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gstreamer-plugins-bad-codecs.changes Changed

gstreamer-plugins-bad-codecs.spec Changed

gst-plugins-bad-1.22.5.tar.xz/ChangeLog -> gst-plugins-bad-1.22.6.tar.xz/ChangeLog Changed

@@ -1,3 +1,287 @@
+=== release 1.22.6 ===
+
+2023-09-20 18:10:57 +0100  Tim-Philipp Müller <tim@centricular.com>
+
+	* NEWS:
+	* RELEASE:
+	* gst-plugins-bad.doap:
+	* meson.build:
+	  Release 1.22.6
+
+2023-08-10 15:47:03 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/mxf/mxfd10.c:
+	  mxfdemux: Check number of channels for AES3 audio
+	  Only up to 8 channels are allowed and using a higher number would cause
+	  integer overflows when copying the data, and lead to out of bound
+	  writes.
+	  Also check that each buffer is at least 4 bytes long to avoid another
+	  overflow.
+	  Fixes ZDI-CAN-21661, CVE-2023-40475
+	  Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/2897
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5365>
+
+2023-08-10 15:45:01 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/mxf/mxfup.c:
+	  mxfdemux: Fix integer overflow causing out of bounds writes when handling invalid uncompressed video
+	  Check ahead of time when parsing the track information whether
+	  width, height and bpp are valid and usable without overflows.
+	  Fixes ZDI-CAN-21660, CVE-2023-40474
+	  Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/2896
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5365>
+
+2023-08-09 12:49:19 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+	* gst-libs/gst/codecparsers/gsth265parser.c:
+	  h265parser: Fix possible overflow using max_sub_layers_minus1
+	  This fixes a possible overflow that can be triggered by an invalid value of
+	  max_sub_layers_minus1 being set in the bitstream. The bitstream uses 3 bits,
+	  but the allowed range is 0 to 6 only.
+	  Fixes ZDI-CAN-21768, CVE-2023-40476
+	  Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/2895
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5366>
+
+2023-09-11 18:12:28 +0200  Hugues Fruchet <hugues.fruchet@foss.st.com>
+
+	* gst-libs/gst/wayland/gstwlwindow.c:
+	  waylandsink: Fix cropping for video with non-square aspect ratio
+	  Padding of unaligned content is still visible at right with some aspect-ratio.
+	  Fix this by giving the original content resolution to wp_viewport_set_source()
+	  instead of pixel aspect ratio scaled one.
+	  Fixes !5259
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5360>
+
+2023-09-14 01:18:59 +0900  Seungha Yang <seungha@centricular.com>
+
+	* gst-libs/gst/codecs/gsth264decoder.c:
+	* gst-libs/gst/codecs/gsth264picture.h:
+	  h264decoder: Update latency dynamically
+	  The actual number of reorder frames is unknown
+	  unless frame reordering is disabled
+	  (e.g., POC type 2 or constrained-* profiles).
+	  Also derived maximum DPB size or max_num_reorder_frames in VUI
+	  is not the upper bound of output delay.
+	  Fixes: https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/2702
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5341>
+
+2023-08-16 13:43:56 +0000  L. E. Segovia <amy@amyspark.me>
+
+	* sys/applemedia/avsamplevideosink.m:
+	  applemedia: Also fix inconsistent pixel format definition for NV12
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5203>
+
+2023-08-15 21:45:56 +0000  L. E. Segovia <amy@amyspark.me>
+
+	* sys/applemedia/coremediabuffer.c:
+	* sys/applemedia/corevideobuffer.c:
+	  applemedia: Fix pixel format for I420
+	  In Intel Macs, using full range 8-bit 4:2:0 YCbCr results in a failure on
+	  initialization. I've validated this to be the correct pixel format with FFmpeg:
+	  https://github.com/FFmpeg/FFmpeg/blob/8653dcaf7d665b15b40ea9a560c8171b0914a882/libavutil/hwcontext_videotoolbox.c#L45
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5203>
+
+2023-08-24 12:12:09 +0200  Víctor Manuel Jáquez Leal <vjaquez@igalia.com>
+
+	* sys/va/gstvadecoder.c:
+	* sys/va/gstvadisplay_priv.c:
+	  va: Fix in error logs functions mismatches
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5340>
+
+2023-09-15 11:06:52 +1000  Matthew Waters <matthew@centricular.com>
+
+	* sys/androidmedia/gstamcvideoenc.c:
+	  androidmedia/enc: handle codec-data before popping GstVideoCodecFrames
+	  Issue is that when amc was producing a codec-data buffer, a
+	  GstVideoCodecFrame was being popped off the internal queue.  This meant
+	  that the codec-data was being associated with the first input frame and
+	  the second (first encoded buffer) output buffer with the second input
+	  frame.  At the end (assuming one input produces one output which seems
+	  to hold in my testing and how the encoder is currently implemented)
+	  there would be an input frame missing and would be pushed without any
+	  timing information.  This would lead to e.g. muxers rejecting the buffer
+	  without PTS and failing to mux.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5335>
+
+2023-09-15 11:06:07 +1000  Matthew Waters <matthew@centricular.com>
+
+	* sys/androidmedia/gstamc-constants.h:
+	* sys/androidmedia/gstamcvideoenc.c:
+	  androidmedia/enc: add fixme log about partial frames
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5335>
+
+2023-09-16 03:13:33 +0900  Seungha Yang <seungha@centricular.com>
+
+	* gst-libs/gst/codecparsers/gstav1parser.c:
+	  av1parser: Fix segmentation params update
+	  Even if the segmentation feature value is not updated,
+	  the parsed "segmentation_update_map" and "segmentation_temporal_update"
+	  values should not be cleared as it's referenced during lower
+	  level bitstream parsing. Also, don't use assert() in parser
+	  unless it's clearly impossible condition.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5336>
+
+2023-08-24 17:40:42 +0200  Jan Alexander Steffens (heftig) <jan.steffens@ltnglobal.com>
+
+	* gst/rtmp2/rtmp/rtmpclient.c:
+	  rtmp2: Allow NULL flash version, omitting the field
+	  rtmpsink omits it by default. Allow us to do the same.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5248>
+
+2023-09-05 14:15:04 +0200  Thomas Schneider <thomas.schneider@voliro.ch>
+
+	* sys/androidmedia/gstamc.c:
+	  androidmedia: fix hevc codec profile registration
+	  Fix the codec registration logic such that all supported
