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

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

+Tue May 17 16:38:34 UTC 2022 - Bjørn Lie <zaitor@opensuse.org>

+

+- Update to version 1.20.2

+

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

 Sat Mar 26 01:04:36 UTC 2022 - Bjørn Lie <zaitor@opensuse.org>

 - Update to version 1.20.1

​

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

 %define _version 1.20.0

 Name:           gstreamer-plugins-bad-codecs

-Version:        1.20.1

+Version:        1.20.2

 Release:        0

 Summary:        Codecs/plugins for gstreamer-plugins-bad

 License:        LGPL-2.1-or-later

​

@@ -1,7 +1,425 @@

+=== release 1.20.2 ===

+

+2022-05-02 23:29:25 +0100  Tim-Philipp Müller <tim@centricular.com>

+

+   * NEWS:

+   * RELEASE:

+   * gst-plugins-bad.doap:

+   * meson.build:

+     Release 1.20.2

+

+2022-05-02 23:29:19 +0100  Tim-Philipp Müller <tim@centricular.com>

+

+   * ChangeLog:

+     Update ChangeLogs for 1.20.2

+

+2022-04-23 04:00:21 +0900  Seungha Yang <seungha@centricular.com>

+

+   * sys/d3d11/gstd3d11window_win32.cpp:

+     d3d11videosink: Fix for unhandled mouse double click events

+     Only window created with CS_DBLCLKS style can receive those mouse

+     double click events, so we need to use the style for internal/external

+     windows can get double click events.

+     Also, passthrough mouse events to parent window in the same message pumping

+     threads instead of manually forwarding each mouse event.

+     Fixes: https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1172

+     Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2286>

+

+2022-04-08 04:24:36 +0900  Seungha Yang <seungha@centricular.com>

+

+   * sys/d3d11/gstd3d11window_win32.cpp:

+     d3d11window: Use ANSI version WIN32 API explicitly

+     We were using ANSI version APIs implicitly because UNICODE is not

+     defined by ourselves. But potentially it can be broken if user

+     defines UNICODE.

+     Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2286>

+

+2022-04-29 19:08:46 +0300  Sebastian Dröge <sebastian@centricular.com>

+

+   * gst/interlace/gstinterlace.c:

+     interlace: Also handle a missing "interlace-mode" field as progressive

+     Otherwise caps negotiation will fail in situations that are supposed

+     to work, like:

+     "video/x-raw,framerate=(fraction)60/1" ! interlace field-pattern=0 ! "video/x-raw,framerate=(fraction)30/1"

+     Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2337>

+

+2022-04-29 19:08:32 +0300  Sebastian Dröge <sebastian@centricular.com>

+

+   * gst/interlace/gstinterlace.c:

+     interlace: Add some more debug output to the getcaps function

+     Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2337>

+

+2022-04-27 15:24:50 +0200  Edward Hervey <edward@centricular.com>

+

+   * gst/mpegtsdemux/mpegtsbase.c:

+     mpegts: Handle "empty" PMT gracefully

+     Some streams have 2 PMT sections in a single TS packet. The first one is "valid"

+     but doesn't contain/define any streams. That causes an unrecoverable issue when

+     we try to activate the 2nd (valid) PMT.

+     Instead of doing that, pre-emptively refuse to process PMT without any streams

+     present within. We still do post that section on the bus to inform applications.

+     Fixes #1181

+     Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2327>

+

+2022-04-28 00:31:49 +0900  Seungha Yang <seungha@centricular.com>

+

+   * sys/nvcodec/gstnvvp9dec.c:

+     nvvp9sldec: Increase DPB size to cover render delay

+     This should've included in the previous MR

+     https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/987

+     already, but missed.

+     Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2324>

+

+2022-04-24 00:02:18 +0800  He Junyan <junyan.he@intel.com>

+

+   * sys/va/gstvabasedec.c:

+     va: dec: Use gst_buffer_pool_config_set_va_alignment() to set alignment.

+     Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2301>

+

+2022-04-23 23:59:08 +0800  He Junyan <junyan.he@intel.com>

+

+   * sys/va/gstvapool.h:

+     va: pool: Replace all tabs with spaces in header file.

+     Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2301>

+

+2022-04-23 23:52:34 +0800  He Junyan <junyan.he@intel.com>

+

+   * sys/va/gstvapool.c:

+     va: pool: Delete the GST_BUFFER_POOL_OPTION_VIDEO_ALIGNMENT.

+     The va pool is used for GPU side surface/image, its alignment should

+     not be changed arbitrarily by others. So we decide not to expose the

+     GST_BUFFER_POOL_OPTION_VIDEO_ALIGNMENT flag anymore.

+     Instead, user can call gst_buffer_pool_config_set_va_alignment() to

+     set its surface/image alignment.

