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gstreamer-plugins-bad-codecs.changes Changed

gstreamer-plugins-bad-codecs.spec Changed

gst-plugins-bad-1.24.8.tar.xz/ChangeLog -> gst-plugins-bad-1.24.9.tar.xz/ChangeLog Changed

@@ -1,3 +1,238 @@
+=== release 1.24.9 ===
+
+2024-10-30 20:33:30 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* NEWS:
+	* RELEASE:
+	* gst-plugins-bad.doap:
+	* meson.build:
+	  Release 1.24.9
+
+2024-10-29 11:29:05 +0100  Edward Hervey <edward@centricular.com>
+
+	* ext/srt/gstsrtobject.c:
+	  srt: Don't attempt to reconnect on authentication failures
+	  This is a fatal issue which can't be recovered
+	  Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1550
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7797>
+
+2024-10-14 11:26:20 +0200  Emil Ljungdahl <emillj@axis.com>
+
+	* ext/webrtc/gstwebrtcbin.c:
+	  webrtcbin: Clean up bin elements when datachannel is removed
+	  When a datachannel within a session is removed after proper close,
+	  reference to the error_ignore_bin elements of the datachannel
+	  appsrc/appsink were left in webrtcbin.
+	  This caused the bin-objects to be left and not freed until the whole
+	  webrtc session was terminated. Among other things that includes a thread
+	  from the appsrc.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7791>
+
+2024-09-24 13:55:39 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+	* docs/plugins/gst_plugins_cache.json:
+	* gst-libs/gst/wayland/gstwlvideoformat.c:
+	* gst-libs/gst/wayland/gstwlvideoformat.h:
+	  wayland: Add NV15 support
+	  This format, which maps to NV12_10LE40 in GStreamer is produced by Rockchip
+	  video decoders when decoding 4:2:0 10 bit content.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7789>
+
+2024-10-22 23:41:13 +0900  Seungha Yang <seungha@centricular.com>
+
+	* gst-libs/gst/codecs/gstvp8decoder.c:
+	  vp8decoder: Fix resolution change handling
+	  Do not store resolution in set_format() so that resolution change
+	  can be detected on keyframe as intended.
+	  Fixes: https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/3928
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7787>
+
+2024-10-25 16:37:15 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/rtmp2/gstrtmp2sink.c:
+	  rtmp2sink: Initialize base_ts / last_ts with the actual first observed timestamp
+	  Initializing it with zero can falsely trigger the overflow / underflow detection
+	  code if the first observed timestamp is a big integer.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7786>
+
+2024-09-30 15:51:04 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/mpegtsdemux/mpegtspacketizer.c:
+	  mpegtsdemux: Handle PTS/DTS wraparound with ignore-pcr=true
+	  The wraparound handling code assumes that the PCR gets updated regularly for
+	  being able to detect wraparounds. With ignore-pcr=true that was not the case and
+	  it stayed initialized at 1h forever.
+	  To avoid this problem, update the fake PCR whenever the PTS advanced by more
+	  than 5s, and also detect wraparounds in these fake PCRs.
+	  Problem can be reproduced with
+	  $ gst-launch-1.0 videotestsrc pattern=black ! video/x-raw,framerate=1/5 ! \
+	  x264enc speed-preset=ultrafast tune=zerolatency ! mpegtsmux ! \
+	  tsdemux ignore-pcr=true ! fakesink
+	  which restarts timestamps at 0 after around 26h30m.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7763>
+
+2024-10-26 11:42:48 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst-libs/gst/play/gstplay.c:
+	* gst-libs/gst/play/gstplay.h:
+	  play: Improve play message API inconsistencies
+	  * Check for the correct message type when parsing
+	  * Use correct field name for warning message details
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7779>
+
+2024-05-28 04:14:15 +0530  Nirbheek Chauhan <nirbheek@centricular.com>
+
+	* gst-libs/gst/va/meson.build:
+	* meson.build:
+	* sys/msdk/gstmsdkallocator_libva.c:
+	* sys/msdk/gstmsdkcaps.c:
+	* sys/msdk/gstmsdkdec.c:
+	* sys/msdk/gstmsdkenc.c:
+	* sys/msdk/gstmsdkvpp.c:
+	* sys/msdk/gstmsdkvpputil.c:
+	* sys/msdk/meson.build:
+	  msdk: Fix libdrm dependency detection and usage
+	  drm_fourcc.h should be picked up via the pkgconfig include, not the
+	  system includedir directly.
+	  Also consolidate the libdrm usage in va and msdk.
+	  All this allows it to be picked up consistently (via the subproject,
+	  for example).
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7760>
+
+2024-05-27 18:50:23 +0530  Nirbheek Chauhan <nirbheek@centricular.com>
+
+	* gst-libs/gst/va/meson.build:
+	  meson: Don't use fallback: kwarg for libva deps
+	  This will cause a fallback even when the `va` option is `auto`, not
+	  giving the user a chance to provide the dependency via the system, and
+	  likely building this feature unnecessarily.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7760>
+
+2024-05-27 18:43:33 +0530  Nirbheek Chauhan <nirbheek@centricular.com>
+
+	* gst-libs/gst/va/gstvavideoformat.h:
+	  va: Fix libdrm include
+	  The libdrm/ prefix should not be used, it will be provided by the
+	  pkgconfig file. Also HAVE_LIBDRM is necessary.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7760>
+
+2024-10-23 14:28:30 +0200  Peter Stensson <petest@axis.com>
+
+	* gst/codectimestamper/gstcodectimestamper.c:
+	* tests/check/elements/h264timestamper.c:
+	  codectimestamper: Fix gint wraparound in pts_compare_func
+	  The diff between compared timestamps might be outside the gint range
+	  resulting in wrong sorting results. This patch corrects that by
+	  comparing the timestamps and then returning -1, 0 or 1 depending on the
+	  result.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7737>
+
+2024-10-24 14:40:23 +0200  Andoni Morales Alastruey <ylatuya@gmail.com>
+
+	* sys/applemedia/vtdec.c:
+	  vtdec: add support for level 6 6.1 and 6.2
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7735>
+
+2024-10-11 11:58:37 -0400  Víctor Manuel Jáquez Leal <vjaquez@igalia.com>
+
+	* gst-libs/gst/codecparsers/gsth264bitwriter.c:
+	* gst-libs/gst/codecparsers/gsth265bitwriter.c:
+	  h26xbitwriter: false have_space if aligning fails on aud
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7697>
+
+2024-10-15 16:44:27 +0800  He Junyan <junyan.he@intel.com>
+
+	* sys/va/gstvaav1enc.c:
+	  vaav1enc: Do not enable palette mode by default
+	  Palette mode should only be enabled only when we know that the content
+	  of the picture is simple. For example, only white letters on black
+	  screen in SCC mode. So, by default, we need to disable it.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7696>
+
+2024-09-27 18:01:53 +0200  Benjamin Gaignard <benjamin.gaignard@collabora.com>
+
+	* sys/kms/gstkmssink.c:
+	  kmssink: Add IMX-DCSS auto-detection
+	  Add IMX DCSS display controller into list of
+	  auto-detected modules.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7695>
+
+2024-10-09 12:32:34 -0400  Francisco Javier Velázquez-García <francisco.velazquez@appear.net>
+
+	* ext/srt/gstsrtsink.c:
+	  srtsink: Add guard for null error when SRT open fails
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7687>
+
+2024-10-09 12:08:10 -0400  Francisco Javier Velázquez-García <francisco.velazquez@appear.net>
+
+	* ext/srt/gstsrtobject.c:
+	  srtsink: Register SRT listen callback before binding socket
+	  This change https://github.com/Haivision/srt/pull/2683 forces us to
+	  call `srt_listen_callback` before `srt_listen`.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7687>
+
+2024-10-11 12:07:27 -0400  Víctor Manuel Jáquez Leal <vjaquez@igalia.com>
+
+	* gst-libs/gst/codecparsers/gstav1bitwriter.c:
+	* gst-libs/gst/codecparsers/gsth264bitwriter.c:
+	* gst-libs/gst/codecparsers/gsth265bitwriter.c:
+	  codecparsers: add debug categories to bitwriters
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7667>
+
+2024-10-09 17:16:46 -0400  Víctor Manuel Jáquez Leal <vjaquez@igalia.com>
+
+	* tests/check/elements/vapostproc.c:
+	  tests: va: fix vapostproc test for DMABuf
+	  Now it picks the first format in the template srcpad list and do
+	  the convertion. Also the format size is reduced because not all
+	  drives support 4K as DMABuf (radeonsi).
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7654>
+
+2024-10-14 15:31:54 +0200  Stéphane Cerveau <scerveau@igalia.com>
+
+	* gst/videofilters/gstscenechange.c:
+	  scenechange: fix memory leak
+	  A reference to the last buffer(oldbuf) was kept
+	  leading to a memory leak on stop.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7666>
+
+2024-10-13 23:04:58 +0530  Nirbheek Chauhan <nirbheek@centricular.com>