+	  profiles are available instead of just the first in the
+	  list.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5313>
+
+2023-09-03 13:21:30 +0000  Akihiro Sagawa <sagawa.aki@gmail.com>
+
+	* gst-libs/gst/codecparsers/gstmpegvideoparser.c:
+	  codecparsers: Fix MPEG-1 aspect ratio table
+	  The values defined in ISO/IEC 11172-2 are different from those used so far.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5300>
+
+2023-08-29 14:55:03 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+	* gst-libs/gst/wayland/gstwlwindow.c:
+	  waylandsink: Crop surfaces to their display width height
+	  Setting the surface source rectangle has been omitted so far. As a side effect
+	  surface created with padded width/height are being scaled down. Fix this using
+	  the viewporter source rectangle configuration. This can later be enhanced
+	  to support crop meta.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5298>
+
+2023-08-28 20:58:22 +0900  Seungha Yang <seungha@centricular.com>
+
+	* sys/d3d11/gstd3d11convert.cpp:
+	  d3d11convert: Passthrough allocation query on same caps
+	  Since d3d11convert and its variant elements does not enable basetransform's
+	  passthrough, passthrough allocation query needs to be handled
+	  manually in order to respect downstream element's min/max buffer
+	  requirement.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5297>
+
+2023-08-23 19:27:43 +0900  Seungha Yang <seungha@centricular.com>
+
+	* sys/nvcodec/gstnvencoder.cpp:
+	  nvencoder: Fix negotiation error when interlace-mode is unspecified
+	  Use GST_PAD_SET_ACCEPT_INTERSECT() to accept caps without interlace-mode
+	  field
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5240>
+
+2023-08-23 18:07:31 +0900  Seungha Yang <seungha@centricular.com>
+
+	* sys/win32ipc/protocol/win32ipcpipeserver.cpp:
+	  win32ipc: Fix pipe handle leak
+	  Named pipe handle must be closed if it's no longer needed
+	  Fixes: https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/2923
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5231>
+
+2023-08-22 02:57:24 +0900  Seungha Yang <seungha@centricular.com>
+
+	* gst/videoparsers/gsth265parse.c:
+	* tests/check/elements/h265parse.c:
+	  h265parse: Allow partially broken hvcC data
+	  Ignores parsing error on the last nalu of the array if the nalu type
+	  is not VPS/SPS/PPS
+	  Fixes: https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/2905
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5220>
+
+2023-08-18 18:21:18 +1000  Jan Schmidt <jan@centricular.com>
+
+	* ext/mdns/gstmicrodnsdevice.c:
+	  mdns: Fix a crash on context error
+	  Make sure not to free the microdns provider context until the
+	  device provider asks it to stop. Fixes a crash if there is
+	  an error (such as MDNS port being busy) that makes the
+	  mdns listener exit early.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5207>
+
+2023-08-18 09:27:36 +0200  Guillaume Desmottes <guillaume.desmottes@onestream.live>
+
+	* gst/rtmp2/gstrtmp2sink.c:
+	  rtmp2sink: fix crash if message conversion failed
+	  The message pointer is not set so we can't display it in logs.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5202>
+
+2023-08-08 01:29:40 +0900  Seungha Yang <seungha@centricular.com>
+
+	* ext/hls/gsthlssink2.c:
+	  hlssink2: Always use forward slash separator
+	  g_build_filename() will insert back slash on Windows, and resulting
+	  playlist will contain media segment path with back slash if
+	  "playlist-root" property is specified
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5161>
+
+2023-08-05 01:39:29 +1000  Jan Schmidt <jan@centricular.com>
+
+	* gst/audiolatency/gstaudiolatency.c:
+	  audiolatency: Fix event refcounting bug handling latency events
+	  Fix a refcounting bug introduced in
+	  https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5146
+	  If upstream returns FALSE when processing a latency event, it will
+	  be unreffed an extra time
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5152>
+
+2023-08-04 17:25:26 +1000  Jan Schmidt <jan@centricular.com>
+
+	* gst/audiolatency/gstaudiolatency.c:
+	  audiolatency: Forward latency query and event upstream
+	  Make sure the pipeline still configures the latency that it would configure
+	  if audiolatency was not in the pipeline.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5152>
+
+2023-07-19 18:08:32 -0500  Ryan Pavlik <ryan.pavlik@collabora.com>
+
+	* sys/androidmedia/gstjniutils.c:
+	  androidmedia: Add more null checks (of env) to JNI utilities
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5142>
+
+2023-07-22 10:42:39 +0100  Philippe Normand <philn@igalia.com>
+
+	* gst/transcode/gsttranscodebin.c:
+	  transcodebin: Fixes for upstream selectable support
+	  The upstream selectable query was not performed in all situations where we
+	  handle the stream-start event. This could potentially lead to unlinked pads
+	  between decodebin3 and encodebin later on.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5141>
+
+2023-07-19 18:08:32 -0500  Ryan Pavlik <ryan.pavlik@collabora.com>
+
+	* sys/androidmedia/gstjniutils.c:
+	  androidmedia: Add more null checks to JNI utilities
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5136>
+
+2023-07-28 14:02:22 -0500  Ryan Pavlik <ryan.pavlik@collabora.com>
+
+	* sys/androidmedia/gstamcvideodec.c:
+	  androidmedia: Fix typo
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5136>
+
+2023-07-19 18:08:16 -0500  Ryan Pavlik <ryan.pavlik@collabora.com>
+
+	* sys/androidmedia/gst-android-hardware-camera.c:
+	  androidmedia: Clear err if we don't have an optional camera field/constant
+	  Fixes startup on devices where those fields/constants are not found.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5136>
+
+2023-07-26 16:45:18 -0500  Ryan Pavlik <ryan.pavlik@collabora.com>
+
+	* ext/webrtc/gstwebrtcbin.c:
+	  webrtc: Fix docs for create-data-channel action signal
+	  Initial line of the doc comment was incorrect, so the nicely written
+	  docs were not being extracted.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5134>
+
+2023-07-20 16:57:47 +0100  Tim-Philipp Müller <tim@centricular.com>
+
+	* meson.build:
+	  Back to development
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5082>
+
 === release 1.22.5 ===
 
