@@ -1,4 +1,9 @@

 -------------------------------------------------------------------

+Sat Sep 28 08:31:55 UTC 2024 - Bjørn Lie <zaitor@opensuse.org>

+

+- Update to version 1.24.8

+

+-------------------------------------------------------------------

 Sat Aug 24 19:53:47 UTC 2024 - Bjørn Lie <zaitor@opensuse.org>

 - Update to version 1.24.7

​

@@ -7,7 +7,7 @@

 %define _version 1.24.0

 Name:           gstreamer-plugins-bad-codecs

-Version:        1.24.7

+Version:        1.24.8

 Release:        0

 Summary:        Codecs/plugins for gstreamer-plugins-bad

 License:        LGPL-2.1-or-later

​

@@ -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>

​

@@ -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

​

@@ -1,4 +1,4 @@

-This is GStreamer gst-plugins-bad 1.24.7.

+This is GStreamer gst-plugins-bad 1.24.8.

 The GStreamer team is thrilled to announce a new major feature release

 of your favourite cross-platform multimedia framework!

​

@@ -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"

                     }

             }

​

@@ -45,13 +45,13 @@

 #define MR660  0

 #define MR885  1

 #define MR1265 2

-#define MR1425 2

-#define MR1585 3

-#define MR1825 4

-#define MR1985 5

-#define MR2305 6

-#define MR2385 7

-#define MRDTX  8

+#define MR1425 3

+#define MR1585 4

+#define MR1825 5

+#define MR1985 6

+#define MR2305 7

+#define MR2385 8

+#define MRDTX  9

 #define L_FRAME16k      320     /* Frame size at 16kHz  */

​

@@ -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);

​

@@ -4820,7 +4820,7 @@

   webrtc->priv->last_generated_offer = NULL;

   if (webrtc->priv->last_generated_answer)

     gst_webrtc_session_description_free (webrtc->priv->last_generated_answer);

-  {

+  if (ret) {

     GstSDPMessage *copy;

     gst_sdp_message_copy (ret, &copy);

     webrtc->priv->last_generated_answer =

​

@@ -41,7 +41,7 @@

  * ### Save the first 50 video frames generated for the GStreamer website as PNG files in /tmp

  *

  * ```

- * LIBGL_ALWAYS_SOFTWARE=true gst-launch-1.0 -v wpesrc num-buffers=50 location="https://gstreamer.freedesktop.org" ! videoconvert ! pngenc ! multifilesink location=/tmp/snapshot-%05d.png

+ * LIBGL_ALWAYS_SOFTWARE=true gst-launch-1.0 -v wpevideosrc num-buffers=50 location="https://gstreamer.freedesktop.org" ! videoconvert ! pngenc ! multifilesink location=/tmp/snapshot-%05d.png

  * ```

  *

  *

​

@@ -1917,7 +1917,7 @@

       gst_event_parse_qos (event, NULL, NULL, &diff, &timestamp);

       /* Only take into account lateness if late */

       if (diff > 0)

-        earliest_time = timestamp + 2 * diff;

+        earliest_time = timestamp + MIN (2 * diff, GST_SECOND);

       else

         earliest_time = timestamp;

​

@@ -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),

​

@@ -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);

   }

​

@@ -33,7 +33,8 @@

   if (GST_MESSAGE_TYPE (msg) == GST_MESSAGE_NEED_CONTEXT &&

       gst_message_parse_context_type (msg, &type)) {

-    return !g_strcmp0 (type, GST_WL_DISPLAY_HANDLE_CONTEXT_TYPE);

+    return !g_strcmp0 (type, GST_WL_DISPLAY_HANDLE_CONTEXT_TYPE) ||

+        !g_strcmp0 (type, GST_WL_DISPLAY_HANDLE_LEGACY_CONTEXT_TYPE);

   }

   return FALSE;

@@ -60,6 +61,7 @@

   s = gst_context_get_structure (context);

   if (gst_structure_get (s, "display", G_TYPE_POINTER, &display, NULL))

     return display;

+  /* backward compatibility */

   if (gst_structure_get (s, "handle", G_TYPE_POINTER, &display, NULL))

     return display;

   return NULL;

​

@@ -26,7 +26,10 @@

 /* The type of GstContext used to pass the wl_display pointer

  * from the application to the sink */

-#define GST_WL_DISPLAY_HANDLE_CONTEXT_TYPE "GstWlDisplayHandleContextType"

