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here is the log from the commit of package gstreamer-plugins-base for 
openSUSE:Factory checked in at 2023-09-29 21:12:57
++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
Comparing /work/SRC/openSUSE:Factory/gstreamer-plugins-base (Old)
 and      /work/SRC/openSUSE:Factory/.gstreamer-plugins-base.new.28202 (New)
++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++

Package is "gstreamer-plugins-base"

Fri Sep 29 21:12:57 2023 rev:97 rq:1113994 version:1.22.6

Changes:
--------
--- 
/work/SRC/openSUSE:Factory/gstreamer-plugins-base/gstreamer-plugins-base.changes
    2023-09-04 22:53:15.243071506 +0200
+++ 
/work/SRC/openSUSE:Factory/.gstreamer-plugins-base.new.28202/gstreamer-plugins-base.changes
 2023-09-29 21:13:31.717650749 +0200
@@ -1,0 +2,14 @@
+Fri Sep 22 11:36:10 UTC 2023 - Bjørn Lie <bjorn....@gmail.com>
+
+- Update to version 1.22.6:
+  + audio: Make sure to stop ringbuffer on error
+  + decodebin3:
+    - Avoid identity, sinkpad, parsebin leakage when reset input
+    - Ensure the slot is unlinked before linking to decoder
+  + sdp:
+    - Fix wrong debug log error message for missing clock-rate in
+      caps
+    - Parse zero clock-rate as default
+- Rebase reduce-required-meson.patch
+
+-------------------------------------------------------------------

Old:
----
  gst-plugins-base-1.22.5.tar.xz

New:
----
  gst-plugins-base-1.22.6.tar.xz

++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++

Other differences:
------------------
++++++ gstreamer-plugins-base.spec ++++++
--- /var/tmp/diff_new_pack.ZIxCcS/_old  2023-09-29 21:13:32.973696067 +0200
+++ /var/tmp/diff_new_pack.ZIxCcS/_new  2023-09-29 21:13:32.977696212 +0200
@@ -20,7 +20,7 @@
 %define gst_branch 1.0
 %define gstreamer_req_version %(echo %{version} | sed -e "s/+.*//")
 Name:           gstreamer-plugins-base
-Version:        1.22.5
+Version:        1.22.6
 Release:        0
 Summary:        GStreamer Streaming-Media Framework Plug-Ins
 License:        GPL-2.0-or-later AND LGPL-2.1-or-later

