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gstreamer-plugins-bad-codecs.changes Changed

gstreamer-plugins-bad-codecs.spec Changed

gst-plugins-bad-1.20.3.tar.xz/ChangeLog -> gst-plugins-bad-1.20.4.tar.xz/ChangeLog Changed

@@ -1,7 +1,387 @@
+=== release 1.20.4 ===
+
+2022-10-12 16:39:47 +0100  Tim-Philipp Müller <tim@centricular.com>
+
+	* NEWS:
+	* RELEASE:
+	* gst-plugins-bad.doap:
+	* meson.build:
+	  Release 1.20.4
+
+2022-10-12 16:39:40 +0100  Tim-Philipp Müller <tim@centricular.com>
+
+	* ChangeLog:
+	  Update ChangeLogs for 1.20.4
+
+2022-08-08 23:37:11 +0900  Seungha Yang <seungha@centricular.com>
+
+	* gst/mxf/mxfaes-bwf.c:
+	  mxfdemux: Always calculate BlockAlign of raw audio
+	  Workaround for nBlockAlign and nBitsPerSample mismatch. Always
+	  use the formula described in the specification for BlockAlign value
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3149>
+
+2022-09-14 00:58:37 +0900  Seungha Yang <seungha@centricular.com>
+
+	* sys/nvcodec/gstnvdec.c:
+	  nvdec: Fix for HEVC decoding when coded resolution is larger than display resolution
+	  As documented in the SDK header, we should set coded width/height
+	  values to the corresponding decoder configuration option,
+	  instead of display resolution
+	  Fixes: https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1438
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3143>
+
+2022-09-22 22:39:31 +0900  Sangchul Lee <sc11.lee@samsung.com>
+
+	* ext/webrtc/gstwebrtcbin.c:
+	  webrtcbin: Fix pointer dereference before null check
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3133>
+
+2022-10-05 15:59:03 +0900  Sangchul Lee <sc11.lee@samsung.com>
+
+	* ext/webrtc/gstwebrtcice.c:
+	  webrtc/nice: Make sure to return NULL when validating turn server fails
+	  It affects 'add-turn-server' signal action and 'turn-server' property
+	  of webrtcbin.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3124>
+
+2022-09-20 23:31:45 +0300  Mart Raudsepp <mart@leio.tech>
+
+	* gst/mpegtsdemux/mpegtsbase.c:
+	  tsdemux: Don't trigger a program change when falling back to ignore-pcr behaviour
+	  Since commit a79a756b79aa1675e we could change to ignore-pcr automatically at 500ms
+	  into a live stream when no PCR is seen by then. However the stream counting in
+	  program change detection was wrongly considering ignore-pcr programs to have a
+	  separate PCR PID, even though we are actually ignoring the PCR PID completely,
+	  resulting in an erroneous program switch getting triggered from the different
+	  stream count. This in turn would send an EOS and switch out the pads for what
+	  actually is still the same program, while we intended to simply apply a
+	  workaround for broken encoders.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3089>
+
+2022-03-25 14:25:02 +1100  Andrew Pritchard <andrew@vivi.io>
+
+	* sys/androidmedia/jni/gstamcsurfacetexture-jni.c:
+	  Fix GstAmcSurfaceTexture segfault Check that `self` and `self->callback` are defined. `self` can be set to `NULL` in `remove_listener`, and `self->callback` can be set to `NULL` inside `gst_amc_surface_texture_jni_set_on_frame_available_callback`. This can cause a segfault since the Java object can outlive the C object, and call the callback after `remove_listener` is called.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3056>
+
+2022-08-25 14:24:25 +0200  Piotr Brzeziński <piotr@centricular.com>
+
+	* sys/applemedia/avfvideosrc.m:
+	  avfvideosrc: Fix wrong default framerate value
+	  Current default G_MAXINT is not a correct value under any circumstances.
+	  This creates an issue with screen capture, during which we currently do
+	  not get any framerate info causing G_MAXINT to show up, where elements
+	  downstream can possibly misbehave - for example, `vtenc` causes
+	  a kernel panic.
+	  Replace with 30/1 to avoid such scenarios.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2946>
+
+2022-08-20 16:15:15 +0100  Philippe Normand <philn@igalia.com>
+
+	* ext/openh264/gstopenh264dec.cpp:
+	* ext/openh264/gstopenh264enc.cpp:
+	  openh264: Register debug categories earlier
+	  Otherwise the GST_ERROR message logged in case of ABI mismatch would be done on
+	  an uninitialized category.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2924>
+
+2022-08-20 16:57:27 +0100  Philippe Normand <philn@igalia.com>
+
+	* ext/openh264/gstopenh264enc.cpp:
+	  openh264enc: Fix constrained-high encoding
+	  constrained-high is high without B-frames, there is no EProfileIdc for this, so
+	  assume high instead of hitting an assert down the line.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2921>
+
+2022-08-14 22:31:29 -0400  Daniel Morin <daniel.morin@collabora.com>
+
+	* gst-libs/gst/play/gstplay-media-info.h:
+	* gst-libs/gst/play/gstplay-signal-adapter.h:
+	* gst-libs/gst/play/gstplay-video-overlay-video-renderer.h:
+	* gst-libs/gst/play/gstplay-video-renderer.h:
+	* gst-libs/gst/play/gstplay-visualization.h:
+	  gst-play: missing cleanup for g_autoptr
+	  Without this change cleanup function for g_autoptr is not defined for
+	  GstPlayMediaInfo, GstPlaySignalAdapter, GstPlayVideoRenderer,
+	  GstPlayVideoOverlayVideoRenderer and GstPlayVisualization. Cleanup
+	  function was defined in gstplay.h, but missing in other header files.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2904>
+
+2022-08-13 12:24:37 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst-libs/gst/player/gstplayer-media-info.c:
+	  player: Don't leak wrapped video info
+	  Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1373
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2881>
+
+2022-08-13 11:50:20 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst-libs/gst/play/gstplay.c:
+	  play: Make ownership of video-sink clearer in combination with floating references
+	  And correctly handle the case of VideoRenderer::create_video_sink() not
+	  actually returning a floating reference, which might be tricky for some
+	  bindings.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2881>
+
+2022-08-13 11:49:08 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst-libs/gst/play/gstplay.c:
+	  play: Fix object construction
+	  Ideally new() functions should simply call g_object_new() and not much
+	  else, so let's do that here and handle all the construction properly in
+	  a GObject way.
+	  Now a play object created via g_object_new() is actually usable.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2881>
+
+2022-08-13 11:39:59 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst-libs/gst/player/gstplayer.c:
+	  player: Fix object construction
+	  Ideally new() functions should simply call g_object_new() and not much
+	  else, so let's do that here and handle all the construction properly in
+	  a GObject way.
+	  Now a player object created via g_object_new() is actually usable.
+	  In addition, also fix the video-renderer property so that reading it
+	  returns an object of the correct type.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2881>
+
+2022-08-13 11:30:35 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst-libs/gst/player/gstplayer.c:
+	  player: Release signal adapter on finalize
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2881>
+
+2022-08-12 18:24:41 +0300  Matthias Clasen <mclasen@redhat.com>
+
+	* gst-libs/gst/player/gstplayer.c:
+	  gstplayer: Plug a memory leak
+	  This was showing up as a memory leak in GTK's
+	  gstreamer media backend:
+	  40 bytes in 1 blocks are definitely lost in loss record 18,487 of 40,868
+	  at 0x484586F: malloc (vg_replace_malloc.c:381)
+	  by 0x50D5278: g_malloc (gmem.c:125)
+	  by 0x50EDBA5: g_slice_alloc (gslice.c:1072)
+	  by 0x50EFBCC: g_slice_alloc0 (gslice.c:1098)
+	  by 0x51F2F45: g_type_create_instance (gtype.c:1911)
+	  by 0x51DAE37: g_object_new_internal (gobject.c:2011)
+	  by 0x51DC080: g_object_new_with_properties (gobject.c:2181)
+	  by 0x51DCB20: g_object_new (gobject.c:1821)
+	  by 0x9855F86: UnknownInlinedFun (gstplayer-wrapped-video-renderer.c:109)
+	  by 0x9855F86: gst_player_new (gstplayer.c:579)
+	  Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1374
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2876>
+
+2022-07-30 02:29:49 +0530  Nirbheek Chauhan <nirbheek@centricular.com>
+
+	* ext/sctp/usrsctp/meson.build:
+	  meson: Don't pass -Werror to vendored code
+	  Do it the correct way with libusrsctp -- override the option so that
+	  it's done in a compiler-agnostic and future-proof way.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2818>
+
+2022-05-25 18:40:30 +0530  Nirbheek Chauhan <nirbheek@centricular.com>
+
+	* gst/rtmp2/gstrtmp2locationhandler.c:
+	* gst/rtmp2/rtmp/rtmpclient.c:
+	  rtsp+rtmp: Forward warning added to tls-validation-flags to our users
+	  With the 2.72 release, glib-networking developers have decided that
+	  TLS certificate validation cannot be implemented correctly by them, so
+	  they've deprecated it.
+	  In a nutshell: a cert can have several validation errors, but there
+	  are no guarantees that the TLS backend will return all those errors,
+	  and things are made even more complicated by the fact that the list of
+	  errors might refer to certs that are added for backwards-compat and
+	  won't actually be used by the TLS library.
+	  Our best option is to ignore the deprecation and pass the warning onto
+	  users so they can make an appropriate security decision regarding
+	  this.
+	  We can't deprecate the tls-validation-flags property because it is
+	  very useful when connecting to RTSP cameras that will never get
+	  updates to fix certificate errors.
+	  Relevant upstream merge requests / issues:
+	  https://gitlab.gnome.org/GNOME/glib/-/merge_requests/2214
+	  https://gitlab.gnome.org/GNOME/glib-networking/-/issues/179
+	  https://gitlab.gnome.org/GNOME/glib-networking/-/merge_requests/193
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2818>
+
+2022-05-25 16:03:22 +0530  Nirbheek Chauhan <nirbheek@centricular.com>
+
+	* ext/dtls/gstdtlscertificate.c:
+	  dtls: Disable OpenSSL 3.0 deprecation warnings for now
+	  Fedora 36 ships with OpenSSL 3.0, which deprecates all low-level APIs,
+	  so this code needs to be rewritten. There is no easy fix in the
+	  porting guide, and it recommends disabling the warnings if you can't
+	  use the high-level API.
+	  https://wiki.openssl.org/index.php/OpenSSL_3.0#Upgrading_to_OpenSSL_3.0_from_OpenSSL_1.1.1
+	  Here's the replacement API:
+	  https://www.openssl.org/docs/man3.0/man7/migration_guide.html#Deprecated-low-level-object-creation
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2818>
+
+2022-07-29 02:36:40 +0900  Seungha Yang <seungha@centricular.com>
+
+	* sys/wasapi2/gstwasapi2ringbuffer.cpp:
+	  wasapi2: Fix initial mute/volume setting
+	  Fix up volume/mute change flag setting
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2817>
+
+2022-07-21 16:11:03 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/audiobuffersplit/gstaudiobuffersplit.c:
+	  audiobuffersplit: Actually store number of samples to drop in gapless mode
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2783>
+
+2022-07-21 16:10:18 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/audiobuffersplit/gstaudiobuffersplit.c:
+	  audiobuffersplit: Use input running time for comparison instead of the currently tracked running time
+	  Otherwise gapless mode would do completely wrong calculations on
+	  discontinuities and cause input/output to drift slowly.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2783>
+
+2022-07-21 13:38:22 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/audiobuffersplit/gstaudiobuffersplit.c:
+	  audiobuffersplit: Combine two if expressions to reduce indentation
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2783>
+
+2022-07-06 16:14:13 +0300  Jordan Petridis <jordan@centricular.com>
+
+	* ext/openmpt/gstopenmptdec.c:
+	  openmpt: update from now deprecated api
+	  https://lib.openmpt.org/doc/classopenmpt_1_1module.html#ab2695af0baa274054f5687741fa7c05b
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2779>
+
+2022-06-30 11:04:29 +0200  Ignazio Pillai <ignazp@amazon.com>
+
+	* sys/wasapi/gstwasapiutil.c:
+	  wasapi: Implement default audio channel mask
+	  Some multichannel capture devices does not provide a channel mask value
+	  which will result in a pipeline failure due to the empty channel mask.
+	  Implemented the same fix used for wasapi2
+	  Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1204
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2714>
+
+2022-07-07 02:17:56 +0900  Seungha Yang <seungha@centricular.com>
+
+	* gst/proxy/gstproxysink.c:
+	  proxysink: Fix GstProxySrc leak
+	  Clear weak pointer to peer src when disposing.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2774>
+
+2022-07-06 03:14:25 +0900  Seungha Yang <seungha@centricular.com>
+
+	* gst/proxy/gstproxysink.c:
+	* gst/proxy/gstproxysink.h:
+	* tests/check/elements/proxysink.c:
+	* tests/check/meson.build:
+	  proxysink: Make sure stream-start and caps events are forwarded
+	  There might be a sequence of event and buffer flow:
+	  - Got stream-start/caps/segment events
+	  - Got flush events
+	  - And then buffers with a new segment event
+	  In the above case, stream-start and caps event might not be reached to
+	  peer proxysrc if peer proxysrc is not ready to receive them.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2774>
+
+2022-06-30 09:09:02 +0300  Sebastian Dröge <sebastian@centricular.com>
+
+	* ext/webrtc/gstwebrtcbin.c:
+	  webrtcbin: Reject caps that are not valid for creating an SDP media.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2705>
+
+2022-06-29 10:55:13 +0100  Tim-Philipp Müller <tim@centricular.com>
+
+	* meson.build:
+	  coding style: allow declarations after statement
+	  See https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1243/
+	  and https://gitlab.freedesktop.org/gstreamer/gstreamer-project/-/issues/78
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2702>
+
+2022-06-28 17:40:56 +0900  Seungha Yang <seungha@centricular.com>
+
+	* sys/d3d11/gstd3d11videosink.cpp:
+	  d3d11videosink: Fix for force-aspect-ratio setting when rendering on shared texture
+	  Set specified force-aspect-ratio value on window object
+	  in case of shared texture rendering as well
+	  Fixes: https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1304
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2701>
+
+2022-06-25 19:50:10 +0100  Tim-Philipp Müller <tim@centricular.com>
+
+	* tests/check/meson.build:
+	  tests: skip unit tests for dependency-less elements that have been disabled
+	  Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/1136
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2672>
+
+2022-06-28 01:29:06 +0100  Tim-Philipp Müller <tim@centricular.com>
+
+	* ext/opus/gstopusheader.h:
+	  dv, opusparse: fix duplicate symbols in static build
+	  Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/1262
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2673>
+
+2022-06-23 14:31:10 +0200  Víctor Manuel Jáquez Leal <vjaquez@igalia.com>
+
+	* sys/va/gstvaallocator.c:
+	  va: allocator: Use always lseek to get dmabuf size.
+	  Gallium drivers historically have reported strange dmabuf sizes, from always
+	  zero to the whole frame (multiple fds). The simplest solution is to use lseek
+	  SEEK_END to get the prime descriptor size.
+	  Also the allocator raises a warning if both values differ in order to report
+	  it to driver.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2657>
+
+2022-06-08 09:02:52 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+	* sys/va/gstvaallocator.c:
+	  va: allocator: Fix translation of VADRMPRIMESurfaceDescriptor
+	  VADRMPRIMESurfaceDescriptor structure describes the offsets from the
+	  point of view of the specific handle (DMABuf). While GstVideoInfo
+	  (and the meta) describes offsets from the point of the view of the
+	  GstBuffer, an aggregate of all the GstMemory (1 per handle).
+	  This changes combined with Mesa Fix(https://gitlab.freedesktop.org/mesa/mesa/-/merge_requests/16813)
+	  fixes decoding failure with AMD driver.
+	  Fixes #1223
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2657>
+
+2022-03-24 21:39:30 +0800  He Junyan <junyan.he@intel.com>
+
+	* sys/va/gstvah265dec.c:
+	  va: h265dec: Fix a crash because of missing reference frame.
+	  Some problematic H265 stream may miss the reference frame in the DPB,
+	  and get some message like: "No short term reference picture for xxx".
+	  So there may be empty entries in ref_pic_list0/1 when passing to
+	  decode_slice() function of sub class. We need to check the NULL pointer.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2647>
+
+2022-06-18 04:05:53 +0900  Seungha Yang <seungha@centricular.com>
+
+	* sys/d3d11/gstd3d11decoder.h:
+	  d3d11decoder: Check 16K resolution support
+	  16K decoding is supported by some GPUs
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2633>
+
+2022-06-15 15:06:20 -0400  Olivier Crête <olivier.crete@collabora.com>
+
+	* ext/webrtc/gstwebrtcbin.c:
+	  webrtcbin: Limit sink query to sink pads
+	  This allows the reception of streams that don't exactly match
+	  the codec preferences. In particular, the ssrc in the codec preferences
+	  is local sender SSRC, the other side is expected to send a different SSRC.
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2632>
+
+2022-06-16 00:59:00 +0100  Tim-Philipp Müller <tim@centricular.com>
+
+	* meson.build:
+	  Back to development
+	  Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2619>
+
 === release 1.20.3 ===
 
