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gstreamer-plugins-bad-codecs.changes Changed

gstreamer-plugins-bad-codecs.spec Changed

gst-plugins-bad-1.24.7.tar.xz/ChangeLog -> gst-plugins-bad-1.24.8.tar.xz/ChangeLog Changed

@@ -1,3 +1,171 @@
+=== release 1.24.8 ===
+
+2024-09-19 12:01:21 +0200  Tim-Philipp Müller <tim@centricular.com>
+
+	* NEWS:
+	* RELEASE:
+	* gst-plugins-bad.doap:
+	* meson.build:
+	  Release 1.24.8
+
+2024-09-17 14:48:03 +0200  Benjamin Gaignard <benjamin.gaignard@collabora.com>
+
+	* sys/v4l2codecs/gstv4l2codech265dec.c:
+	  v4l2codecs: h265: Minimize memory allocation
+	  Be smarter when allocating sink and source memory pools to reduce the
+	  memory footprint. Use gst_v4l2_decoder_get_render_delay() to know the
+	  need number of buffers for downstream element.
+	  Handle errors in case of memory allocation failures.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7546>
+
+2024-09-17 18:31:30 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/mpegtsmux/gstbasetsmux.c:
+	  mpegtsmux: Fix refcounting issue when selecting the best pad
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7539>
+
+2024-09-11 08:40:42 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/mpegtsmux/gstbasetsmux.c:
+	  mpegtsmux: Wait for data on all pads before deciding on a best pad unless timing out
+	  This makes sure that if upstream has different latencies that we're still
+	  outputting buffers with increasining timestamps across the different streams
+	  unless buffers are arriving after the latency deadline.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7516>
+
+2024-09-05 22:07:24 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst-libs/gst/adaptivedemux/gstadaptivedemux.c:
+	  video: Don't overshoot QoS earliest time by a factor of 2
+	  By setting the earliest time to timestamp + 2 * diff there would be a difference
+	  of 1 * diff between the current clock time and the earliest time the element
+	  would let through in the future. If e.g. a frame is arriving 30s late at the
+	  sink, then not just all frames up to that point would be dropped but also 30s of
+	  frames after the current clock time.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7518>
+
+2024-08-30 14:59:14 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+	* gst-libs/gst/wayland/gstwldisplay.c:
+	* gst-libs/gst/wayland/gstwlshmallocator.c:
+	* gst-libs/gst/wayland/gstwlshmallocator.h:
+	  wayland: Set a debug category for the shm allocator
+	  None was set, which meant the debug was associated with default.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7494>
+
+2024-09-09 16:27:43 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+	* ext/wayland/gstwaylandsink.c:
+	* gst-libs/gst/wayland/gstwlcontext.c:
+	* gst-libs/gst/wayland/gstwlcontext.h:
+	  wayland: Fix ABI break in WL context type name
+	  While transforming the internals of waylandsink into a library, the
+	  context type name was accidentally changed, causing an ABI break. Change
+	  it back to its original (as used by the libgstgl), and add support for
+	  the misnamed version as a backward compatibility measure.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7494>
+
+2023-07-18 17:34:54 +0200  Michael Tretter <m.tretter@pengutronix.de>
+
+	* sys/uvcgadget/gstuvcsink.c:
+	  uvcsink: set cur_caps to upstream selected caps
+	  If the UVC gadget announces multiple formats in the descriptors the uvcsink
+	  doesn't select the actual format but let's the UVC hosts select the format.
+	  If the GStreamer pipeline is started before a UVC host selected the format,
+	  upstream decides on a format until the UVC host has decided. In this case, the
+	  current format needs to be set based on the caps from the caps event to be able
+	  to detect if the format selection by the UVC host requires a format change on
+	  the GStreamer pipeline.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7484>
+
+2024-09-05 15:04:33 +0200  Michael Tretter <m.tretter@pengutronix.de>
+
+	* sys/uvcgadget/gstuvcsink.c:
+	  uvcsink: skip comparison with prev_caps if they are not set
