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

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

+Fri Sep 22 17:25:19 UTC 2023 - Bjørn Lie <zaitor@opensuse.org>

+

+- Update to version 1.22.6

+

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

 Sat Aug 12 09:49:32 UTC 2023 - Bjørn Lie <zaitor@opensuse.org>

 - Update to version 1.22.5

​

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

 %define _version 1.22.0

 Name:           gstreamer-plugins-bad-codecs

-Version:        1.22.5

+Version:        1.22.6

 Release:        0

 Summary:        Codecs/plugins for gstreamer-plugins-bad

 License:        LGPL-2.1-or-later

​

@@ -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>

​

@@ -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

​

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

-This is GStreamer gst-plugins-bad 1.22.5.

+This is GStreamer gst-plugins-bad 1.22.6.

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

 of your favourite cross-platform multimedia framework!

​

@@ -447,7 +447,9 @@

             entry_location = g_path_get_basename (sink->current_location);

           } else {

             gchar *name = g_path_get_basename (sink->current_location);

-            entry_location = g_build_filename (sink->playlist_root, name, NULL);

+            /* g_build_filename() will insert back slash on Windows */

+            entry_location = g_build_path ("/",

+                sink->playlist_root, name, NULL);

             g_free (name);

           }

​

@@ -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;

​

@@ -8945,7 +8945,7 @@

       G_CALLBACK (gst_webrtc_bin_add_turn_server), NULL, NULL, NULL,

       G_TYPE_BOOLEAN, 1, G_TYPE_STRING);

-  /*

+  /**

    * GstWebRTCBin::create-data-channel:

    * @object: the #GstWebRTCBin

    * @label: the label for the data channel

​

@@ -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;

​

@@ -1845,6 +1845,7 @@

   READ_UINT8 (&nr, vps->max_layers_minus1, 6);

   READ_UINT8 (&nr, vps->max_sub_layers_minus1, 3);

+  CHECK_ALLOWED (vps->max_sub_layers_minus1, 0, 6);

   READ_UINT8 (&nr, vps->temporal_id_nesting_flag, 1);

   /* skip reserved_0xffff_16bits */

@@ -2015,6 +2016,7 @@

   READ_UINT8 (&nr, sps->vps_id, 4);

   READ_UINT8 (&nr, sps->max_sub_layers_minus1, 3);

+  CHECK_ALLOWED (sps->max_sub_layers_minus1, 0, 6);

   READ_UINT8 (&nr, sps->temporal_id_nesting_flag, 1);

   if (!gst_h265_parse_profile_tier_level (&sps->profile_tier_level, &nr,

​

@@ -150,13 +150,13 @@

     {64, 45},                   /* 3, 0.7031 16:9 625 line */

     {10000, 7615},              /* 4, 0.7615 */

     {10000, 8055},              /* 5, 0.8055 */

-    {32, 27},                   /* 6, 0.8437 */

+    {32, 27},                   /* 6, 0.8437 16:9 525 line */

     {10000, 8935},              /* 7, 0.8935 */

-    {10000, 9375},              /* 8, 0.9375 */

+    {10000, 9157},              /* 8, 0.9157 CCIR601, 625 line */

     {10000, 9815},              /* 9, 0.9815 */

     {10000, 10255},             /* 10, 1.0255 */

     {10000, 10695},             /* 11, 1.0695 */

-    {8, 9},                     /* 12, 1.125 */

+    {10000, 10950},             /* 12, 1.0950 CCIR601, 525 line */

     {10000, 11575},             /* 13, 1.1575 */

     {10000, 12015},             /* 14, 1.2015 */

     {0, 0},                     /* 15, invalid */

​

@@ -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) {

​

@@ -155,6 +155,9 @@

   GstH264DecRefPicMarking dec_ref_pic_marking;

+  /* Set by decoder to trace the number of delayed output pictures */

+  guint32 reorder_frame_number;

+

   /* For interlaced decoding */

   gboolean second_field;

   GstH264Picture * other_field;

​

@@ -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);

 }

​

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

  <release>

   <Version>

+   <revision>1.22.6</revision>

+   <branch>1.22</branch>

+   <name></name>

+   <created>2023-09-20</created>

+   <file-release rdf:resource="https://gstreamer.freedesktop.org/src/gst-plugins-bad/gst-plugins-bad-1.22.6.tar.xz" />

+  </Version>

+ </release>

+

+ <release>

+  <Version>

    <revision>1.22.5</revision>

    <branch>1.22</branch>

    <name></name>

​

@@ -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)

​

@@ -119,7 +119,7 @@

   gst_buffer_map (buffer, &map, GST_MAP_READ);

   /* Now transform raw AES3 into raw audio, see SMPTE 331M */

-  if ((map.size - 4) % 32 != 0) {

+  if (map.size < 4 || (map.size - 4) % 32 != 0) {

     gst_buffer_unmap (buffer, &map);

     GST_ERROR ("Invalid D10 sound essence buffer size");

     return GST_FLOW_ERROR;

@@ -219,6 +219,7 @@

     GstAudioFormat audio_format;

     if (s->channel_count == 0 ||

+        s->channel_count > 8 ||

         s->quantization_bits == 0 ||

         s->audio_sampling_rate.n == 0 || s->audio_sampling_rate.d == 0) {

       GST_ERROR ("Invalid descriptor");

​

@@ -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;

 }

​

@@ -821,7 +821,7 @@

 gst_rtmp2_sink_render (GstBaseSink * sink, GstBuffer * buffer)

 {

   GstRtmp2Sink *self = GST_RTMP2_SINK (sink);

-  GstBuffer *message;

+  GstBuffer *message = NULL;

   GstFlowReturn ret;

   if (G_UNLIKELY (should_drop_header (self, buffer))) {

@@ -833,7 +833,7 @@

   if (G_UNLIKELY (!buffer_to_message (self, buffer, &message))) {

     GST_ELEMENT_ERROR (self, STREAM, FAILED, ("Failed to convert FLV to RTMP"),

-        ("Failed to convert %" GST_PTR_FORMAT, message));

+        ("Failed to convert %" GST_PTR_FORMAT, buffer));

     return GST_FLOW_ERROR;

   }

​

@@ -602,13 +602,6 @@

     goto out;

   }

-  if (!flash_ver) {

-    g_task_return_new_error (task, G_IO_ERROR, G_IO_ERROR_NOT_INITIALIZED,

-        "Flash version is not set");

-    g_object_unref (task);

-    goto out;

-  }

-

   if (data->auth_query) {

     const gchar *query = data->auth_query;

     appstr = g_strdup_printf ("%s?%s", app, query);

@@ -649,9 +642,11 @@

     gst_amf_node_append_field_string (node, "type", "nonprivate", -1);

   }

-  /* "Flash Player version. It is the same string as returned by the

-   * ApplicationScript getversion () function." */

-  gst_amf_node_append_field_string (node, "flashVer", flash_ver, -1);

+  if (flash_ver) {

+    /* "Flash Player version. It is the same string as returned by the

+     * ApplicationScript getversion () function." */

+    gst_amf_node_append_field_string (node, "flashVer", flash_ver, -1);

+  }

   /* "URL of the source SWF file making the connection."

    * XXX: libavformat sends "swfUrl" here, if provided. */

​

@@ -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"