 2022-06-15 23:36:18 +0100  Tim-Philipp Müller <tim@centricular.com>
 
+	* ChangeLog:
 	* NEWS:
 	* RELEASE:
 	* gst-plugins-bad.doap:

gst-plugins-bad-1.20.3.tar.xz/NEWS -> gst-plugins-bad-1.20.4.tar.xz/NEWS Changed

@@ -2,13 +2,13 @@
 
 GStreamer 1.20.0 was originally released on 3 February 2022.
 
-The latest bug-fix release in the 1.20 series is 1.20.3 and was released
-on 15 June 2022.
+The latest bug-fix release in the 1.20 series is 1.20.4 and was released
+on 12 October 2022.
 
 See https://gstreamer.freedesktop.org/releases/1.20/ for the latest
 version of this document.
 
-Last updated: Monday 15 June 2022, 17:00 UTC (log)
+Last updated: Wednesday 12 October 2022, 15:00 UTC (log)
 
 Introduction
 
@@ -757,7 +757,8 @@
 
 -   matroskamux: accept in-band SPS/PPS/VPS for H.264 and H.265
     (i.e. stream-format avc3 and hev1) which allows on-the-fly
-    profile/level/resolution changes.
+    profile/level changes, and from 1.20.4 onwards also resolution
+    changes.
 
 -   matroskamux: new "cluster-timestamp-offset" property, useful for use
     cases where the container timestamps should map to some absolute
@@ -909,7 +910,7 @@
 -   vp8enc: finish support for temporal scalability: two new properties
     ("temporal-scalability-layer-flags",
     "temporal-scalability-layer-sync-flags") and a unit change on the
-    "temporal-scalability-target-bitrate" property (now expects bps);
+    "temporal-scalability-target-bitrate” property (now expects bps);
     also make temporal scalability details available to RTP payloaders
     as buffer metadata.
 
@@ -1754,7 +1755,7 @@
     names of internal elements anyway.
 
 -   The vp8enc element now expects bps (bits per second) for the
-    "temporal-scalability-target-bitrate" property, which is consistent
+    "temporal-scalability-target-bitrate” property, which is consistent
     with the "target-bitrate" property. Since additional configuration
     is required with modern libvpx to make temporal scaling work anyway,
     chances are that very few people will have been using this property
@@ -2503,15 +2504,293 @@
 -   List of Merge Requests applied in 1.20.3
 -   List of Issues fixed in 1.20.3
 