++++++ gst-plugins-base-1.22.5.tar.xz -> gst-plugins-base-1.22.6.tar.xz ++++++
diff -urN '--exclude=CVS' '--exclude=.cvsignore' '--exclude=.svn' 
'--exclude=.svnignore' old/gst-plugins-base-1.22.5/ChangeLog 
new/gst-plugins-base-1.22.6/ChangeLog
--- old/gst-plugins-base-1.22.5/ChangeLog       2023-07-20 16:23:40.389558000 
+0200
+++ new/gst-plugins-base-1.22.6/ChangeLog       2023-09-20 19:12:48.543076300 
+0200
@@ -1,3 +1,75 @@
+=== release 1.22.6 ===
+
+2023-09-20 18:10:57 +0100  Tim-Philipp Müller <t...@centricular.com>
+
+       * NEWS:
+       * RELEASE:
+       * gst-plugins-base.doap:
+       * meson.build:
+         Release 1.22.6
+
+2023-08-26 15:12:05 +0200  Stephan Seitz <sse...@nvidia.com>
+
+       * gst-libs/gst/sdp/gstsdpmessage.c:
+         sdp: fix wrong error message for missing clock-rate in caps
+         When using `gst_sdp_media_set_media_from_caps` on `application/x-rtp` 
caps
+         without `clock-rate` it wrongly reports missing payload type even if 
`payload`
+         is present in the caps.
+         This seems to be a copy&paste error from the error message for 
missing payload
+         type.
+         When using payload=10, both `clock-rate` and some other media 
properties are
+         defined by the RTP standard so I was wondering whether I could omit 
`clock-rate`
+         and was confused about the error message.
+         Part-of: 
<https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5252>
+
+2023-08-19 01:00:16 +1000  Jan Schmidt <j...@centricular.com>
+
+       * gst-libs/gst/audio/gstaudiobasesrc.c:
+       * gst-libs/gst/audio/gstaudioringbuffer.c:
+         audio: Make sure to stop ringbuffer on error
+         Add gst_audio_ring_buffer_set_errored() that will mark the
+         ringbuffer as errored only if it is currently started or paused,
+         so gst_audio_ringbuffer_stop() can be sure that the error
+         state means that the ringbuffer was started and needs stop called.
+         Fixes a crash with osxaudiosrc if the source element posts
+         an error, because the ringbuffer would not get stopped and CoreAudio
+         would continue trying to do callbacks.
+         Also, anywhere that modifies the ringbuffer state, make sure to
+         use atomic operations, to guarantee their visibility
+         Part-of: 
<https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5216>
+
+2023-08-09 17:35:39 -0400  Olivier Crête <olivier.cr...@collabora.com>
+
+       * gst-libs/gst/sdp/gstsdpmessage.c:
+         sdpmessage: Parse zero clock-rate as default
+         It seems there is at least one broken RTSP server out there that 
returns a clock-rate of 0.
+         Let's just ignore it and use the default in that case.
+         Part-of: 
<https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5199>
+
+2023-08-01 15:14:29 +0100  Philippe Normand <ph...@igalia.com>
+
+       * gst/playback/gstdecodebin3.c:
+         decodebin3: Ensure the slot is unlinked before linking to decoder
+         When switching from a raw stream to an encoded stream we need to make 
sure the
+         slot is unlinked, there is code in place for this but it wasn't 
triggered
+         because the slot being reconfigured wasn't advertised as linked 
beforehand.
+         Part-of: 
<https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5133>
+
+2023-05-18 16:08:03 +0800  Haihua Hu <jared...@nxp.com>
+
+       * gst/playback/gstdecodebin3.c:
+         decodebin3: avoid identity, sinkpad, parsebin leakage when reset input
+         when reset_input, need remove identity/parsebin from decodebin3
+         when release_pad, need call free or reset input if collection
+         didn't change
+         Part-of: 
<https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5086>
+
+2023-07-20 16:57:47 +0100  Tim-Philipp Müller <t...@centricular.com>
+
+       * meson.build:
+         Back to development
+         Part-of: 
<https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5082>
+
 === release 1.22.5 ===
 
 2023-07-20 15:22:48 +0100  Tim-Philipp Müller <t...@centricular.com>
diff -urN '--exclude=CVS' '--exclude=.cvsignore' '--exclude=.svn' 
'--exclude=.svnignore' old/gst-plugins-base-1.22.5/NEWS 
new/gst-plugins-base-1.22.6/NEWS
--- old/gst-plugins-base-1.22.5/NEWS    2023-07-20 16:22:48.000000000 +0200
+++ new/gst-plugins-base-1.22.6/NEWS    2023-09-20 19:10:57.000000000 +0200
@@ -2189,6 +2189,183 @@
 -   List of Merge Requests applied in 1.22.5
 -   List of Issues fixed in 1.22.5
 