+     Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2301>

+

+2022-04-23 23:40:14 +0800  He Junyan <junyan.he@intel.com>

+

+   * sys/va/gstvapool.c:

+   * sys/va/gstvapool.h:

+     va: pool: Add set_va_alignment() API.

+     We want to use gst_buffer_pool_config_set_va_alignment() to replace

+     gst_buffer_pool_config_get_video_alignment(). The later one is specific

+     for GST_BUFFER_POOL_OPTION_VIDEO_ALIGNMENT option.

+     Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2301>

+

+2022-04-26 01:16:06 +0800  He Junyan <junyan.he@intel.com>

+

+   * gst/videoparsers/gstav1parse.c:

+     av1parse: Fix a latent memory leak in colorimetry setting.

+     Also delete the useless "have_cinfo" judgement.

+     Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2297>

+

+2022-04-24 22:54:58 +0800  He Junyan <junyan.he@intel.com>

+

+   * gst/videoparsers/gstav1parse.c:

+     av1parse: Correct the meaning of color_range flag.

+     According to spec:

+     color range equal to 0 shall be referred to as the studio swing

+     representation and color range equal to 1 shall be referred to as

+     the full swing representation.

+     The current status is just the opposite.

+     Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2297>

+

+2022-04-22 17:44:06 +0200  Stéphane Cerveau <scerveau@collabora.com>

+

+   * ext/rsvg/gstrsvgdec.h:

+     rvsg: fix cairo include

+     Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2280>

+

+2022-04-21 00:38:37 +0900  Sangchul Lee <sc11.lee@samsung.com>

+

+   * ext/webrtc/gstwebrtcbin.c:

+     webrtcbin: Avoid access of freed memory

+     Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2258>

+

+2022-04-20 08:09:58 +0900  Wonchul Lee <wonchul.dev@gmail.com>

+

+   * ext/dash/gstdashsink.c:

+     dashsink: Unlock when failed to get content

+     Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2246>

+

+2022-04-18 17:50:22 +0900  Camilo Celis Guzman <camilo@pexip.com>

+

+   * gst/videoparsers/gstav1parse.c:

+     gstav1parse: fixup various possible logical errors

+     Found via an analyzed build for Clang. Specifically we had:

+     gstav1parse.c1850,11 in gst_av1_parse_detect_stream_format: Logic error: The left operand of '==' is a garbage value

+     gstav1parse.c1606,11 in gst_av1_parse_handle_to_small_and_equal_align: Logic error: The left operand of '==' is a garbage value

+     Also a couple of false-positives:

+     gstav1parse.c1398,24 in gst_av1_parse_handle_one_obu: Logic error: Branch condition evaluates to a garbage value

+     gstav1parse.c1440,37 in gst_av1_parse_handle_one_obu: Logic error: The left operand of '-' is a garbage value

+     Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2240>

+

+2022-04-12 01:01:23 +0900  Seungha Yang <seungha@centricular.com>

+

+   * sys/d3d11/gstd3d11decoder.cpp:

+   * sys/d3d11/meson.build:

+     win32: Enable high-resolution timer for MinGW build

+     timeapi.h is missing in our MinGW toolchain. Include mmsystem.h

+     header instead, which defines struct and APIs in case of our MinGW

+     toolchain. Note that in case of native Windows10 SDK (MSVC build),

+     mmsystem.h will include timeapi.h

+     Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2167>

+

+2022-04-14 08:23:51 +0200  Edward Hervey <edward@centricular.com>

+

+   * gst/mpegtsdemux/gstmpegdefs.h:

+   * gst/mpegtsdemux/tsdemux.c:

+     tsdemux: Fix AC-4 detection

+     This regression was introduced by

+     https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1102

+     and has been present since 1.18

+     * Stream Type 0x06 is defined in the base mpeg-ts specification as Private PES

+     Packets. Determining the content should be solely based on descriptors found

+     within the PMT.

+     * This was abused in that commit by defining a "bluray-only" stream type for AC4

+     : `ST_BD_AUDIO_AC4`

+     * This should be entirely handled in the regular private pes handling further

+     down in the code

+     Fixes #1154

+     Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2176>

+

+2022-04-11 10:32:40 +0200  Edward Hervey <edward@centricular.com>

+

+   * gst/mxf/mxfdemux.c:

+     mxfdemux: Fix issue with re-syncing

+     In case of re-syncing (i.e. moving to another partition to avoid too much of an

+     interleave), there was previously no checks to figure out whether a given

+     partition was already fully handled (i.e. when coming across it again after a

+     previous resync).

+     In order to handle this at least for single-track partitions, check whether we

+     have reached the essence track duration, and if so skip the partition.