gst-plugins-bad-1.24.8.tar.xz/NEWS -> gst-plugins-bad-1.24.9.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 1.24.8 and was released on 19 September 2024.
+The latest bug-fix release in the stable 1.24 series is 1.24.9 and was released on 30 October 2024.
 
 See https://gstreamer.freedesktop.org/releases/1.24/ for the latest version of this document.
 
-Last updated: Thursday 19 September 2024, 00:30 UTC (log)
+Last updated: Wednesday 30 October 2024, 20:00 UTC (log)
 
 ## Introduction
 
@@ -2450,8 +2450,6 @@
 -   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
 
@@ -2461,6 +2459,8 @@
 -   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
+-   v4l2bufferpool: actually queue back the empty buffer flagged LAST
+-   v4l2videoenc: unref buffer pool after usage properly
 -   v4l2: encoder: Add dynamic framerate support
 
 gst-plugins-bad
@@ -2547,6 +2547,185 @@
 -   List of Merge Requests applied in 1.24.8
 -   List of Issues fixed in 1.24.8
 
+1.24.9
+
+The ninth 1.24 bug-fix release (1.24.9) was released on 30 October 2024.
+
+This release only contains bugfixes and a security fix and it should be safe to update from 1.24.x.
+
+Highlighted bugfixes in 1.24.9
+
+-   gst-rtsp-server security fix
+-   GstAggregator start time selection and latency query fixes for force-live mode
+-   audioconvert: fix dynamic handling of mix matrix, and accept custom upstream event for setting one
+-   encodebin: fix parser selection for encoders that support multiple codecs
+-   flvmux improvments for pipelines where timestamps don’t start at 0
+-   glcontext: egl: Unrestrict the support base DRM formats
+-   kms: Add IMX-DCSS auto-detection in sink and fix stride with planar formats in allocator
+-   macOS main application event loop fixes
+-   mpegtsdemux: Handle PTS/DTS wraparound with ignore-pcr=true
+-   playbin3, decodebin3, parsebin, urisourcebin: fix races, and improve stability and stream-collection handling
+-   rtpmanager: fix early RTCP SR generation for sparse streams like metadata
+-   qml6glsrc: Reduce capture delay
+-   qtdemux: fix parsing of rotation matrix with 180 degree rotation
+-   rtpav1depay: added wait-for-keyframe and request-keyframe properties
+-   srt: make work with newer libsrt versions and don’t re-connect on authentication failure
+-   v4l2 fixes and improvement
+-   webrtcsink, webrtcbin and whepsrc fixes
+-   cerbero: fix Python 3.13 compatibility, g-i with newer setuptools, bootstrap on Arch Linux; iOS build fixes
+-   Ship qroverlay plugin in binary packages - Various bug fixes, memory leak fixes, and other stability and reliability
+    improvements
+-   Various bug fixes, build fixes, memory leak fixes, and other stability and reliability improvements
+
+gstreamer
+
+-   aggregator: fix start time selection first with force-live
+-   aggregator: fix live query when force-live is TRUE
+-   parse-launch: Make sure children are bins before recursing in
+-   macos: Fix race conditions in cocoa/application main event loop
+-   multiqueue: Do not unref the query we get in pad->query
+
+gst-plugins-base
+
+-   audioconvert: fix dynamic handling of mix matrix, accept custom upstream event for setting one
+-   playback: Fix a variety of decodebin3/parsebin/urisourcebin races
+-   playbin3: prevent crashing trying to play a corrupted mp4 file (WARNING : HIGH PITCHED CORRUPTED SOUND)
+-   urisourcebin: Ensure all stream-start are handled
+-   urisourcebin: Allow more cases for posting stream-collection
+-   decodebin3: Make update/posting of collection messages atomic
+-   decodebin3: send selected stream message as long as not all the tracks can’t select decoders
+-   urisourcebin/parsebin: Improve collection creation and handling
+-   encodebasebin: Miscellaneous fixes
+-   allocators: drmdumb: Fix bpp value for P010
+-   gldownload: use gst_gl_sync_meta_wait_cpu()
+-   Revert “meson: Fix invalid include flag in uninstalled gl pc file”
+-   gl: Fix configure error when libdrm is a subproject
+-   glcontext: egl: Unrestrict the support base DRM formats
+-   exiftag: Check the result of gst_date_time_new_local_time(), fixes criticals with malformed EXIF tags
+
+gst-plugins-good
+
+-   flvmux: Use first running time on the initial header instead of 0
+-   rtpmanager: skip RTPSources which are not ready in the RTCP generation
+-   rtppassthroughpay: Fix reading clock-rate and payload type from caps
+-   qml6glsrc: Reduce capture delay
+-   qtdemux: fix parsing of matrix with 180 rotation
+-   qtdemux: Check fourcc of a second CEA608 atom instead of assuming it’s cdt2
+-   qtdemux: Skip zero-sized boxes instead of stopping to look at further boxes
+-   twcc: Handle wrapping of reference time
+-   v4l2object: append non colorimetry structure to probed caps
+-   v4l2: Various fixes and improvement
+
+gst-plugins-bad
+
+-   avfdeviceprovider: Fix leak from the GstCaps
+-   codecparsers: add debug categories to bitwriters
+-   codectimestamper: Fix gint wraparound in pts_compare_func
+-   dvxa: Explicitly use cpp_std=c++11
+-   GstPlay: message parsing and documentation improvements
+-   h26xbitwriter: false have_space if aligning fails on aud
+-   kmsallocator: fix stride with planar formats
+-   kmssink: Add IMX-DCSS auto-detection
+-   mpegtsdemux: Handle PTS/DTS wraparound with ignore-pcr=true
+-   rtmp2sink: Initialize base_ts / last_ts with the actual first observed timestamp
+-   scenechange: fix memory leak
+-   srtsink: Register SRT listen callback before binding socket