+1.22.6
+
+The sixth 1.22 bug-fix release (1.22.6) was released on 20 September
+2023.
+
+This release only contains bugfixes and security fixes and it should be
+safe to update from 1.22.x.
+
+Highlighted bugfixes in 1.22.6
+
+-   Security fixes for the MXF demuxer and H.265 video parser
+-   Fix latency regression in H.264 hardware decoder base class
+-   androidmedia: fix HEVC codec profile registration and fix coded_data
+    handling
+-   decodebin3: fix switching from a raw stream to an encoded stream
+-   gst-inspect: prettier and more correct signal and action signals
+    printing
+-   rtmp2: Allow NULL flash version, omitting the field, for better RTMP
+    server compatibility
+-   rtspsrc: better compatibility with buggy RTSP servers that don’t set
+    a clock-rate
+-   rtpjitterbuffer: fix integer overflow that led to more packets being
+    declared lost than have been lost
+-   v4l2: fix video encoding regression on RPi and fix support for left
+    and top padding
+-   waylandsink: Crop surfaces to their display width height
+-   cerbero: recognise Manjaro; add Rust support for MSVC ARM64; cmake
+    detection fixes
+-   various bug fixes, build fixes, memory leak fixes, and other
+    stability and reliability improvements
+
+gstreamer
+
+-   gst-inspect: prettier and more correct signal printing, and print
+    action signals in g_signal_emit_by_name() format
+-   gst-launch: Disable fault signal handlers on macOS
+
+gst-plugins-base
+
+-   audio: Make sure to stop ringbuffer on error
+-   decodebin3: avoid identity, sinkpad, parsebin leakage when reset
+    input
+-   decodebin3: Ensure the slot is unlinked before linking to decoder
+-   sdp: fix wrong debug log error message for missing clock-rate in
+    caps
+-   sdp: Parse zero clock-rate as default
+
+gst-plugins-good
+
+-   adaptivedemux2: fix memory leak
+-   pulsedeviceprovider: fix incorrect usage of GST_ELEMENT_ERROR
+-   qt: Unbreak build with qt-egl enabled but viv_fb missing
+-   qt: Fix searching of qt5/qt6 tools with qmake in Meson
+-   qtdemux: Fix premature EOS when some files are played in push mode
+-   qtdemux: attach cbcs crypt info at the right moment
+-   rtpjitterbuffer: Avoid integer overflow in max saveable packets
+    calculation with negative offset
+-   videoflip: fix concurrent access when modifying the tag list
+-   v4l2: allocator: Don’t close foreign dmabuf
+-   v4l2: bufferpool: Fix large encoded stream regression
+-   v4l2: bufferpool: Problems when checking for truncated buffer
+-   v4l2: Fix support for left and top padding
+-   v4l2object: clear format lists if source change event is received
+
+gst-plugins-bad
+
+-   androidmedia/enc: handle codec-data before popping
+    GstVideoCodecFrames
+-   androidmedia: fix hevc codec profile registration
+-   androidmedia: Small fixes
+-   androidmedia: Add more null checks (of env) to JNI utilities
+-   applemedia: Fix pixel format for I420 and NV12
+-   audiolatency: Forward latency query and event upstream
+-   av1parser: Fix segmentation params update
+-   codecparsers: Fix MPEG-1 aspect ratio table
+-   d3d11convert: Passthrough allocation query on same caps
+-   h264decoder: Update latency dynamically
+-   h265parser: Allow partially broken hvcC data
+-   h265parser: Fix possible overflow using max_sub_layers_minus1
+-   hlssink2: Always use forward slash separator
+-   mdns: Fix a crash on context error
+-   mxfdemux: Fix integer overflow causing out of bounds writes when
+    handling invalid uncompressed video and check channels for AES3
+-   nvencoder: Fix negotiation error when interlace-mode is unspecified
+-   rtmp2: Allow NULL flash version, omitting the field
+-   rtmp2sink: fix crash if message conversion failed
+-   transcodebin: Fixes for upstream selectable support
+-   va: Fix in error logs functions mismatches
+-   waylandsink: Crop surfaces to their display width height
+-   waylandsink: Fix cropping for video with non-square aspect ratio
+-   webrtc: Fix docs for create-data-channel action signal
+-   win32ipc: Fix pipe handle leak
+
+gst-plugins-ugly
+
+-   No changes
+
+gst-plugins-rs
+
+-   fallbackswitch: locking/deadlock fixes
+-   onvifmetadataparse: Skip metadata frames with unrepresentable UTC
+    time
+-   transcriberbin: Configure audioresample in front of transcriber
+-   webrtcsink: Propagate GstContext messages
+-   webrtcsink: Add support for d3d11 memory and qsvh264enc
+-   webrtcsink: fix TWCC extension adding
+-   webrtcsink: don’t forget to setup encoders for discoveries
+-   webrtcsink: NVIDIA V4L2 encoders always require NVMM memory
+-   meson: Fix handling of optional deps, and don’t require Python 3.8
+
+gst-libav
+
+-   No changes
+
+gst-rtsp-server
+
+-   No changes
+
+gstreamer-vaapi
+
+-   No changes
+
+gstreamer-sharp
+
+-   No changes
+
+gst-omx
+
+-   No changes
+
+gst-python
+
+-   No changes
+
+gst-editing-services
+
+-   No changes
+
+gst-validate + gst-integration-testsuites
+
+-   gst-validate: Disable fault signal handlers on macOS
+
+gst-examples
+
+-   No changes
+
+Development build environment
+
+-   macos-bison: Update to 3.8.2 and add an ARM64 build
+-   wrap: update libpsl to 0.21.2
+
+Cerbero build tool and packaging changes in 1.22.6
+
+-   Add Rust support for MSVC ARM64
+-   Recognise PERL5LIB as a joinable Unix variable
+-   Recognise Manjaro as an Arch derivative
+-   Fix picking up cmake from build-tools
+
+Contributors to 1.22.6
+
+Akihiro Sagawa, Alicia Boya García, Guillaume Desmottes, Haihua Hu,
+Hugues Fruchet, Ivan Molodetskikh, Jan Alexander Steffens (heftig), Jan
+Schmidt, L. E. Segovia, Mathieu Duponchelle, Matthew Waters, Ming Qian,
+Nicolas Dufresne, Nirbheek Chauhan, Olivier Blin, Olivier Crête,
+Philippe Normand, Piotr Brzeziński, Robert Ayrapetyan, Ryan Pavlik,
+Sebastian Dröge, Seungha Yang, Stéphane Cerveau, Stephan Seitz, Thomas
+Schneider, Tim-Philipp Müller, Víctor Manuel Jáquez Leal, Wang Chuan,
+Xabier Rodriguez Calvar,
+
+… 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 1.22.6
+
+-   List of Merge Requests applied in 1.22.6
+-   List of Issues fixed in 1.22.6
+
 Schedule for 1.24
 