+	  The uvcsink may be put into the READY state to start listening for UVC requests.
+	  Therefore, the UVC host may set a streaming format before the GStreamer pipeline
+	  is started and the uvcsink received a caps event. In this case, prev_caps will
+	  be NULL.
+	  If the EVENT_CAPS has not been received, skip the check if the format needs to
+	  be changed, since the sink will be started with the format selected by the UVC
+	  host, anyway.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7484>
+
+2024-09-03 17:00:09 +0200  Piotr Brzeziński <piotr@centricular.com>
+
+	* sys/applemedia/vtenc.c:
+	* sys/applemedia/vtenc.h:
+	  vtenc: Restart encoding session when certain errors are detected
+	  Sometimes under certain loads, VT can error out with kVTVideoEncoderMalfunctionErr or kVTVideoEncoderNotAvailableNowErr.
+	  These have been reported to happen more often than usual if CopyProperty/SetProperty() is used close to the encode call.
+	  Both can be worked around by restarting the encoding session.
+	  These errors can be returned either directly from VTCompressionSessionEncodeFrame() or later in the encoding callback.
+	  This patch handles both scenarios the same way - a session restart is be attempted on the next encode_frame() call.
+	  If the error is returned immediately by the encode call, it's possible that some correct frames will still be given to
+	  the output callback, but for simplicity (+ because I wasn't able to verify this scenario) let's just discard those.
+	  In addition, this commit also simplifies the beach/drop logic in enqueue_buffer.
+	  Related bug reports in other projects:
+	  http://www.openradar.me/45889262
+	  https://github.com/aws/amazon-chime-sdk-ios/issues/170#issuecomment-741908622
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7449>
+
+2024-09-06 10:51:01 +0200  Edward Hervey <edward@centricular.com>
+
+	* gst-libs/gst/play/gstplay.c:
+	  gstplay: Name the different bus
+	  Makes it clearer when reading logs which one is which
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7471>
+
+2024-09-02 18:25:56 +0900  Hou Qi <qi.hou@nxp.com>
+
+	* gst-libs/gst/play/gstplay.c:
+	  gstplay: check whether stream is seekable before seeking when state change
+	  If state is changing from playing to paused, and rate is reset to 1
+	  which causes seek position is valid, current code will do seek for
+	  streams that are not seekable. So need to check whether stream is
+	  seekable before seeking.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7448>
+
+2024-08-29 12:01:30 +0100  Philippe Normand <philn@igalia.com>
+
+	* ext/webrtc/gstwebrtcbin.c:
+	* tests/check/elements/webrtcbin.c:
+	  webrtcbin: Prevent crash when attempting to set answer on invalid SDP
+	  If the pending remote description has an invalid BUNDLE group _parse_bundle()
+	  triggers early return from _create_answer_task(), before ret has been
+	  initialized, so it needs to be checked before attempting to call
+	  gst_sdp_message_copy().
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7436>
+
+2024-08-21 13:23:36 +0100  Francis Quiers <fquiers@cisco.com>
+
+	* docs/plugins/gst_plugins_cache.json:
+	* ext/voamrwbenc/gstvoamrwbenc.c:
+	  voamrwbenc: fix list of bitrates
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7418>
+
+2024-08-20 22:09:13 +1000  Jan Schmidt <jan@centricular.com>
+
+	* gst-libs/gst/player/gstplayer.c:
+	  gstplayer: Check GstPlayerSignalDispatcher type
+	  Before trying to retrieve a GMainContext from a provided
+	  GstPlayerSignalDispatcher, check that it is actually
+	  GstPlayerGMainContextSignalDispatcher. If not, use the
+	  default GMainContext for dispatching signals via the adapter
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7398>
+
+2024-08-21 09:19:39 +0200  Guillaume Desmottes <guillaume.desmottes@onestream.live>
+
+	* ext/wpe/gstwpesrcbin.cpp:
+	  wpe: fix gst-launch example
+	  wpesrc does not have num-buffers property but wpevideosrc does.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7397>
+
+2024-08-21 12:33:28 +0100  Tim-Philipp Müller <tim@centricular.com>
+
+	* meson.build:
+	  Back to development after 1.24.7
+
 === release 1.24.7 ===
 