+-   srt: Don’t attempt to reconnect on authentication failures
+-   tests: va: fix vapostproc test for DMABuf
+-   tests: lc3: Allocate the same size for the buffer and the data
+-   va: Fix libdrm include, plus meson and wrap changes
+-   vaav1enc: Do not enable palette mode by default
+-   vp8decoder: Fix resolution change handling
+-   vtdec: add support for level 6 6.1 and 6.2
+-   wayland: Add NV15 support
+-   webrtcbin: Clean up bin elements when datachannel is removed
+
+gst-plugins-ugly
+
+-   No changes
+
+GStreamer Rust plugins
+
+-   Build: turn lto off for dev profile for faster dev builds
+-   fmp4 hls_live example: Don’t set header-update-mode=update, no need to update the header in live mode
+-   gtk4paintablesink: Don’t check for a GL context when filtering dmabuf caps
+-   livesync: Log latency query results when handling latency query too
+-   onvifmetadatapay: Set output caps earlier, so upstream can send gap events earlier
+-   rtpav1depay: Add wait-for-keyframe and request-keyframe properties
+-   spotify: tweak dependencies
+-   transcriberbin: fix panic during gst-inspect-1.0 -u
+-   webrtcsink: fix segment format mismatch with remote offer
+-   webrtcsink: fix assertions when finalizing
+-   webrtcsink: Fix typo in “turn-servers” property description
+-   whepsrc: Fix incorrect default caps
+
+gst-libav
+
+-   avviddec: Unlock video decoder stream lock temporarily while finishing frames
+
+gst-rtsp-server
+
+-   rtsp-server: Remove pointless assertions that can happen if client provides invalid rates (security fix)
+
+gstreamer-vaapi
+
+-   No changes
+
+gstreamer-sharp
+
+-   No changes
+
+gst-omx
+
+-   No changes
+
+gst-python
+
+-   No changes
+
+gst-editing-services
+
+-   ges: Fix name of GESFrameCompositionMeta API type (which caused gobject-introspection failures at build time)
+
+gst-devtools, gst-validate + gst-integration-testsuites
+
+-   validate: Ignore flaky dash playbin3 issue
+-   validate: Blacklist more netsim tests
+
+gst-examples
+
+-   No changes
+
+Development build environment
+
+-   No changes
+
+Cerbero build tool and packaging changes in 1.24.9
+
+-   Fix Python 3.13 compatibility
+-   gobject-introspection: Import patch to build against newer setuptools
+-   Switch from wget to curl on Fedora 40 and newer
+-   bootstrap: Add missing dependencies on Arch Linux
+-   harfbuzz: Add CXXFLAGS to fix broken build on iOS
+-   openssl.recipe: Stop using non-existent domain in primary tarball url
+-   gst-plugins-bad: ship qroverlay plugin
+
+Contributors to 1.24.9

gst-plugins-bad-1.24.8.tar.xz/RELEASE -> gst-plugins-bad-1.24.9.tar.xz/RELEASE Changed

gst-plugins-bad-1.24.8.tar.xz/docs/plugins/gst_plugins_cache.json -> gst-plugins-bad-1.24.9.tar.xz/docs/plugins/gst_plugins_cache.json Changed

@@ -36081,7 +36081,7 @@
                 "long-name": "Gtk Wayland Video Sink",
                 "pad-templates": {
                     "sink": {
-                        "caps": "video/x-raw:\n         format: { AYUV, RGBA, ARGB, BGRA, ABGR, P010_10LE, v308, RGBx, xRGB, BGRx, xBGR, RGB, BGR, Y42B, NV16, NV61, YUY2, YVYU, UYVY, I420, YV12, NV12, NV21, Y41B, YUV9, YVU9, BGR16, RGB16 }\n          width:  1, 2147483647 \n         height:  1, 2147483647 \n      framerate:  0/1, 2147483647/1 \n\nvideo/x-raw(memory:DMABuf):\n         format: DMA_DRM\n          width:  1, 2147483647 \n         height:  1, 2147483647 \n      framerate:  0/1, 2147483647/1 \n",
+                        "caps": "video/x-raw:\n         format: { AYUV, RGBA, ARGB, BGRA, ABGR, P010_10LE, NV12_10LE40, v308, RGBx, xRGB, BGRx, xBGR, RGB, BGR, Y42B, NV16, NV61, YUY2, YVYU, UYVY, I420, YV12, NV12, NV21, Y41B, YUV9, YVU9, BGR16, RGB16 }\n          width:  1, 2147483647 \n         height:  1, 2147483647 \n      framerate:  0/1, 2147483647/1 \n\nvideo/x-raw(memory:DMABuf):\n         format: DMA_DRM\n          width:  1, 2147483647 \n         height:  1, 2147483647 \n      framerate:  0/1, 2147483647/1 \n",
                         "direction": "sink",
                         "presence": "always"
                     }
@@ -245349,7 +245349,7 @@
                 "long-name": "wayland video sink",
                 "pad-templates": {
                     "sink": {
-                        "caps": "video/x-raw:\n         format: { AYUV, RGBA, ARGB, BGRA, ABGR, P010_10LE, v308, RGBx, xRGB, BGRx, xBGR, RGB, BGR, Y42B, NV16, NV61, YUY2, YVYU, UYVY, I420, YV12, NV12, NV21, Y41B, YUV9, YVU9, BGR16, RGB16 }\n          width:  1, 2147483647 \n         height:  1, 2147483647 \n      framerate:  0/1, 2147483647/1 \n\nvideo/x-raw(memory:DMABuf):\n         format: DMA_DRM\n          width:  1, 2147483647 \n         height:  1, 2147483647 \n      framerate:  0/1, 2147483647/1 \n",
+                        "caps": "video/x-raw:\n         format: { AYUV, RGBA, ARGB, BGRA, ABGR, P010_10LE, NV12_10LE40, v308, RGBx, xRGB, BGRx, xBGR, RGB, BGR, Y42B, NV16, NV61, YUY2, YVYU, UYVY, I420, YV12, NV12, NV21, Y41B, YUV9, YVU9, BGR16, RGB16 }\n          width:  1, 2147483647 \n         height:  1, 2147483647 \n      framerate:  0/1, 2147483647/1 \n\nvideo/x-raw(memory:DMABuf):\n         format: DMA_DRM\n          width:  1, 2147483647 \n         height:  1, 2147483647 \n      framerate:  0/1, 2147483647/1 \n",
                         "direction": "sink",
                         "presence": "always"
                     }

gst-plugins-bad-1.24.8.tar.xz/ext/srt/gstsrtobject.c -> gst-plugins-bad-1.24.9.tar.xz/ext/srt/gstsrtobject.c Changed