 Our next major feature release will be 1.24, and 1.23 will be the
diff -urN '--exclude=CVS' '--exclude=.cvsignore' '--exclude=.svn' 
'--exclude=.svnignore' old/gst-plugins-base-1.22.5/RELEASE 
new/gst-plugins-base-1.22.6/RELEASE
--- old/gst-plugins-base-1.22.5/RELEASE 2023-07-20 16:22:48.000000000 +0200
+++ new/gst-plugins-base-1.22.6/RELEASE 2023-09-20 19:10:57.000000000 +0200
@@ -1,4 +1,4 @@
-This is GStreamer gst-plugins-base 1.22.5.
+This is GStreamer gst-plugins-base 1.22.6.
 
 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-plugins-base-1.22.5/gst/playback/gstdecodebin3.c 
new/gst-plugins-base-1.22.6/gst/playback/gstdecodebin3.c
--- old/gst-plugins-base-1.22.5/gst/playback/gstdecodebin3.c    2023-07-20 
16:22:48.000000000 +0200
+++ new/gst-plugins-base-1.22.6/gst/playback/gstdecodebin3.c    2023-09-20 
19:10:57.000000000 +0200
@@ -1127,37 +1127,17 @@
   gst_element_post_message (GST_ELEMENT_CAST (dbin), msg);
   update_requested_selection (dbin);
 
-  gst_ghost_pad_set_target (GST_GHOST_PAD (input->ghost_sink), NULL);
   if (input->parsebin) {
-    gst_bin_remove (GST_BIN (dbin), input->parsebin);
-    gst_element_set_state (input->parsebin, GST_STATE_NULL);
-    g_signal_handler_disconnect (input->parsebin, input->pad_removed_sigid);
-    g_signal_handler_disconnect (input->parsebin, input->pad_added_sigid);
-    g_signal_handler_disconnect (input->parsebin, input->drained_sigid);
     gst_pad_remove_probe (input->parsebin_sink, probe_id);
-    gst_object_unref (input->parsebin);
-    gst_object_unref (input->parsebin_sink);
-
-    input->parsebin = NULL;
-    input->parsebin_sink = NULL;
-  }
-  if (input->identity) {
-    GstPad *idpad = gst_element_get_static_pad (input->identity, "src");
-    DecodebinInputStream *stream = find_input_stream_for_pad (dbin, idpad);
-    gst_object_unref (idpad);
-    remove_input_stream (dbin, stream);
-    gst_element_set_state (input->identity, GST_STATE_NULL);
-    gst_bin_remove (GST_BIN (dbin), input->identity);
-    gst_object_unref (input->identity);
-    input->identity = NULL;
   }
 
