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Hello community,
here is the log from the commit of package gstreamer-rtsp-server for
openSUSE:Factory checked in at 2021-04-12 12:36:34
++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
Comparing /work/SRC/openSUSE:Factory/gstreamer-rtsp-server (Old)
and /work/SRC/openSUSE:Factory/.gstreamer-rtsp-server.new.2401 (New)
++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
Package is "gstreamer-rtsp-server"
Mon Apr 12 12:36:34 2021 rev:29 rq:883609 version:1.18.4
Changes:
--------
---
/work/SRC/openSUSE:Factory/gstreamer-rtsp-server/gstreamer-rtsp-server.changes
2021-01-21 21:54:23.237770982 +0100
+++
/work/SRC/openSUSE:Factory/.gstreamer-rtsp-server.new.2401/gstreamer-rtsp-server.changes
2021-04-12 12:39:25.729513255 +0200
@@ -1,0 +2,10 @@
+Wed Mar 31 16:21:16 UTC 2021 - Antonio Larrosa <alarr...@suse.com>
+
+- Update to version 1.18.4:
+ + 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
+
+-------------------------------------------------------------------
Old:
----
gst-rtsp-server-1.18.3.tar.xz
New:
----
gst-rtsp-server-1.18.4.tar.xz
++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
Other differences:
------------------
++++++ gstreamer-rtsp-server.spec ++++++
--- /var/tmp/diff_new_pack.TgxD1y/_old 2021-04-12 12:39:26.389514021 +0200
+++ /var/tmp/diff_new_pack.TgxD1y/_new 2021-04-12 12:39:26.389514021 +0200
@@ -20,7 +20,7 @@
%define _name gst-rtsp-server
Name: gstreamer-rtsp-server
-Version: 1.18.3
+Version: 1.18.4
Release: 0
Summary: GStreamer-based RTSP server library
License: LGPL-2.0-or-later
++++++ _service ++++++
--- /var/tmp/diff_new_pack.TgxD1y/_old 2021-04-12 12:39:26.409514045 +0200
+++ /var/tmp/diff_new_pack.TgxD1y/_new 2021-04-12 12:39:26.409514045 +0200
@@ -9,7 +9,7 @@
<!--
<param name="changesgenerate">enable</param>
-->
- <param name="revision">1.18.3</param>
+ <param name="revision">1.18.4</param>
<param name="scm">git</param>
</service>
<service name="recompress" mode="disabled">
++++++ gst-rtsp-server-1.18.3.tar.xz -> gst-rtsp-server-1.18.4.tar.xz ++++++
diff -urN '--exclude=CVS' '--exclude=.cvsignore' '--exclude=.svn'
'--exclude=.svnignore' old/gst-rtsp-server-1.18.3/ChangeLog
new/gst-rtsp-server-1.18.4/ChangeLog
--- old/gst-rtsp-server-1.18.3/ChangeLog 2021-01-13 22:12:06.000000000
+0100
+++ new/gst-rtsp-server-1.18.4/ChangeLog 2021-03-15 18:49:54.000000000
+0100
@@ -1,3 +1,87 @@
+=== release 1.18.4 ===
+
+2021-03-15 17:49:53 +0000 Tim-Philipp M??ller <t...@centricular.com>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * gst-rtsp-server.doap:
+ * meson.build:
+ Release 1.18.4
+
+2021-02-15 12:07:15 +0000 Tim-Philipp M??ller <t...@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 M??ller <t...@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 M??ller <t...@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 M??ller <t...@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 <bra...@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 <bra...@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 M??ller <t...@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 M??ller <t...@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 -urN '--exclude=CVS' '--exclude=.cvsignore' '--exclude=.svn'
'--exclude=.svnignore' old/gst-rtsp-server-1.18.3/NEWS
new/gst-rtsp-server-1.18.4/NEWS
--- old/gst-rtsp-server-1.18.3/NEWS 2021-01-13 22:12:06.000000000 +0100