 2023-07-20 15:22:48 +0100  Tim-Philipp Müller <tim@centricular.com>

gst-plugins-bad-1.22.5.tar.xz/NEWS -> gst-plugins-bad-1.22.6.tar.xz/NEWS Changed

@@ -2189,6 +2189,183 @@
 -   List of Merge Requests applied in 1.22.5
 -   List of Issues fixed in 1.22.5
 
+1.22.6
+
+The sixth 1.22 bug-fix release (1.22.6) was released on 20 September
+2023.
+
+This release only contains bugfixes and security fixes and it should be
+safe to update from 1.22.x.
+
+Highlighted bugfixes in 1.22.6
+
+-   Security fixes for the MXF demuxer and H.265 video parser
+-   Fix latency regression in H.264 hardware decoder base class
+-   androidmedia: fix HEVC codec profile registration and fix coded_data
+    handling
+-   decodebin3: fix switching from a raw stream to an encoded stream
+-   gst-inspect: prettier and more correct signal and action signals
+    printing
+-   rtmp2: Allow NULL flash version, omitting the field, for better RTMP
+    server compatibility
+-   rtspsrc: better compatibility with buggy RTSP servers that don’t set
+    a clock-rate
+-   rtpjitterbuffer: fix integer overflow that led to more packets being
+    declared lost than have been lost
+-   v4l2: fix video encoding regression on RPi and fix support for left
+    and top padding
+-   waylandsink: Crop surfaces to their display width height
+-   cerbero: recognise Manjaro; add Rust support for MSVC ARM64; cmake
+    detection fixes
+-   various bug fixes, build fixes, memory leak fixes, and other
+    stability and reliability improvements
+
+gstreamer
+
+-   gst-inspect: prettier and more correct signal printing, and print
+    action signals in g_signal_emit_by_name() format
+-   gst-launch: Disable fault signal handlers on macOS
+
+gst-plugins-base
+
+-   audio: Make sure to stop ringbuffer on error
+-   decodebin3: avoid identity, sinkpad, parsebin leakage when reset
+    input
+-   decodebin3: Ensure the slot is unlinked before linking to decoder
+-   sdp: fix wrong debug log error message for missing clock-rate in
+    caps
+-   sdp: Parse zero clock-rate as default
+
+gst-plugins-good
+
+-   adaptivedemux2: fix memory leak
+-   pulsedeviceprovider: fix incorrect usage of GST_ELEMENT_ERROR
+-   qt: Unbreak build with qt-egl enabled but viv_fb missing
+-   qt: Fix searching of qt5/qt6 tools with qmake in Meson
+-   qtdemux: Fix premature EOS when some files are played in push mode
+-   qtdemux: attach cbcs crypt info at the right moment
+-   rtpjitterbuffer: Avoid integer overflow in max saveable packets
+    calculation with negative offset
+-   videoflip: fix concurrent access when modifying the tag list
+-   v4l2: allocator: Don’t close foreign dmabuf
+-   v4l2: bufferpool: Fix large encoded stream regression
+-   v4l2: bufferpool: Problems when checking for truncated buffer
+-   v4l2: Fix support for left and top padding
+-   v4l2object: clear format lists if source change event is received
+
+gst-plugins-bad
+
+-   androidmedia/enc: handle codec-data before popping
+    GstVideoCodecFrames
+-   androidmedia: fix hevc codec profile registration
+-   androidmedia: Small fixes
+-   androidmedia: Add more null checks (of env) to JNI utilities
+-   applemedia: Fix pixel format for I420 and NV12
+-   audiolatency: Forward latency query and event upstream
+-   av1parser: Fix segmentation params update
+-   codecparsers: Fix MPEG-1 aspect ratio table
+-   d3d11convert: Passthrough allocation query on same caps
+-   h264decoder: Update latency dynamically
+-   h265parser: Allow partially broken hvcC data
+-   h265parser: Fix possible overflow using max_sub_layers_minus1
+-   hlssink2: Always use forward slash separator
+-   mdns: Fix a crash on context error
+-   mxfdemux: Fix integer overflow causing out of bounds writes when
+    handling invalid uncompressed video and check channels for AES3
+-   nvencoder: Fix negotiation error when interlace-mode is unspecified
+-   rtmp2: Allow NULL flash version, omitting the field
+-   rtmp2sink: fix crash if message conversion failed
+-   transcodebin: Fixes for upstream selectable support
+-   va: Fix in error logs functions mismatches
+-   waylandsink: Crop surfaces to their display width height
+-   waylandsink: Fix cropping for video with non-square aspect ratio
+-   webrtc: Fix docs for create-data-channel action signal
+-   win32ipc: Fix pipe handle leak
+
+gst-plugins-ugly
+
+-   No changes
+
+gst-plugins-rs
+
+-   fallbackswitch: locking/deadlock fixes
+-   onvifmetadataparse: Skip metadata frames with unrepresentable UTC
+    time
+-   transcriberbin: Configure audioresample in front of transcriber
+-   webrtcsink: Propagate GstContext messages
+-   webrtcsink: Add support for d3d11 memory and qsvh264enc
+-   webrtcsink: fix TWCC extension adding
+-   webrtcsink: don’t forget to setup encoders for discoveries
+-   webrtcsink: NVIDIA V4L2 encoders always require NVMM memory
+-   meson: Fix handling of optional deps, and don’t require Python 3.8
+
+gst-libav
+
+-   No changes
+
+gst-rtsp-server
+
+-   No changes
+
+gstreamer-vaapi
+
+-   No changes
+
+gstreamer-sharp
+
+-   No changes
+
+gst-omx
+
+-   No changes
+
+gst-python
+
+-   No changes
+
+gst-editing-services
+
+-   No changes
+
+gst-validate + gst-integration-testsuites
+
+-   gst-validate: Disable fault signal handlers on macOS
+
+gst-examples
+
+-   No changes
+
+Development build environment
+
+-   macos-bison: Update to 3.8.2 and add an ARM64 build
+-   wrap: update libpsl to 0.21.2
+
+Cerbero build tool and packaging changes in 1.22.6
+
+-   Add Rust support for MSVC ARM64
+-   Recognise PERL5LIB as a joinable Unix variable
+-   Recognise Manjaro as an Arch derivative
+-   Fix picking up cmake from build-tools
+
+Contributors to 1.22.6
+
+Akihiro Sagawa, Alicia Boya García, Guillaume Desmottes, Haihua Hu,
+Hugues Fruchet, Ivan Molodetskikh, Jan Alexander Steffens (heftig), Jan
+Schmidt, L. E. Segovia, Mathieu Duponchelle, Matthew Waters, Ming Qian,
+Nicolas Dufresne, Nirbheek Chauhan, Olivier Blin, Olivier Crête,
+Philippe Normand, Piotr Brzeziński, Robert Ayrapetyan, Ryan Pavlik,
+Sebastian Dröge, Seungha Yang, Stéphane Cerveau, Stephan Seitz, Thomas
+Schneider, Tim-Philipp Müller, Víctor Manuel Jáquez Leal, Wang Chuan,
+Xabier Rodriguez Calvar,
+
+… and many others who have contributed bug reports, translations, sent
+suggestions or helped testing. Thank you all!
+
+List of merge requests and issues fixed in 1.22.6
+
+-   List of Merge Requests applied in 1.22.6
+-   List of Issues fixed in 1.22.6
+
 Schedule for 1.24
 
 Our next major feature release will be 1.24, and 1.23 will be the

gst-plugins-bad-1.22.5.tar.xz/RELEASE -> gst-plugins-bad-1.22.6.tar.xz/RELEASE Changed

gst-plugins-bad-1.22.5.tar.xz/ext/hls/gsthlssink2.c -> gst-plugins-bad-1.22.6.tar.xz/ext/hls/gsthlssink2.c Changed

gst-plugins-bad-1.22.5.tar.xz/ext/mdns/gstmicrodnsdevice.c -> gst-plugins-bad-1.22.6.tar.xz/ext/mdns/gstmicrodnsdevice.c Changed

@@ -44,6 +44,7 @@
 struct _ListenerContext
 {
   GMutex lock;
+  GCond stop_cond;
   GstDeviceProvider *provider;
 
   /* The following fields are protected by @lock */
@@ -362,9 +363,18 @@
 done:
   GST_INFO_OBJECT (ctx->provider, "Done listening");
 
+  /* Wait until we're told to stop, or gst_mdns_device_provider_stop()
+     can access a freed context */
+  g_mutex_lock (&ctx->lock);
+  while (!ctx->stop) {
+    g_cond_wait (&ctx->stop_cond, &ctx->lock);
+  }
+  g_mutex_unlock (&ctx->lock);
+
   g_sequence_free (ctx->last_seen_devices);
   g_hash_table_unref (ctx->devices);
   g_mutex_clear (&ctx->lock);
+  g_cond_clear (&ctx->stop_cond);
   g_free (ctx);
 
   return NULL;
@@ -385,6 +395,7 @@
   ListenerContext *ctx = g_new0 (ListenerContext, 1);
 
   g_mutex_init (&ctx->lock);
+  g_cond_init (&ctx->stop_cond);
   ctx->provider = provider;
   ctx->devices =
       g_hash_table_new_full (g_str_hash, g_str_equal, g_free, g_object_unref);
@@ -405,6 +416,7 @@
 
   g_mutex_lock (&self->current_ctx->lock);
   self->current_ctx->stop = true;
+  g_cond_broadcast (&self->current_ctx->stop_cond);
   g_mutex_unlock (&self->current_ctx->lock);
 
   self->current_ctx = NULL;

gst-plugins-bad-1.22.5.tar.xz/ext/webrtc/gstwebrtcbin.c -> gst-plugins-bad-1.22.6.tar.xz/ext/webrtc/gstwebrtcbin.c Changed

gst-plugins-bad-1.22.5.tar.xz/gst-libs/gst/codecparsers/gstav1parser.c -> gst-plugins-bad-1.22.6.tar.xz/gst-libs/gst/codecparsers/gstav1parser.c Changed

@@ -2129,20 +2129,37 @@
         }
       }
     } else {
+      gint8 ref_idx;
+      GstAV1SegmenationParams *ref_seg_params;
+
       /* Copy it from prime_ref */
-      g_assert (frame_header->primary_ref_frame != GST_AV1_PRIMARY_REF_NONE);
-      g_assert (parser->state.ref_info.
-          entryframe_header->ref_frame_idxframe_header->primary_ref_frame.
-          ref_valid);
-      memcpy (seg_params,
-          &parser->state.ref_info.
-          entryframe_header->ref_frame_idxframe_header->
-                  primary_ref_frame.ref_segmentation_params,
-          sizeof (GstAV1SegmenationParams));
+      if (frame_header->primary_ref_frame >= GST_AV1_PRIMARY_REF_NONE) {
+        GST_WARNING ("Invalid primary_ref_frame %d",
+            frame_header->primary_ref_frame);
+        return GST_AV1_PARSER_BITSTREAM_ERROR;
+      }
 