+beach:
   if (!input->is_main) {
     dbin->other_inputs = g_list_remove (dbin->other_inputs, input);
     free_input (dbin, input);
-  }
+  } else
+    reset_input (dbin, input);
 
-beach:
   INPUT_UNLOCK (dbin);
   return;
 }
@@ -1175,6 +1155,7 @@
     g_signal_handler_disconnect (input->parsebin, input->pad_added_sigid);
     g_signal_handler_disconnect (input->parsebin, input->drained_sigid);
     gst_element_set_state (input->parsebin, GST_STATE_NULL);
+    gst_bin_remove (GST_BIN (dbin), input->parsebin);
     gst_clear_object (&input->parsebin);
     gst_clear_object (&input->parsebin_sink);
   }
@@ -1184,6 +1165,7 @@
     gst_object_unref (idpad);
     remove_input_stream (dbin, stream);
     gst_element_set_state (input->identity, GST_STATE_NULL);
+    gst_bin_remove (GST_BIN (dbin), input->identity);
     gst_clear_object (&input->identity);
   }
   if (input->collection)
@@ -2927,6 +2909,9 @@
     ret = FALSE;
     goto cleanup;
   }
+
+  output->linked = TRUE;
+
   if (output->src_exposed == FALSE) {
     GstEvent *stream_start;
 
diff -urN '--exclude=CVS' '--exclude=.cvsignore' '--exclude=.svn' 
'--exclude=.svnignore' 
old/gst-plugins-base-1.22.5/gst-libs/gst/audio/gstaudiobasesrc.c 
new/gst-plugins-base-1.22.6/gst-libs/gst/audio/gstaudiobasesrc.c
--- old/gst-plugins-base-1.22.5/gst-libs/gst/audio/gstaudiobasesrc.c    
2023-07-20 16:22:48.000000000 +0200
+++ new/gst-plugins-base-1.22.6/gst-libs/gst/audio/gstaudiobasesrc.c    
2023-09-20 19:10:57.000000000 +0200
@@ -45,6 +45,10 @@
 GST_DEBUG_CATEGORY_STATIC (gst_audio_base_src_debug);
 #define GST_CAT_DEFAULT gst_audio_base_src_debug
 
+/* This function is public in >= 1.23, but internal in 1.22 */
+G_GNUC_INTERNAL
+    void __gst_audio_ring_buffer_set_errored (GstAudioRingBuffer * buf);
+
 struct _GstAudioBaseSrcPrivate
 {
   /* the clock slaving algorithm in use */
@@ -1229,7 +1233,7 @@
      * flow error message */
     ret = GST_ELEMENT_CLASS (parent_class)->post_message (element, message);
 
-    g_atomic_int_set (&ringbuffer->state, GST_AUDIO_RING_BUFFER_STATE_ERROR);
+    __gst_audio_ring_buffer_set_errored (ringbuffer);
     GST_AUDIO_RING_BUFFER_SIGNAL (ringbuffer);
     gst_object_unref (ringbuffer);
   } else {
diff -urN '--exclude=CVS' '--exclude=.cvsignore' '--exclude=.svn' 
'--exclude=.svnignore' 
old/gst-plugins-base-1.22.5/gst-libs/gst/audio/gstaudioringbuffer.c 
new/gst-plugins-base-1.22.6/gst-libs/gst/audio/gstaudioringbuffer.c
--- old/gst-plugins-base-1.22.5/gst-libs/gst/audio/gstaudioringbuffer.c 
2023-07-20 16:22:48.000000000 +0200
+++ new/gst-plugins-base-1.22.6/gst-libs/gst/audio/gstaudioringbuffer.c 
2023-09-20 19:10:57.000000000 +0200
@@ -81,7 +81,7 @@
 {
   ringbuffer->open = FALSE;
   ringbuffer->acquired = FALSE;
-  ringbuffer->state = GST_AUDIO_RING_BUFFER_STATE_STOPPED;
+  g_atomic_int_set (&ringbuffer->state, GST_AUDIO_RING_BUFFER_STATE_STOPPED);
   g_cond_init (&ringbuffer->cond);
   ringbuffer->waiting = 0;
   ringbuffer->empty_seg = NULL;
@@ -1005,7 +1005,7 @@
   }
 