+++ new/gst-rtsp-server-1.18.4/NEWS 2021-03-15 18:49:54.000000000 +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 1.18.3 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 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 Garc??a, 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
+Bj??rkstr??m, Marijn Suijten, Matthew Waters, Paul Goulpi??, Philipp Zabel,
+Rafa?? Dzi??giel, Sebastian Dr??ge, Seungha Yang, Staz M, St??phane Cerveau,
+Thibault Saunier, Tim-Philipp M??ller, V??ctor Manuel J??quez 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 @@
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 -urN '--exclude=CVS' '--exclude=.cvsignore' '--exclude=.svn'
'--exclude=.svnignore' old/gst-rtsp-server-1.18.3/RELEASE
new/gst-rtsp-server-1.18.4/RELEASE
--- old/gst-rtsp-server-1.18.3/RELEASE 2021-01-13 22:12:06.000000000 +0100
+++ new/gst-rtsp-server-1.18.4/RELEASE 2021-03-15 18:49:54.000000000 +0100
@@ -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 -urN '--exclude=CVS' '--exclude=.cvsignore' '--exclude=.svn'
'--exclude=.svnignore' old/gst-rtsp-server-1.18.3/docs/gst_plugins_cache.json
new/gst-rtsp-server-1.18.4/docs/gst_plugins_cache.json
--- old/gst-rtsp-server-1.18.3/docs/gst_plugins_cache.json 2021-01-13
22:12:06.000000000 +0100
+++ new/gst-rtsp-server-1.18.4/docs/gst_plugins_cache.json 2021-03-15
18:49:54.000000000 +0100
@@ -321,7 +321,7 @@
"construct": false,
"construct-only": false,
"controllable": false,
- "default": "GStreamer/1.18.3",
+ "default": "GStreamer/1.18.4",
"mutable": "null",
"readable": true,
"type": "gchararray",
diff -urN '--exclude=CVS' '--exclude=.cvsignore' '--exclude=.svn'
'--exclude=.svnignore' old/gst-rtsp-server-1.18.3/gst/rtsp-server/rtsp-client.c
new/gst-rtsp-server-1.18.4/gst/rtsp-server/rtsp-client.c
--- old/gst-rtsp-server-1.18.3/gst/rtsp-server/rtsp-client.c 2021-01-13
22:12:06.000000000 +0100
+++ new/gst-rtsp-server-1.18.4/gst/rtsp-server/rtsp-client.c 2021-03-15
18:49:54.000000000 +0100
@@ -1369,6 +1369,55 @@
return TRUE;
}
+/* The cleanup_transports function is called from handle_teardown_request() to
+ * remove any stream transports from the newly closed session that were added
to
+ * priv->transports in handle_setup_request(). This is done to avoid forwarding
+ * data from the client on a channel that we just closed.
+ */
+static void
+cleanup_transports (GstRTSPClient * client, GPtrArray * transports)
+{
+ GstRTSPClientPrivate *priv = client->priv;
+ GstRTSPStreamTransport *stream_transport;
+ const GstRTSPTransport *rtsp_transport;
+ guint i;
+
+ GST_LOG_OBJECT (client, "potentially removing %u transports",
+ transports->len);
+
+ for (i = 0; i < transports->len; i++) {
+ stream_transport = g_ptr_array_index (transports, i);
+ if (stream_transport == NULL) {
+ GST_LOG_OBJECT (client, "stream transport %u is NULL, continue", i);
+ continue;
+ }
+
+ rtsp_transport = gst_rtsp_stream_transport_get_transport
(stream_transport);
+ if (rtsp_transport == NULL) {
+ GST_LOG_OBJECT (client, "RTSP transport %u is NULL, continue", i);
+ continue;
+ }
+
+ /* priv->transport only stores transports where RTP is tunneled over RTSP
*/
+ if (rtsp_transport->lower_transport == GST_RTSP_LOWER_TRANS_TCP) {
+ if (!g_hash_table_remove (priv->transports,
+ GINT_TO_POINTER (rtsp_transport->interleaved.min))) {
+ GST_WARNING_OBJECT (client,
+ "failed removing transport with key '%d' from priv->transports",
+ rtsp_transport->interleaved.min);
+ }
+ if (!g_hash_table_remove (priv->transports,
+ GINT_TO_POINTER (rtsp_transport->interleaved.max))) {
+ GST_WARNING_OBJECT (client,