+#define GST_WL_DISPLAY_HANDLE_CONTEXT_TYPE "GstWaylandDisplayHandleContextType"

+

+/* Accidental naming, used for implementing backward compatibility */

+#define GST_WL_DISPLAY_HANDLE_LEGACY_CONTEXT_TYPE "GstWlDisplayHandleContextType"

 GST_WL_API

 gboolean gst_is_wl_display_handle_need_context_message (GstMessage * msg);

​

@@ -98,6 +98,7 @@

   g_rec_mutex_init (&priv->sync_mutex);

   gst_wl_linux_dmabuf_init_once ();

+  gst_wl_shm_init_once ();

   gst_shm_allocator_init_once ();

   gst_wl_videoformat_init_once ();

 }

​

@@ -26,6 +26,19 @@

 #include "gstwlshmallocator.h"

+GST_DEBUG_CATEGORY (gst_wl_shm_debug);

+#define GST_CAT_DEFAULT gst_wl_shm_debug

+

+void

+gst_wl_shm_init_once (void)

+{

+  static gsize _init = 0;

+

+  if (g_once_init_enter (&_init)) {

+    GST_DEBUG_CATEGORY_INIT (gst_wl_shm_debug, "wl_shm", 0, "wl_shm library");

+    g_once_init_leave (&_init, 1);

+  }

+}

 static gboolean

 gst_wl_shm_validate_video_info (const GstVideoInfo * vinfo)

​

@@ -30,6 +30,9 @@

 G_BEGIN_DECLS

 GST_WL_API

+void gst_wl_shm_init_once (void);

+

+GST_WL_API

 struct wl_buffer * gst_wl_shm_memory_construct_wl_buffer (GstMemory * mem,

     GstWlDisplay * display, const GstVideoInfo * info);

​

@@ -35,6 +35,16 @@

  <release>

   <Version>

+   <revision>1.24.8</revision>

+   <branch>1.24</branch>

+   <name></name>

+   <created>2024-09-19</created>

+   <file-release rdf:resource="https://gstreamer.freedesktop.org/src/gst-plugins-bad/gst-plugins-bad-1.24.8.tar.xz" />

+  </Version>

+ </release>

+

+ <release>

+  <Version>

    <revision>1.24.7</revision>

    <branch>1.24</branch>

    <name></name>

​

@@ -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 */

​

@@ -1,5 +1,5 @@

 project('gst-plugins-bad', 'c', 'cpp',

-  version : '1.24.7',

+  version : '1.24.8',

   meson_version : '>= 1.1',

   default_options :  'warning_level=1',

                       'buildtype=debugoptimized' )

​

@@ -6,9 +6,9 @@

 #, fuzzy

 msgid ""

 msgstr ""

-"Project-Id-Version: gst-plugins-bad-1.24.7\n"

+"Project-Id-Version: gst-plugins-bad-1.24.8\n"

 "Report-Msgid-Bugs-To: \n"

-"POT-Creation-Date: 2024-08-21 12:27+0100\n"

+"POT-Creation-Date: 2024-09-19 12:02+0200\n"

 "PO-Revision-Date: YEAR-MO-DA HO:MI+ZONE\n"

 "Last-Translator: FULL NAME <EMAIL@ADDRESS>\n"

 "Language-Team: LANGUAGE <LL@li.org>\n"

​

@@ -6,9 +6,9 @@

 #, fuzzy

 msgid ""

 msgstr ""

-"Project-Id-Version: gst-plugins-bad-1.24.7\n"

+"Project-Id-Version: gst-plugins-bad-1.24.8\n"

 "Report-Msgid-Bugs-To: \n"

-"POT-Creation-Date: 2024-08-21 12:27+0100\n"

+"POT-Creation-Date: 2024-09-19 12:02+0200\n"

 "PO-Revision-Date: YEAR-MO-DA HO:MI+ZONE\n"

 "Last-Translator: FULL NAME <EMAIL@ADDRESS>\n"

 "Language-Team: LANGUAGE <LL@li.org>\n"

​

@@ -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);

​

@@ -92,6 +92,10 @@

   gboolean negotiate_downstream;

   gboolean is_flushing;

   gboolean pause_task;

+

+  /* If we get an EncoderMalfunctionErr or similar, we restart the session

+   * before the next encode call */

+  gboolean require_restart;

 };

 void gst_vtenc_register_elements (GstPlugin * plugin);

​

@@ -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:

​

@@ -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:

​

@@ -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);

​