   if (G_UNLIKELY (!res)) {
-    buf->state = GST_AUDIO_RING_BUFFER_STATE_PAUSED;
+    g_atomic_int_set (&buf->state, GST_AUDIO_RING_BUFFER_STATE_PAUSED);
     GST_DEBUG_OBJECT (buf, "failed to start");
   } else {
     GST_DEBUG_OBJECT (buf, "started");
@@ -1036,6 +1036,40 @@
   }
 }
 
+G_GNUC_INTERNAL
+    void __gst_audio_ring_buffer_set_errored (GstAudioRingBuffer * buf);
+
+/* __gst_audio_ring_buffer_set_errored:
+ * @buf: the #GstAudioRingBuffer that has encountered an error
+ *
+ * Mark the ringbuffer as errored after it has started.
+ *
+ * MT safe.
+
+ * Since: 1.24 (internal in 1.22)
+ */
+void
+__gst_audio_ring_buffer_set_errored (GstAudioRingBuffer * buf)
+{
+  gboolean res;
+
+  /* If started set to errored */
+  res = g_atomic_int_compare_and_exchange (&buf->state,
+      GST_AUDIO_RING_BUFFER_STATE_STARTED, GST_AUDIO_RING_BUFFER_STATE_ERROR);
+  if (!res) {
+    GST_DEBUG_OBJECT (buf, "ringbuffer was not started, checking paused");
+    res = g_atomic_int_compare_and_exchange (&buf->state,
+        GST_AUDIO_RING_BUFFER_STATE_PAUSED, GST_AUDIO_RING_BUFFER_STATE_ERROR);
+  }
+  if (res) {
+    GST_DEBUG_OBJECT (buf, "ringbuffer is errored");
+  } else {
+    GST_DEBUG_OBJECT (buf,
+        "Could not mark ringbuffer as errored. It must have been stopped or 
already errored (was state %d)",
+        g_atomic_int_get (&buf->state));
+  }
+}
+
 static gboolean
 gst_audio_ring_buffer_pause_unlocked (GstAudioRingBuffer * buf)
 {
@@ -1060,7 +1094,8 @@
     res = rclass->pause (buf);
 
   if (G_UNLIKELY (!res)) {
-    buf->state = GST_AUDIO_RING_BUFFER_STATE_STARTED;
+    /* Restore started state */
+    g_atomic_int_set (&buf->state, GST_AUDIO_RING_BUFFER_STATE_STARTED);
     GST_DEBUG_OBJECT (buf, "failed to pause");
   } else {
     GST_DEBUG_OBJECT (buf, "paused");
@@ -1071,7 +1106,7 @@
 not_started:
   {
     /* was not started */
-    GST_DEBUG_OBJECT (buf, "was not started");
+    GST_DEBUG_OBJECT (buf, "was not started (state %d)", buf->state);
     return TRUE;
   }
 }
@@ -1153,9 +1188,16 @@
         GST_AUDIO_RING_BUFFER_STATE_PAUSED,
         GST_AUDIO_RING_BUFFER_STATE_STOPPED);
     if (!res) {
-      /* was not paused either, must have been stopped then */
+      GST_DEBUG_OBJECT (buf, "was not paused, try errored");
+      res = g_atomic_int_compare_and_exchange (&buf->state,
+          GST_AUDIO_RING_BUFFER_STATE_ERROR,
+          GST_AUDIO_RING_BUFFER_STATE_STOPPED);
+    }
+    if (!res) {
+      /* was not paused or stopped either, must have been stopped then */
       res = TRUE;
-      GST_DEBUG_OBJECT (buf, "was not paused, must have been stopped");
+      GST_DEBUG_OBJECT (buf,
+          "was not paused or errored, must have been stopped");
       goto done;
     }
   }
@@ -1169,7 +1211,7 @@
     res = rclass->stop (buf);
 