 2024-08-21 12:25:15 +0100  Tim-Philipp Müller <tim@centricular.com>

gst-plugins-bad-1.24.7.tar.xz/NEWS -> gst-plugins-bad-1.24.8.tar.xz/NEWS Changed

@@ -2,11 +2,11 @@
 
 GStreamer 1.24.0 was originally released on 4 March 2024.
 
-The latest bug-fix release in the stable 1.24 series is 1.24.6 and was released on 21 August 2024.
+The latest bug-fix release in the stable 1.24 series is 1.24.8 and was released on 19 September 2024.
 
 See https://gstreamer.freedesktop.org/releases/1.24/ for the latest version of this document.
 
-Last updated: Tuesday 20 August 2024, 20:30 UTC (log)
+Last updated: Thursday 19 September 2024, 00:30 UTC (log)
 
 ## Introduction
 
@@ -2416,6 +2416,137 @@
 -   List of Merge Requests applied in 1.24.7
 -   List of Issues fixed in 1.24.7
 
+1.24.8
+
+The eigth 1.24 bug-fix release (1.24.8) was released on 19 September 2024.
+
+This release only contains bugfixes and it should be safe to update from 1.24.x.
+
+Highlighted bugfixes in 1.24.8
+
+-   decodebin3: collection handling fixes
+-   encodebin: Fix pad removal (and smart rendering in gst-editing-services)
+-   glimagesink: Fix cannot resize viewport when video size changed in caps
+-   matroskamux, webmmux: fix firefox compatibility issue with Opus audio streams
+-   mpegtsmux: Wait for data on all pads before deciding on a best pad unless timing out
+-   splitmuxsink: Override LATENCY query to pretend to downstream that we’re not live
+-   video: QoS event handling improvements
+-   voamrwbenc: fix list of bitrates
+-   vtenc: Restart encoding session when certain errors are detected
+-   wayland: Fix ABI break in WL context type name
+-   webrtcbin: Prevent crash when attempting to set answer on invalid SDP
+-   cerbero: ship vp8/vp9 software encoders again, which went missing in 1.24.7; ship transcode plugin
+-   Various bug fixes, memory leak fixes, and other stability and reliability improvements
+
+gstreamer
+
+-   clock: Fix unchecked overflows in linear regression code
+-   meta: Add missing include of gststructure.h
+-   pad: Check data NULL-ness when probes are stopped
+-   aggregator: Immediately return NONE from simple_get_next_time() on non-TIME segments
+
+gst-plugins-base
+
+-   decodebin3: Fix collection identity check
+-   encodebin: Fix pad removal
+-   glimagesink: Fix cannot resize viewport when video size changed in caps
+-   v4l2bufferpool: actually queue back the empty buffer flagged LAST
+-   v4l2videoenc: unref buffer pool after usage properly
+-   video: Don’t overshoot QoS earliest time by a factor of 2
+-   meson: gst-play: link to libm
+
+gst-plugins-good
+
+-   jackaudiosrc: actually use the queried ports from JACK
+-   matroskamux: Include end padding in the block duration for Opus streams, fixing firefox compatibility
+-   osxaudio: Avoid dangling pointer on shutdown
+-   splitmuxsink: Override LATENCY query to pretend to downstream that we’re not live
+-   v4l2: encoder: Add dynamic framerate support
+
+gst-plugins-bad
+
+-   GstPlay: Name the different bus
+-   GstPlay: check whether stream is seekable before seeking when state change
+-   GstPlayer: Check GstPlayerSignalDispatcher type
+-   mpegtsmux: Wait for data on all pads before deciding on a best pad unless timing out
+-   mpegtsmux: Fix refcounting issue when selecting the best pad
+-   uvcsink: fix caps event handling
+-   v4l2codecs: h265: Minimize memory allocation
+-   voamrwbenc: fix list of bitrates
+-   vtenc: Restart encoding session when certain errors are detected
+-   wayland: Fix ABI break in WL context type name
+-   webrtcbin: Prevent crash when attempting to set answer on invalid SDP
+-   wpe: fix gst-launch example
+
+gst-plugins-ugly
+
+-   No changes
+
+GStreamer Rust plugins
+
+-   No changes
+
+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
+
+-   discoverer-manager: Fix race leading to assertion when stopping
+-   structured-interface: Fix memory leak of invalid fields GList
+
+gst-devtools, gst-validate + gst-integration-testsuites
+
+-   pad-monitor: Fix remaining pad function data handling
+-   pad-monitor: Fix pad function data properly
+
+gst-examples
+
+-   No changes
+
+Development build environment
+
+-   meson: Update openjpeg wrap to 2.5.2, fixes a warning
+
+Cerbero build tool and packaging changes in 1.24.8
+
+-   No vp8 / vp9 encoders packaged (regression)
+-   libvpx: Fix codec detection to fix vp8enc/vp9enc elements not being shipped
+-   gst-plugins-bad: Add missing transcode plugin
+
+Contributors to 1.24.8
+
+Andoni Morales Alastruey, Arun Raghavan, Benjamin Gaignard, Carlos Bentzen, Chao Guo, Edward Hervey, Francis Quiers, Guillaume
+Desmottes, Hou Qi, Jan Schmidt,, L. E. Segovia, Michael Tretter, Nicolas Dufresne, Nirbheek Chauhan, Peter Kjellerstedt,
+Philippe Normand, Piotr Brzeziński, Randy Li (ayaka), Sebastian Dröge, Thibault Saunier, Tim-Philipp Müller, Wim Taymans,
+
+… 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 1.24.8
+
+-   List of Merge Requests applied in 1.24.8
+-   List of Issues fixed in 1.24.8
+
 Schedule for 1.26
 
 Our next major feature release will be 1.26, and 1.25 will be the unstable development version leading up to the stable 1.26

gst-plugins-bad-1.24.7.tar.xz/RELEASE -> gst-plugins-bad-1.24.8.tar.xz/RELEASE Changed

gst-plugins-bad-1.24.7.tar.xz/docs/plugins/gst_plugins_cache.json -> gst-plugins-bad-1.24.8.tar.xz/docs/plugins/gst_plugins_cache.json Changed

@@ -244388,37 +244388,37 @@
                     {
                         "desc": "MR1425",
                         "name": "MR1425",
-                        "value": "2"
+                        "value": "3"
                     },
                     {
                         "desc": "MR1585",
                         "name": "MR1585",
-                        "value": "3"
+                        "value": "4"
                     },
                     {
                         "desc": "MR1825",
                         "name": "MR1825",
-                        "value": "4"
+                        "value": "5"
                     },
                     {
                         "desc": "MR1985",
                         "name": "MR1985",
-                        "value": "5"
+                        "value": "6"
                     },
                     {
                         "desc": "MR2305",
                         "name": "MR2305",
-                        "value": "6"
+                        "value": "7"
                     },
                     {
                         "desc": "MR2385",
                         "name": "MR2385",
-                        "value": "7"
+                        "value": "8"
                     },
                     {
                         "desc": "MRDTX",
                         "name": "MRDTX",
-                        "value": "8"
+                        "value": "9"
                     }
                 
             }

gst-plugins-bad-1.24.7.tar.xz/ext/voamrwbenc/gstvoamrwbenc.c -> gst-plugins-bad-1.24.8.tar.xz/ext/voamrwbenc/gstvoamrwbenc.c Changed

gst-plugins-bad-1.24.7.tar.xz/ext/wayland/gstwaylandsink.c -> gst-plugins-bad-1.24.8.tar.xz/ext/wayland/gstwaylandsink.c Changed