+1.20.4
+
+The fourth 1.20 bug-fix release (1.20.4) was released on 12 October
+2022.
+
+This release only contains bugfixes and it should be safe to upgrade
+from 1.20.x.
+
+Highlighted bugfixes in 1.20.4
+
+-   avaudiodec: fix playback issue with WMA files, would throw an error
+    at EOS with FFmpeg 5.x
+-   Fix deadlock when loading gst-editing-services plugin
+-   Fix input buffering capacity in live mode for aggregator,
+    video/audio aggregator subclasses, muxers
+-   glimagesink: fix crash on Android
+-   subtitle handling and subtitle overlay fixes
+-   matroska-mux: allow width + height changes for avc3|hev1|vp8|vp9
+-   rtspsrc: fix control url handling for spec compliant servers and add
+    fallback for incompliant servers
+-   WebRTC fixes
+-   RTP retransmission fixes
+-   video: fixes for formats with 4x subsampling and horizontal co-sited
+    chroma (Y41B, YUV9, YVU9 and IYU9)
+-   macOS build and packaging fixes, in particular fix finding of gio
+    modules on macOS for https/TLS support
+-   Performance improvements
+-   Miscellaneous bug fixes, memory leak fixes, and other stability and
+    reliability improvements
+
+gstreamer
+
+-   buffer: drop parent meta in deep copy/foreach_metadata
+-   devicemonitor: Use a sync bus handler for the provider to avoid
+    accumulating all messages until the provider is stopped
+-   element: Fix requesting of pads with string templates
+-   gst: Protect initialization state with a recursive mutex
+-   gst: add missing define guard for build without gstreamer debug
+    logging support
+-   gst_init: Initialize static plugins just before dynamic plugins
+-   info: Parse “NONE” as a valid level name
+-   meta: Set the parent refcount of the GstStructure correctly
+-   pluginloader: Don’t hang on short reads/writes
+-   tracers: leaks: fix potentially invalid memory access when trying to
+    detect object type
+-   tracers: leaks: fix object-refings.class flags
+-   uri: When setting the same string again do nothing
+-   value: Don’t loop forever when serializing invalid flag
+
+Base Libraries
+
+-   aggregator: fix input buffering in live mode (was too low before in
+    many cases)
+-   aggregator: fix reversed active/flushing arguments in debug log
+    output
+-   aggregator: Reset EOS flag after receiving a stream-start event
+
+Core Elements
+
+-   queue2: Hold the lock when modifying sinkresult
+-   queue2: Fix deadlock when deactivate is called in pull mode
+
+gst-plugins-base
+
+-   decodebin3: fix mutex leaks
+-   decodebin3: Fix memory issues with active selection list
+-   decodebin3, uridecodebin3, urisourcebin: Event handling fixes
+-   decodebin3: fix EOS event sequence
+-   parsebin: Avoid crash with unknown streams
+-   parsebin: SIGSEGV during HLS stream using souphttpsrc
+-   glimagesink: only allow setting the GL display/context if it is a
+    valid value
+-   glimagesink: segfault on android devices
+-   gstgl: Fix several memory leaks in macOS
+-   opusenc: improve inband-fec property documentation
+-   playsink: Hold a reference to the soft volume element
+-   pbutils: descriptions: fix gst_pb_utils_get_caps_description_flags()
+-   rtspurl: Use gst_uri_join_strings() in
+    gst_rtsp_url_get_request_uri_with_control() instead of a
+    hand-crafted, wrong version
+-   rtspconnection: protect cancellable by a mutex
+-   sdpmessage: Don’t set SDP medias from caps without
+    media/payload/clock-rate fields
+-   samiparse: fix handling of self-closing tags
+-   ssaparse: include required system headers for isspace() and sscanf()
+    functions
+-   subparse: fix crash when parsing invalid timestamps in mpl2
+-   subparse fixes
+-   textoverlay: Don’t miscalculate text running times
+-   videoaggregator: always convert when user provides converter-config
+-   video: Fix scaling in 4x horizontal co-sited chroma (Y41B, YUV9,
+    YVU9 and IYU9)
+-   xmptag: register musicbrainz tags during init to fix critical in
+    jpegparse
+-   xvimagesink: fix image leaks in error code path
+-   tests: skip unit tests for dependency-less elements that have been
+    disabled
+
+Tools
+
+-   No changes
+
+gst-plugins-good
+
+-   alpha: fix stride issue when out buffer has padding on right
+-   isoff: Fix earliest pts field parse issue
+-   matroska-mux: allow width + height changes for avc3|hev1|vp8|vp9
+-   qt: Fix another instance of Qt/GStreamer both defining GLsync
+    differently
+-   qtdemux: Avoid crash on reconfiguring.
+-   qtdemux: guard against timestamp calculation overflow in gap event
+    loop
+-   qtdemux: Don’t use invalid values from failed trex parsing
+-   qtdemux: possible endless loop
+-   rtpjitterbuffer: Only unschedule timers for late packets if they’re
+    not RTX packets and only once
+-   rtpjitterbuffer: remove lost timer for out of order packets
+-   rtspsrc: SETUP generates 400 Bad Request
+-   rtspsrc: Retry SETUP with non-compliant URL resolution on “Bad
+    Request” and “Not found”
+-   rtpst2022-1-fecenc: Drain column packets on EOS
+-   rtpvp8depay: If configured to wait for keyframes after packet loss,
+    also do that if incomplete frames are detected
+-   splitmuxsink: Don’t crash on EOS without buffer
+-   splitmuxsrc: Stop pad task before cleanup
+-   splitmuxsrc: don’t consider unlinked pads when deactivating part
+-   soup: libsoup3 makes audio streaming stop
+-   v4l2: fix critical when unreferencign buffer with no data
+-   v4l2bufferpool: Fix debug trace
+-   v4l2object: Add support for Apple’s full-range bt709 colorspace
+    variant 1:3:5:1
+-   v4l2videocodec: workaround for failure to fully drain frames
+    preceding MIDSTREAM renegotiation
+-   v4l2allocator: Fix invalid imported dmabuf fd
+-   videoflip: Fix caps negotiation when method is selected
+-   build failure trying to build jack examples
+-   examples: don’t try and build jack examples if jack was disabled
+-   tests: skip unit tests for dependency-less elements that have been
+    disabled
+
+gst-plugins-bad
+
+-   amcvideodec: fix GstAmcSurfaceTexture segfault
+-   audiobuffersplit: Fix drift that was introduced by wrong
+    calculations in gapless mode
+-   avfvideosrc: Fix wrong default framerate value
+-   audiovisualizer: fix buffer mapping to not increase refcount
+-   d3d11decoder: Check 16K resolution support
+-   d3d11videosink: Fix for force-aspect-ratio setting when rendering on
+    shared texture
+-   mxfdemux: Always calculate BlockAlign of raw audio to work around
+    files with broken BlockAlign field in the headers
+-   nvdec: Fix for HEVC decoding when coded resolution is larger than
+    display resolution
+-   openh264: Register debug categories earlier
+-   openh264enc: Fix constrained-high encoding
+-   openmpt: update from now deprecated api
+-   GstPlay: missing cleanup for g_autoptr
+-   player/play: Fix object construction and various leaks
+-   player: Plug a memory leak
+-   proxysink: Make sure stream-start and caps events are forwarded, and
+    fix memory leak
+-   tsdemux: Don’t trigger a program change when falling back to
+    ignore-pcr behaviour
+-   va: allocator: Fix translation of VADRMPRIMESurfaceDescriptor
+-   va: h265dec: Fix a crash because of missing reference frame.
+-   vah265dec: Decoder segfaults on seek
+-   wasapi: Implement default audio channel mask
+-   wasapi2: Fix initial mute/volume setting
+-   webrtcbin: Limit sink query to sink pads
+-   webrtcbin: Fix pointer dereference before null check
+-   webrtc: Make sure to return NULL when validating TURN server fails
+-   tests: skip unit tests for dependency-less elements that have been
+    disabled
+
+gst-plugins-ugly
+
+-   tests: skip unit tests for dependency-less elements that have been
+    disabled
+
+gst-libav
+
+-   avauddec: fix regression with WMA files, would throw an error at EOS
+-   avauddec: fix unnecessary reconfiguration if the audio layout isn’t
+    specified
+-   libav: Fix for APNG encoder property registration
+-   Failure to decode end of WMA file
+
+gst-rtsp-server
+
+-   gst-rtsp-server: Fix pushing backlog to client
+-   rtsp-server: stream: Don’t loop forever if binding to the multicast
+    address fails
+
+gstreamer-vaapi
+
+-   vaapi: Handle when no encoders/decoders available.
+-   vaapi: Crash in gst_vaapidecode_class_init() when no
+    decoders/encoders available
+
+gstreamer-sharp
+
+-   No changes
+
+gst-omx
+
+-   No changes
+
+gst-python
+
+-   python: Do not call gst_init when it is already is_initialized
+
+gst-editing-services
+
+-   Deadlock in ges because of recursive gst_init() call
+-   ges/gstframepositioner: don’t create one compositor per frame meta
+-   nle: clear seek event properly
+
+gst-examples:
+
+-   examples/webrtc/signalling: Fix compatibility with Python 3.10
+
+Development build environment + gst-full build
+
+-   build: Fix some compiler warnings by upgrading wraps
+-   dv, opusparse: fix duplicate symbols in static build
+-   Fix fedora 36 warnings - OpenSSL 3.0 deprecations + GLib 2.72
+    tls-validation deprecations
+-   Various macOS build fixes
+-   meson: Improve certifi documentation on macOS
+
+Cerbero build tool and packaging changes in 1.20.4
+
+-   Add Ubuntu 22.04 Jammy Jellyfish
+-   Add gst-rtsp-server library to the macOS framework
+-   cerbero: Quick fix for gen-cache breakage
+-   macos: Fix the install_name for the GStreamer framework
+-   Download using powershell on Windows and rework download func
+-   macos: Add arm64 to the metadata for the installer
+-   cerbero: Allow building on Linux ARM64
+-   pkg-config.recipe: Add to core platform files list
+-   git: Fix issue with last security patch
+-   distros: Fix CentOS allowance
+-   cerbero: Print working directory for commands that are run
+-   cerbero: Fix license property usage example
+-   Fix issue getting distro_version in Debian Bookworm
+-   glib: Fix gio modules loading on macOS
+-   cmake: Fix macOS ARM64 -> x86_64 cross-compilation
+-   Fix logo display in macOS installer
+-   openssl.recipe: Fix segfault on latest macOS
+-   msvc: Fix for broken CRT linking at application project because of
+    MSVCRT linking
+-   cerbero: Do not add rpaths that already exist on macOS
+-   android: fix build with android gradle plugin 7.2
+-   macOS framework is unusable starting from 1.18.0
+
+Contributors to 1.20.4
+
+Adrian Fiergolski, Aleksandr Slobodeniuk, Andoni Morales Alastruey,
+Andrew Pritchard, Bruce Liang, Corentin Damman, Daniel Morin, Edward
+Hervey, Elliot Chen, Fabian Orccon, fduncanh, Guillaume Desmottes,
+Haihua Hu, He Junyan, Ignazio Pillai, James Cowgill, James Hilliard, Jan
+Alexander Steffens (heftig), Jan Schmidt, Jianhui Dai, Jonas Danielsson,
+Jordan Petridis, Khem Raj, Krystian Wojtas, Martin Dørum, Mart Raudsepp,
+Mathieu Duponchelle, Matthew Waters, Matthias Clasen, Nicolas Dufresne,
+Nirbheek Chauhan, Olivier Crête, Paweł Stawicki, Philippe Normand,
+Philipp Zabel, Piotr Brzeziński, Rafael Caricio, Rafael Sobral, Raul
+Tambre, Ruben Gonzalez, Sangchul Lee, Sebastian Dröge, Seungha Yang,
+Stéphane Cerveau, Thibault Saunier, Tim-Philipp Müller, Tristan
+Matthews, Víctor Manuel Jáquez Leal, Xavier Claessens, Zhiyuan Liu,
+
+… 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 1.20.4
+
+-   List of Merge Requests applied in 1.20.4
+-   List of Issues fixed in 1.20.4
+
 Schedule for 1.22
 
 Our next major feature release will be 1.22, and 1.21 will be the
 unstable development version leading up to the stable 1.22 release. The
 development of 1.21/1.22 will happen in the git main branch.
 
-The plan for the 1.22 development cycle is yet to be confirmed. Assuming
-no major project-wide reorganisations in the 1.22 cycle we might try and
-aim for a release around August 2022.
+The plan for the 1.22 development cycle is now confirmed, and we aim for
+a 1.22.0 release in December 2022.
 