   if (G_UNLIKELY (!res)) {
-    buf->state = GST_AUDIO_RING_BUFFER_STATE_STARTED;
+    g_atomic_int_set (&buf->state, GST_AUDIO_RING_BUFFER_STATE_STARTED);
     GST_DEBUG_OBJECT (buf, "failed to stop");
   } else {
     GST_DEBUG_OBJECT (buf, "stopped");
diff -urN '--exclude=CVS' '--exclude=.cvsignore' '--exclude=.svn' 
'--exclude=.svnignore' 
old/gst-plugins-base-1.22.5/gst-libs/gst/sdp/gstsdpmessage.c 
new/gst-plugins-base-1.22.6/gst-libs/gst/sdp/gstsdpmessage.c
--- old/gst-plugins-base-1.22.5/gst-libs/gst/sdp/gstsdpmessage.c        
2023-07-20 16:22:48.000000000 +0200
+++ new/gst-plugins-base-1.22.6/gst-libs/gst/sdp/gstsdpmessage.c        
2023-09-20 19:10:57.000000000 +0200
@@ -3614,7 +3614,9 @@
 
   /* check if we have a rate, if not, we need to look up the rate from the
    * default rates based on the payload types. */
-  if (rate == -1) {
+  /* Some broken RTSP server puts a rate of 0, also use the default in that
+   * case */
+  if (rate <= 0) {
     const GstRTPPayloadInfo *info;
 
     if (GST_RTP_PAYLOAD_IS_DYNAMIC (pt)) {
@@ -3822,7 +3824,7 @@
 
   /* get clock-rate, media type and params for the rtpmap attribute */
   if (!gst_structure_get_int (s, "clock-rate", &caps_rate)) {
-    GST_ERROR ("ignoring stream without payload type");
+    GST_ERROR ("ignoring stream without clock rate");
     goto error;
   }
   caps_enc = gst_structure_get_string (s, "encoding-name");
diff -urN '--exclude=CVS' '--exclude=.cvsignore' '--exclude=.svn' 
'--exclude=.svnignore' old/gst-plugins-base-1.22.5/gst-plugins-base.doap 
new/gst-plugins-base-1.22.6/gst-plugins-base.doap
--- old/gst-plugins-base-1.22.5/gst-plugins-base.doap   2023-07-20 
16:22:48.000000000 +0200
+++ new/gst-plugins-base-1.22.6/gst-plugins-base.doap   2023-09-20 
19:10:57.000000000 +0200
@@ -36,6 +36,16 @@
 
  <release>
   <Version>
+   <revision>1.22.6</revision>
+   <branch>1.22</branch>
+   <name></name>
+   <created>2023-09-20</created>
+   <file-release 
rdf:resource="https://gstreamer.freedesktop.org/src/gst-plugins-base/gst-plugins-base-1.22.6.tar.xz";
 />
+  </Version>
+ </release>
+
+ <release>
+  <Version>
    <revision>1.22.5</revision>
    <branch>1.22</branch>
    <name></name>
diff -urN '--exclude=CVS' '--exclude=.cvsignore' '--exclude=.svn' 
'--exclude=.svnignore' old/gst-plugins-base-1.22.5/meson.build 
new/gst-plugins-base-1.22.6/meson.build
--- old/gst-plugins-base-1.22.5/meson.build     2023-07-20 16:22:48.000000000 
+0200
+++ new/gst-plugins-base-1.22.6/meson.build     2023-09-20 19:10:57.000000000 
+0200
@@ -1,5 +1,5 @@
 project('gst-plugins-base', 'c',
-  version : '1.22.5',
+  version : '1.22.6',
   meson_version : '>= 0.62',
   default_options : [ 'warning_level=1',
                       'buildtype=debugoptimized' ])
diff -urN '--exclude=CVS' '--exclude=.cvsignore' '--exclude=.svn' 
'--exclude=.svnignore' old/gst-plugins-base-1.22.5/po/gst-plugins-base-1.0.pot 
new/gst-plugins-base-1.22.6/po/gst-plugins-base-1.0.pot
--- old/gst-plugins-base-1.22.5/po/gst-plugins-base-1.0.pot     2023-07-20 
16:23:40.065552700 +0200
+++ new/gst-plugins-base-1.22.6/po/gst-plugins-base-1.0.pot     2023-09-20 
19:12:48.075071000 +0200
@@ -6,9 +6,9 @@
 #, fuzzy
 msgid ""
 msgstr ""
-"Project-Id-Version: gst-plugins-base-1.22.5\n"
+"Project-Id-Version: gst-plugins-base-1.22.6\n"
 "Report-Msgid-Bugs-To: \n"
-"POT-Creation-Date: 2023-07-20 15:23+0100\n"
+"POT-Creation-Date: 2023-09-20 18:12+0100\n"
 "PO-Revision-Date: YEAR-MO-DA HO:MI+ZONE\n"
 "Last-Translator: FULL NAME <EMAIL@ADDRESS>\n"
 "Language-Team: LANGUAGE <l...@li.org>\n"
@@ -209,7 +209,7 @@
 msgid "Error while sending data to \"%s:%d\"."
 msgstr ""
 