@@ -732,6 +732,8 @@
    * Boolean to choose whether to automatically reconnect.  If TRUE, an element
    * in caller mode will try to reconnect instead of reporting an error.
    *
+   * This will be ignored for authentication failures.
+   *
    * Since: 1.22
    *
    */
@@ -1198,19 +1200,22 @@
   }
   poll_added = TRUE;
 
-  GST_DEBUG_OBJECT (srtobject->element, "Starting to listen on bind socket");
-  if (srt_listen (sock, 1) == SRT_ERROR) {
-    g_set_error (error, GST_RESOURCE_ERROR,
-        GST_RESOURCE_ERROR_OPEN_READ_WRITE, "Cannot listen on bind socket: %s",
-        srt_getlasterror_str ());
-    goto failed;
-  }
-
   srtobject->sock = sock;
 
   /* Register the SRT listen callback */
   if (srt_listen_callback (srtobject->sock,
           (srt_listen_callback_fn *) srt_listen_callback_func, srtobject)) {
+    g_set_error (error, GST_RESOURCE_ERROR,
+        GST_RESOURCE_ERROR_OPEN_READ_WRITE,
+        "Failed to register SRT listen callback: %s", srt_getlasterror_str ());
+    goto failed;
+  }
+
+  GST_DEBUG_OBJECT (srtobject->element, "Starting to listen on bind socket");
+  if (srt_listen (sock, 1) == SRT_ERROR) {
+    g_set_error (error, GST_RESOURCE_ERROR,
+        GST_RESOURCE_ERROR_OPEN_READ_WRITE, "Cannot listen on bind socket: %s",
+        srt_getlasterror_str ());
     goto failed;
   }
 
@@ -1597,6 +1602,8 @@
         g_set_error (&internal_error, GST_RESOURCE_ERROR,
             GST_RESOURCE_ERROR_NOT_AUTHORIZED,
             "Failed to authenticate: %" REASON_FORMAT, REASON_ARGS (reason));
+        /* Failure to authenticate are fatal */
+        auto_reconnect = FALSE;
       } else {
         g_set_error (&internal_error, GST_RESOURCE_ERROR,
             GST_RESOURCE_ERROR_READ,

gst-plugins-bad-1.24.8.tar.xz/ext/srt/gstsrtsink.c -> gst-plugins-bad-1.24.9.tar.xz/ext/srt/gstsrtsink.c Changed

gst-plugins-bad-1.24.8.tar.xz/ext/webrtc/gstwebrtcbin.c -> gst-plugins-bad-1.24.9.tar.xz/ext/webrtc/gstwebrtcbin.c Changed

gst-plugins-bad-1.24.8.tar.xz/gst-libs/gst/codecparsers/gstav1bitwriter.c -> gst-plugins-bad-1.24.9.tar.xz/gst-libs/gst/codecparsers/gstav1bitwriter.c Changed

gst-plugins-bad-1.24.8.tar.xz/gst-libs/gst/codecparsers/gsth264bitwriter.c -> gst-plugins-bad-1.24.9.tar.xz/gst-libs/gst/codecparsers/gsth264bitwriter.c Changed

gst-plugins-bad-1.24.8.tar.xz/gst-libs/gst/codecparsers/gsth265bitwriter.c -> gst-plugins-bad-1.24.9.tar.xz/gst-libs/gst/codecparsers/gsth265bitwriter.c Changed

gst-plugins-bad-1.24.8.tar.xz/gst-libs/gst/codecs/gstvp8decoder.c -> gst-plugins-bad-1.24.9.tar.xz/gst-libs/gst/codecs/gstvp8decoder.c Changed

gst-plugins-bad-1.24.8.tar.xz/gst-libs/gst/dxva/meson.build -> gst-plugins-bad-1.24.9.tar.xz/gst-libs/gst/dxva/meson.build Changed

gst-plugins-bad-1.24.8.tar.xz/gst-libs/gst/play/gstplay.c -> gst-plugins-bad-1.24.9.tar.xz/gst-libs/gst/play/gstplay.c Changed

@@ -4721,9 +4721,12 @@
   return g_str_equal (gst_structure_get_name (data), GST_PLAY_MESSAGE_DATA);
 }
 
-#define PARSE_MESSAGE_FIELD(msg, field, value_type, value) G_STMT_START { \
+#define PARSE_MESSAGE_FIELD(msg, expected_msg_type, field, value_type, value) G_STMT_START { \
     const GstStructure *data = NULL;                                      \
+    GstPlayMessage msg_type;                                              \
     g_return_if_fail (gst_play_is_play_message (msg));                    \
+    gst_play_message_parse_type (msg, &msg_type);                         \
+    g_return_if_fail (msg_type == expected_msg_type);                     \
     data = gst_message_get_structure (msg);                               \
     gst_structure_get (data, field, value_type, value, NULL);             \
 } G_STMT_END
@@ -4740,8 +4743,11 @@
 void
 gst_play_message_parse_type (GstMessage * msg, GstPlayMessage * type)
 {
-  PARSE_MESSAGE_FIELD (msg, GST_PLAY_MESSAGE_DATA_TYPE,
-      GST_TYPE_PLAY_MESSAGE, type);
+  const GstStructure *data = NULL;
+  g_return_if_fail (gst_play_is_play_message (msg));
+  data = gst_message_get_structure (msg);
+  gst_structure_get (data, GST_PLAY_MESSAGE_DATA_TYPE, GST_TYPE_PLAY_MESSAGE,
+      type, NULL);
 }
 