+     Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2162>

+

+2022-04-11 10:31:15 +0200  Edward Hervey <edward@centricular.com>

+

+   * gst/mxf/mxfdemux.c:

+     mxfdemux: Don't double-increase track position

+     The essence track position should only be overriden if we sucesfully switched to

​

@@ -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 1.20.1 and was released

-on 14 March 2022.

+The latest bug-fix release in the 1.20 series is 1.20.2 and was released

+on 2 May 2022.

 See https://gstreamer.freedesktop.org/releases/1.20/ for the latest

 version of this document.

-Last updated: Monday 14 March 2022, 00:30 UTC (log)

+Last updated: Monday 2 May 2022, 0:30 UTC (log)

 Introduction

@@ -2076,6 +2076,221 @@

 -   List of Merge Requests applied in 1.20.1

 -   List of Issues fixed in 1.20.1

+1.20.2

+

+The second 1.20 bug-fix release (1.20.2) was released on 2 May 2022.

+

+This release only contains bugfixes and it should be safe to update from

+1.20.x.

+

+Highlighted bugfixes in 1.20.2

+

+-   avviddec: Remove vc1/wmv3 override and fix crashes on WMV files with

+    FFMPEG 5.0+

+-   macOS: fix plugin discovery for GStreamer installed via brew and fix

+    loading of Rust plugins

+-   rtpbasepayload: various header extension handling fixes

+-   rtpopusdepay: fix regression in stereo input handling if

+    sprop-stereo is not advertised

+-   rtspclientsink: fix possible shutdown deadlock

+-   mpegts: gracefully handle “empty” program maps and fix AC-4

+    detection

+-   mxfdemux: Handle empty VANC packets and fix EOS handling

+-   playbin3: various playbin3, uridecodebin3, and playsink fixes

+-   ptpclock: fix initial sync-up with certain devices

+-   gltransformation: let graphene alloc its structures memory aligned

+-   webrtcbin fixes and webrtc sendrecv example improvements

+-   video4linux2: various fixes including some fixes for Raspberry Pi

+    users

+-   videorate segment handling fixes and other fixes

+-   nvh264dec, nvh265dec: Fix broken key-unit trick modes and reverse

+    playback

+-   wpe: Reintroduce persistent WebContext

+-   cerbero: Make it easier to consume 1.20.1 macOS GStreamer .pkgs

+-   build fixes and gobject annotation fixes

+-   bug fixes, security fixes, memory leak fixes, and other stability

+    and reliability improvements

+

+gstreamer

+

+-   devicemonitor: clean up signal handlers and hidden providers list

+-   Leaks tracer: fix pthread_atfork return value check leading to bogus

+    warning in log

+-   Rust plugins: Not picked up by the plugin loader on macOS

+-   Failed to use plugins of latest GStreamer version 1.20.x installed

+    by brew on macOS

+-   ptpclock: Allow at least 100ms delay between Sync/Follow_Up and

+    Delay_Req/Delay_Resp messages. Fixes problems acquiring initial sync

+    with certain devices

+-   meson: Add -Wl,-rpath,${libdir} on macOS

+-   registry: skip Rust dep builddirs when searching for plugins

+    recursively

+

+gst-plugins-base

+

+-   appsrc: Clarify buffer ref semantics in signals documentation

+-   appsrc: fix annotations for bindings

+-   typefind: Skip extension parsing for data:// URIs, fixing regression

+    with mp4 files serialised to data uris

+-   playbin3: various fixes

+-   playbin3: fix missing lock when unknown stream type in pad-removed

+    cb

+-   decodebin3: fix collection leaks

+-   decodebin3: Don’t duplicate stream selections

+-   discoverer: chain up to parent finalize methods in all our types to

+    fix memory leaks

+-   glmixerbin: slightly better pad/element creation

+-   gltransformation: let graphene alloc its structures memory aligned

+-   ogg: fix possible buffer overrun

+-   rtpbasepayload: Don’t write header extensions if there’s no

+    corresponding…

+-   rtpbasepayload: always store input buffer meta before negotiation

+-   rtpbasepayload: fix transfer annotation for push and push_list

+-   subparse: don’t try to index string with -1

+-   riff-media: fix memory leak after usage for g_strjoin()

+-   playbin/playbin3: Allow setting a NULL URI

+-   playsink: Complete reconfiguration on pad release.