 1.22 will be backwards-compatible to the stable 1.20, 1.18, 1.16, 1.14,
 1.12, 1.10, 1.8, 1.6, 1.4, 1.2 and 1.0 release series.

gst-plugins-bad-1.20.3.tar.xz/RELEASE -> gst-plugins-bad-1.20.4.tar.xz/RELEASE Changed

gst-plugins-bad-1.20.3.tar.xz/ext/dtls/gstdtlscertificate.c -> gst-plugins-bad-1.20.4.tar.xz/ext/dtls/gstdtlscertificate.c Changed

@@ -221,14 +221,24 @@
 #if OPENSSL_VERSION_NUMBER < 0x10100001L
   rsa = RSA_generate_key (2048, RSA_F4, NULL, NULL);
 #else
+  /*
+   * OpenSSL 3.0 deprecated all low-level APIs, so we need to rewrite this code
+   * to get rid of the warnings. The porting guide explicitly recommends
+   * disabling the warnings if this is not feasible, so let's do that for now:
+   * https://wiki.openssl.org/index.php/OpenSSL_3.0#Upgrading_to_OpenSSL_3.0_from_OpenSSL_1.1.1
+   */
+  G_GNUC_BEGIN_IGNORE_DEPRECATIONS;
   rsa = RSA_new ();
+  G_GNUC_END_IGNORE_DEPRECATIONS;
   if (rsa != NULL) {
     BIGNUM *e = BN_new ();
+    G_GNUC_BEGIN_IGNORE_DEPRECATIONS;
     if (e == NULL || !BN_set_word (e, RSA_F4)
         || !RSA_generate_key_ex (rsa, 2048, e, NULL)) {
       RSA_free (rsa);
       rsa = NULL;
     }
+    G_GNUC_END_IGNORE_DEPRECATIONS;
     if (e)
       BN_free (e);
   }
@@ -236,16 +246,20 @@
 
   if (!rsa) {
     GST_WARNING_OBJECT (self, "failed to generate RSA");
+    G_GNUC_BEGIN_IGNORE_DEPRECATIONS;
     EVP_PKEY_free (priv->private_key);
+    G_GNUC_END_IGNORE_DEPRECATIONS;
     priv->private_key = NULL;
     X509_free (priv->x509);
     priv->x509 = NULL;
     return;
   }
 
+  G_GNUC_BEGIN_IGNORE_DEPRECATIONS;
   if (!EVP_PKEY_assign_RSA (priv->private_key, rsa)) {
     GST_WARNING_OBJECT (self, "failed to assign RSA");
     RSA_free (rsa);
+    G_GNUC_END_IGNORE_DEPRECATIONS;
     rsa = NULL;
     EVP_PKEY_free (priv->private_key);
     priv->private_key = NULL;
@@ -259,7 +273,9 @@
 
   /* Set a random 64 bit integer as serial number */
   serial_number = BN_new ();
+  G_GNUC_BEGIN_IGNORE_DEPRECATIONS;
   BN_pseudo_rand (serial_number, 64, 0, 0);
+  G_GNUC_END_IGNORE_DEPRECATIONS;
   asn1_serial_number = X509_get_serialNumber (priv->x509);
   BN_to_ASN1_INTEGER (serial_number, asn1_serial_number);
   BN_free (serial_number);

gst-plugins-bad-1.20.3.tar.xz/ext/openh264/gstopenh264dec.cpp -> gst-plugins-bad-1.20.4.tar.xz/ext/openh264/gstopenh264dec.cpp Changed

gst-plugins-bad-1.20.3.tar.xz/ext/openh264/gstopenh264enc.cpp -> gst-plugins-bad-1.20.4.tar.xz/ext/openh264/gstopenh264enc.cpp Changed

@@ -234,10 +234,7 @@
 /* class initialization */
 
 G_DEFINE_TYPE_WITH_CODE (GstOpenh264Enc, gst_openh264enc,
-    GST_TYPE_VIDEO_ENCODER,
-    G_IMPLEMENT_INTERFACE (GST_TYPE_PRESET, NULL);
-    GST_DEBUG_CATEGORY_INIT (gst_openh264enc_debug_category, "openh264enc", 0,
-        "debug category for openh264enc element"));
+    GST_TYPE_VIDEO_ENCODER, G_IMPLEMENT_INTERFACE (GST_TYPE_PRESET, NULL));
 GST_ELEMENT_REGISTER_DEFINE_CUSTOM (openh264enc, openh264enc_element_init);
 
 static void
@@ -711,11 +708,12 @@
 
   gst_structure_set (s, "profile", G_TYPE_STRING, profile, NULL);
   if (!g_strcmp0 (profile, "constrained-baseline") ||
-       !g_strcmp0 (profile, "baseline"))
+      !g_strcmp0 (profile, "baseline"))
     return PRO_BASELINE;
-   else if (!g_strcmp0 (profile, "main"))
+  else if (!g_strcmp0 (profile, "main"))
     return PRO_MAIN;
-   else if (!g_strcmp0 (profile, "high"))
+  else if (!g_strcmp0 (profile, "high") ||
+      !g_strcmp0 (profile, "constrained-high"))
     return PRO_HIGH;
 
   g_assert_not_reached ();
@@ -1056,13 +1054,16 @@
 
   return GST_FLOW_OK;
 }
+
 static gboolean
 openh264enc_element_init (GstPlugin * plugin)
 {
+  GST_DEBUG_CATEGORY_INIT (gst_openh264enc_debug_category, "openh264enc", 0,
+      "debug category for openh264enc element");
   if (openh264_element_init (plugin))
     return gst_element_register (plugin, "openh264enc", GST_RANK_MARGINAL,
-                                 GST_TYPE_OPENH264ENC);
+        GST_TYPE_OPENH264ENC);
 
- GST_ERROR ("Incorrect library version loaded, expecting %s", g_strCodecVer);
- return FALSE;
+  GST_ERROR ("Incorrect library version loaded, expecting %s", g_strCodecVer);
+  return FALSE;
 }

gst-plugins-bad-1.20.3.tar.xz/ext/openmpt/gstopenmptdec.c -> gst-plugins-bad-1.20.4.tar.xz/ext/openmpt/gstopenmptdec.c Changed

gst-plugins-bad-1.20.3.tar.xz/ext/opus/gstopusheader.h -> gst-plugins-bad-1.20.4.tar.xz/ext/opus/gstopusheader.h Changed

gst-plugins-bad-1.20.3.tar.xz/ext/sctp/usrsctp/meson.build -> gst-plugins-bad-1.20.4.tar.xz/ext/sctp/usrsctp/meson.build Changed

gst-plugins-bad-1.20.3.tar.xz/ext/webrtc/gstwebrtcbin.c -> gst-plugins-bad-1.20.4.tar.xz/ext/webrtc/gstwebrtcbin.c Changed

@@ -488,8 +488,10 @@
       direction, "template", template, NULL);
   gst_object_unref (template);
 
-  gst_pad_set_event_function (GST_PAD (pad), gst_webrtcbin_sink_event);
-  gst_pad_set_query_function (GST_PAD (pad), gst_webrtcbin_sink_query);
+  if (direction == GST_PAD_SINK) {
+    gst_pad_set_event_function (GST_PAD (pad), gst_webrtcbin_sink_event);
+    gst_pad_set_query_function (GST_PAD (pad), gst_webrtcbin_sink_query);
+  }
 
   gst_pad_add_probe (GST_PAD (pad), GST_PAD_PROBE_TYPE_BUFFER |
       GST_PAD_PROBE_TYPE_BUFFER_LIST, webrtc_bin_pad_buffer_cb, NULL, NULL);
@@ -3085,7 +3087,15 @@
 
     /* this only looks at the first structure so we loop over the given caps
      * and add each structure inside it piecemeal */
-    gst_sdp_media_set_media_from_caps (format, media);
+    if (gst_sdp_media_set_media_from_caps (format, media) != GST_SDP_OK) {
+      GST_ERROR_OBJECT (webrtc,
+          "Failed to build media from caps %" GST_PTR_FORMAT
+          " for transceiver %" GST_PTR_FORMAT, format, trans);
+      gst_caps_unref (caps);
+      gst_caps_unref (format);
+      gst_structure_free (extmap);
+      return FALSE;
+    }
 
     gst_caps_unref (format);
   }
@@ -4125,7 +4135,13 @@
         }
       }
 
-      gst_sdp_media_set_media_from_caps (answer_caps, media);
+      if (gst_sdp_media_set_media_from_caps (answer_caps, media) != GST_SDP_OK) {
+        GST_WARNING_OBJECT (webrtc,
+            "Could not build media from caps %" GST_PTR_FORMAT, answer_caps);
+        gst_clear_caps (&answer_caps);
+        gst_clear_caps (&offer_caps);
+        goto rejected;
+      }
 
       _get_rtx_target_pt_and_ssrc_from_caps (answer_caps, &target_pt,
           &target_ssrc);
@@ -5746,6 +5762,12 @@
         continue;
       }
 
+      if (!pad->trans) {
+        GST_LOG_OBJECT (pad, "doesn't have a transceiver");
+        tmp = tmp->next;
+        continue;
+      }
+
       if (pad->trans->mline >= gst_sdp_message_medias_len (sd->sdp->sdp)) {
         GST_DEBUG_OBJECT (pad, "not mentioned in this description. Skipping");
         tmp = tmp->next;
@@ -5761,12 +5783,6 @@
         tmp = tmp->next;
         continue;
       }
-
-      if (!pad->trans) {
-        GST_LOG_OBJECT (pad, "doesn't have a transceiver");
-        tmp = tmp->next;
-        continue;
-      }
 
       new_dir = pad->trans->direction;
       if (new_dir != GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDONLY &&

gst-plugins-bad-1.20.3.tar.xz/ext/webrtc/gstwebrtcice.c -> gst-plugins-bad-1.20.4.tar.xz/ext/webrtc/gstwebrtcice.c Changed

gst-plugins-bad-1.20.3.tar.xz/gst-libs/gst/play/gstplay-media-info.h -> gst-plugins-bad-1.20.4.tar.xz/gst-libs/gst/play/gstplay-media-info.h Changed

gst-plugins-bad-1.20.3.tar.xz/gst-libs/gst/play/gstplay-signal-adapter.h -> gst-plugins-bad-1.20.4.tar.xz/gst-libs/gst/play/gstplay-signal-adapter.h Changed

gst-plugins-bad-1.20.3.tar.xz/gst-libs/gst/play/gstplay-video-overlay-video-renderer.h -> gst-plugins-bad-1.20.4.tar.xz/gst-libs/gst/play/gstplay-video-overlay-video-renderer.h Changed

gst-plugins-bad-1.20.3.tar.xz/gst-libs/gst/play/gstplay-video-renderer.h -> gst-plugins-bad-1.20.4.tar.xz/gst-libs/gst/play/gstplay-video-renderer.h Changed

gst-plugins-bad-1.20.3.tar.xz/gst-libs/gst/play/gstplay-visualization.h -> gst-plugins-bad-1.20.4.tar.xz/gst-libs/gst/play/gstplay-visualization.h Changed

gst-plugins-bad-1.20.3.tar.xz/gst-libs/gst/play/gstplay.c -> gst-plugins-bad-1.20.4.tar.xz/gst-libs/gst/play/gstplay.c Changed

@@ -212,6 +212,8 @@
 
 static gpointer gst_play_main (gpointer data);
 
+static void gst_play_set_playbin_video_sink (GstPlay * self);
+
 static void gst_play_seek_internal_locked (GstPlay * self);
 static void gst_play_stop_internal (GstPlay * self, gboolean transient);
 static gboolean gst_play_pause_internal (gpointer user_data);
@@ -509,6 +511,8 @@
   self->thread = g_thread_new ("GstPlay", gst_play_main, self);
   while (!self->loop || !g_main_loop_is_running (self->loop))
     g_cond_wait (&self->cond, &self->lock);
+
+  gst_play_set_playbin_video_sink (self);
   g_mutex_unlock (&self->lock);
 
   G_OBJECT_CLASS (parent_class)->constructed (object);
@@ -594,11 +598,16 @@
 {
   GstElement *video_sink = NULL;
 
-  if (self->video_renderer != NULL)
+  if (self->video_renderer != NULL) {
     video_sink =
         gst_play_video_renderer_create_video_sink (self->video_renderer, self);
-  if (video_sink)
+  }
+
+  if (video_sink) {
+    gst_object_ref_sink (video_sink);
     g_object_set (self->playbin, "video-sink", video_sink, NULL);
+    gst_object_unref (video_sink);
+  }
 }
 
 static void
@@ -612,7 +621,14 @@
       g_mutex_lock (&self->lock);
       g_clear_object (&self->video_renderer);
       self->video_renderer = g_value_dup_object (value);
-      gst_play_set_playbin_video_sink (self);
+
+      // When the video_renderer is a GstPlayerWrappedVideoRenderer it cannot be set
+      // at construction time because it requires a valid pipeline which is created
+      // only after GstPlay has been constructed. That is why the video renderer is
+      // set *after* GstPlay has been constructed.
+      if (self->thread) {
+        gst_play_set_playbin_video_sink (self);
+      }
       g_mutex_unlock (&self->lock);
       break;
     case PROP_URI:{
@@ -2648,15 +2664,8 @@
 
   g_once (&once, gst_play_init_once, NULL);
 