-      seg_params->segmentation_update_map = 0;
-      seg_params->segmentation_temporal_update = 0;
-      seg_params->segmentation_update_data = 0;
+      ref_idx = frame_header->ref_frame_idxframe_header->primary_ref_frame;
+      if (ref_idx >= GST_AV1_NUM_REF_FRAMES || ref_idx < 0) {
+        GST_WARNING ("Invalid ref_frame_idx %d", ref_idx);
+        return GST_AV1_PARSER_BITSTREAM_ERROR;
+      }
+
+      if (!parser->state.ref_info.entryref_idx.ref_valid) {
+        GST_WARNING ("Reference frame at index %d is unavailable", ref_idx);
+        return GST_AV1_PARSER_BITSTREAM_ERROR;
+      }
+
+      ref_seg_params =
+          &parser->state.ref_info.entryref_idx.ref_segmentation_params;
+
+      for (i = 0; i < GST_AV1_MAX_SEGMENTS; i++) {
+        for (j = 0; j < GST_AV1_SEG_LVL_MAX; j++) {
+          seg_params->feature_enabledij =
+              ref_seg_params->feature_enabledij;
+          seg_params->feature_dataij = ref_seg_params->feature_dataij;
+        }
+      }
     }
   } else {
     seg_params->segmentation_update_map = 0;

gst-plugins-bad-1.22.5.tar.xz/gst-libs/gst/codecparsers/gsth265parser.c -> gst-plugins-bad-1.22.6.tar.xz/gst-libs/gst/codecparsers/gsth265parser.c Changed

gst-plugins-bad-1.22.5.tar.xz/gst-libs/gst/codecparsers/gstmpegvideoparser.c -> gst-plugins-bad-1.22.6.tar.xz/gst-libs/gst/codecparsers/gstmpegvideoparser.c Changed

gst-plugins-bad-1.22.5.tar.xz/gst-libs/gst/codecs/gsth264decoder.c -> gst-plugins-bad-1.22.6.tar.xz/gst-libs/gst/codecs/gsth264decoder.c Changed

@@ -152,6 +152,12 @@
   GstQueueArray *output_queue;
 
   gboolean input_state_changed;
+
+  /* Latency report params */
+  guint32 max_reorder_count;
+  guint32 last_reorder_frame_number;
+  gint fps_n;
+  gint fps_d;
 };
 
 typedef struct
@@ -407,6 +413,17 @@
 }
 
 static void
+gst_h264_decoder_reset_latency_infos (GstH264Decoder * self)
+{
+  GstH264DecoderPrivate *priv = self->priv;
+
+  priv->max_reorder_count = 0;
+  priv->last_reorder_frame_number = 0;
+  priv->fps_n = 25;
+  priv->fps_d = 1;
+}
+
+static void
 gst_h264_decoder_reset (GstH264Decoder * self)
 {
   GstH264DecoderPrivate *priv = self->priv;
@@ -420,6 +437,8 @@
   priv->width = 0;
   priv->height = 0;
   priv->nal_length_size = 4;
+
+  gst_h264_decoder_reset_latency_infos (self);
 }
 
 static gboolean
@@ -1285,6 +1304,9 @@
         gst_h264_picture_unref (picture);
         return ret;
       }
+
+      priv->last_reorder_frame_number++;
+      picture->reorder_frame_number = priv->last_reorder_frame_number;
     }
 
     /* This allows accessing the frame from the picture. */
@@ -1788,6 +1810,28 @@
 
   priv->last_output_poc = picture->pic_order_cnt;
 
+  if (priv->last_reorder_frame_number > picture->reorder_frame_number) {
+    guint64 diff = priv->last_reorder_frame_number -
+        picture->reorder_frame_number;
+    guint64 total_delay = diff + priv->preferred_output_delay;
+    if (diff > priv->max_reorder_count && total_delay < G_MAXUINT32) {
+      GstClockTime latency;
+
+      priv->max_reorder_count = (guint32) diff;
+      latency = gst_util_uint64_scale_int (GST_SECOND * total_delay,
+          priv->fps_d, priv->fps_n);
+
+      if (latency != G_MAXUINT64) {
+        GST_DEBUG_OBJECT (self, "Updating latency to %" GST_TIME_FORMAT
+            ", reorder count: %" G_GUINT64_FORMAT ", output-delay: %u",
+            GST_TIME_ARGS (latency), diff, priv->preferred_output_delay);
+
+        gst_video_decoder_set_latency (GST_VIDEO_DECODER (self),
+            latency, latency);
+      }
+    }
+  }
+
   frame = gst_video_decoder_get_frame (GST_VIDEO_DECODER (self),
       picture->system_frame_number);
 
@@ -2326,16 +2370,14 @@
 
   bump_level = get_bump_level (self);
   if (bump_level != GST_H264_DPB_BUMP_NORMAL_LATENCY) {
-    if (sps->pic_order_cnt_type == 2) {
-      /* POC type 2 has does not allow frame reordering */
-      frames_delay = 0;
-    } else {
-      guint32 max_reorder_frames =
-          gst_h264_dpb_get_max_num_reorder_frames (priv->dpb);
-      frames_delay = MIN (max_dpb_size, max_reorder_frames);
-    }
+    GST_DEBUG_OBJECT (self, "Actual latency will be updated later");
+    frames_delay = 0;
   }
 
+  priv->max_reorder_count = frames_delay;
+  priv->fps_n = fps_n;
+  priv->fps_d = fps_d;
+
   /* Consider output delay wanted by subclass */
   frames_delay += priv->preferred_output_delay;
 
@@ -2450,6 +2492,8 @@
     if (ret != GST_FLOW_OK)
       return ret;
 
+    gst_h264_decoder_reset_latency_infos (self);
+
     g_assert (klass->new_sequence);
 
     if (klass->get_preferred_output_delay) {

gst-plugins-bad-1.22.5.tar.xz/gst-libs/gst/codecs/gsth264picture.h -> gst-plugins-bad-1.22.6.tar.xz/gst-libs/gst/codecs/gsth264picture.h Changed

gst-plugins-bad-1.22.5.tar.xz/gst-libs/gst/wayland/gstwlwindow.c -> gst-plugins-bad-1.22.6.tar.xz/gst-libs/gst/wayland/gstwlwindow.c Changed

@@ -63,6 +63,9 @@
   /* the size of the video in the buffers */
   gint video_width, video_height;
 
+  /* video width scaled according to par */
+  gint scaled_width;
+
   enum wl_output_transform buffer_transform;
 
   /* when this is not set both the area_surface and the video_surface are not
@@ -431,7 +434,7 @@
     case WL_OUTPUT_TRANSFORM_180:
     case WL_OUTPUT_TRANSFORM_FLIPPED:
     case WL_OUTPUT_TRANSFORM_FLIPPED_180:
-      src.w = priv->video_width;
+      src.w = priv->scaled_width;
       src.h = priv->video_height;
       break;
     case WL_OUTPUT_TRANSFORM_90:
@@ -439,7 +442,7 @@
     case WL_OUTPUT_TRANSFORM_FLIPPED_90:
     case WL_OUTPUT_TRANSFORM_FLIPPED_270:
       src.w = priv->video_height;
-      src.h = priv->video_width;
+      src.h = priv->scaled_width;
       break;
   }
 
@@ -449,6 +452,9 @@
   /* center the video_subsurface inside area_subsurface */
   if (priv->video_viewport) {
     gst_video_center_rect (&src, &dst, &res, TRUE);
+    wp_viewport_set_source (priv->video_viewport, wl_fixed_from_int (0),
+        wl_fixed_from_int (0), wl_fixed_from_int (priv->video_width),
+        wl_fixed_from_int (priv->video_height));
     wp_viewport_set_destination (priv->video_viewport, res.w, res.h);
   } else {
     gst_video_center_rect (&src, &dst, &res, FALSE);
@@ -494,8 +500,9 @@
   GstWlWindowPrivate *priv = gst_wl_window_get_instance_private (self);
 
   if (G_UNLIKELY (info)) {
-    priv->video_width =
+    priv->scaled_width =
         gst_util_uint64_scale_int_round (info->width, info->par_n, info->par_d);
+    priv->video_width = info->width;
     priv->video_height = info->height;
 
     wl_subsurface_set_sync (priv->video_subsurface);
@@ -608,14 +615,14 @@
   if (!priv->configured)
     return;
 