+ "failed removing transport with key '%d' from priv->transports",
+ rtsp_transport->interleaved.max);
+ }
+ } else {
+ GST_LOG_OBJECT (client, "transport %u not RTP/RTSP, skip it", i);
+ }
+ }
+}
+
static gboolean
handle_teardown_request (GstRTSPClient * client, GstRTSPContext * ctx)
{
@@ -1382,6 +1431,7 @@
gint matched;
gboolean keep_session;
GstRTSPStatusCode sig_result;
+ GPtrArray *session_media_transports;
if (!ctx->session)
goto no_session;
@@ -1417,6 +1467,10 @@
goto sig_failed;
}
+ /* get a reference to the transports in the session media so we can clean up
+ * our priv->transports before returning */
+ session_media_transports = gst_rtsp_session_media_get_transports (sessmedia);
+
/* we emit the signal before closing the connection */
g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_TEARDOWN_REQUEST],
0, ctx);
@@ -1442,6 +1496,12 @@
gst_rtsp_media_unlock (media);
g_object_unref (media);
+ /* remove all transports that were present in the session media which we just
+ * unmanaged from the priv->transports array, so we do not try to handle data
+ * on channels that were just closed */
+ cleanup_transports (client, session_media_transports);
+ g_ptr_array_unref (session_media_transports);
+
return TRUE;
/* ERRORS */
diff -urN '--exclude=CVS' '--exclude=.cvsignore' '--exclude=.svn'
'--exclude=.svnignore' old/gst-rtsp-server-1.18.3/gst/rtsp-server/rtsp-stream.c
new/gst-rtsp-server-1.18.4/gst/rtsp-server/rtsp-stream.c
--- old/gst-rtsp-server-1.18.3/gst/rtsp-server/rtsp-stream.c 2021-01-13
22:12:06.000000000 +0100
+++ new/gst-rtsp-server-1.18.4/gst/rtsp-server/rtsp-stream.c 2021-03-15
18:49:54.000000000 +0100
@@ -2702,11 +2702,10 @@
send_func (GstRTSPStream * stream)
{
GstRTSPStreamPrivate *priv = stream->priv;
- gboolean cont = TRUE;
g_mutex_lock (&priv->send_lock);
- while (cont) {
+ while (priv->continue_sending) {
int i;
int idx = -1;
guint cookie;
@@ -2732,10 +2731,9 @@
g_mutex_unlock (&priv->lock);
g_mutex_lock (&priv->send_lock);
- while (cookie == priv->send_cookie) {
+ while (cookie == priv->send_cookie && priv->continue_sending) {
g_cond_wait (&priv->send_cond, &priv->send_lock);
}
- cont = priv->continue_sending;
}
g_mutex_unlock (&priv->send_lock);
diff -urN '--exclude=CVS' '--exclude=.cvsignore' '--exclude=.svn'
'--exclude=.svnignore'
old/gst-rtsp-server-1.18.3/gst/rtsp-sink/gstrtspclientsink.c
new/gst-rtsp-server-1.18.4/gst/rtsp-sink/gstrtspclientsink.c
--- old/gst-rtsp-server-1.18.3/gst/rtsp-sink/gstrtspclientsink.c
2021-01-13 22:12:06.000000000 +0100
+++ new/gst-rtsp-server-1.18.4/gst/rtsp-sink/gstrtspclientsink.c
2021-03-15 18:49:54.000000000 +0100
@@ -1071,6 +1071,8 @@
goto out;
}
+ GST_MINI_OBJECT_FLAG_SET (caps, GST_MINI_OBJECT_FLAG_MAY_BE_LEAKED);
+
out:
g_once_init_leave (&ret, caps);
}
@@ -4340,6 +4342,18 @@
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)
{
@@ -4374,14 +4388,25 @@
* parts yet. */
gst_rtsp_client_sink_collect_streams (sink);
+ if (gst_rtsp_client_sink_is_stopping (sink)) {
+ GST_INFO_OBJECT (sink, "task stopped, shutting down");
+ return GST_RTSP_EINTR;
+ }
+
g_mutex_lock (&sink->block_streams_lock);
/* Wait for streams to be blocked */
- while (sink->n_streams_blocked < g_list_length (sink->contexts)) {
+ while (sink->n_streams_blocked < g_list_length (sink->contexts)
+ && !gst_rtsp_client_sink_is_stopping (sink)) {
GST_DEBUG_OBJECT (sink, "waiting for streams to be blocked");
g_cond_wait (&sink->block_streams_cond, &sink->block_streams_lock);
}
g_mutex_unlock (&sink->block_streams_lock);