-  self = g_object_new (GST_TYPE_PLAY, NULL);
+  self = g_object_new (GST_TYPE_PLAY, "video-renderer", video_renderer, NULL);
 
-  // When the video_renderer is a GstPlayerWrappedVideoRenderer it cannot be set
-  // at construction time because it requires a valid pipeline which is created
-  // only after GstPlay has been constructed. That is why the video renderer is
-  // set *after* GstPlay has been constructed.
-  if (video_renderer != NULL) {
-    g_object_set (self, "video-renderer", video_renderer, NULL);
-  }
   gst_object_ref_sink (self);
 
   if (video_renderer)

gst-plugins-bad-1.20.3.tar.xz/gst-libs/gst/player/gstplayer-media-info.c -> gst-plugins-bad-1.20.4.tar.xz/gst-libs/gst/player/gstplayer-media-info.c Changed

gst-plugins-bad-1.20.3.tar.xz/gst-libs/gst/player/gstplayer.c -> gst-plugins-bad-1.20.4.tar.xz/gst-libs/gst/player/gstplayer.c Changed

@@ -157,6 +157,7 @@
 
   /* legacy */
   GstPlayerSignalDispatcher *signal_dispatcher;
+  GstPlayerVideoRenderer *video_renderer;
 };
 
 struct _GstPlayerClass
@@ -175,11 +176,12 @@
     const GValue * value, GParamSpec * pspec);
 static void gst_player_get_property (GObject * object, guint prop_id,
     GValue * value, GParamSpec * pspec);
+static void gst_player_constructed (GObject * object);
 
 static void
-gst_player_init (G_GNUC_UNUSED GstPlayer * self)
+gst_player_init (GstPlayer * self)
 {
-
+  self->play = gst_play_new (NULL);
 }
 
 static void
@@ -207,12 +209,13 @@
   gobject_class->set_property = gst_player_set_property;
   gobject_class->get_property = gst_player_get_property;
   gobject_class->finalize = gst_player_finalize;
+  gobject_class->constructed = gst_player_constructed;
 
   param_specsPROP_VIDEO_RENDERER =
       g_param_spec_object ("video-renderer",
       "Video Renderer", "Video renderer to use for rendering videos",
       GST_TYPE_PLAYER_VIDEO_RENDERER,
-      G_PARAM_READABLE | G_PARAM_STATIC_STRINGS);
+      G_PARAM_READWRITE | G_PARAM_CONSTRUCT_ONLY | G_PARAM_STATIC_STRINGS);
 
   param_specsPROP_SIGNAL_DISPATCHER =
       g_param_spec_object ("signal-dispatcher",
@@ -377,6 +380,10 @@
 
   if (self->signal_dispatcher)
     g_object_unref (self->signal_dispatcher);
+  if (self->video_renderer)
+    g_object_unref (self->video_renderer);
+  if (self->signal_adapter)
+    g_object_unref (self->signal_adapter);
   if (self->play)
     gst_object_unref (self->play);
 
@@ -393,6 +400,9 @@
     case PROP_SIGNAL_DISPATCHER:
       self->signal_dispatcher = g_value_dup_object (value);
       break;
+    case PROP_VIDEO_RENDERER:
+      self->video_renderer = g_value_dup_object (value);
+      break;
     default:
       g_object_set_property (G_OBJECT (self->play),
           g_param_spec_get_name (pspec), value);
@@ -407,6 +417,9 @@
   GstPlayer *self = GST_PLAYER (object);
 
   switch (prop_id) {
+    case PROP_VIDEO_RENDERER:
+      g_value_set_object (value, self->video_renderer);
+      break;
     case PROP_MEDIA_INFO:
       g_value_take_object (value, gst_player_get_media_info (self));
       break;
@@ -544,47 +557,27 @@
   g_signal_emit (self, signalsSIGNAL_SEEK_DONE, 0, time);
 }
 
-/**
- * gst_player_new:
- * @video_renderer: (transfer full) (allow-none): GstPlayerVideoRenderer to use
- * @signal_dispatcher: (transfer full) (allow-none): GstPlayerSignalDispatcher to use
- *
- * Creates a new #GstPlayer instance that uses @signal_dispatcher to dispatch
- * signals to some event loop system, or emits signals directly if NULL is
- * passed. See gst_player_g_main_context_signal_dispatcher_new().
- *
- * Video is going to be rendered by @video_renderer, or if %NULL is provided
- * no special video set up will be done and some default handling will be
- * performed.
- *
- * Returns: (transfer full): a new #GstPlayer instance
- */
-GstPlayer *
-gst_player_new (GstPlayerVideoRenderer * video_renderer,
-    GstPlayerSignalDispatcher * signal_dispatcher)
+static void
+gst_player_constructed (GObject * object)
 {
-  static GOnce once = G_ONCE_INIT;
-  GstPlayer *self;
+  GstPlayer *self = GST_PLAYER (object);
   GstPlayerVideoRenderer *renderer = NULL;
 
-  g_once (&once, gst_player_init_once, NULL);
-
-  self =
-      g_object_new (GST_TYPE_PLAYER, "signal-dispatcher", signal_dispatcher,
-      NULL);
-
-  self->play = gst_play_new (NULL);
+  G_OBJECT_CLASS (parent_class)->constructed (object);
 
-  if (video_renderer != NULL) {
-    renderer = gst_player_wrapped_video_renderer_new (video_renderer, self);
+  if (self->video_renderer != NULL) {
+    renderer =
+        gst_player_wrapped_video_renderer_new (self->video_renderer, self);
     g_object_set (self->play, "video-renderer",
         GST_PLAY_VIDEO_RENDERER (renderer), NULL);
+    g_object_unref (renderer);
   }
 
-  if (signal_dispatcher != NULL) {
+  if (self->signal_dispatcher != NULL) {
     GMainContext *context = NULL;
 
-    g_object_get (signal_dispatcher, "application-context", &context, NULL);
+    g_object_get (self->signal_dispatcher, "application-context", &context,
+        NULL);
     self->signal_adapter =
         gst_play_signal_adapter_new_with_main_context (self->play, context);
     g_main_context_unref (context);
@@ -592,8 +585,6 @@
     self->signal_adapter = gst_play_signal_adapter_new (self->play);
   }
 
-  gst_object_ref_sink (self);
-
   g_signal_connect (self->signal_adapter, "uri-loaded",
       G_CALLBACK (uri_loaded_cb), self);
   g_signal_connect (self->signal_adapter, "position-updated",
@@ -619,6 +610,37 @@
       self);
   g_signal_connect (self->signal_adapter, "seek-done",
       G_CALLBACK (seek_done_cb), self);
+}
+
+/**
+ * gst_player_new:
+ * @video_renderer: (transfer full) (allow-none): GstPlayerVideoRenderer to use
+ * @signal_dispatcher: (transfer full) (allow-none): GstPlayerSignalDispatcher to use
+ *
+ * Creates a new #GstPlayer instance that uses @signal_dispatcher to dispatch
+ * signals to some event loop system, or emits signals directly if NULL is
+ * passed. See gst_player_g_main_context_signal_dispatcher_new().
+ *
+ * Video is going to be rendered by @video_renderer, or if %NULL is provided
+ * no special video set up will be done and some default handling will be
+ * performed.
+ *
+ * Returns: (transfer full): a new #GstPlayer instance
+ */
+GstPlayer *
+gst_player_new (GstPlayerVideoRenderer * video_renderer,
+    GstPlayerSignalDispatcher * signal_dispatcher)
+{
+  static GOnce once = G_ONCE_INIT;
+  GstPlayer *self;
+
+  g_once (&once, gst_player_init_once, NULL);
+
+  self =
+      g_object_new (GST_TYPE_PLAYER, "signal-dispatcher", signal_dispatcher,
+      "video-renderer", video_renderer, NULL);
+
+  gst_object_ref_sink (self);
 
   if (video_renderer)
     g_object_unref (video_renderer);

gst-plugins-bad-1.20.3.tar.xz/gst-plugins-bad.doap -> gst-plugins-bad-1.20.4.tar.xz/gst-plugins-bad.doap Changed

gst-plugins-bad-1.20.3.tar.xz/gst/audiobuffersplit/gstaudiobuffersplit.c -> gst-plugins-bad-1.20.4.tar.xz/gst/audiobuffersplit/gstaudiobuffersplit.c Changed

@@ -540,84 +540,81 @@
         avail_samples, GST_SECOND, rate * self->in_segment.rate);
   }
 
-  if (self->gapless) {
-    if (self->current_offset != -1) {
-      GST_DEBUG_OBJECT (self,
-          "Got discont in gapless mode: Current running time %" GST_TIME_FORMAT
-          ", current end running time %" GST_TIME_FORMAT
-          ", running time after discont %" GST_TIME_FORMAT,
-          GST_TIME_ARGS (current_rt),
-          GST_TIME_ARGS (current_rt_end), GST_TIME_ARGS (input_rt));
+  if (self->gapless && self->current_offset != -1) {
+    GST_DEBUG_OBJECT (self,
+        "Got discont in gapless mode: Current running time %" GST_TIME_FORMAT
+        ", current end running time %" GST_TIME_FORMAT
+        ", running time after discont %" GST_TIME_FORMAT,
+        GST_TIME_ARGS (current_rt),
+        GST_TIME_ARGS (current_rt_end), GST_TIME_ARGS (input_rt));
 
-      new_offset =
-          gst_util_uint64_scale (current_rt - self->resync_rt,
-          rate * ABS (self->in_segment.rate), GST_SECOND);
-      if (current_rt < self->resync_rt) {
-        guint64 drop_samples;
+    new_offset =
+        gst_util_uint64_scale (input_rt - self->resync_rt,
+        rate * ABS (self->in_segment.rate), GST_SECOND);
+    if (input_rt < self->resync_rt) {
+      guint64 drop_samples;
 
-        new_offset =
-            gst_util_uint64_scale (self->resync_rt -
-            current_rt, rate * ABS (self->in_segment.rate), GST_SECOND);
-        drop_samples = self->current_offset + avail_samples + new_offset;
+      new_offset =
+          gst_util_uint64_scale (self->resync_rt -
+          input_rt, rate * ABS (self->in_segment.rate), GST_SECOND);
+      drop_samples = self->current_offset + avail_samples + new_offset;
 