 /**
@@ -4749,7 +4755,7 @@
  * @msg: A #GstMessage
  * @duration: (out) (optional): the resulting duration
  *
- * Parse the given duration @msg and extract the corresponding #GstClockTime
+ * Parse the given duration-changed @msg and extract the corresponding #GstClockTime
  *
  * Since: 1.20
  */
@@ -4757,8 +4763,8 @@
 gst_play_message_parse_duration_updated (GstMessage * msg,
     GstClockTime * duration)
 {
-  PARSE_MESSAGE_FIELD (msg, GST_PLAY_MESSAGE_DATA_DURATION,
-      GST_TYPE_CLOCK_TIME, duration);
+  PARSE_MESSAGE_FIELD (msg, GST_PLAY_MESSAGE_DURATION_CHANGED,
+      GST_PLAY_MESSAGE_DATA_DURATION, GST_TYPE_CLOCK_TIME, duration);
 }
 
 /**
@@ -4766,7 +4772,7 @@
  * @msg: A #GstMessage
  * @position: (out) (optional): the resulting position
  *
- * Parse the given position @msg and extract the corresponding #GstClockTime
+ * Parse the given position-updated @msg and extract the corresponding #GstClockTime
  *
  * Since: 1.20
  */
@@ -4774,8 +4780,8 @@
 gst_play_message_parse_position_updated (GstMessage * msg,
     GstClockTime * position)
 {
-  PARSE_MESSAGE_FIELD (msg, GST_PLAY_MESSAGE_DATA_POSITION,
-      GST_TYPE_CLOCK_TIME, position);
+  PARSE_MESSAGE_FIELD (msg, GST_PLAY_MESSAGE_POSITION_UPDATED,
+      GST_PLAY_MESSAGE_DATA_POSITION, GST_TYPE_CLOCK_TIME, position);
 }
 
 /**
@@ -4783,15 +4789,15 @@
  * @msg: A #GstMessage
  * @state: (out) (optional): the resulting play state
  *
- * Parse the given state @msg and extract the corresponding #GstPlayState
+ * Parse the given state-changed @msg and extract the corresponding #GstPlayState
  *
  * Since: 1.20
  */
 void
 gst_play_message_parse_state_changed (GstMessage * msg, GstPlayState * state)
 {
-  PARSE_MESSAGE_FIELD (msg, GST_PLAY_MESSAGE_DATA_PLAY_STATE,
-      GST_TYPE_PLAY_STATE, state);
+  PARSE_MESSAGE_FIELD (msg, GST_PLAY_MESSAGE_STATE_CHANGED,
+      GST_PLAY_MESSAGE_DATA_PLAY_STATE, GST_TYPE_PLAY_STATE, state);
 }
 
 /**
@@ -4799,15 +4805,15 @@
  * @msg: A #GstMessage
  * @percent: (out) (optional): the resulting buffering percent
  *
- * Parse the given buffering-percent @msg and extract the corresponding value
+ * Parse the given buffering @msg and extract the corresponding value
  *
  * Since: 1.20
  */
 void
 gst_play_message_parse_buffering_percent (GstMessage * msg, guint * percent)
 {
-  PARSE_MESSAGE_FIELD (msg, GST_PLAY_MESSAGE_DATA_BUFFERING_PERCENT,
-      G_TYPE_UINT, percent);
+  PARSE_MESSAGE_FIELD (msg, GST_PLAY_MESSAGE_BUFFERING,
+      GST_PLAY_MESSAGE_DATA_BUFFERING_PERCENT, G_TYPE_UINT, percent);
 }
 
 /**
@@ -4824,9 +4830,10 @@
 gst_play_message_parse_error (GstMessage * msg, GError ** error,
     GstStructure ** details)
 {
-  PARSE_MESSAGE_FIELD (msg, GST_PLAY_MESSAGE_DATA_ERROR, G_TYPE_ERROR, error);
-  PARSE_MESSAGE_FIELD (msg, GST_PLAY_MESSAGE_DATA_ERROR_DETAILS,
-      GST_TYPE_STRUCTURE, details);
+  PARSE_MESSAGE_FIELD (msg, GST_PLAY_MESSAGE_ERROR, GST_PLAY_MESSAGE_DATA_ERROR,
+      G_TYPE_ERROR, error);
+  PARSE_MESSAGE_FIELD (msg, GST_PLAY_MESSAGE_ERROR,
+      GST_PLAY_MESSAGE_DATA_ERROR_DETAILS, GST_TYPE_STRUCTURE, details);
 }
 
 /**
@@ -4835,7 +4842,7 @@
  * @error: (out) (optional) (transfer full): the resulting warning
  * @details: (out) (optional) (nullable) (transfer full): A #GstStructure containing additional details about the warning
  *
- * Parse the given error @msg and extract the corresponding #GError warning
+ * Parse the given warning @msg and extract the corresponding #GError
  *
  * Since: 1.20
  */
@@ -4843,9 +4850,10 @@
 gst_play_message_parse_warning (GstMessage * msg, GError ** error,
     GstStructure ** details)
 {
-  PARSE_MESSAGE_FIELD (msg, GST_PLAY_MESSAGE_DATA_WARNING, G_TYPE_ERROR, error);
-  PARSE_MESSAGE_FIELD (msg, GST_PLAY_MESSAGE_DATA_WARNING, GST_TYPE_STRUCTURE,
-      details);
+  PARSE_MESSAGE_FIELD (msg, GST_PLAY_MESSAGE_WARNING,
+      GST_PLAY_MESSAGE_DATA_WARNING, G_TYPE_ERROR, error);
+  PARSE_MESSAGE_FIELD (msg, GST_PLAY_MESSAGE_WARNING,
+      GST_PLAY_MESSAGE_DATA_WARNING_DETAILS, GST_TYPE_STRUCTURE, details);
 }
 
 /**
@@ -4854,7 +4862,7 @@
  * @width: (out) (optional): the resulting video width
  * @height: (out) (optional): the resulting video height
  *
- * Parse the given @msg and extract the corresponding video dimensions
+ * Parse the given video-dimensions-changed @msg and extract the corresponding video dimensions
  *
  * Since: 1.20
  */
@@ -4862,10 +4870,10 @@
 gst_play_message_parse_video_dimensions_changed (GstMessage * msg,
     guint * width, guint * height)
 {
-  PARSE_MESSAGE_FIELD (msg, GST_PLAY_MESSAGE_DATA_VIDEO_WIDTH,
-      G_TYPE_UINT, width);
-  PARSE_MESSAGE_FIELD (msg, GST_PLAY_MESSAGE_DATA_VIDEO_HEIGHT,
-      G_TYPE_UINT, height);
+  PARSE_MESSAGE_FIELD (msg, GST_PLAY_MESSAGE_VIDEO_DIMENSIONS_CHANGED,
+      GST_PLAY_MESSAGE_DATA_VIDEO_WIDTH, G_TYPE_UINT, width);
+  PARSE_MESSAGE_FIELD (msg, GST_PLAY_MESSAGE_VIDEO_DIMENSIONS_CHANGED,
+      GST_PLAY_MESSAGE_DATA_VIDEO_HEIGHT, G_TYPE_UINT, height);
 }
 