+-   parsebin: Expose streams of unknown type

+-   pbutils: Fix wmv screen description detection

+-   subparse: don’t deref a potentially NULL variable

+-   rawvideoparse: set format from caps in

+    gst_raw_video_parse_set_config_from_caps

+-   videodecoder: release stream lock after handling gap events

+-   videorate: fix assertion when pushing last and only buffer without

+    duration

+-   videorate: Revert “don’t reset on segment update” to fix segment

+    handling regressions

+-   gst-play-1.0, gst-launch-1.0: Enable win32 high-resolution timer

+    also for MinGW build

+

+gst-plugins-good

+

+-   deinterlace: silence unused-but-set werror from imported code

+-   qtdemux: fix leak of channel_mapping

+-   rtpopusdepay: missing sprop-stereo should not assume mono

+-   rtpjitterbuffer: Fix invalid memory access in

+    rtp_jitter_buffer_pop()

+-   rtpptdemux: fix leak of caps when ignoring a pt

+-   rtpredenc: quieten warning about ignoring header extensions

+-   soup: Fix pre-processor macros in souploader for libsoup-3.0

+-   twcc: Note that twcc-stats packet loss counts reordering as loss +

+    add some logging

+-   video4linux2: Manual backports for RPi users

+-   wavparse: handle URI query in any parse state, fixing audio track

+    selection issue in GES

+-   wavparse: Unset DISCONT buffer flag for divided into multiple

+    buffers in push mode

+

+gst-plugins-bad

+

+-   av1parse: Fix several issues about the colorimetry.

+-   av1parse: fix up various possible logic errors

+-   dashsink: fix missing mutex unlock in error code path when failing

+    to get content

+-   d3d11videosink: Fix for unhandled mouse double click events

+-   interlace: Also handle a missing “interlace-mode” field as

+    progressive

+-   msdk: fix build with MSVC

+-   mxfdemux: Fix issues at EOS

+-   mxfdemux: Handle empty VANC packets

+-   nvh264dec, nvh265dec: Fix broken key-unit trick and reverse playback

+-   nvvp9sldec: Increase DPB size to cover render delay

+-   rvsg: fix cairo include

+-   tsdemux: Fix AC-4 detection in MPEG-TS

+-   tsdemux: Handle “empty” PMT gracefully

+-   va: pool: don’t advertise the GST_BUFFER_POOL_OPTION_VIDEO_ALIGNMENT

+    option any more

+-   v4l2codecs: Fix memory leak

+-   v4l2videodec: set frame duration according to framerate

+-   webrtcbin: Update documentation of ‘get-stats’ action signal

+-   webrtcbin: Check data channel transport for notifying

+    ‘ice-gathering-state’

+-   webrtcbin: Avoid access of freed memory

+-   wpe: Reintroduce persistent WebContext

+-   Build: use CMake to find some openssl and exr deps

+-   Fix multiple “unused-but-set variable” compiler warnings

+

+gst-plugins-ugly

+

+-   x264enc: Don’t try to fixate ANY allowed caps

+

+gst-libav

+

+-   video decoders: fix frame leak on negotiation error

+-   Fix build on systems without C++ compiler

+-   avviddec: Remove vc1/wmv3 override (fixing crash with FFmpeg 5

+-   Segfaults on ASF/WMV files with FFMPEG 5.0+

+

+gst-rtsp-server

+

+-   rtspclientsink: fix possible shutdown deadlock in collect_streams()

+-   Minor spelling fixes

+

+gstreamer-vaapi

+

+-   No changes

+

+gstreamer-sharp

+

+-   No changes

+

+gst-omx

+

+-   No changes

+

+gst-python

+

+-   Fix build on systems without C++ compiler

+

+gst-editing-services

+

+-   License clarification: GES is released under the LGPL2+ license

+

+gst-examples:

+

+-   Fix build on macOS with gtk+-quartz-3.0

+-   player android: add missing dummy.cpp

+-   player android: update for android changes

+-   webrtc_sendrecv.py: Link pads instead of elements

+-   webrtc_sendrecv.py: Implement all negotiation modes + bugfixes

+

+Development build environment + gst-full build

​

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

-This is GStreamer gst-plugins-bad 1.20.1.

+This is GStreamer gst-plugins-bad 1.20.2.

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

 of your favourite cross-platform multimedia framework!

​

@@ -854,8 +854,10 @@

   g_mutex_lock (&sink->mpd_lock);

   gst_dash_sink_generate_mpd_content (sink, current_stream);

-  if (!gst_mpd_client_get_xml_content (sink->mpd_client, &mpd_content, &size))

+  if (!gst_mpd_client_get_xml_content (sink->mpd_client, &mpd_content, &size)) {

+    g_mutex_unlock (&sink->mpd_lock);

     return;

+  }

   g_mutex_unlock (&sink->mpd_lock);

   if (sink->mpd_root_path)

​

@@ -1,4 +1,5 @@

-openexr_dep = dependency('OpenEXR', required: get_option('openexr'))

+openexr_dep = dependency('OpenEXR', modules: 'OpenEXR::IlmImf', required: get_option('openexr'))