-  if (priv->video_width != 0) {
+  if (priv->scaled_width != 0) {
     wl_subsurface_set_sync (priv->video_subsurface);
     gst_wl_window_resize_video_surface (self, TRUE);
   }
 
   wl_surface_commit (priv->area_surface_wrapper);
 
-  if (priv->video_width != 0)
+  if (priv->scaled_width != 0)
     wl_subsurface_set_desync (priv->video_subsurface);
 }

gst-plugins-bad-1.22.5.tar.xz/gst-plugins-bad.doap -> gst-plugins-bad-1.22.6.tar.xz/gst-plugins-bad.doap Changed

gst-plugins-bad-1.22.5.tar.xz/gst/audiolatency/gstaudiolatency.c -> gst-plugins-bad-1.22.6.tar.xz/gst/audiolatency/gstaudiolatency.c Changed

@@ -227,7 +227,9 @@
 
   templ = gst_static_pad_template_get (&src_template);
   srcpad = gst_element_get_static_pad (self->audiosrc, "src");
-  gst_pad_add_probe (srcpad, GST_PAD_PROBE_TYPE_BUFFER,
+  gst_pad_add_probe (srcpad,
+      GST_PAD_PROBE_TYPE_BUFFER | GST_PAD_PROBE_TYPE_QUERY_UPSTREAM |
+      GST_PAD_PROBE_TYPE_EVENT_UPSTREAM,
       (GstPadProbeCallback) gst_audiolatency_src_probe, self, NULL);
 
   self->srcpad = gst_ghost_pad_new_from_template ("src", srcpad, templ);
@@ -383,7 +385,7 @@
 }
 
 static GstPadProbeReturn
-gst_audiolatency_src_probe (GstPad * pad, GstPadProbeInfo * info,
+gst_audiolatency_src_probe_buffer (GstPad * pad, GstPadProbeInfo * info,
     gpointer user_data)
 {
   GstAudioLatency *self = user_data;
@@ -425,6 +427,46 @@
   return GST_PAD_PROBE_OK;
 }
 
+static GstPadProbeReturn
+gst_audiolatency_src_probe (GstPad * pad, GstPadProbeInfo * info,
+    gpointer user_data)
+{
+  GstAudioLatency *self = user_data;
+
+  if (info->type & GST_PAD_PROBE_TYPE_BUFFER) {
+    return gst_audiolatency_src_probe_buffer (pad, info, user_data);
+  } else if (info->type & GST_PAD_PROBE_TYPE_QUERY_UPSTREAM) {
+    GstQuery *query = gst_pad_probe_info_get_query (info);
+
+    /* Forward latency query to the upstream sinkpad */
+    if (GST_QUERY_TYPE (query) == GST_QUERY_LATENCY) {
+      gboolean res = gst_pad_peer_query (self->sinkpad, query);
+      GST_LOG_OBJECT (self,
+          "Forwarded latency query to sinkpad. Result %d %" GST_PTR_FORMAT, res,
+          query);
+      return res ? GST_PAD_PROBE_HANDLED : GST_PAD_PROBE_DROP;
+    }
+  } else if (info->type & GST_PAD_PROBE_TYPE_EVENT_UPSTREAM) {
+    GstEvent *event = gst_pad_probe_info_get_event (info);
+
+    if (GST_EVENT_TYPE (event) == GST_EVENT_LATENCY) {
+      gboolean res = gst_pad_push_event (self->sinkpad, event);
+
+      GST_LOG_OBJECT (self,
+          "Forwarded latency event to sinkpad. Result %d %" GST_PTR_FORMAT, res,
+          event);
+      if (!res) {
+        /* This doesn't actually do anything - pad probe handling ignores
+         * it, but maybe one day */
+        GST_PAD_PROBE_INFO_FLOW_RETURN (info) = GST_FLOW_ERROR;
+      }
+      return GST_PAD_PROBE_HANDLED;
+    }
+  }
+
+  return GST_PAD_PROBE_OK;
+}
+
 static GstFlowReturn
 gst_audiolatency_sink_chain (GstPad * pad, GstObject * parent,
     GstBuffer * buffer)

gst-plugins-bad-1.22.5.tar.xz/gst/mxf/mxfd10.c -> gst-plugins-bad-1.22.6.tar.xz/gst/mxf/mxfd10.c Changed

gst-plugins-bad-1.22.5.tar.xz/gst/mxf/mxfup.c -> gst-plugins-bad-1.22.6.tar.xz/gst/mxf/mxfup.c Changed

@@ -134,6 +134,8 @@
     gpointer mapping_data, GstBuffer ** outbuf)
 {
   MXFUPMappingData *data = mapping_data;
+  gsize expected_in_stride = 0, out_stride = 0;
+  gsize expected_in_size = 0, out_size = 0;
 
   /* SMPTE 384M 7.1 */
   if (key->u12 != 0x15 || (key->u14 != 0x01 && key->u14 != 0x02
@@ -162,22 +164,25 @@
     }
   }
 
-  if (gst_buffer_get_size (buffer) != data->bpp * data->width * data->height) {
+  // Checked for overflows when parsing the descriptor
+  expected_in_stride = data->bpp * data->width;
+  out_stride = GST_ROUND_UP_4 (expected_in_stride);
+  expected_in_size = expected_in_stride * data->height;
+  out_size = out_stride * data->height;
+
+  if (gst_buffer_get_size (buffer) != expected_in_size) {
     GST_ERROR ("Invalid buffer size");
     gst_buffer_unref (buffer);
     return GST_FLOW_ERROR;
   }
 
-  if (data->bpp != 4
-      || GST_ROUND_UP_4 (data->width * data->bpp) != data->width * data->bpp) {
+  if (data->bpp != 4 || out_stride != expected_in_stride) {
     guint y;
     GstBuffer *ret;
     GstMapInfo inmap, outmap;
     guint8 *indata, *outdata;
 
-    ret =
-        gst_buffer_new_and_alloc (GST_ROUND_UP_4 (data->width * data->bpp) *
-        data->height);
+    ret = gst_buffer_new_and_alloc (out_size);
     gst_buffer_map (buffer, &inmap, GST_MAP_READ);
     gst_buffer_map (ret, &outmap, GST_MAP_WRITE);
     indata = inmap.data;
@@ -185,8 +190,8 @@
 
     for (y = 0; y < data->height; y++) {
       memcpy (outdata, indata, data->width * data->bpp);
-      outdata += GST_ROUND_UP_4 (data->width * data->bpp);
-      indata += data->width * data->bpp;
+      outdata += out_stride;
+      indata += expected_in_stride;
     }
 
     gst_buffer_unmap (buffer, &inmap);
@@ -394,6 +399,36 @@
     return NULL;
   }
 
+  if (caps) {
+    MXFUPMappingData *data = *mapping_data;
+    gsize expected_in_stride = 0, out_stride = 0;
+    gsize expected_in_size = 0, out_size = 0;
+
+    // Do some checking of the parameters to see if they're valid and
+    // we can actually work with them.
+    if (data->image_start_offset > data->image_end_offset) {
+      GST_WARNING ("Invalid image start/end offset");
+      g_free (data);
+      *mapping_data = NULL;
+      gst_clear_caps (&caps);
+
+      return NULL;
+    }
+
+    if (!g_size_checked_mul (&expected_in_stride, data->bpp, data->width) ||
+        (out_stride = GST_ROUND_UP_4 (expected_in_stride)) < expected_in_stride
+        || !g_size_checked_mul (&expected_in_size, expected_in_stride,
+            data->height)
+        || !g_size_checked_mul (&out_size, out_stride, data->height)) {
+      GST_ERROR ("Invalid resolution or bit depth");
+      g_free (data);
+      *mapping_data = NULL;
+      gst_clear_caps (&caps);
+
+      return NULL;
+    }
+  }
+
   return caps;
 }

gst-plugins-bad-1.22.5.tar.xz/gst/rtmp2/gstrtmp2sink.c -> gst-plugins-bad-1.22.6.tar.xz/gst/rtmp2/gstrtmp2sink.c Changed

gst-plugins-bad-1.22.5.tar.xz/gst/rtmp2/rtmp/rtmpclient.c -> gst-plugins-bad-1.22.6.tar.xz/gst/rtmp2/rtmp/rtmpclient.c Changed

gst-plugins-bad-1.22.5.tar.xz/gst/transcode/gsttranscodebin.c -> gst-plugins-bad-1.22.6.tar.xz/gst/transcode/gsttranscodebin.c Changed