 /**
@@ -4873,7 +4881,7 @@
  * @msg: A #GstMessage
  * @info: (out) (optional) (transfer full): the resulting media info
  *
- * Parse the given @msg and extract the corresponding media information
+ * Parse the given media-info-updated @msg and extract the corresponding media information
  *
  * Since: 1.20
  */
@@ -4881,8 +4889,8 @@
 gst_play_message_parse_media_info_updated (GstMessage * msg,
     GstPlayMediaInfo ** info)
 {
-  PARSE_MESSAGE_FIELD (msg, GST_PLAY_MESSAGE_DATA_MEDIA_INFO,
-      GST_TYPE_PLAY_MEDIA_INFO, info);
+  PARSE_MESSAGE_FIELD (msg, GST_PLAY_MESSAGE_MEDIA_INFO_UPDATED,
+      GST_PLAY_MESSAGE_DATA_MEDIA_INFO, GST_TYPE_PLAY_MEDIA_INFO, info);
 }
 
 /**
@@ -4890,15 +4898,15 @@
  * @msg: A #GstMessage
  * @volume: (out) (optional): the resulting audio volume
  *
- * Parse the given @msg and extract the corresponding audio volume
+ * Parse the given volume-changed @msg and extract the corresponding audio volume
  *
  * Since: 1.20
  */
 void
 gst_play_message_parse_volume_changed (GstMessage * msg, gdouble * volume)
 {
-  PARSE_MESSAGE_FIELD (msg, GST_PLAY_MESSAGE_DATA_VOLUME, G_TYPE_DOUBLE,

gst-plugins-bad-1.24.8.tar.xz/gst-libs/gst/play/gstplay.h -> gst-plugins-bad-1.24.9.tar.xz/gst-libs/gst/play/gstplay.h Changed

gst-plugins-bad-1.24.8.tar.xz/gst-libs/gst/va/gstvavideoformat.h -> gst-plugins-bad-1.24.9.tar.xz/gst-libs/gst/va/gstvavideoformat.h Changed

gst-plugins-bad-1.24.8.tar.xz/gst-libs/gst/va/meson.build -> gst-plugins-bad-1.24.9.tar.xz/gst-libs/gst/va/meson.build Changed

@@ -69,31 +69,26 @@
   libva_req = '>= 1.18'
 endif
 
-libva_dep = dependency('libva', version: libva_req, required: va_required,
-    fallback: 'libva', 'libva_dep')
+libva_dep = dependency('libva', version: libva_req, required: va_required)
 if not libva_dep.found()
   subdir_done()
 endif
 
 if host_system == 'linux'
-  libva_drm_dep = dependency('libva-drm', version: libva_req, required: va_required,
-      fallback: 'libva', 'libva_drm_dep')
+  libva_drm_dep = dependency('libva-drm', version: libva_req, required: va_required)
   if not libva_drm_dep.found()
     subdir_done()
   endif
   platform_deps += libva_drm_dep
 
-  libdrm_dep = dependency('libdrm', version: '>=2.4', required: get_option('drm'))
   if libdrm_dep.found()
     platform_deps += libdrm_dep
-    extra_args += '-DHAVE_LIBDRM'
   endif
 
   va_sources += va_linux_sources
   va_headers += va_linux_headers
 else
-  libva_win32_dep = dependency('libva-win32', version: libva_req, required: va_required,
-      fallback: 'libva', 'libva_win32_dep')
+  libva_win32_dep = dependency('libva-win32', version: libva_req, required: va_required)
   if not libva_win32_dep.found()
     subdir_done()
   endif

gst-plugins-bad-1.24.8.tar.xz/gst-libs/gst/wayland/gstwlvideoformat.c -> gst-plugins-bad-1.24.9.tar.xz/gst-libs/gst/wayland/gstwlvideoformat.c Changed

gst-plugins-bad-1.24.8.tar.xz/gst-libs/gst/wayland/gstwlvideoformat.h -> gst-plugins-bad-1.24.9.tar.xz/gst-libs/gst/wayland/gstwlvideoformat.h Changed

@@ -37,13 +37,13 @@
  * Since: 1.24
  */
 #if G_BYTE_ORDER == G_BIG_ENDIAN
-#define GST_WL_VIDEO_FORMATS "{ AYUV, RGBA, ARGB, BGRA, ABGR, P010_10LE, v308, " \
-    "RGBx, xRGB, BGRx, xBGR, RGB, BGR, Y42B, NV16, NV61, YUY2, YVYU, UYVY, " \
-    "I420, YV12, NV12, NV21, Y41B, YUV9, YVU9, BGR16, RGB16 }"
+#define GST_WL_VIDEO_FORMATS "{ AYUV, RGBA, ARGB, BGRA, ABGR, P010_10LE, " \
+    "NV12_10LE40, v308, RGBx, xRGB, BGRx, xBGR, RGB, BGR, Y42B, NV16, NV61, " \
+    "YUY2, YVYU, UYVY, I420, YV12, NV12, NV21, Y41B, YUV9, YVU9, BGR16, RGB16 }"
 #elif G_BYTE_ORDER == G_LITTLE_ENDIAN
-#define GST_WL_VIDEO_FORMATS "{ AYUV, RGBA, ARGB, BGRA, ABGR, P010_10LE, v308, " \
-    "RGBx, xRGB, BGRx, xBGR, RGB, BGR, Y42B, NV16, NV61, YUY2, YVYU, UYVY, " \
-    "I420, YV12, NV12, NV21, Y41B, YUV9, YVU9, BGR16, RGB16 }"
+#define GST_WL_VIDEO_FORMATS "{ AYUV, RGBA, ARGB, BGRA, ABGR, P010_10LE, " \
+    "NV12_10LE40, v308, RGBx, xRGB, BGRx, xBGR, RGB, BGR, Y42B, NV16, NV61, " \
+    "YUY2, YVYU, UYVY, I420, YV12, NV12, NV21, Y41B, YUV9, YVU9, BGR16, RGB16 }"
 #endif
 
 GST_WL_API

gst-plugins-bad-1.24.8.tar.xz/gst-plugins-bad.doap -> gst-plugins-bad-1.24.9.tar.xz/gst-plugins-bad.doap Changed

gst-plugins-bad-1.24.8.tar.xz/gst/codectimestamper/gstcodectimestamper.c -> gst-plugins-bad-1.24.9.tar.xz/gst/codectimestamper/gstcodectimestamper.c Changed

gst-plugins-bad-1.24.8.tar.xz/gst/mpegtsdemux/mpegtspacketizer.c -> gst-plugins-bad-1.24.9.tar.xz/gst/mpegtsdemux/mpegtspacketizer.c Changed

@@ -2261,10 +2261,24 @@
   PACKETIZER_GROUP_LOCK (packetizer);
   pcrtable = get_pcr_table (packetizer, pcr_pid);
 