+

 if openexr_dep.found()

   openexr_override_options = 

   # Older versions of openexr fail to build with -Werror when using GCC >= 9.2

​

@@ -114,7 +114,7 @@

   guint32 start_code;

   gboolean STD_buffer_bound_scale G_GNUC_UNUSED;

-  guint16 STD_buffer_size_bound;

+  guint16 STD_buffer_size_bound G_GNUC_UNUSED;

   const guint8 *data;

   gint avail, datalen;

   gboolean have_size = FALSE;

​

@@ -24,7 +24,7 @@

 #include <gst/base/gstadapter.h>

 #include <gst/video/video.h>

-#include <cairo/cairo.h>

+#include <cairo.h>

 #include <librsvg/rsvg.h>

​

@@ -1102,7 +1102,11 @@

 {

 #define STATE(val) GST_WEBRTC_ICE_GATHERING_STATE_ ## val

   GstWebRTCICEGatheringState any_state = 0;

-  gboolean all_completed = webrtc->priv->transceivers->len > 0;

+  GstWebRTCICEGatheringState ice_state;

+  GstWebRTCDTLSTransport *dtls_transport;

+  GstWebRTCICETransport *transport;

+  gboolean all_completed = webrtc->priv->transceivers->len > 0 ||

+      webrtc->priv->data_channel_transport;

   int i;

   for (i = 0; i < webrtc->priv->transceivers->len; i++) {

@@ -1110,9 +1114,6 @@

         g_ptr_array_index (webrtc->priv->transceivers, i);

     WebRTCTransceiver *trans = WEBRTC_TRANSCEIVER (rtp_trans);

     TransportStream *stream = trans->stream;

-    GstWebRTCDTLSTransport *dtls_transport;

-    GstWebRTCICETransport *transport;

-    GstWebRTCICEGatheringState ice_state;

     if (rtp_trans->stopped || stream == NULL) {

       GST_TRACE_OBJECT (webrtc, "transceiver %p stopped or unassociated",

@@ -1143,6 +1144,20 @@

       all_completed = FALSE;

   }

+  /* check data channel transport gathering state */

+  if (all_completed && webrtc->priv->data_channel_transport) {

+    if ((dtls_transport = webrtc->priv->data_channel_transport->transport)) {

+      transport = dtls_transport->transport;

+      g_object_get (transport, "gathering-state", &ice_state, NULL);

+      GST_TRACE_OBJECT (webrtc,

+          "data channel transport %p gathering state: 0x%x", dtls_transport,

+          ice_state);

+      any_state |= (1 << ice_state);

+      if (ice_state != STATE (COMPLETE))

+        all_completed = FALSE;

+    }

+  }

+

   GST_TRACE_OBJECT (webrtc, "ICE gathering state: 0x%x", any_state);

   /* Any of the RTCIceTransport s are in the gathering state. */

@@ -1813,7 +1828,6 @@

       caps = _query_pad_caps (webrtc, rtp_trans, pad, filter, error);

     }

-    gst_object_unref (pad);

     if (*error)

       goto out;

@@ -1867,6 +1881,8 @@

 out:

+  if (pad)

+    gst_object_unref (pad);

   if (codec_preferences)

     gst_caps_unref (codec_preferences);

@@ -4010,7 +4026,7 @@

             goto rejected;

           }

-          GST_TRACE_OBJECT (webrtc, "trying to compare %" GST_PTR_FORMAT

+          GST_LOG_OBJECT (webrtc, "trying to compare %" GST_PTR_FORMAT

               " and %" GST_PTR_FORMAT, offer_caps, trans_caps);

           /* FIXME: technically this is a little overreaching as some fields we

@@ -4041,6 +4057,8 @@

         /* if no transceiver, then we only receive that stream and respond with

          * the intersection with the transceivers codec preferences caps */

         answer_dir = GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_RECVONLY;

+        GST_WARNING_OBJECT (webrtc, "did not find compatible transceiver for "

+            "offer caps %" GST_PTR_FORMAT ", will only receive", offer_caps);

       }

       if (!rtp_trans) {

@@ -7679,25 +7697,29 @@

    *  "ssrc"                G_TYPE_STRING               the rtp sequence src in use

    *  "transport-id"        G_TYPE_STRING               identifier for the associated RTCTransportStats for this stream

    *  "codec-id"            G_TYPE_STRING               identifier for the associated RTCCodecStats for this stream

-   *  "fir-count"           G_TYPE_UINT                 FIR requests received by the sender (only for local statistics)

-   *  "pli-count"           G_TYPE_UINT                 PLI requests received by the sender (only for local statistics)

-   *  "nack-count"          G_TYPE_UINT                 NACK requests received by the sender (only for local statistics)