@@ -124,6 +124,9 @@
 
 #define DEFAULT_AVOID_REENCODING   FALSE
 
+GST_DEBUG_CATEGORY_STATIC(gst_transcodebin_debug);
+#define GST_CAT_DEFAULT gst_transcodebin_debug
+
 G_DEFINE_TYPE (GstTranscodeBin, gst_transcode_bin, GST_TYPE_BIN);
 GST_ELEMENT_REGISTER_DEFINE_WITH_CODE (transcodebin, "transcodebin", GST_RANK_NONE,
       GST_TYPE_TRANSCODE_BIN, transcodebin_element_init (plugin));
@@ -303,6 +306,9 @@
   TranscodingStream *res = NULL;
 
   GST_OBJECT_LOCK (self);
+  GST_DEBUG_OBJECT (self,
+      "Looking for stream %s in %u existing transcoding streams",
+      stream_id, self->transcoding_streams->len);
   for (i = 0; i < self->transcoding_streams->len; i = i + 1) {
     TranscodingStream *s = self->transcoding_streams->pdatai;
 
@@ -317,6 +323,7 @@
 
 done:
   GST_OBJECT_UNLOCK (self);
+  GST_DEBUG_OBJECT (self, "Look-up result: %p", res);
 
   return res;
 }
@@ -327,6 +334,9 @@
   TranscodingStream *res = NULL;
   GstPad *encodebin_pad = get_encodebin_pad_from_stream (self, stream);
 
+  GST_DEBUG_OBJECT (self,
+      "Encodebin pad for stream %" GST_PTR_FORMAT " : %" GST_PTR_FORMAT, stream,
+      encodebin_pad);
   if (encodebin_pad) {
     GST_INFO_OBJECT (self,
         "Going to transcode stream %s (encodebin pad: %" GST_PTR_FORMAT ")",
@@ -410,6 +420,31 @@
   }
 }
 
+static void
+query_upstream_selectable (GstTranscodeBin * self, GstPad * pad)
+{
+  GstQuery *query;
+  gboolean result;
+
+  /* Query whether upstream can handle stream selection or not */
+  query = gst_query_new_selectable ();
+  result = GST_PAD_IS_SINK (pad) ? gst_pad_peer_query (pad, query)
+      : gst_pad_query (pad, query);
+  if (result) {
+    GST_FIXME_OBJECT (self,
+        "We force `transcodebin` to upstream selection"
+        " mode if *any* of the inputs is. This means things might break if"
+        " there's a mix");
+    gst_query_parse_selectable (query, &self->upstream_selected);
+    GST_DEBUG_OBJECT (pad, "Upstream is selectable : %d",
+        self->upstream_selected);
+  } else {
+    self->upstream_selected = FALSE;
+    GST_DEBUG_OBJECT (pad, "Upstream does not handle SELECTABLE query");
+  }
+  gst_query_unref (query);
+}
+
 static GstPadProbeReturn
 wait_stream_start_probe (GstPad * pad,
     GstPadProbeInfo * info, GstTranscodeBin * self)
@@ -420,6 +455,7 @@
   GST_INFO_OBJECT (self,
       "Got pad %" GST_PTR_FORMAT " with stream:: %" GST_PTR_FORMAT, pad,
       info->data);
+  query_upstream_selectable (self, pad);
   gst_transcode_bin_link_encodebin_pad (self, pad, info->data);
 
   return GST_PAD_PROBE_REMOVE;
@@ -440,6 +476,7 @@
     gst_event_parse_stream_start (sstart_event, &stream_id);
     GST_INFO_OBJECT (self, "Got pad %" GST_PTR_FORMAT " with stream ID: %s",
         pad, stream_id);
+    query_upstream_selectable (self, pad);
     gst_transcode_bin_link_encodebin_pad (self, pad, sstart_event);
     return;
   }
@@ -773,25 +810,8 @@
 
   switch (GST_EVENT_TYPE (event)) {
     case GST_EVENT_STREAM_START:
-    {
-      GstQuery *q = gst_query_new_selectable ();
-
-      /* Query whether upstream can handle stream selection or not */
-      if (gst_pad_peer_query (sinkpad, q)) {
-        GST_FIXME_OBJECT (self, "We force `transcodebin` to upstream selection"
-            " mode if *any* of the inputs is. This means things might break if"
-            " there's a mix");
-        gst_query_parse_selectable (q, &self->upstream_selected);
-        GST_DEBUG_OBJECT (sinkpad, "Upstream is selectable : %d",
-            self->upstream_selected);
-      } else {
-        self->upstream_selected = FALSE;
-        GST_DEBUG_OBJECT (sinkpad, "Upstream does not handle SELECTABLE query");
-      }
-      gst_query_unref (q);
-
+      query_upstream_selectable (self, sinkpad);
       break;
-    }
     default:
       break;
   }
@@ -997,6 +1017,9 @@
   object_class->get_property = gst_transcode_bin_get_property;
   object_class->set_property = gst_transcode_bin_set_property;
 
+  GST_DEBUG_CATEGORY_INIT (gst_transcodebin_debug, "transcodebin", 0,
+      "Transcodebin element");
+
   gstelement_klass = (GstElementClass *) klass;
   gstelement_klass->change_state =
       GST_DEBUG_FUNCPTR (gst_transcode_bin_change_state);

gst-plugins-bad-1.22.5.tar.xz/gst/videoparsers/gsth265parse.c -> gst-plugins-bad-1.22.6.tar.xz/gst/videoparsers/gsth265parse.c Changed

@@ -3180,11 +3180,14 @@
     off = 23;
 
     for (i = 0; i < num_nal_arrays; i++) {
+      guint8 nalu_type;
+
       if (off + 3 >= size) {
         gst_buffer_unmap (codec_data, &map);
         goto hvcc_too_small;
       }
 
+      nalu_type = dataoff & 0x3f;
       num_nals = GST_READ_UINT16_BE (data + off + 1);
       off += 3;
       for (j = 0; j < num_nals; j++) {
@@ -3192,6 +3195,15 @@
             data, off, size, 2, &nalu);
 
         if (parseres != GST_H265_PARSER_OK) {
+          if (i + 1 == num_nal_arrays && j + 1 == num_nals &&
+              nalu_type != GST_H265_NAL_VPS && nalu_type != GST_H265_NAL_SPS &&
+              nalu_type != GST_H265_NAL_PPS) {
+            GST_WARNING_OBJECT (h265parse,
+                "Couldn't parse the last nalu, type %d at array %d / %d",
+                nalu_type, i, j);
+            goto codec_data_done;
+          }
+          GST_ERROR ("aaa, %d", nalu_type);
           gst_buffer_unmap (codec_data, &map);
           goto hvcc_too_small;
         }
@@ -3200,6 +3212,7 @@
         off = nalu.offset + nalu.size;
       }
     }
+  codec_data_done:
     gst_buffer_unmap (codec_data, &map);
 
     /* don't confuse codec_data with inband vps/sps/pps */

gst-plugins-bad-1.22.5.tar.xz/meson.build -> gst-plugins-bad-1.22.6.tar.xz/meson.build Changed

gst-plugins-bad-1.22.5.tar.xz/po/gst-plugins-bad-1.0.pot -> gst-plugins-bad-1.22.6.tar.xz/po/gst-plugins-bad-1.0.pot Changed

gst-plugins-bad-1.22.5.tar.xz/po/gst-plugins-bad.pot -> gst-plugins-bad-1.22.6.tar.xz/po/gst-plugins-bad.pot Changed

gst-plugins-bad-1.22.5.tar.xz/sys/androidmedia/gst-android-hardware-camera.c -> gst-plugins-bad-1.22.6.tar.xz/sys/androidmedia/gst-android-hardware-camera.c Changed