-  if (!GST_CLOCK_TIME_IS_VALID (pcrtable->base_time) && pcr_pid == 0x1fff &&
-      GST_CLOCK_TIME_IS_VALID (packetizer->last_in_time)) {
-    pcrtable->base_time = packetizer->last_in_time;
-    pcrtable->base_pcrtime = pts;
+  if (pcr_pid == 0x1fff && GST_CLOCK_TIME_IS_VALID (packetizer->last_in_time)) {
+    if (!GST_CLOCK_TIME_IS_VALID (pcrtable->base_time)) {
+      pcrtable->base_time = packetizer->last_in_time;
+      pcrtable->base_pcrtime = pts;
+    } else if (check_diff) {
+      /* Handle discont and wraparound */
+      guint64 tmp_pts = pts + pcrtable->pcroffset + packetizer->extra_shift;
+      if (pcrtable->base_pcrtime < tmp_pts
+          && tmp_pts - pcrtable->base_pcrtime >= 5 * GST_SECOND) {
+        guint64 diff = tmp_pts - pcrtable->base_pcrtime - 2 * GST_SECOND;
+
+        pcrtable->base_time += diff;
+        pcrtable->base_pcrtime += diff;
+      } else if (pcrtable->base_pcrtime > tmp_pts
+          && pcrtable->base_pcrtime > PCR_GST_MAX_VALUE / 2) {
+        pcrtable->pcroffset += PCR_GST_MAX_VALUE;
+      }
+    }
   }
 
   /* Use clock skew if present */

gst-plugins-bad-1.24.8.tar.xz/gst/rtmp2/gstrtmp2sink.c -> gst-plugins-bad-1.24.9.tar.xz/gst/rtmp2/gstrtmp2sink.c Changed

@@ -533,8 +533,8 @@
   self->running = TRUE;
   self->cancellable = g_cancellable_new ();
   self->stream_id = 0;
-  self->last_ts = 0;
-  self->base_ts = 0;
+  self->last_ts = GST_CLOCK_TIME_NONE;
+  self->base_ts = GST_CLOCK_TIME_NONE;
 
   if (async) {
     gst_task_start (self->task);
@@ -700,27 +700,31 @@
 
     /* flvmux timestamps roll over after about 49 days */
     timestamp = header.timestamp;
-    if (timestamp + self->base_ts + G_MAXINT32 < self->last_ts) {
-      GST_WARNING_OBJECT (self, "Timestamp regression %" G_GUINT64_FORMAT
-          " -> %" G_GUINT64_FORMAT "; assuming overflow", self->last_ts,
-          timestamp + self->base_ts);
-      self->base_ts += G_MAXUINT32;
-      self->base_ts += 1;
-    } else if (timestamp + self->base_ts > self->last_ts + G_MAXINT32) {
-      GST_WARNING_OBJECT (self, "Timestamp jump %" G_GUINT64_FORMAT
-          " -> %" G_GUINT64_FORMAT "; assuming underflow", self->last_ts,
-          timestamp + self->base_ts);
-      if (self->base_ts > 0) {
-        self->base_ts -= G_MAXUINT32;
-        self->base_ts -= 1;
-      } else {
-        GST_WARNING_OBJECT (self, "Cannot regress further;"
-            " forcing timestamp to zero");
-        timestamp = 0;
+    if (self->base_ts == GST_CLOCK_TIME_NONE) {
+      self->last_ts = self->base_ts = timestamp;
+    } else {
+      if (timestamp + self->base_ts + G_MAXINT32 < self->last_ts) {
+        GST_WARNING_OBJECT (self, "Timestamp regression %" G_GUINT64_FORMAT
+            " -> %" G_GUINT64_FORMAT "; assuming overflow", self->last_ts,
+            timestamp + self->base_ts);
+        self->base_ts += G_MAXUINT32;
+        self->base_ts += 1;
+      } else if (timestamp + self->base_ts > self->last_ts + G_MAXINT32) {
+        GST_WARNING_OBJECT (self, "Timestamp jump %" G_GUINT64_FORMAT
+            " -> %" G_GUINT64_FORMAT "; assuming underflow", self->last_ts,
+            timestamp + self->base_ts);
+        if (self->base_ts > 0) {
+          self->base_ts -= G_MAXUINT32;
+          self->base_ts -= 1;
+        } else {
+          GST_WARNING_OBJECT (self, "Cannot regress further;"
+              " forcing timestamp to zero");
+          timestamp = 0;
+        }
       }
+      timestamp += self->base_ts;
+      self->last_ts = timestamp;
     }
-    timestamp += self->base_ts;
-    self->last_ts = timestamp;
 
     gst_buffer_unmap (buffer, &info);
   }

gst-plugins-bad-1.24.8.tar.xz/gst/videofilters/gstscenechange.c -> gst-plugins-bad-1.24.9.tar.xz/gst/videofilters/gstscenechange.c Changed

@@ -60,7 +60,7 @@
  * terms of accuracy, schro's implementation has about 2-3 false positives
  * or false negatives per 100 scene changes.  This implementation has
  * about 5 per 100.  The threshold is tuned for minimum total false
- * positives or negatives, on the assumption that the badness of a 
+ * positives or negatives, on the assumption that the badness of a
  * false negative is the same as a false positive.
  *
  * This algorithm is pretty much at its limit for error rate.  I
@@ -92,6 +92,7 @@
 
 static GstFlowReturn gst_scene_change_transform_frame_ip (GstVideoFilter *
     filter, GstVideoFrame * frame);
+static gboolean gst_scenechange_stop (GstBaseTransform * base);
 
 #undef TESTING
 #ifdef TESTING
@@ -119,6 +120,7 @@
 gst_scene_change_class_init (GstSceneChangeClass * klass)
 {
   GstVideoFilterClass *video_filter_class = GST_VIDEO_FILTER_CLASS (klass);
+  GstBaseTransformClass *trans_class = GST_BASE_TRANSFORM_CLASS (klass);
 
   gst_element_class_add_pad_template (GST_ELEMENT_CLASS (klass),
       gst_pad_template_new ("src", GST_PAD_SRC, GST_PAD_ALWAYS,
@@ -135,6 +137,7 @@
   video_filter_class->transform_frame_ip =
       GST_DEBUG_FUNCPTR (gst_scene_change_transform_frame_ip);
 
+  trans_class->stop = GST_DEBUG_FUNCPTR (gst_scenechange_stop);
 }
 
 static void
@@ -142,7 +145,6 @@
 {
 }
 
-
 static double
 get_frame_score (GstVideoFrame * f1, GstVideoFrame * f2)
 {
@@ -259,10 +261,15 @@
   return GST_FLOW_OK;
 }
 
+static gboolean
+gst_scenechange_stop (GstBaseTransform * base)
+{
+  GstSceneChange *scenechange = GST_SCENE_CHANGE (base);
 
+  gst_clear_buffer (&scenechange->oldbuf);
 