@@ -365,35 +365,62 @@
 }
 
 static gboolean
-gst_wayland_sink_find_display (GstWaylandSink * self)
+gst_wayland_sink_query_context (GstWaylandSink * self, const gchar * type)
 {
   GstQuery *query;
+  gboolean ret;
+
+  query = gst_query_new_context (type);
+  ret = gst_pad_peer_query (GST_VIDEO_SINK_PAD (self), query);
+
+  if (ret) {
+    GstContext *context = NULL;
+    gst_query_parse_context (query, &context);
+    gst_wayland_sink_set_display_from_context (self, context);
+  }
+
+  gst_query_unref (query);
+  return ret;
+}
+
+static gboolean
+gst_wayland_sink_post_need_context (GstWaylandSink * self, const gchar * type)
+{
   GstMessage *msg;
-  GstContext *context = NULL;
+
+  /* now ask the application to set the display handle */
+  msg = gst_message_new_need_context (GST_OBJECT_CAST (self), type);
+
+  g_mutex_unlock (&self->display_lock);
+  gst_element_post_message (GST_ELEMENT_CAST (self), msg);
+  /* at this point we expect gst_wayland_sink_set_context
+   * to get called and fill self->display */
+  g_mutex_lock (&self->display_lock);
+
+  return self->display != NULL;
+}
+
+static gboolean
+gst_wayland_sink_find_display (GstWaylandSink * self)
+{
   GError *error = NULL;
   gboolean ret = TRUE;
 
   g_mutex_lock (&self->display_lock);
 
   if (!self->display) {
-    /* first query upstream for the needed display handle */
-    query = gst_query_new_context (GST_WL_DISPLAY_HANDLE_CONTEXT_TYPE);
-    if (gst_pad_peer_query (GST_VIDEO_SINK_PAD (self), query)) {
-      gst_query_parse_context (query, &context);
-      gst_wayland_sink_set_display_from_context (self, context);
+    if (!gst_wayland_sink_query_context (self,
+            GST_WL_DISPLAY_HANDLE_CONTEXT_TYPE)) {
+      gst_wayland_sink_query_context (self,
+          GST_WL_DISPLAY_HANDLE_LEGACY_CONTEXT_TYPE);
     }
-    gst_query_unref (query);
 
     if (G_LIKELY (!self->display)) {
-      /* now ask the application to set the display handle */
-      msg = gst_message_new_need_context (GST_OBJECT_CAST (self),
-          GST_WL_DISPLAY_HANDLE_CONTEXT_TYPE);
-
-      g_mutex_unlock (&self->display_lock);
-      gst_element_post_message (GST_ELEMENT_CAST (self), msg);
-      /* at this point we expect gst_wayland_sink_set_context
-       * to get called and fill self->display */
-      g_mutex_lock (&self->display_lock);
+      if (!gst_wayland_sink_post_need_context (self,
+              GST_WL_DISPLAY_HANDLE_CONTEXT_TYPE)) {
+        gst_wayland_sink_post_need_context (self,
+            GST_WL_DISPLAY_HANDLE_LEGACY_CONTEXT_TYPE);
+      }
 
       if (!self->display) {
         /* if the application didn't set a display, let's create it ourselves */
@@ -480,7 +507,9 @@
   GstWaylandSink *self = GST_WAYLAND_SINK (element);
 
   if (gst_context_has_context_type (context,
-          GST_WL_DISPLAY_HANDLE_CONTEXT_TYPE)) {
+          GST_WL_DISPLAY_HANDLE_CONTEXT_TYPE) ||
+      gst_context_has_context_type (context,
+          GST_WL_DISPLAY_HANDLE_LEGACY_CONTEXT_TYPE)) {
     g_mutex_lock (&self->display_lock);
     if (G_LIKELY (!self->display)) {
       gst_wayland_sink_set_display_from_context (self, context);

gst-plugins-bad-1.24.7.tar.xz/ext/webrtc/gstwebrtcbin.c -> gst-plugins-bad-1.24.8.tar.xz/ext/webrtc/gstwebrtcbin.c Changed

gst-plugins-bad-1.24.7.tar.xz/ext/wpe/gstwpesrcbin.cpp -> gst-plugins-bad-1.24.8.tar.xz/ext/wpe/gstwpesrcbin.cpp Changed

gst-plugins-bad-1.24.7.tar.xz/gst-libs/gst/adaptivedemux/gstadaptivedemux.c -> gst-plugins-bad-1.24.8.tar.xz/gst-libs/gst/adaptivedemux/gstadaptivedemux.c Changed

gst-plugins-bad-1.24.7.tar.xz/gst-libs/gst/play/gstplay.c -> gst-plugins-bad-1.24.8.tar.xz/gst-libs/gst/play/gstplay.c Changed

@@ -314,6 +314,7 @@
   self->context = g_main_context_new ();
   self->loop = g_main_loop_new (self->context, FALSE);
   self->api_bus = gst_bus_new ();
+  gst_object_set_name (GST_OBJECT (self->api_bus), "api_bus");
 
   /* *INDENT-OFF* */
   self->config = gst_structure_new_id (QUARK_CONFIG,
@@ -1422,8 +1423,12 @@
       }
 
       if (self->seek_position != GST_CLOCK_TIME_NONE) {
-        GST_DEBUG_OBJECT (self, "Seeking now that we reached PAUSED state");
-        gst_play_seek_internal_locked (self);
+        if (!self->media_info->seekable) {
+          GST_DEBUG_OBJECT (self, "Media is not seekable");
+        } else {
+          GST_DEBUG_OBJECT (self, "Seeking now that we reached PAUSED state");
+          gst_play_seek_internal_locked (self);
+        }
         g_mutex_unlock (&self->lock);
       } else if (!self->seek_pending) {
         g_mutex_unlock (&self->lock);
@@ -2627,6 +2632,7 @@
   }
 
   self->bus = bus = gst_element_get_bus (self->playbin);
+  gst_object_set_name (GST_OBJECT (self->bus), "playbin_bus");
   gst_bus_add_signal_watch (bus);
 
   g_signal_connect (G_OBJECT (bus), "message::error", G_CALLBACK (error_cb),

gst-plugins-bad-1.24.7.tar.xz/gst-libs/gst/player/gstplayer.c -> gst-plugins-bad-1.24.8.tar.xz/gst-libs/gst/player/gstplayer.c Changed