    *

    * RTCReceivedStreamStats supported fields (https://w3c.github.io/webrtc-stats/#receivedrtpstats-dict*)

    *

-   *  "packets-received"     G_TYPE_UINT64              number of packets received (only for local inbound)

-   *  "bytes-received"       G_TYPE_UINT64              number of bytes received (only for local inbound)

-   *  "packets-lost"         G_TYPE_UINT                number of packets lost

-   *  "jitter"               G_TYPE_DOUBLE              packet jitter measured in secondss

+   *  "packets-received"    G_TYPE_UINT64               number of packets received (only for local inbound)

+   *  "packets-lost"        G_TYPE_UINT64               number of packets lost

+   *  "packets-discarded"   G_TYPE_UINT64               number of packets discarded

+   *  "packets-repaired"    G_TYPE_UINT64               number of packets repaired

+   *  "jitter"              G_TYPE_DOUBLE               packet jitter measured in seconds

    *

    * RTCInboundRTPStreamStats supported fields (https://w3c.github.io/webrtc-stats/#inboundrtpstats-dict*)

    *

    *  "remote-id"           G_TYPE_STRING               identifier for the associated RTCRemoteOutboundRTPStreamStats

+   *  "bytes-received"      G_TYPE_UINT64               number of bytes received (only for local inbound)

+   *  "packets-duplicated"  G_TYPE_UINT64               number of packets duplicated

+   *  "fir-count"           G_TYPE_UINT                 FIR packets sent by the receiver

+   *  "pli-count"           G_TYPE_UINT                 PLI packets sent by the receiver

+   *  "nack-count"          G_TYPE_UINT                 NACK packets sent by the receiver

    *

    * RTCRemoteInboundRTPStreamStats supported fields (https://w3c.github.io/webrtc-stats/#remoteinboundrtpstats-dict*)

    *

    *  "local-id"            G_TYPE_STRING               identifier for the associated RTCOutboundRTPSTreamStats

    *  "round-trip-time"     G_TYPE_DOUBLE               round trip time of packets measured in seconds

+   *  "fraction-lost"       G_TYPE_DOUBLE               fraction packet loss

    *

    * RTCSentRTPStreamStats supported fields (https://w3c.github.io/webrtc-stats/#sentrtpstats-dict*)

    *

@@ -7707,10 +7729,14 @@

    * RTCOutboundRTPStreamStats supported fields (https://w3c.github.io/webrtc-stats/#outboundrtpstats-dict*)

    *

    *  "remote-id"           G_TYPE_STRING               identifier for the associated RTCRemoteInboundRTPSTreamStats

+   *  "fir-count"           G_TYPE_UINT                 FIR packets received by the sender

+   *  "pli-count"           G_TYPE_UINT                 PLI packets received by the sender

+   *  "nack-count"          G_TYPE_UINT                 NACK packets received by the sender

    *

    * RTCRemoteOutboundRTPStreamStats supported fields (https://w3c.github.io/webrtc-stats/#remoteoutboundrtpstats-dict*)

    *

    *  "local-id"            G_TYPE_STRING               identifier for the associated RTCInboundRTPSTreamStats

+   *  "remote-timestamp"    G_TYPE_DOUBLE               remote timestamp the statistics were sent by the remote

    *

    */

   gst_webrtc_bin_signalsGET_STATS_SIGNAL =

​

@@ -346,11 +346,13 @@

     WPEView* view = nullptr;

     dispatch(&() mutable {

-        auto* manager = webkit_website_data_manager_new_ephemeral();

-        auto web_context = webkit_web_context_new_with_website_data_manager(manager);

-        g_object_unref(manager);

-

-        view = new WPEView(web_context, src, context, display, width, height);

+        if (!glib.web_context) {

+            auto *manager = webkit_website_data_manager_new_ephemeral();

+            glib.web_context =

+                webkit_web_context_new_with_website_data_manager(manager);

+            g_object_unref(manager);

+        }

+        view = new WPEView(glib.web_context, src, context, display, width, height);

     });

     if (view && view->hasUri()) {

​

@@ -203,7 +203,7 @@

 _parse_eit (GstMpegtsSection * section)

 {

   GstMpegtsEIT *eit = NULL;

-  guint i = 0, allocated_events = 12;

+  guint allocated_events = 12;

   guint8 *data, *end, *duration_ptr;

   guint16 descriptors_loop_length;

@@ -267,8 +267,6 @@

     if (event->descriptors == NULL)

       goto error;

     data += descriptors_loop_length;

-

-    i += 1;

   }

   if (data != end - 4) {

@@ -362,7 +360,7 @@

 _parse_bat (GstMpegtsSection * section)

 {

   GstMpegtsBAT *bat = NULL;

-  guint i = 0, allocated_streams = 12;