@@ -979,6 +979,7 @@
     }
   } else {
     android_hardware_camera_parameters.EFFECT_EMBOSS = NULL;
+    g_clear_error (&err);
   }
 
   fieldID =
@@ -1000,6 +1001,7 @@
     }
   } else {
     android_hardware_camera_parameters.EFFECT_SKETCH = NULL;
+    g_clear_error (&err);
   }
 
   fieldID =
@@ -1021,6 +1023,7 @@
     }
   } else {
     android_hardware_camera_parameters.EFFECT_NEON = NULL;
+    g_clear_error (&err);
   }
 
   fieldID =
@@ -1519,6 +1522,7 @@
     }
   } else {
     android_hardware_camera_parameters.SCENE_MODE_BACKLIGHT = NULL;
+    g_clear_error (&err);
   }
 
   fieldID =
@@ -1540,6 +1544,7 @@
     }
   } else {
     android_hardware_camera_parameters.SCENE_MODE_FLOWERS = NULL;
+    g_clear_error (&err);
   }
 
   fieldID =
@@ -1561,6 +1566,7 @@
     }
   } else {
     android_hardware_camera_parameters.SCENE_MODE_AR = NULL;
+    g_clear_error (&err);
   }
 
   fieldID =
@@ -1582,6 +1588,7 @@
     }
   } else {
     android_hardware_camera_parameters.SCENE_MODE_HDR = NULL;
+    g_clear_error (&err);
   }
 
   fieldID =

gst-plugins-bad-1.22.5.tar.xz/sys/androidmedia/gstamc-constants.h -> gst-plugins-bad-1.22.6.tar.xz/sys/androidmedia/gstamc-constants.h Changed

gst-plugins-bad-1.22.5.tar.xz/sys/androidmedia/gstamc.c -> gst-plugins-bad-1.22.6.tar.xz/sys/androidmedia/gstamc.c Changed

gst-plugins-bad-1.22.5.tar.xz/sys/androidmedia/gstamcvideodec.c -> gst-plugins-bad-1.22.6.tar.xz/sys/androidmedia/gstamcvideodec.c Changed

gst-plugins-bad-1.22.5.tar.xz/sys/androidmedia/gstamcvideoenc.c -> gst-plugins-bad-1.22.6.tar.xz/sys/androidmedia/gstamcvideoenc.c Changed

@@ -877,6 +877,16 @@
   if (best)
     gst_video_codec_frame_ref (best);
 
+  GST_DEBUG_OBJECT (self, "found best %p from %u frames", best,
+      g_list_length (frames));
+  if (best) {
+    GST_LOG_OBJECT (self, "best %p (input pts %" GST_TIME_FORMAT " dts %"
+        GST_TIME_FORMAT " frame no %" G_GUINT32_FORMAT " buffer %"
+        GST_PTR_FORMAT, best, GST_TIME_ARGS (best->pts),
+        GST_TIME_ARGS (best->dts), best->system_frame_number,
+        best->input_buffer);
+  }
+
   g_list_foreach (frames, (GFunc) gst_video_codec_frame_unref, NULL);
   g_list_free (frames);
 
@@ -954,56 +964,14 @@
   GstFlowReturn flow_ret = GST_FLOW_OK;
   GstVideoEncoder *encoder = GST_VIDEO_ENCODER_CAST (self);
 
-  /* The BUFFER_FLAG_CODEC_CONFIG logic is borrowed from
-   * gst-omx. see *_handle_output_frame in
-   * gstomxvideoenc.c and gstomxh264enc.c */
-  if ((buffer_info->flags & BUFFER_FLAG_CODEC_CONFIG)
-      && buffer_info->size > 0) {
-
-    if (self->codec_data_in_bytestream) {
-      if (buffer_info->size > 4 &&
-          GST_READ_UINT32_BE (buf->data + buffer_info->offset) == 0x00000001) {
-        GList *l = NULL;
-        GstBuffer *hdrs;
-
-        GST_DEBUG_OBJECT (self, "got codecconfig in byte-stream format");
-
-        hdrs = gst_buffer_new_and_alloc (buffer_info->size);
-        gst_buffer_fill (hdrs, 0, buf->data + buffer_info->offset,
-            buffer_info->size);
-        GST_BUFFER_PTS (hdrs) =
-            gst_util_uint64_scale (buffer_info->presentation_time_us,
-            GST_USECOND, 1);
-
-        l = g_list_append (l, hdrs);
-        gst_video_encoder_set_headers (encoder, l);
-      }
-    } else {
-      GstBuffer *codec_data;
-      GstVideoCodecState *output_state =
-          gst_video_encoder_get_output_state (GST_VIDEO_ENCODER (self));
-
-      GST_DEBUG_OBJECT (self, "Handling codec data");
-
-      codec_data = gst_buffer_new_and_alloc (buffer_info->size);
-      gst_buffer_fill (codec_data, 0, buf->data + buffer_info->offset,
-          buffer_info->size);
-      output_state->codec_data = codec_data;
-      gst_video_codec_state_unref (output_state);
-
-      if (!gst_video_encoder_negotiate (encoder)) {
-        gst_video_codec_frame_unref (frame);
-        return GST_FLOW_NOT_NEGOTIATED;
-      }
-
-      return GST_FLOW_OK;
-    }
-  }
-
   if (buffer_info->size > 0) {
     GstBuffer *out_buf;
     GstPad *srcpad;
 
+    if (buffer_info->flags & BUFFER_FLAG_PARTIAL_FRAME) {
+      GST_FIXME_OBJECT (self, "partial frames are currently not handled");
+    }
+
     srcpad = GST_VIDEO_ENCODER_SRC_PAD (encoder);
     out_buf =
         gst_video_encoder_allocate_output_buffer (encoder, buffer_info->size);
@@ -1023,7 +991,10 @@
        * caps and filling it
        */
 
-      GST_ERROR_OBJECT (self, "No corresponding frame found");
+      GST_ERROR_OBJECT (self, "No corresponding frame found: buffer pts: %"
+          GST_TIME_FORMAT " presentation_time_us %" G_GUINT64_FORMAT,
+          GST_TIME_ARGS (GST_BUFFER_PTS (out_buf)),
+          (guint64) buffer_info->presentation_time_us);
       flow_ret = gst_pad_push (srcpad, out_buf);
     }
   } else if (frame) {
@@ -1036,9 +1007,10 @@
 static void
 gst_amc_video_enc_loop (GstAmcVideoEnc * self)
 {
+  GstVideoEncoder *encoder = GST_VIDEO_ENCODER_CAST (self);
   GstVideoCodecFrame *frame;
   GstFlowReturn flow_ret = GST_FLOW_OK;
-  gboolean is_eos;
+  gboolean is_eos, is_codec_data;
   GstAmcBufferInfo buffer_info;
   GstAmcBuffer *buf;
   gint idx;
@@ -1148,14 +1120,62 @@
     goto got_null_output_buffer;
   }
 