@@ -49,6 +49,7 @@
 
 #include "gstplayer.h"
 #include "gstplayer-signal-dispatcher-private.h"
+#include "gstplayer-g-main-context-signal-dispatcher.h"
 #include "gstplayer-video-renderer-private.h"
 #include "gstplayer-media-info-private.h"
 #include "gstplayer-wrapped-video-renderer-private.h"
@@ -576,11 +577,16 @@
   if (self->signal_dispatcher != NULL) {
     GMainContext *context = NULL;
 
-    g_object_get (self->signal_dispatcher, "application-context", &context,
-        NULL);
-    self->signal_adapter =
-        gst_play_signal_adapter_new_with_main_context (self->play, context);
-    g_main_context_unref (context);
+    if (GST_IS_PLAYER_G_MAIN_CONTEXT_SIGNAL_DISPATCHER
+        (self->signal_dispatcher)) {
+      g_object_get (self->signal_dispatcher, "application-context", &context,
+          NULL);
+      self->signal_adapter =
+          gst_play_signal_adapter_new_with_main_context (self->play, context);
+      g_main_context_unref (context);
+    } else {
+      self->signal_adapter = gst_play_signal_adapter_new (self->play);
+    }
   } else {
     self->signal_adapter = gst_play_signal_adapter_new_sync_emit (self->play);
   }

gst-plugins-bad-1.24.7.tar.xz/gst-libs/gst/wayland/gstwlcontext.c -> gst-plugins-bad-1.24.8.tar.xz/gst-libs/gst/wayland/gstwlcontext.c Changed

gst-plugins-bad-1.24.7.tar.xz/gst-libs/gst/wayland/gstwlcontext.h -> gst-plugins-bad-1.24.8.tar.xz/gst-libs/gst/wayland/gstwlcontext.h Changed

gst-plugins-bad-1.24.7.tar.xz/gst-libs/gst/wayland/gstwldisplay.c -> gst-plugins-bad-1.24.8.tar.xz/gst-libs/gst/wayland/gstwldisplay.c Changed

gst-plugins-bad-1.24.7.tar.xz/gst-libs/gst/wayland/gstwlshmallocator.c -> gst-plugins-bad-1.24.8.tar.xz/gst-libs/gst/wayland/gstwlshmallocator.c Changed

gst-plugins-bad-1.24.7.tar.xz/gst-libs/gst/wayland/gstwlshmallocator.h -> gst-plugins-bad-1.24.8.tar.xz/gst-libs/gst/wayland/gstwlshmallocator.h Changed

gst-plugins-bad-1.24.7.tar.xz/gst-plugins-bad.doap -> gst-plugins-bad-1.24.8.tar.xz/gst-plugins-bad.doap Changed

gst-plugins-bad-1.24.7.tar.xz/gst/mpegtsmux/gstbasetsmux.c -> gst-plugins-bad-1.24.8.tar.xz/gst/mpegtsmux/gstbasetsmux.c Changed

@@ -2449,7 +2449,7 @@
 }
 
 static GstBaseTsMuxPad *
-gst_base_ts_mux_find_best_pad (GstAggregator * aggregator)
+gst_base_ts_mux_find_best_pad (GstAggregator * aggregator, gboolean timeout)
 {
   GstBaseTsMuxPad *best = NULL;
   GstClockTime best_ts = GST_CLOCK_TIME_NONE;
@@ -2463,8 +2463,14 @@
     GstBuffer *buffer;
 
     buffer = gst_aggregator_pad_peek_buffer (apad);
-    if (!buffer)
+    if (!buffer) {
+      if (!timeout && !GST_PAD_IS_EOS (apad)) {
+        best = NULL;
+        best_ts = GST_CLOCK_TIME_NONE;
+        break;
+      }
       continue;
+    }
     if (best_ts == GST_CLOCK_TIME_NONE) {
       best = tpad;
       best_ts = GST_BUFFER_DTS_OR_PTS (buffer);
@@ -2518,7 +2524,7 @@
 {
   GstBaseTsMux *mux = GST_BASE_TS_MUX (agg);
   GstFlowReturn ret = GST_FLOW_OK;
-  GstBaseTsMuxPad *best = gst_base_ts_mux_find_best_pad (agg);
+  GstBaseTsMuxPad *best = gst_base_ts_mux_find_best_pad (agg, timeout);
   GstCaps *caps;
 
   /* set caps on the srcpad if no caps were set yet */

gst-plugins-bad-1.24.7.tar.xz/meson.build -> gst-plugins-bad-1.24.8.tar.xz/meson.build Changed

gst-plugins-bad-1.24.7.tar.xz/po/gst-plugins-bad-1.0.pot -> gst-plugins-bad-1.24.8.tar.xz/po/gst-plugins-bad-1.0.pot Changed

gst-plugins-bad-1.24.7.tar.xz/po/gst-plugins-bad.pot -> gst-plugins-bad-1.24.8.tar.xz/po/gst-plugins-bad.pot Changed

gst-plugins-bad-1.24.7.tar.xz/sys/applemedia/vtenc.c -> gst-plugins-bad-1.24.8.tar.xz/sys/applemedia/vtenc.c Changed