+  guint allocated_streams = 12;

   guint8 *data, *end, *entry_begin;

   guint16 descriptors_loop_length, transport_stream_loop_length;

@@ -444,7 +442,6 @@

     data += descriptors_loop_length;

-    i += 1;

     transport_stream_loop_length -= data - entry_begin;

   }

@@ -540,7 +537,7 @@

 _parse_nit (GstMpegtsSection * section)

 {

   GstMpegtsNIT *nit = NULL;

-  guint i = 0, allocated_streams = 12;

+  guint allocated_streams = 12;

   guint8 *data, *end, *entry_begin;

   guint16 descriptors_loop_length, transport_stream_loop_length;

@@ -625,7 +622,6 @@

     data += descriptors_loop_length;

-    i += 1;

     transport_stream_loop_length -= data - entry_begin;

   }

@@ -882,7 +878,7 @@

 _parse_sdt (GstMpegtsSection * section)

 {

   GstMpegtsSDT *sdt = NULL;

-  guint i = 0, allocated_services = 8;

+  guint allocated_services = 8;

   guint8 *data, *end, *entry_begin;

   guint tmp;

   guint sdt_info_length;

@@ -954,7 +950,6 @@

     data += descriptors_loop_length;

     sdt_info_length -= data - entry_begin;

-    i += 1;

   }

   if (data != end - 4) {

@@ -1313,7 +1308,7 @@

 _parse_sit (GstMpegtsSection * section)

 {

   GstMpegtsSIT *sit = NULL;

-  guint i = 0, allocated_services = 8;

+  guint allocated_services = 8;

   guint8 *data, *end, *entry_begin;

   guint sit_info_length;

   guint descriptors_loop_length;

@@ -1373,7 +1368,6 @@

     data += descriptors_loop_length;

     sit_info_length -= data - entry_begin;

-    i += 1;

   }

   if (data != end - 4) {

​

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

  <release>

   <Version>

+   <revision>1.20.2</revision>

+   <branch>1.20</branch>

+   <name></name>

+   <created>2022-05-02</created>

+   <file-release rdf:resource="https://gstreamer.freedesktop.org/src/gst-plugins-bad/gst-plugins-bad-1.20.2.tar.xz" />

+  </Version>

+ </release>

+

+ <release>

+  <Version>

    <revision>1.20.1</revision>

    <branch>1.20</branch>

    <name></name>

​

@@ -775,7 +775,8 @@

     interlace_mode = gst_structure_get_string (s, "interlace-mode");

     /* Do not double the framerate for interlaced - we will either passthrough

      * or fail to negotiate */

-    if (skip_progressive && (g_strcmp0 (interlace_mode, "progressive") != 0))

+    if (skip_progressive && (interlace_mode

+            && g_strcmp0 (interlace_mode, "progressive") != 0))

       continue;

     if (G_VALUE_TYPE (val) == GST_TYPE_FRACTION) {

@@ -891,6 +892,8 @@

   top_field_first = interlace->top_field_first;

   GST_OBJECT_UNLOCK (interlace);

+  GST_DEBUG_OBJECT (pad, "Querying caps with filter %" GST_PTR_FORMAT, filter);

+

   if (filter != NULL) {

     clean_filter = gst_caps_copy (filter);

     if (pattern == GST_INTERLACE_PATTERN_1_1) {

@@ -928,9 +931,12 @@

     }

   }

+  GST_DEBUG_OBJECT (pad, "Querying peer with filter %" GST_PTR_FORMAT,

+      clean_filter);

   tcaps = gst_pad_get_pad_template_caps (otherpad);

   othercaps = gst_pad_peer_query_caps (otherpad, clean_filter);

   othercaps = gst_caps_make_writable (othercaps);

+  GST_DEBUG_OBJECT (pad, "Other caps %" GST_PTR_FORMAT, othercaps);

   if (othercaps) {

     if (pattern == GST_INTERLACE_PATTERN_2_2) {

       for (i = 0; i < gst_caps_get_size (othercaps); ++i) {

​

@@ -114,7 +114,7 @@

   guint32 start_code;

   gboolean STD_buffer_bound_scale G_GNUC_UNUSED;

-  guint16 STD_buffer_size_bound;

+  guint16 STD_buffer_size_bound G_GNUC_UNUSED;

   const guint8 *data;

   gint avail, datalen;

   gboolean have_size = FALSE;

​

@@ -56,7 +56,6 @@

 #define ST_PS_DVD_SUBPICTURE            0xff

 /* Blu-ray related (registration: 'HDMV'*/

-#define ST_BD_AUDIO_AC4                 0x06

 #define ST_BD_AUDIO_LPCM                0x80

 #define ST_BD_AUDIO_AC3                 0x81

 #define ST_BD_AUDIO_DTS                 0x82

​

@@ -1190,6 +1190,12 @@

     return TRUE;

