->blocked_running_time = GST_CLOCK_TIME_NONE;
priv->blocked_clock_rate = 0;
- priv->blocked_id[i] = gst_pad_add_probe (priv->send_src[i],
- GST_PAD_PROBE_TYPE_BLOCK | GST_PAD_PROBE_TYPE_BUFFER |
- GST_PAD_PROBE_TYPE_BUFFER_LIST |
- GST_PAD_PROBE_TYPE_EVENT_DOWNSTREAM, pad_blocking,
- g_object_ref (stream), g_object_unref);
+
+ if (i == 0) {
+ priv->blocked_id[i] = gst_pad_add_probe (priv->send_src[i],
+ GST_PAD_PROBE_TYPE_BLOCK | GST_PAD_PROBE_TYPE_BUFFER |
+ GST_PAD_PROBE_TYPE_BUFFER_LIST |
+ GST_PAD_PROBE_TYPE_EVENT_DOWNSTREAM, rtp_pad_blocking,
+ g_object_ref (stream), g_object_unref);
+ } else {
+ priv->blocked_id[i] = gst_pad_add_probe (priv->send_src[i],
+ GST_PAD_PROBE_TYPE_BLOCK | GST_PAD_PROBE_TYPE_BUFFER |
+ GST_PAD_PROBE_TYPE_BUFFER_LIST |
+ GST_PAD_PROBE_TYPE_EVENT_DOWNSTREAM, rtcp_pad_blocking,
+ g_object_ref (stream), g_object_unref);
+ }
}
}
} else {
diff -urN '--exclude=CVS' '--exclude=.cvsignore' '--exclude=.svn'
'--exclude=.svnignore' old/gst-rtsp-server-1.18.5/gst-rtsp-server.doap
new/gst-rtsp-server-1.18.6/gst-rtsp-server.doap
--- old/gst-rtsp-server-1.18.5/gst-rtsp-server.doap 2021-09-08
21:04:14.000000000 +0200
+++ new/gst-rtsp-server-1.18.6/gst-rtsp-server.doap 2022-02-02
16:08:18.000000000 +0100
@@ -32,6 +32,16 @@
<release>
<Version>
+ <revision>1.18.6</revision>
+ <branch>1.18</branch>
+ <name></name>
+ <created>2022-02-02</created>
+ <file-release
rdf:resource="https://gstreamer.freedesktop.org/src/gst-rtsp-server/gst-rtsp-server-1.18.6.tar.xz";
/>
+ </Version>
+ </release>
+
+ <release>
+ <Version>
<revision>1.18.5</revision>
<branch>1.18</branch>
<name></name>
diff -urN '--exclude=CVS' '--exclude=.cvsignore' '--exclude=.svn'
'--exclude=.svnignore' old/gst-rtsp-server-1.18.5/meson.build
new/gst-rtsp-server-1.18.6/meson.build
--- old/gst-rtsp-server-1.18.5/meson.build 2021-09-08 21:04:14.000000000
+0200
+++ new/gst-rtsp-server-1.18.6/meson.build 2022-02-02 16:08:18.000000000
+0100
@@ -1,5 +1,5 @@
project('gst-rtsp-server', 'c',
- version : '1.18.5',
+ version : '1.18.6',
meson_version : '>= 0.48',
default_options : ['warning_level=1', 'buildtype=debugoptimized'])