@@ -151,7 +151,7 @@
 
 static gboolean gst_vtenc_start (GstVideoEncoder * enc);
 static gboolean gst_vtenc_stop (GstVideoEncoder * enc);
-static void gst_vtenc_loop (GstVTEnc * self);
+static void gst_vtenc_output_loop (GstVTEnc * self);
 static gboolean gst_vtenc_set_format (GstVideoEncoder * enc,
     GstVideoCodecState * input_state);
 static GstFlowReturn gst_vtenc_handle_frame (GstVideoEncoder * enc,
@@ -780,6 +780,7 @@
 
   self->is_flushing = FALSE;
   self->downstream_ret = GST_FLOW_OK;
+  g_atomic_int_set (&self->require_restart, FALSE);
 
   self->output_queue = gst_queue_array_new (VTENC_OUTPUT_QUEUE_SIZE);
   /* Set clear_func to unref all remaining frames in gst_queue_array_free() */
@@ -789,7 +790,7 @@
   /* Create the output task, but pause it immediately */
   self->pause_task = TRUE;
   if (!gst_pad_start_task (GST_VIDEO_ENCODER_SRC_PAD (enc),
-          (GstTaskFunction) gst_vtenc_loop, self, NULL)) {
+          (GstTaskFunction) gst_vtenc_output_loop, self, NULL)) {
     GST_ERROR_OBJECT (self, "failed to start output thread");
     return FALSE;
   }
@@ -1545,6 +1546,7 @@
 static void
 gst_vtenc_destroy_session (GstVTEnc * self, VTCompressionSessionRef * session)
 {
+  GST_DEBUG_OBJECT (self, "Destroying VT session");
   VTCompressionSessionInvalidate (*session);
   if (*session != NULL) {
     CFRelease (*session);
@@ -1753,6 +1755,49 @@
   }
 }
 
+static Boolean
+gst_vtenc_is_recoverable_error (OSStatus status)
+{
+  return status == kVTVideoEncoderMalfunctionErr
+      || status == kVTVideoEncoderNotAvailableNowErr;
+}
+
+static void
+gst_vtenc_restart_session (GstVTEnc * self)
+{
+  OSStatus status;
+  VTCompressionSessionRef session;
+
+  /* We need to push out all frames still inside the encoder,
+   * otherwise destroy_session() will wait for all callbacks to fire
+   * and very likely deadlock due to the object lock being taken */
+  GST_VIDEO_ENCODER_STREAM_UNLOCK (self);
+  status = VTCompressionSessionCompleteFrames (self->session,
+      kCMTimePositiveInfinity);
+  if (status != noErr) {
+    GST_WARNING_OBJECT (self,
+        "Error when emptying encoder before restart: %d, will retry on next frame encode",
+        (int) status);
+    GST_VIDEO_ENCODER_STREAM_LOCK (self);
+    return;
+  } else {
+    GST_DEBUG_OBJECT (self, "All frames out, restarting encoder session");
+  }
+  GST_VIDEO_ENCODER_STREAM_LOCK (self);
+
+  GST_OBJECT_LOCK (self);
+  gst_vtenc_destroy_session (self, &self->session);
+  GST_OBJECT_UNLOCK (self);
+
+  session = gst_vtenc_create_session (self);
+
+  GST_OBJECT_LOCK (self);
+  self->session = session;
+  GST_OBJECT_UNLOCK (self);
+
+  g_atomic_int_set (&self->require_restart, FALSE);
+}
+
 static GstFlowReturn
 gst_vtenc_encode_frame (GstVTEnc * self, GstVideoCodecFrame * frame)
 {
@@ -1802,6 +1847,11 @@
     }
   }
 
+  /* Flushes all remaining frames out of the encoder
+   * and recreates the encoding session. */
+  if (g_atomic_int_get (&self->require_restart))
+    gst_vtenc_restart_session (self);
+
   if (GST_VIDEO_CODEC_FRAME_IS_FORCE_KEYFRAME (frame)) {
     GST_INFO_OBJECT (self, "received force-keyframe-event, will force intra");
     frame_props = self->keyframe_props;
@@ -1960,9 +2010,16 @@
       GINT_TO_POINTER (frame->system_frame_number), NULL);
   GST_VIDEO_ENCODER_STREAM_LOCK (self);
 
-  if (vt_status != noErr) {
-    GST_WARNING_OBJECT (self, "VTCompressionSessionEncodeFrame returned %d",
-        (int) vt_status);
+  if (gst_vtenc_is_recoverable_error (vt_status)) {
+    GST_ELEMENT_WARNING (self, LIBRARY, ENCODE, (NULL),
+        ("Failed to encode frame %d: %d, restarting session on next frame encode",
+            frame->system_frame_number, (int) vt_status));
+
+    g_atomic_int_set (&self->require_restart, TRUE);
+  } else if (vt_status != noErr) {
+    GST_ELEMENT_ERROR (self, LIBRARY, ENCODE, (NULL),
+        ("Failed to encode frame %d: %d", frame->system_frame_number,
+            (int) vt_status));
   }
 
   gst_video_codec_frame_unref (frame);
@@ -1991,14 +2048,23 @@
 {
   GstVTEnc *self = outputCallbackRefCon;
   GstVideoCodecFrame *frame;
-  gboolean is_flushing;
 
   frame =
       gst_video_encoder_get_frame (GST_VIDEO_ENCODER_CAST (self),
       GPOINTER_TO_INT (sourceFrameRefCon));
 