+      GST_DEBUG_OBJECT (self,
+          "Dropping %" G_GUINT64_FORMAT " samples (%" GST_TIME_FORMAT ")",
+          drop_samples, GST_TIME_ARGS (gst_util_uint64_scale (drop_samples,
+                  GST_SECOND, rate)));
+      self->drop_samples = drop_samples;
+      discont = FALSE;
+    } else if (new_offset > self->current_offset + avail_samples) {
+      guint64 silence_samples =
+          new_offset - (self->current_offset + avail_samples);
+      const GstAudioFormatInfo *info = gst_audio_format_get_info (format);
+      GstClockTime silence_time =
+          gst_util_uint64_scale (silence_samples, GST_SECOND, rate);
+
+      if (silence_time > self->max_silence_time) {
         GST_DEBUG_OBJECT (self,
-            "Dropping %" G_GUINT64_FORMAT " samples (%" GST_TIME_FORMAT ")",
-            drop_samples, GST_TIME_ARGS (gst_util_uint64_scale (drop_samples,
-                    GST_SECOND, rate)));
-        discont = FALSE;
-      } else if (new_offset > self->current_offset + avail_samples) {
-        guint64 silence_samples =
-            new_offset - (self->current_offset + avail_samples);
-        const GstAudioFormatInfo *info = gst_audio_format_get_info (format);
-        GstClockTime silence_time =
-            gst_util_uint64_scale (silence_samples, GST_SECOND, rate);
-
-        if (silence_time > self->max_silence_time) {
-          GST_DEBUG_OBJECT (self,
-              "Not inserting %" G_GUINT64_FORMAT " samples of silence (%"
-              GST_TIME_FORMAT " exceeds maximum %" GST_TIME_FORMAT ")",
-              silence_samples, GST_TIME_ARGS (silence_time),
-              GST_TIME_ARGS (self->max_silence_time));
-        } else {
-          GST_DEBUG_OBJECT (self,
-              "Inserting %" G_GUINT64_FORMAT " samples of silence (%"
-              GST_TIME_FORMAT ")", silence_samples,
-              GST_TIME_ARGS (silence_time));
-
-          /* Insert silence buffers to fill the gap in 1s chunks */
-          while (silence_samples > 0) {
-            guint n_samples = MIN (silence_samples, rate);
-            GstBuffer *silence;
-            GstMapInfo map;
-
-            silence = gst_buffer_new_and_alloc (n_samples * bpf);
-            GST_BUFFER_FLAG_SET (silence, GST_BUFFER_FLAG_GAP);
-            gst_buffer_map (silence, &map, GST_MAP_WRITE);
-            gst_audio_format_info_fill_silence (info, map.data, map.size);
-            gst_buffer_unmap (silence, &map);
-
-            gst_adapter_push (self->adapter, silence);
-            ret =
-                gst_audio_buffer_split_output (self, FALSE, rate, bpf,
-                samples_per_buffer);
-            if (ret != GST_FLOW_OK)
-              return ret;
-
-            silence_samples -= n_samples;
-          }
-          discont = FALSE;
-        }
-      } else if (new_offset < self->current_offset + avail_samples) {
-        guint64 drop_samples =
-            self->current_offset + avail_samples - new_offset;
-
+            "Not inserting %" G_GUINT64_FORMAT " samples of silence (%"
+            GST_TIME_FORMAT " exceeds maximum %" GST_TIME_FORMAT ")",
+            silence_samples, GST_TIME_ARGS (silence_time),
+            GST_TIME_ARGS (self->max_silence_time));
+      } else {
         GST_DEBUG_OBJECT (self,
-            "Dropping %" G_GUINT64_FORMAT " samples (%" GST_TIME_FORMAT ")",
-            drop_samples, GST_TIME_ARGS (gst_util_uint64_scale (drop_samples,
-                    GST_SECOND, rate)));
-        self->drop_samples = drop_samples;
+            "Inserting %" G_GUINT64_FORMAT " samples of silence (%"
+            GST_TIME_FORMAT ")", silence_samples, GST_TIME_ARGS (silence_time));
+
+        /* Insert silence buffers to fill the gap in 1s chunks */
+        while (silence_samples > 0) {
+          guint n_samples = MIN (silence_samples, rate);
+          GstBuffer *silence;
+          GstMapInfo map;
+
+          silence = gst_buffer_new_and_alloc (n_samples * bpf);
+          GST_BUFFER_FLAG_SET (silence, GST_BUFFER_FLAG_GAP);
+          gst_buffer_map (silence, &map, GST_MAP_WRITE);
+          gst_audio_format_info_fill_silence (info, map.data, map.size);
+          gst_buffer_unmap (silence, &map);
+
+          gst_adapter_push (self->adapter, silence);
+          ret =
+              gst_audio_buffer_split_output (self, FALSE, rate, bpf,
+              samples_per_buffer);
+          if (ret != GST_FLOW_OK)
+            return ret;
+
+          silence_samples -= n_samples;
+        }
         discont = FALSE;
       }
+    } else if (new_offset < self->current_offset + avail_samples) {
+      guint64 drop_samples = self->current_offset + avail_samples - new_offset;
+
+      GST_DEBUG_OBJECT (self,
+          "Dropping %" G_GUINT64_FORMAT " samples (%" GST_TIME_FORMAT ")",
+          drop_samples, GST_TIME_ARGS (gst_util_uint64_scale (drop_samples,
+                  GST_SECOND, rate)));
+      self->drop_samples = drop_samples;
+      discont = FALSE;
     }
   }

gst-plugins-bad-1.20.3.tar.xz/gst/mpegtsdemux/mpegtsbase.c -> gst-plugins-bad-1.20.4.tar.xz/gst/mpegtsdemux/mpegtsbase.c Changed

gst-plugins-bad-1.20.3.tar.xz/gst/mxf/mxfaes-bwf.c -> gst-plugins-bad-1.20.4.tar.xz/gst/mxf/mxfaes-bwf.c Changed

@@ -1270,12 +1270,13 @@
       GST_ERROR ("Invalid descriptor");
       return NULL;
     }
-    if (wa_descriptor && wa_descriptor->block_align != 0)
-      block_align = wa_descriptor->block_align;
-    else
-      block_align =
-          (GST_ROUND_UP_8 (descriptor->quantization_bits) *
-          descriptor->channel_count) / 8;
+
+    /* XXX: block align value can be carried via audio essential descriptor but
+     * there are some files with broken block align value.
+     * Calculates the value always */
+    block_align =
+        (GST_ROUND_UP_8 (descriptor->quantization_bits) *
+        descriptor->channel_count) / 8;
 
     audio_format =
         gst_audio_format_build_integer (block_align !=
@@ -1302,12 +1303,12 @@
       return NULL;
     }
 
-    if (wa_descriptor && wa_descriptor->block_align != 0)
-      block_align = wa_descriptor->block_align;
-    else
-      block_align =
-          (GST_ROUND_UP_8 (descriptor->quantization_bits) *
-          descriptor->channel_count) / 8;
+    /* XXX: block align value can be carried via audio essential descriptor but
+     * there are some files with broken block align value.
+     * Calculates the value always */
+    block_align =
+        (GST_ROUND_UP_8 (descriptor->quantization_bits) *
+        descriptor->channel_count) / 8;
 
     audio_format =
         gst_audio_format_build_integer (block_align !=

gst-plugins-bad-1.20.3.tar.xz/gst/proxy/gstproxysink.c -> gst-plugins-bad-1.20.4.tar.xz/gst/proxy/gstproxysink.c Changed

@@ -57,6 +57,7 @@
 GST_ELEMENT_REGISTER_DEFINE (proxysink, "proxysink", GST_RANK_NONE,
     GST_TYPE_PROXY_SINK);
 
+static void gst_proxy_sink_dispose (GObject * object);
 static gboolean gst_proxy_sink_sink_query (GstPad * pad, GstObject * parent,
     GstQuery * query);
 static GstFlowReturn gst_proxy_sink_sink_chain (GstPad * pad,
@@ -76,10 +77,13 @@
 static void
 gst_proxy_sink_class_init (GstProxySinkClass * klass)
 {
-  GstElementClass *gstelement_class = (GstElementClass *) klass;
+  GObjectClass *object_class = G_OBJECT_CLASS (klass);
+  GstElementClass *gstelement_class = GST_ELEMENT_CLASS (klass);
 
   GST_DEBUG_CATEGORY_INIT (gst_proxy_sink_debug, "proxysink", 0, "proxy sink");
 
+  object_class->dispose = gst_proxy_sink_dispose;
+
   gstelement_class->change_state = gst_proxy_sink_change_state;
   gstelement_class->send_event = gst_proxy_sink_send_event;
   gstelement_class->query = gst_proxy_sink_query;
@@ -109,6 +113,16 @@
   GST_OBJECT_FLAG_SET (self, GST_ELEMENT_FLAG_SINK);
 }
 
+static void
+gst_proxy_sink_dispose (GObject * object)
+{
+  GstProxySink *self = GST_PROXY_SINK (object);
+
+  g_weak_ref_clear (&self->proxysrc);
+
+  G_OBJECT_CLASS (parent_class)->dispose (object);
+}
+
 static GstStateChangeReturn
 gst_proxy_sink_change_state (GstElement * element, GstStateChange transition)
 {
@@ -120,6 +134,8 @@
   switch (transition) {
     case GST_STATE_CHANGE_READY_TO_PAUSED:
       self->pending_sticky_events = FALSE;
+      self->sent_stream_start = FALSE;
+      self->sent_caps = FALSE;
       break;
     default:
       break;
@@ -180,6 +196,7 @@
 
 typedef struct
 {
+  GstProxySink *self;
   GstPad *otherpad;
   GstFlowReturn ret;
 } CopyStickyEventsData;
@@ -189,12 +206,46 @@
     gpointer user_data)
 {
   CopyStickyEventsData *data = user_data;
+  GstProxySink *self = data->self;
 
   data->ret = gst_pad_store_sticky_event (data->otherpad, *event);
+  switch (GST_EVENT_TYPE (*event)) {
+    case GST_EVENT_STREAM_START:
+      if (data->ret != GST_FLOW_OK)
+        self->sent_stream_start = FALSE;
+      else
+        self->sent_stream_start = TRUE;
+      break;
+    case GST_EVENT_CAPS:
+      if (data->ret != GST_FLOW_OK)
+        self->sent_caps = FALSE;
+      else
+        self->sent_caps = TRUE;
+      break;
+    default:
+      break;
+  }
 
   return data->ret == GST_FLOW_OK;
 }
 
+static void
+gst_proxy_sink_send_sticky_events (GstProxySink * self, GstPad * pad,
+    GstPad * otherpad)
+{
+  if (self->pending_sticky_events || !self->sent_stream_start ||
+      !self->sent_caps) {
+    CopyStickyEventsData data;
+
+    data.self = self;
+    data.otherpad = otherpad;
+    data.ret = GST_FLOW_OK;
+
+    gst_pad_sticky_events_foreach (pad, copy_sticky_events, &data);
+    self->pending_sticky_events = data.ret != GST_FLOW_OK;
+  }
+}
+
 static gboolean
 gst_proxy_sink_sink_event (GstPad * pad, GstObject * parent, GstEvent * event)
 {
@@ -202,10 +253,11 @@
   GstProxySrc *src;
   gboolean ret = FALSE;
   gboolean sticky = GST_EVENT_IS_STICKY (event);
+  GstEventType event_type = GST_EVENT_TYPE (event);
 
   GST_LOG_OBJECT (pad, "Got %s event", GST_EVENT_TYPE_NAME (event));
 
-  if (GST_EVENT_TYPE (event) == GST_EVENT_FLUSH_STOP)
+  if (event_type == GST_EVENT_FLUSH_STOP)
     self->pending_sticky_events = FALSE;
 
   src = g_weak_ref_get (&self->proxysrc);
@@ -213,17 +265,24 @@
     GstPad *srcpad;
     srcpad = gst_proxy_src_get_internal_srcpad (src);
 