-
-
+  return TRUE;
+}
 
 #ifdef TESTING
 /* This is from ds's personal collection.  No, you can't have it. */

gst-plugins-bad-1.24.8.tar.xz/meson.build -> gst-plugins-bad-1.24.9.tar.xz/meson.build Changed

gst-plugins-bad-1.24.8.tar.xz/po/gst-plugins-bad-1.0.pot -> gst-plugins-bad-1.24.9.tar.xz/po/gst-plugins-bad-1.0.pot Changed

gst-plugins-bad-1.24.8.tar.xz/po/gst-plugins-bad.pot -> gst-plugins-bad-1.24.9.tar.xz/po/gst-plugins-bad.pot Changed

gst-plugins-bad-1.24.8.tar.xz/sys/applemedia/avfdeviceprovider.m -> gst-plugins-bad-1.24.9.tar.xz/sys/applemedia/avfdeviceprovider.m Changed

gst-plugins-bad-1.24.8.tar.xz/sys/applemedia/vtdec.c -> gst-plugins-bad-1.24.9.tar.xz/sys/applemedia/vtdec.c Changed

gst-plugins-bad-1.24.8.tar.xz/sys/kms/gstkmsallocator.c -> gst-plugins-bad-1.24.9.tar.xz/sys/kms/gstkmsallocator.c Changed

gst-plugins-bad-1.24.8.tar.xz/sys/kms/gstkmssink.c -> gst-plugins-bad-1.24.9.tar.xz/sys/kms/gstkmssink.c Changed

gst-plugins-bad-1.24.8.tar.xz/sys/msdk/gstmsdkallocator_libva.c -> gst-plugins-bad-1.24.9.tar.xz/sys/msdk/gstmsdkallocator_libva.c Changed

gst-plugins-bad-1.24.8.tar.xz/sys/msdk/gstmsdkcaps.c -> gst-plugins-bad-1.24.9.tar.xz/sys/msdk/gstmsdkcaps.c Changed

gst-plugins-bad-1.24.8.tar.xz/sys/msdk/gstmsdkdec.c -> gst-plugins-bad-1.24.9.tar.xz/sys/msdk/gstmsdkdec.c Changed

gst-plugins-bad-1.24.8.tar.xz/sys/msdk/gstmsdkenc.c -> gst-plugins-bad-1.24.9.tar.xz/sys/msdk/gstmsdkenc.c Changed

gst-plugins-bad-1.24.8.tar.xz/sys/msdk/gstmsdkvpp.c -> gst-plugins-bad-1.24.9.tar.xz/sys/msdk/gstmsdkvpp.c Changed

gst-plugins-bad-1.24.8.tar.xz/sys/msdk/gstmsdkvpputil.c -> gst-plugins-bad-1.24.9.tar.xz/sys/msdk/gstmsdkvpputil.c Changed

gst-plugins-bad-1.24.8.tar.xz/sys/msdk/meson.build -> gst-plugins-bad-1.24.9.tar.xz/sys/msdk/meson.build Changed

gst-plugins-bad-1.24.8.tar.xz/sys/va/gstvaav1enc.c -> gst-plugins-bad-1.24.9.tar.xz/sys/va/gstvaav1enc.c Changed

gst-plugins-bad-1.24.8.tar.xz/tests/check/elements/h264timestamper.c -> gst-plugins-bad-1.24.9.tar.xz/tests/check/elements/h264timestamper.c Changed

gst-plugins-bad-1.24.8.tar.xz/tests/check/elements/lc3.c -> gst-plugins-bad-1.24.9.tar.xz/tests/check/elements/lc3.c Changed

gst-plugins-bad-1.24.8.tar.xz/tests/check/elements/vapostproc.c -> gst-plugins-bad-1.24.9.tar.xz/tests/check/elements/vapostproc.c Changed

@@ -62,18 +62,73 @@
 
 GST_END_TEST;
 
+static GstCaps *
+get_drmdma_format (void)
+{
+  GstElement *vpp;
+  GstCaps *templ, *allowed_caps, *drm_caps = NULL;
+  GstPad *srcpad;
+  guint i;
+
+  vpp = gst_element_factory_make ("vapostproc", NULL);
+  if (!vpp)
+    return NULL;
+  srcpad = gst_element_get_static_pad (vpp, "src");
+  fail_unless (srcpad != NULL);
+  templ = gst_pad_get_pad_template_caps (srcpad);
+  fail_unless (templ != NULL);
+
+  allowed_caps = gst_caps_normalize (templ);
+
+  for (i = 0; i < gst_caps_get_size (allowed_caps); ++i) {
+    GstStructure *new_structure;
+    GstStructure *structure;
+
+    /* non-dmabuf caps don't describe drm-format: skip them */
+    structure = gst_caps_get_structure (allowed_caps, i);
+    if (!gst_structure_has_field (structure, "drm-format"))
+      continue;
+
+    drm_caps = gst_caps_new_empty ();
+    new_structure = gst_structure_copy (structure);
+    gst_structure_set (new_structure, "framerate", GST_TYPE_FRACTION,
+        1, 1, NULL);
+    gst_structure_remove_field (new_structure, "width");
+    gst_structure_remove_field (new_structure, "height");
+    gst_caps_append_structure (drm_caps, new_structure);
+    gst_caps_set_features_simple (drm_caps,
+        gst_caps_features_new_single ("memory:DMABuf"));
+
+    GST_DEBUG ("have caps %" GST_PTR_FORMAT, drm_caps);
+    /* should be fixed without width/height */
+    fail_unless (gst_caps_is_fixed (drm_caps));
+    break;
+  }
+
+  gst_caps_unref (allowed_caps);
+  gst_object_unref (srcpad);
+  gst_object_unref (vpp);
+
+  return drm_caps;
+}
+
 GST_START_TEST (dmabuf_copy)
 {
   GstHarness *h;
   GstBuffer *buf, *buf_copy;
   gboolean ret;
+  GstCaps *drm_caps;
 
   h = gst_harness_new_parse ("videotestsrc num-buffers=1 ! "
       "video/x-raw, width=(int)1024, height=(int)768 ! vapostproc");
   ck_assert (h);
 
-  gst_harness_set_sink_caps_str (h,
-      "video/x-raw(memory:DMABuf), format=(string)NV12, width=(int)3840, height=(int)2160");
+  drm_caps = get_drmdma_format ();
+  ck_assert (drm_caps);
+  gst_caps_set_simple (drm_caps, "width", G_TYPE_INT, 1600, "height",
+      G_TYPE_INT, 1200, NULL);
+
+  gst_harness_set_sink_caps (h, drm_caps);
 
   gst_harness_add_propose_allocation_meta (h, GST_VIDEO_META_API_TYPE, NULL);
   gst_harness_play (h);

Difference Between Revision 5 and Essentials / gstreamer-plugins-bad-codecs