+  if (g_atomic_int_get (&self->require_restart)) {
+    GST_DEBUG_OBJECT (self, "Ignoring frame because of scheduled restart");
+    goto drop;
+  }
+
   if (status != noErr) {
-    if (frame) {
+    if (gst_vtenc_is_recoverable_error (status)) {
+      GST_ELEMENT_WARNING (self, LIBRARY, ENCODE, (NULL),
+          ("Failed to encode frame (%d), restarting session on next frame encode",
+              (int) status));
+      g_atomic_int_set (&self->require_restart, TRUE);
+    } else if (frame) {
       GST_ELEMENT_ERROR (self, LIBRARY, ENCODE, (NULL),
           ("Failed to encode frame %d: %d", frame->system_frame_number,
               (int) status));
@@ -2006,26 +2072,29 @@
       GST_ELEMENT_ERROR (self, LIBRARY, ENCODE, (NULL),
           ("Failed to encode (frame unknown): %d", (int) status));
     }
-    goto beach;
+
+    goto drop;
   }
 
   if (!frame) {
     GST_WARNING_OBJECT (self, "No corresponding frame found!");
-    goto beach;
+    return;
   }
 
   g_mutex_lock (&self->queue_mutex);
-  is_flushing = self->is_flushing;
-  g_mutex_unlock (&self->queue_mutex);
-  if (is_flushing) {
+  if (self->is_flushing) {
     GST_DEBUG_OBJECT (self, "Ignoring frame %d because we're flushing",
         frame->system_frame_number);
-    goto beach;
+
+    gst_video_codec_frame_unref (frame);
+    g_mutex_unlock (&self->queue_mutex);
+    return;
   }
+  g_mutex_unlock (&self->queue_mutex);
 
   /* This may happen if we don't have enough bitrate */
   if (sampleBuffer == NULL)
-    goto beach;
+    goto drop;
 
   if (gst_vtenc_buffer_is_keyframe (self, sampleBuffer))
     GST_VIDEO_CODEC_FRAME_SET_SYNC_POINT (frame);
@@ -2043,28 +2112,18 @@
       VTENC_OUTPUT_QUEUE_SIZE) {
     g_cond_wait (&self->queue_cond, &self->queue_mutex);
   }
-  g_mutex_unlock (&self->queue_mutex);
-
-beach:
-  if (!frame)
-    return;
-
-  g_mutex_lock (&self->queue_mutex);
-  if (self->is_flushing) {
-    /* We can discard the frame here, no need to have the output loop do that */
-    gst_video_codec_frame_unref (frame);
-    g_mutex_unlock (&self->queue_mutex);
-    return;
-  }
 
-  /* Buffer-less frames will be discarded in the output loop */
   gst_queue_array_push_tail (self->output_queue, frame);
   g_cond_signal (&self->queue_cond);

gst-plugins-bad-1.24.7.tar.xz/sys/applemedia/vtenc.h -> gst-plugins-bad-1.24.8.tar.xz/sys/applemedia/vtenc.h Changed

gst-plugins-bad-1.24.7.tar.xz/sys/uvcgadget/gstuvcsink.c -> gst-plugins-bad-1.24.8.tar.xz/sys/uvcgadget/gstuvcsink.c Changed

@@ -248,6 +248,22 @@
     case GST_EVENT_CAPS:
       GST_DEBUG_OBJECT (self, "Handling %" GST_PTR_FORMAT, event);
 
+      /* If the UVC host did not yet commit a format, the cur_caps may contain
+       * all probed caps. In this case, the element is not able to detect, if
+       * the caps have changed when the stream is enabled. Take the caps from
+       * upstream to be able to detect a change.
+       */
+      if (!GST_CAPS_IS_SIMPLE (self->cur_caps)) {
+        GstCaps *caps;
+
+        gst_event_parse_caps (event, &caps);
+        gst_caps_replace (&self->cur_caps, caps);
+
+        GST_DEBUG_OBJECT (self,
+            "UVC host didn't select a format, yet. Using upstream %"
+            GST_PTR_FORMAT, self->cur_caps);
+      }
+
       /* EVENT CAPS signals that the buffers after the event will use new caps.
        * If the UVC host requested a new format, we now must start the stream.
        */
@@ -624,12 +640,14 @@
           gst_caps_unref (configured_caps);
 
           prev_caps = gst_pad_get_current_caps (self->sinkpad);
-          if (!gst_caps_is_subset (prev_caps, self->cur_caps)) {
-            self->caps_changed = TRUE;
-            GST_DEBUG_OBJECT (self,
-                "caps changed from %" GST_PTR_FORMAT, prev_caps);
+          if (prev_caps) {
+            if (!gst_caps_is_subset (prev_caps, self->cur_caps)) {
+              self->caps_changed = TRUE;
+              GST_DEBUG_OBJECT (self,
+                  "caps changed from %" GST_PTR_FORMAT, prev_caps);
+            }
+            gst_caps_unref (prev_caps);
           }
-          gst_caps_unref (prev_caps);
         }
         break;
       default:

gst-plugins-bad-1.24.7.tar.xz/sys/v4l2codecs/gstv4l2codech265dec.c -> gst-plugins-bad-1.24.8.tar.xz/sys/v4l2codecs/gstv4l2codech265dec.c Changed