-    if (sticky && self->pending_sticky_events) {
-      CopyStickyEventsData data = { srcpad, GST_FLOW_OK };
-
-      gst_pad_sticky_events_foreach (pad, copy_sticky_events, &data);
-      self->pending_sticky_events = data.ret != GST_FLOW_OK;
-    }
+    if (sticky)
+      gst_proxy_sink_send_sticky_events (self, pad, srcpad);
 
     ret = gst_pad_push_event (srcpad, event);
     gst_object_unref (srcpad);
     gst_object_unref (src);
 
+    switch (event_type) {
+      case GST_EVENT_STREAM_START:
+        self->sent_stream_start = ret;
+        break;
+      case GST_EVENT_CAPS:
+        self->sent_caps = ret;
+        break;
+      default:
+        break;
+    }
+
     if (!ret && sticky) {
       self->pending_sticky_events = TRUE;
       ret = TRUE;
@@ -250,12 +309,7 @@
     GstPad *srcpad;
     srcpad = gst_proxy_src_get_internal_srcpad (src);
 
-    if (self->pending_sticky_events) {
-      CopyStickyEventsData data = { srcpad, GST_FLOW_OK };
-
-      gst_pad_sticky_events_foreach (pad, copy_sticky_events, &data);
-      self->pending_sticky_events = data.ret != GST_FLOW_OK;
-    }
+    gst_proxy_sink_send_sticky_events (self, pad, srcpad);
 
     ret = gst_pad_push (srcpad, buffer);
     gst_object_unref (srcpad);
@@ -286,12 +340,7 @@
     GstPad *srcpad;
     srcpad = gst_proxy_src_get_internal_srcpad (src);
 
-    if (self->pending_sticky_events) {
-      CopyStickyEventsData data = { srcpad, GST_FLOW_OK };
-
-      gst_pad_sticky_events_foreach (pad, copy_sticky_events, &data);
-      self->pending_sticky_events = data.ret != GST_FLOW_OK;
-    }
+    gst_proxy_sink_send_sticky_events (self, pad, srcpad);
 
     ret = gst_pad_push_list (srcpad, list);
     gst_object_unref (srcpad);

gst-plugins-bad-1.20.3.tar.xz/gst/proxy/gstproxysink.h -> gst-plugins-bad-1.20.4.tar.xz/gst/proxy/gstproxysink.h Changed

gst-plugins-bad-1.20.3.tar.xz/gst/rtmp2/gstrtmp2locationhandler.c -> gst-plugins-bad-1.20.4.tar.xz/gst/rtmp2/gstrtmp2locationhandler.c Changed

@@ -84,6 +84,22 @@
   g_object_interface_install_property (iface, g_param_spec_uint ("timeout",
           "Timeout", "RTMP timeout in seconds", 0, G_MAXUINT, DEFAULT_TIMEOUT,
           G_PARAM_CONSTRUCT | G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+  /**
+   * GstRtmpLocationHandler::tls-validation-flags:
+   *
+   * TLS certificate validation flags used to validate server
+   * certificate.
+   *
+   * GLib guarantees that if certificate verification fails, at least one
+   * error will be set, but it does not guarantee that all possible errors
+   * will be set. Accordingly, you may not safely decide to ignore any
+   * particular type of error.
+   *
+   * For example, it would be incorrect to mask %G_TLS_CERTIFICATE_EXPIRED if
+   * you want to allow expired certificates, because this could potentially be
+   * the only error flag set even if other problems exist with the
+   * certificate.
+   */
   g_object_interface_install_property (iface,
       g_param_spec_flags ("tls-validation-flags", "TLS validation flags",
           "TLS validation flags to use", G_TYPE_TLS_CERTIFICATE_FLAGS,

gst-plugins-bad-1.20.3.tar.xz/gst/rtmp2/rtmp/rtmpclient.c -> gst-plugins-bad-1.20.4.tar.xz/gst/rtmp2/rtmp/rtmpclient.c Changed

gst-plugins-bad-1.20.3.tar.xz/meson.build -> gst-plugins-bad-1.20.4.tar.xz/meson.build Changed

gst-plugins-bad-1.20.3.tar.xz/po/gst-plugins-bad-1.0.pot -> gst-plugins-bad-1.20.4.tar.xz/po/gst-plugins-bad-1.0.pot Changed

gst-plugins-bad-1.20.3.tar.xz/sys/androidmedia/jni/gstamcsurfacetexture-jni.c -> gst-plugins-bad-1.20.4.tar.xz/sys/androidmedia/jni/gstamcsurfacetexture-jni.c Changed

gst-plugins-bad-1.20.3.tar.xz/sys/applemedia/avfvideosrc.m -> gst-plugins-bad-1.20.4.tar.xz/sys/applemedia/avfvideosrc.m Changed

gst-plugins-bad-1.20.3.tar.xz/sys/d3d11/gstd3d11decoder.h -> gst-plugins-bad-1.20.4.tar.xz/sys/d3d11/gstd3d11decoder.h Changed

gst-plugins-bad-1.20.3.tar.xz/sys/d3d11/gstd3d11videosink.cpp -> gst-plugins-bad-1.20.4.tar.xz/sys/d3d11/gstd3d11videosink.cpp Changed

gst-plugins-bad-1.20.3.tar.xz/sys/nvcodec/gstnvdec.c -> gst-plugins-bad-1.20.4.tar.xz/sys/nvcodec/gstnvdec.c Changed

gst-plugins-bad-1.20.3.tar.xz/sys/va/gstvaallocator.c -> gst-plugins-bad-1.20.4.tar.xz/sys/va/gstvaallocator.c Changed

@@ -513,6 +513,7 @@
   VASurfaceID surface;
   guint32 i, fourcc, rt_format, export_flags;
   GDestroyNotify buffer_destroy = NULL;
+  gsize object_offset4;
 
   g_return_val_if_fail (GST_IS_VA_DMABUF_ALLOCATOR (allocator), FALSE);
 
@@ -588,11 +589,19 @@
 
   for (i = 0; i < desc.num_objects; i++) {
     gint fd = desc.objectsi.fd;
-    gsize size = desc.objectsi.size > 0 ?
-        desc.objectsi.size : _get_fd_size (fd);
+    /* don't rely on prime descriptor reported size since gallium drivers report
+     * different values */
+    gsize size = _get_fd_size (fd);
     GstMemory *mem = gst_dmabuf_allocator_alloc (allocator, fd, size);
     guint64 *drm_mod = g_new (guint64, 1);
 
+    if (size != desc.objectsi.size) {
+      GST_WARNING_OBJECT (self, "driver bug: fd size (%" G_GSIZE_FORMAT
+          ") differs from object descriptor size (%" G_GUINT32_FORMAT ")",
+          size, desc.objectsi.size);
+    }
+
+    object_offseti = gst_buffer_get_size (buffer);
     gst_buffer_append_memory (buffer, mem);
     buf->memsi = mem;
 
@@ -615,18 +624,22 @@
         drm_mod, g_free);
 
     if (G_UNLIKELY (info))
-      GST_VIDEO_INFO_SIZE (info) += size;
+      GST_VIDEO_INFO_PLANE_OFFSET (info, i) = GST_VIDEO_INFO_SIZE (info);
 
     GST_LOG_OBJECT (self, "buffer %p: new dmabuf %d / surface %#x %dx%d "
         "size %" G_GSIZE_FORMAT " drm mod %#lx", buffer, fd, surface,
         GST_VIDEO_INFO_WIDTH (&self->info), GST_VIDEO_INFO_HEIGHT (&self->info),
-        GST_VIDEO_INFO_SIZE (&self->info), *drm_mod);
+        size, *drm_mod);
   }
 
   if (G_UNLIKELY (info)) {
+    GST_VIDEO_INFO_SIZE (info) = gst_buffer_get_size (buffer);
+
     for (i = 0; i < desc.num_layers; i++) {
       g_assert (desc.layersi.num_planes == 1);
-      GST_VIDEO_INFO_PLANE_OFFSET (info, i) = desc.layersi.offset0;
+      GST_VIDEO_INFO_PLANE_OFFSET (info, i) =
+          object_offsetdesc.layersi.object_index0 +
+          desc.layersi.offset0;
       GST_VIDEO_INFO_PLANE_STRIDE (info, i) = desc.layersi.pitch0;
     }
   } else {

gst-plugins-bad-1.20.3.tar.xz/sys/va/gstvah265dec.c -> gst-plugins-bad-1.20.4.tar.xz/sys/va/gstvah265dec.c Changed

gst-plugins-bad-1.20.3.tar.xz/sys/wasapi/gstwasapiutil.c -> gst-plugins-bad-1.20.4.tar.xz/sys/wasapi/gstwasapiutil.c Changed

@@ -82,6 +82,72 @@
   {0xa7, 0xbf, 0xad, 0xdc, 0xa7, 0xc2, 0x60, 0xe2}
 };
 
+/* Desktop only defines */
+#ifndef KSAUDIO_SPEAKER_MONO
+#define KSAUDIO_SPEAKER_MONO            (SPEAKER_FRONT_CENTER)
+#endif
+#ifndef KSAUDIO_SPEAKER_1POINT1
+#define KSAUDIO_SPEAKER_1POINT1         (SPEAKER_FRONT_CENTER | SPEAKER_LOW_FREQUENCY)
+#endif
+#ifndef KSAUDIO_SPEAKER_STEREO
+#define KSAUDIO_SPEAKER_STEREO          (SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT)
+#endif
+#ifndef KSAUDIO_SPEAKER_2POINT1
+#define KSAUDIO_SPEAKER_2POINT1         (SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | SPEAKER_LOW_FREQUENCY)
+#endif
+#ifndef KSAUDIO_SPEAKER_3POINT0
+#define KSAUDIO_SPEAKER_3POINT0         (SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | SPEAKER_FRONT_CENTER)
+#endif
+#ifndef KSAUDIO_SPEAKER_3POINT1
+#define KSAUDIO_SPEAKER_3POINT1         (SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | \
+                                         SPEAKER_FRONT_CENTER | SPEAKER_LOW_FREQUENCY)
+#endif
+#ifndef KSAUDIO_SPEAKER_QUAD
+#define KSAUDIO_SPEAKER_QUAD            (SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | \
+                                         SPEAKER_BACK_LEFT  | SPEAKER_BACK_RIGHT)
+#endif
+#define KSAUDIO_SPEAKER_SURROUND        (SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | \
+                                         SPEAKER_FRONT_CENTER | SPEAKER_BACK_CENTER)
+#ifndef KSAUDIO_SPEAKER_5POINT0
+#define KSAUDIO_SPEAKER_5POINT0         (SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | SPEAKER_FRONT_CENTER | \
+                                         SPEAKER_SIDE_LEFT  | SPEAKER_SIDE_RIGHT)
+#endif
+#define KSAUDIO_SPEAKER_5POINT1         (SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | \
+                                         SPEAKER_FRONT_CENTER | SPEAKER_LOW_FREQUENCY | \
+                                         SPEAKER_BACK_LEFT  | SPEAKER_BACK_RIGHT)
+#ifndef KSAUDIO_SPEAKER_7POINT0
+#define KSAUDIO_SPEAKER_7POINT0         (SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | SPEAKER_FRONT_CENTER | \
+                                         SPEAKER_BACK_LEFT | SPEAKER_BACK_RIGHT | \
+                                         SPEAKER_SIDE_LEFT | SPEAKER_SIDE_RIGHT)
+#endif
+#ifndef KSAUDIO_SPEAKER_7POINT1
+#define KSAUDIO_SPEAKER_7POINT1         (SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | \
+                                         SPEAKER_FRONT_CENTER | SPEAKER_LOW_FREQUENCY | \
+                                         SPEAKER_BACK_LEFT | SPEAKER_BACK_RIGHT | \
+                                         SPEAKER_FRONT_LEFT_OF_CENTER | SPEAKER_FRONT_RIGHT_OF_CENTER)
+#endif
+
+static DWORD default_ch_masks = {
+  0,
+  KSAUDIO_SPEAKER_MONO,
+  /* 2ch */
+  KSAUDIO_SPEAKER_STEREO,
+  /* 2.1ch */
+  /* KSAUDIO_SPEAKER_3POINT0 ? */
+  KSAUDIO_SPEAKER_2POINT1,
+  /* 4ch */
+  /* KSAUDIO_SPEAKER_3POINT1 or KSAUDIO_SPEAKER_SURROUND ? */
+  KSAUDIO_SPEAKER_QUAD,
+  /* 5ch */
+  KSAUDIO_SPEAKER_5POINT0,
+  /* 5.1ch */
+  KSAUDIO_SPEAKER_5POINT1,
+  /* 7ch */
+  KSAUDIO_SPEAKER_7POINT0,
+  /* 7.1ch */
+  KSAUDIO_SPEAKER_7POINT1,
+};
+
 /* *INDENT-OFF* */
 static struct
 {
@@ -705,6 +771,17 @@
   DWORD dwChannelMask = format->dwChannelMask;
   GstAudioChannelPosition *pos = NULL;
 