-  frame =
-      _find_nearest_frame (self,
-      gst_util_uint64_scale (buffer_info.presentation_time_us, GST_USECOND, 1));
+  is_codec_data = FALSE;
+  /* The BUFFER_FLAG_CODEC_CONFIG logic is borrowed from
+   * gst-omx. see *_handle_output_frame in
+   * gstomxvideoenc.c and gstomxh264enc.c */
+  if ((buffer_info.flags & BUFFER_FLAG_CODEC_CONFIG)
+      && buffer_info.size > 0) {
+
+    if (self->codec_data_in_bytestream) {
+      if (buffer_info.size > 4 &&
+          GST_READ_UINT32_BE (buf->data + buffer_info.offset) == 0x00000001) {
+        GList *l = NULL;
+        GstBuffer *hdrs;
+
+        GST_DEBUG_OBJECT (self, "got codecconfig in byte-stream format");
+
+        hdrs = gst_buffer_new_and_alloc (buffer_info.size);
+        gst_buffer_fill (hdrs, 0, buf->data + buffer_info.offset,
+            buffer_info.size);
+        GST_BUFFER_PTS (hdrs) =
+            gst_util_uint64_scale (buffer_info.presentation_time_us,
+            GST_USECOND, 1);
+
+        l = g_list_append (l, hdrs);
+        gst_video_encoder_set_headers (encoder, l);
+        is_codec_data = TRUE;
+      }
+    } else {
+      GstBuffer *codec_data;
+      GstVideoCodecState *output_state =
+          gst_video_encoder_get_output_state (GST_VIDEO_ENCODER (self));
+
+      GST_DEBUG_OBJECT (self, "Handling codec data");
+
+      codec_data = gst_buffer_new_and_alloc (buffer_info.size);
+      gst_buffer_fill (codec_data, 0, buf->data + buffer_info.offset,
+          buffer_info.size);
+      output_state->codec_data = codec_data;
+      gst_video_codec_state_unref (output_state);
+      is_codec_data = TRUE;
+
+      if (!gst_video_encoder_negotiate (encoder))
+        flow_ret = GST_FLOW_NOT_NEGOTIATED;
+    }
+  }
 
   is_eos = ! !(buffer_info.flags & BUFFER_FLAG_END_OF_STREAM);
 
-  flow_ret =
-      gst_amc_video_enc_handle_output_frame (self, buf, &buffer_info, frame);
+  if (flow_ret == GST_FLOW_OK && !is_codec_data) {
+    frame =
+        _find_nearest_frame (self,
+        gst_util_uint64_scale (buffer_info.presentation_time_us, GST_USECOND,
+            1));
+
+    flow_ret =
+        gst_amc_video_enc_handle_output_frame (self, buf, &buffer_info, frame);
+  }
 
   gst_amc_buffer_free (buf);
   buf = NULL;

gst-plugins-bad-1.22.5.tar.xz/sys/androidmedia/gstjniutils.c -> gst-plugins-bad-1.22.6.tar.xz/sys/androidmedia/gstjniutils.c Changed

gst-plugins-bad-1.22.5.tar.xz/sys/applemedia/avsamplevideosink.m -> gst-plugins-bad-1.22.6.tar.xz/sys/applemedia/avsamplevideosink.m Changed

gst-plugins-bad-1.22.5.tar.xz/sys/applemedia/coremediabuffer.c -> gst-plugins-bad-1.22.6.tar.xz/sys/applemedia/coremediabuffer.c Changed

gst-plugins-bad-1.22.5.tar.xz/sys/applemedia/corevideobuffer.c -> gst-plugins-bad-1.22.6.tar.xz/sys/applemedia/corevideobuffer.c Changed

gst-plugins-bad-1.22.5.tar.xz/sys/d3d11/gstd3d11convert.cpp -> gst-plugins-bad-1.22.6.tar.xz/sys/d3d11/gstd3d11convert.cpp Changed

gst-plugins-bad-1.22.5.tar.xz/sys/nvcodec/gstnvencoder.cpp -> gst-plugins-bad-1.22.6.tar.xz/sys/nvcodec/gstnvencoder.cpp Changed

gst-plugins-bad-1.22.5.tar.xz/sys/va/gstvadecoder.c -> gst-plugins-bad-1.22.6.tar.xz/sys/va/gstvadecoder.c Changed

gst-plugins-bad-1.22.5.tar.xz/sys/va/gstvadisplay_priv.c -> gst-plugins-bad-1.22.6.tar.xz/sys/va/gstvadisplay_priv.c Changed

gst-plugins-bad-1.22.5.tar.xz/sys/win32ipc/protocol/win32ipcpipeserver.cpp -> gst-plugins-bad-1.22.6.tar.xz/sys/win32ipc/protocol/win32ipcpipeserver.cpp Changed

gst-plugins-bad-1.22.5.tar.xz/tests/check/elements/h265parse.c -> gst-plugins-bad-1.22.6.tar.xz/tests/check/elements/h265parse.c Changed

@@ -1116,6 +1116,68 @@
 
 GST_END_TEST;
 
+GST_START_TEST (test_invalid_sei_in_hvcc)
+{
+  GstHarness *h;
+  GstCaps *caps;
+  GstBuffer *codec_data;
+  /* Consists of 4 arrays (VPS, SPS, PPS, SEI -> broken) and each array contains
+   * single nalu
+   * Captured from the log at
+   * https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/2905
+   */
+  static const guint8 hvcc_data = {
+    0x01, 0x01, 0x01, 0x01, 0x60, 0x00, 0x90, 0x00, 0x00, 0x00, 0x00, 0x00,
+    0x00, 0xf0, 0x00, 0xfc, 0xfd, 0xf8, 0xf8, 0x00, 0x00, 0x0f, 0x04, 0x20,
+    0x00, 0x01, 0x00, 0x17, 0x40, 0x01, 0x0c, 0x01, 0xff, 0xff, 0x01, 0x60,
+    0x00, 0x00, 0x03, 0x00, 0x80, 0x00, 0x00, 0x03, 0x00, 0x00, 0x03, 0x00,
+    0x7b, 0xac, 0x09, 0x21, 0x00, 0x01, 0x00, 0x42, 0x42, 0x01, 0x01, 0x01,
+    0x60, 0x00, 0x00, 0x03, 0x00, 0x80, 0x00, 0x00, 0x03, 0x00, 0x00, 0x03,
+    0x00, 0x7b, 0xa0, 0x02, 0x80, 0x80, 0x2d, 0x1f, 0xe3, 0x6b, 0xbb, 0x53,
+    0x77, 0x72, 0x5d, 0x60, 0x2d, 0xc0, 0x40, 0x40, 0x41, 0x00, 0x00, 0x03,
+    0x03, 0xe8, 0x00, 0x00, 0x4e, 0x20, 0x72, 0x1d, 0xee, 0x51, 0x00, 0x05,
+    0xdc, 0x00, 0x00, 0x1a, 0x5e, 0x00, 0x00, 0x2e, 0xe0, 0x00, 0x00, 0xd2,
+    0xf0, 0x08, 0x22, 0x00, 0x01, 0x00, 0x0b, 0x44, 0x01, 0xc1, 0x72, 0xb0,
+    0x9c, 0x38, 0x77, 0x06, 0x0c, 0x24, 0x27, 0x00, 0x01, 0x00, 0x00, 0x00
+  };
+
+  caps = gst_caps_new_simple ("video/x-h265", "stream-format", G_TYPE_STRING,
+      "hvc1", "alignment", G_TYPE_STRING, "au", NULL);
+  codec_data = gst_buffer_new_memdup (hvcc_data, sizeof (hvcc_data));
+  gst_caps_set_simple (caps, "codec_data", GST_TYPE_BUFFER, codec_data, NULL);
+  gst_buffer_unref (codec_data);
+
+  h = gst_harness_new ("h265parse");
+  gst_harness_set_src_caps (h, caps);
+  gst_harness_push_event (h, gst_event_new_eos ());
+
+  while (TRUE) {
+    GstEvent *event = gst_harness_pull_event (h);
+    fail_unless (event);
+
+    if (GST_EVENT_TYPE (event) == GST_EVENT_CAPS) {
+      GstStructure *s;
+      gint width, height;
+
+      gst_event_parse_caps (event, &caps);
+      s = gst_caps_get_structure (caps, 0);
+
+      fail_unless (gst_structure_get_int (s, "width", &width));
+      fail_unless_equals_int (width, 1280);
+      fail_unless (gst_structure_get_int (s, "height", &height));
+      fail_unless_equals_int (height, 720);
+
+      gst_event_unref (event);
+      break;
+    }
+
+    gst_event_unref (event);
+  }
+
+  gst_harness_teardown (h);
+}
+
+GST_END_TEST;
 
 static Suite *
 h265parse_harnessed_suite (void)
@@ -1152,6 +1214,8 @@
 
   tcase_add_test (tc_chain, test_drain);
 
+  tcase_add_test (tc_chain, test_invalid_sei_in_hvcc);
+
   return s;
 }

Changes of Revision 19