@@ -461,7 +461,7 @@
 {
   GstV4l2CodecH265Dec *self = GST_V4L2_CODEC_H265_DEC (decoder);
   GstCaps *caps = NULL;
-  guint min = 0;
+  guint min = 0, num_bitstream;
 
   if (self->streaming)
     goto no_internal_changes;
@@ -489,10 +489,26 @@
 
   min = MAX (2, min);
 
+  num_bitstream = 1 +
+      MAX (1, gst_v4l2_decoder_get_render_delay (self->decoder));
+
   self->sink_allocator = gst_v4l2_codec_allocator_new (self->decoder,
-      GST_PAD_SINK, self->min_pool_size + 2);
+      GST_PAD_SINK, num_bitstream);
+  if (!self->sink_allocator) {
+    GST_ELEMENT_ERROR (self, RESOURCE, NO_SPACE_LEFT,
+        ("Not enough memory to allocate sink buffers."), (NULL));
+    return FALSE;
+  }
+
   self->src_allocator = gst_v4l2_codec_allocator_new (self->decoder,
-      GST_PAD_SRC, self->min_pool_size + min + 1);
+      GST_PAD_SRC, self->min_pool_size + min);
+  if (!self->src_allocator) {
+    GST_ELEMENT_ERROR (self, RESOURCE, NO_SPACE_LEFT,
+        ("Not enough memory to allocate source buffers."), (NULL));
+    g_clear_object (&self->sink_allocator);
+    return FALSE;
+  }
+
   self->src_pool = gst_v4l2_codec_pool_new (self->src_allocator, &self->vinfo);
 
 no_internal_changes:

gst-plugins-bad-1.24.7.tar.xz/tests/check/elements/webrtcbin.c -> gst-plugins-bad-1.24.8.tar.xz/tests/check/elements/webrtcbin.c Changed

@@ -4538,6 +4538,58 @@
   }
 }
 
+GST_START_TEST (test_invalid_bundle_in_pending_remote_description)
+{
+  GstPromise *promise;
+  struct test_webrtc *t = test_webrtc_new ();
+  const gchar *invalid_bundle = "v=0\r\n\
+o=thisisadapterortc 2683876491 2 IN IP4 127.0.0.1\r\n\
+s=-\r\n\
+t=0 0\r\n\
+a=setup:actpass\r\n\
+a=fingerprint:sha-256 95:B3:DB:24:83:3B:9E:3F:B0:AD:93:2D:EF:73:C9:D2:1C:68:EA:19:C6:F8:73:BA:9A:FA:34:A9:64:69:C0:D8\r\n\
+a=ice-ufrag:ERn4TYI2HSbtKrzQNCdp9wD2EHt4wM2O\r\n\
+a=ice-pwd:QxEZFuCPRwIURJPMSYNNYFr2XFNgqNkG\r\n\
+a=group:BUNDLE \r\n\
+";
+  GstSDPMessage *sdp;
+  const GstStructure *reply;
+  GError *error = NULL;
+
+  t->on_negotiation_needed = NULL;
+  t->on_offer_created = NULL;
+  t->on_answer_created = NULL;
+
+  gst_sdp_message_new_from_text (invalid_bundle, &sdp);
+  GstWebRTCSessionDescription *desc =
+      gst_webrtc_session_description_new (GST_WEBRTC_SDP_TYPE_OFFER,
+      sdp);
+  gst_element_set_state (t->webrtc1, GST_STATE_READY);
+
+  promise = gst_promise_new ();
+  g_signal_emit_by_name (t->webrtc1, "set-remote-description", desc, promise);
+  gst_promise_wait (promise);
+  gst_promise_unref (promise);
+  gst_webrtc_session_description_free (desc);
+
+  /* Creating an answer from SDP with invalid BUNDLE group should trigger no crash. */
+  promise = gst_promise_new ();
+  g_signal_emit_by_name (t->webrtc1, "create-answer", NULL, promise);
+  gst_promise_wait (promise);
+  reply = gst_promise_get_reply (promise);
+  fail_unless (gst_structure_get (reply, "error", G_TYPE_ERROR, &error, NULL));
+  fail_unless (g_error_matches (error, GST_WEBRTC_ERROR,
+          GST_WEBRTC_ERROR_SDP_SYNTAX_ERROR));
+  fail_unless_equals_string (error->message,
+      "Invalid format for BUNDLE group, expected at least one mid (BUNDLE )");
+  g_clear_error (&error);
+  gst_promise_unref (promise);
+
+  test_webrtc_free (t);
+}
+
+GST_END_TEST;
+
 static void
 new_jitterbuffer_set_fast_start (GstElement * rtpbin,
     GstElement * rtpjitterbuffer, guint session_id, guint ssrc,
@@ -5979,6 +6031,7 @@
     tcase_add_test (tc, test_msid);
     tcase_add_test (tc, test_ice_end_of_candidates);
     tcase_add_test (tc, test_sdp_session_setup_attribute);
+    tcase_add_test (tc, test_invalid_bundle_in_pending_remote_description);
     if (sctpenc && sctpdec) {
       tcase_add_test (tc, test_data_channel_create);
       tcase_add_test (tc, test_data_channel_create_two_channels);

Changes of Revision 26