+ if (gst_rtsp_client_sink_is_stopping (sink)) {
+ GST_INFO_OBJECT (sink, "task stopped, shutting down");
+ return GST_RTSP_EINTR;
+ }
+
/* Send announce, then setup for all streams */
gst_sdp_message_init (&sink->cursdp);
sdp = &sink->cursdp;
@@ -4921,6 +4946,10 @@
gst_task_stop (task);
+ g_mutex_lock (&sink->block_streams_lock);
+ g_cond_broadcast (&sink->block_streams_cond);
+ g_mutex_unlock (&sink->block_streams_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-1.18.3/gst-rtsp-server.doap
new/gst-rtsp-server-1.18.4/gst-rtsp-server.doap
--- old/gst-rtsp-server-1.18.3/gst-rtsp-server.doap 2021-01-13
22:12:06.000000000 +0100
+++ new/gst-rtsp-server-1.18.4/gst-rtsp-server.doap 2021-03-15
18:49:54.000000000 +0100
@@ -32,6 +32,16 @@
<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 -urN '--exclude=CVS' '--exclude=.cvsignore' '--exclude=.svn'
'--exclude=.svnignore' old/gst-rtsp-server-1.18.3/meson.build
new/gst-rtsp-server-1.18.4/meson.build
--- old/gst-rtsp-server-1.18.3/meson.build 2021-01-13 22:12:06.000000000
+0100
+++ new/gst-rtsp-server-1.18.4/meson.build 2021-03-15 18:49:54.000000000
+0100
@@ -1,5 +1,5 @@
project('gst-rtsp-server', 'c',
- version : '1.18.3',
+ version : '1.18.4',
meson_version : '>= 0.48',
default_options : ['warning_level=1', 'buildtype=debugoptimized'])
diff -urN '--exclude=CVS' '--exclude=.cvsignore' '--exclude=.svn'
'--exclude=.svnignore'
old/gst-rtsp-server-1.18.3/tests/check/gst/rtspclientsink.c
new/gst-rtsp-server-1.18.4/tests/check/gst/rtspclientsink.c
--- old/gst-rtsp-server-1.18.3/tests/check/gst/rtspclientsink.c 2021-01-13
22:12:06.000000000 +0100
+++ new/gst-rtsp-server-1.18.4/tests/check/gst/rtspclientsink.c 2021-03-15
18:49:54.000000000 +0100
@@ -214,6 +214,7 @@
gst_element_set_state (pipeline, GST_STATE_NULL);
gst_object_unref (pipeline);
+ gst_object_unref (bus);
}
iterate ();
@@ -227,8 +228,57 @@
g_signal_emit_by_name (G_OBJECT (server_sink), "pull-sample", &sample);
GST_INFO ("%2d recv sample: %p", i, sample);
- if (sample)
+ if (sample) {
gst_sample_unref (sample);
+ sample = NULL;
+ }
+ }
+ gst_object_unref (server_sink);
+
+ /* clean up and iterate so the clean-up can finish */
+ stop_server ();
+ iterate ();
+}
+
+GST_END_TEST;
+
+/* Make sure we can shut down rtspclientsink while it's still waiting for
+ * the initial preroll data */
+GST_START_TEST (test_record_no_data)
+{
+
+ start_record_server ("( rtppcmadepay name=depay0 ! fakesink )");
+
+ /* Create an rtspclientsink and send some data */
+ {
+ gchar *uri = get_server_uri (test_port, TEST_MOUNT_POINT);
+ gchar *pipe_str;
+ GstMessage *msg;
+ GstElement *pipeline;
+ GstBus *bus;
+
+ pipe_str = g_strdup_printf ("appsrc caps=audio/x-alaw,rate=8000,channels=1"
+ " ! rtspclientsink name=sink location=%s", uri);
+ g_free (uri);
+
+ pipeline = gst_parse_launch (pipe_str, NULL);
+ g_free (pipe_str);
+
+ fail_unless (pipeline != NULL);
+
+ bus = gst_element_get_bus (pipeline);
+ fail_if (bus == NULL);
+
+ gst_element_set_state (pipeline, GST_STATE_PLAYING);
+
+ /* wait for a bit */
+ msg = gst_bus_poll (bus, GST_MESSAGE_EOS | GST_MESSAGE_ERROR,
+ 500 * GST_MSECOND);
+ fail_unless (msg == NULL);
+
+ gst_element_set_state (pipeline, GST_STATE_NULL);
+ gst_object_unref (pipeline);
+ gst_object_unref (bus);
}
/* clean up and iterate so the clean-up can finish */
@@ -248,6 +298,7 @@
tcase_add_checked_fixture (tc, setup, teardown);
tcase_set_timeout (tc, 120);
tcase_add_test (tc, test_record);
+ tcase_add_test (tc, test_record_no_data);
return s;
}