+  if (nChannels > 2 && !dwChannelMask) {
+    GST_WARNING ("Unknown channel mask value for %d channel stream", nChannels);
+
+    if (nChannels >= G_N_ELEMENTS (default_ch_masks)) {
+      GST_ERROR ("Too many channels %d", nChannels);
+      return 0;
+    }
+
+    dwChannelMask = default_ch_masksnChannels;
+  }
+
   pos = g_new (GstAudioChannelPosition, nChannels);
   gst_wasapi_util_channel_position_all_none (nChannels, pos);

gst-plugins-bad-1.20.3.tar.xz/sys/wasapi2/gstwasapi2ringbuffer.cpp -> gst-plugins-bad-1.20.4.tar.xz/sys/wasapi2/gstwasapi2ringbuffer.cpp Changed

gst-plugins-bad-1.20.4.tar.xz/tests/check/elements/proxysink.c Added

@@ -0,0 +1,110 @@
+/* GStreamer
+ * Copyright (C) 2022 Seungha Yang <seungha@centricular.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#ifdef HAVE_CONFIG_H
+#include "config.h"
+#endif
+
+#include <gst/gst.h>
+#include <gst/check/gstcheck.h>
+#include <gst/check/gstharness.h>
+
+GST_START_TEST (test_flush_before_buffer)
+{
+  GstElement *sink, *src;
+  GstHarness *h_in;
+  GstHarness *h_out;
+  GstEvent *event;
+  GstSegment segment;
+  GstCaps *caps;
+  GstBuffer *buf;
+
+  sink = gst_element_factory_make ("proxysink", NULL);
+  src = gst_element_factory_make ("proxysrc", NULL);
+
+  g_object_set (src, "proxysink", sink, NULL);
+
+  h_in = gst_harness_new_with_element (sink, "sink", NULL);
+  h_out = gst_harness_new_with_element (src, NULL, "src");
+  gst_object_unref (sink);
+  gst_object_unref (src);
+
+  /* Activate only input side first, then push sticky events
+   * without buffer */
+  gst_harness_play (h_in);
+
+  event = gst_event_new_stream_start ("proxy-test-stream-start");
+  fail_unless (gst_harness_push_event (h_in, event));
+
+  caps = gst_caps_from_string ("foo/bar");
+  event = gst_event_new_caps (caps);
+  gst_caps_unref (caps);
+  fail_unless (gst_harness_push_event (h_in, event));
+
+  gst_segment_init (&segment, GST_FORMAT_TIME);
+  event = gst_event_new_segment (&segment);
+  fail_unless (gst_harness_push_event (h_in, event));
+
+  /* Now activate output side, sticky event and buffers should be
+   * serialized */
+  gst_harness_play (h_out);
+
+  event = gst_event_new_flush_start ();
+  fail_unless (gst_harness_push_event (h_in, event));
+
+  event = gst_event_new_flush_stop (TRUE);
+  fail_unless (gst_harness_push_event (h_in, event));
+
+  event = gst_event_new_segment (&segment);
+  fail_unless (gst_harness_push_event (h_in, event));
+
+  buf = gst_buffer_new_and_alloc (4);
+  GST_BUFFER_PTS (buf) = 0;
+  GST_BUFFER_DTS (buf) = 0;
+
+  /* There must be no critical warning regarding
+   * sticky-event and buffer flow order*/
+  fail_unless_equals_int (gst_harness_push (h_in, buf), GST_FLOW_OK);
+
+  event = gst_event_new_eos ();
+  fail_unless (gst_harness_push_event (h_in, event));
+
+  /* make sure everything has been forwarded */
+  fail_unless (gst_harness_pull_until_eos (h_out, &buf));
+  gst_buffer_unref (buf);
+
+  gst_harness_teardown (h_in);
+  gst_harness_teardown (h_out);
+}
+
+GST_END_TEST;
+
+static Suite *
+proxysink_suite (void)
+{
+  Suite *s = suite_create ("proxysink");
+  TCase *tc_basic = tcase_create ("general");
+
+  suite_add_tcase (s, tc_basic);
+  tcase_add_test (tc_basic, test_flush_before_buffer);
+
+  return s;
+}
+
+GST_CHECK_MAIN (proxysink);

gst-plugins-bad-1.20.3.tar.xz/tests/check/meson.build -> gst-plugins-bad-1.20.4.tar.xz/tests/check/meson.build Changed

@@ -23,49 +23,48 @@
 base_tests = 
   'elements/aesenc.c', not aes_dep.found(), aes_dep,
   'elements/aesdec.c', not aes_dep.found(), aes_dep,
-  'elements/aiffparse.c',
-  'elements/asfmux.c',
-  'elements/autoconvert.c',
-  'elements/autovideoconvert.c',
-  'elements/avwait.c',
-  'elements/camerabin.c',
+  'elements/aiffparse.c', get_option('aiff').disabled(),
+  'elements/asfmux.c', get_option('asfmux').disabled(),
+  'elements/autoconvert.c', get_option('autoconvert').disabled(),
+  'elements/autovideoconvert.c', get_option('autoconvert').disabled(),
+  'elements/avwait.c', get_option('timecode').disabled(),
+  'elements/camerabin.c', get_option('camerabin2').disabled(),
   'elements/ccconverter.c', not closedcaption_dep.found(), gstvideo_dep,
   'elements/cccombiner.c', not closedcaption_dep.found(), ,
   'elements/ccextractor.c', not closedcaption_dep.found(), ,
   'elements/cudaconvert.c', false, gmodule_dep, gstgl_dep,
   'elements/cudafilter.c', false, gmodule_dep, gstgl_dep,
   'elements/d3d11colorconvert.c', host_machine.system() != 'windows', ,
-  'elements/gdpdepay.c',
-  'elements/gdppay.c',
+  'elements/gdpdepay.c', get_option('gdp').disabled(),
+  'elements/gdppay.c', get_option('gdp').disabled(),
   'elements/h263parse.c', false, libparser_dep, gstcodecparsers_dep,
   'elements/h264parse.c', false, libparser_dep, gstcodecparsers_dep,
   'elements/h265parse.c', false, libparser_dep, gstcodecparsers_dep,
   'elements/hlsdemux_m3u8.c', not hls_dep.found(), hls_dep,
-  'elements/id3mux.c',
-  'elements/interlace.c',
+  'elements/id3mux.c', get_option('id3tag').disabled(),
+  'elements/interlace.c', get_option('interlace').disabled(),
   'elements/jpeg2000parse.c', false, libparser_dep, gstcodecparsers_dep,
   'elements/line21.c', not closedcaption_dep.found(), ,
   'elements/mfvideosrc.c', host_machine.system() != 'windows', ,
-  'elements/mpegtsdemux.c', false, gstmpegts_dep,
-  'elements/mpegtsmux.c', false, gstmpegts_dep,
+  'elements/mpegtsdemux.c', get_option('mpegtsdemux').disabled(), gstmpegts_dep,
+  'elements/mpegtsmux.c', get_option('mpegtsmux').disabled(), gstmpegts_dep,
   'elements/mpeg4videoparse.c', false, libparser_dep, gstcodecparsers_dep,
   'elements/mpegvideoparse.c', false, libparser_dep, gstcodecparsers_dep,
   'elements/msdkh264enc.c', not have_msdk, msdk_dep,
-  'elements/mxfdemux.c',
-  'elements/mxfmux.c',
-  'elements/nvenc.c', false, gmodule_dep, gstgl_dep,
-  'elements/nvdec.c', not gstgl_dep.found(), gmodule_dep, gstgl_dep,
+  'elements/mxfdemux.c', get_option('mxf').disabled(),
+  'elements/mxfmux.c', get_option('mxf').disabled(),
   'elements/svthevcenc.c', not svthevcenc_dep.found(), svthevcenc_dep,
    'elements/openjpeg.c', not openjpeg_dep.found(), openjpeg_dep,
   'elements/pcapparse.c', false, libparser_dep,
-  'elements/pnm.c',
+  'elements/pnm.c', get_option('pnm').disabled(),
+  'elements/proxysink.c', get_option('proxy').disabled(),
   'elements/ristrtpext.c',
-  'elements/rtponvifparse.c',
-  'elements/rtponviftimestamp.c',
-  'elements/rtpsrc.c',
-  'elements/rtpsink.c',
-  'elements/switchbin.c',
-  'elements/videoframe-audiolevel.c',
+  'elements/rtponvifparse.c', get_option('onvif').disabled(),
+  'elements/rtponviftimestamp.c', get_option('onvif').disabled(),
+  'elements/rtpsrc.c', get_option('rtp').disabled(),
+  'elements/rtpsink.c', get_option('rtp').disabled(),
+  'elements/switchbin.c', get_option('switchbin').disabled(),
+  'elements/videoframe-audiolevel.c', get_option('videoframe_audiolevel').disabled(),
   'elements/viewfinderbin.c',
   'elements/vp9parse.c', false, gstcodecparsers_dep,
   'elements/av1parse.c', false, gstcodecparsers_dep,
@@ -109,7 +108,7 @@
     'elements/avtpcvfdepay.c', not avtp_dep.found(), avtp_dep,
     'elements/avtpsink.c', not avtp_dep.found(), avtp_dep,
     'elements/avtpsrc.c', not avtp_dep.found(), avtp_dep,
-    'elements/clockselect.c',
+    'elements/clockselect.c', get_option('debugutils').disabled(),
     'elements/curlhttpsink.c', not curl_dep.found(), curl_dep,
     'elements/curlhttpsrc.c', not curl_dep.found(), curl_dep, gio_dep,
     'elements/curlfilesink.c',

Changes of Revision 8