-#: gst-libs/gst/audio/gstaudiobasesrc.c:848
+#: gst-libs/gst/audio/gstaudiobasesrc.c:852
 msgid "Can't record audio fast enough"
 msgstr ""
 
diff -urN '--exclude=CVS' '--exclude=.cvsignore' '--exclude=.svn' 
'--exclude=.svnignore' old/gst-plugins-base-1.22.5/po/gst-plugins-base.pot 
new/gst-plugins-base-1.22.6/po/gst-plugins-base.pot
--- old/gst-plugins-base-1.22.5/po/gst-plugins-base.pot 2023-07-20 
16:23:40.065552700 +0200
+++ new/gst-plugins-base-1.22.6/po/gst-plugins-base.pot 2023-09-20 
19:12:48.075071000 +0200
@@ -6,9 +6,9 @@
 #, fuzzy
 msgid ""
 msgstr ""
-"Project-Id-Version: gst-plugins-base-1.22.5\n"
+"Project-Id-Version: gst-plugins-base-1.22.6\n"
 "Report-Msgid-Bugs-To: \n"
-"POT-Creation-Date: 2023-07-20 15:23+0100\n"
+"POT-Creation-Date: 2023-09-20 18:12+0100\n"
 "PO-Revision-Date: YEAR-MO-DA HO:MI+ZONE\n"
 "Last-Translator: FULL NAME <EMAIL@ADDRESS>\n"
 "Language-Team: LANGUAGE <l...@li.org>\n"
@@ -209,7 +209,7 @@
 msgid "Error while sending data to \"%s:%d\"."
 msgstr ""
 
-#: gst-libs/gst/audio/gstaudiobasesrc.c:848
+#: gst-libs/gst/audio/gstaudiobasesrc.c:852
 msgid "Can't record audio fast enough"
 msgstr ""
 

++++++ reduce-required-meson.patch ++++++
--- /var/tmp/diff_new_pack.ZIxCcS/_old  2023-09-29 21:13:33.741723778 +0200
+++ /var/tmp/diff_new_pack.ZIxCcS/_new  2023-09-29 21:13:33.745723922 +0200
@@ -1,10 +1,10 @@
-Index: gst-plugins-base-1.22.5/meson.build
+Index: gst-plugins-base-1.22.6/meson.build
 ===================================================================
---- gst-plugins-base-1.22.5.orig/meson.build
-+++ gst-plugins-base-1.22.5/meson.build
+--- gst-plugins-base-1.22.6.orig/meson.build
++++ gst-plugins-base-1.22.6/meson.build
 @@ -1,6 +1,6 @@
  project('gst-plugins-base', 'c',
-   version : '1.22.5',
+   version : '1.22.6',
 -  meson_version : '>= 0.62',
 +  meson_version : '>= 0.61',
    default_options : [ 'warning_level=1',

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