   }

+  /* Don't attempt to handle pmt without any streams */

+  if (G_UNLIKELY (pmt->streams->len == 0)) {

+    GST_WARNING ("Skipping PMT without any entries");

+    return TRUE;

+  }

+

   program_number = section->subtable_extension;

   GST_DEBUG ("Applying PMT (program_number:%d, pid:0x%04x)",

       program_number, section->pid);

​

@@ -1382,14 +1382,6 @@

         is_audio = TRUE;

         caps = gst_caps_new_empty_simple ("audio/x-eac3");

         break;

-      case ST_BD_AUDIO_AC4:

-        /* Opus also uses 0x06, and there are bad streams that have HDMV registration ID,

-         * but contain an Opus registration id, so check for it */

-        if (bstream->registration_id != DRF_ID_OPUS) {

-          is_audio = TRUE;

-          caps = gst_caps_new_empty_simple ("audio/x-ac4");

-        }

-        break;

       case ST_BD_AUDIO_AC3_TRUE_HD:

         is_audio = TRUE;

         caps = gst_caps_new_empty_simple ("audio/x-true-hd");

​

@@ -2112,7 +2112,7 @@

       GstIterator *iter;

       GValue sinkpad_value = G_VALUE_INIT;

       GstClockTime running_time;

-      gboolean all_headers, done = FALSE, res = FALSE;

+      gboolean all_headers, done = FALSE;

       guint count;

       if (!gst_video_event_is_force_key_unit (event))

​

@@ -3109,11 +3109,11 @@

         ret =

             gst_mxf_demux_pad_set_component (demux, pad,

             pad->current_component_index + 1);

-        if (ret != GST_FLOW_OK && ret != GST_FLOW_EOS) {

-          GST_ERROR_OBJECT (demux, "Switching component failed");

-        } else {

+        if (ret == GST_FLOW_OK) {

           pad->current_essence_track->position =

               pad->current_essence_track_position;

+        } else if (ret != GST_FLOW_EOS) {

+          GST_ERROR_OBJECT (demux, "Switching component failed");

         }

       } else if (etrack->duration > 0

           && pad->current_essence_track_position >= etrack->duration) {

@@ -4136,6 +4136,7 @@

     /* We entered a new partition */

     if (ret == GST_FLOW_OK && mxf_is_partition_pack (&klv.key)) {

       GstMXFDemuxPartition *partition = demux->current_partition;

+      gboolean partition_done = FALSE;

       /* Grab footer metadata if needed */

       if (demux->pull_footer_metadata

@@ -4178,8 +4179,13 @@

                 "Track already at another position : %" G_GINT64_FORMAT,

                 partition->single_track->position);

             if (find_edit_entry (demux, partition->single_track,

-                    partition->single_track->position, FALSE, &entry))

+                    partition->single_track->position, FALSE, &entry)) {

               lowest_offset = entry.offset;

+            } else if (partition->single_track->position >=

+                partition->single_track->duration) {

+              GST_DEBUG_OBJECT (demux, "Track fully consumed, partition done");

+              partition_done = TRUE;

+            }

           }

         } else {

           guint i;

@@ -4207,14 +4213,15 @@

           }

         }

-        if (lowest_offset != G_MAXUINT64) {

+        if (partition_done || lowest_offset != G_MAXUINT64) {

           GstMXFDemuxPartition *next_partition = NULL;

           GList *cur_part = g_list_find (demux->partitions, partition);

           if (cur_part && cur_part->next)

             next_partition = (GstMXFDemuxPartition *) cur_part->next->data;

           /* If we have completely processed this partition, skip to next partition */

-          if (lowest_offset > next_partition->partition.this_partition) {

+          if (partition_done

+              || lowest_offset > next_partition->partition.this_partition) {

             GST_DEBUG_OBJECT (demux,

                 "Partition entirely processed, skipping to next one");

             demux->offset = next_partition->partition.this_partition;

​

@@ -116,7 +116,9 @@

     return GST_FLOW_ERROR;

   }

-  if (gst_buffer_get_size (buffer) < 2) {

+  /* Either there is no data or there is at least room for the 16bit length,

+   * therefore the only invalid packet length is 1 */

+  if (gst_buffer_get_size (buffer) == 1) {

     GST_ERROR ("Invalid VANC essence element size");

     gst_buffer_unref (buffer);

     return GST_FLOW_ERROR;

@@ -125,6 +127,11 @@

   gst_buffer_map (buffer, &map, GST_MAP_READ);

   gst_byte_reader_init (&reader, map.data, map.size);

+  /* Some XDCAM recorders store empty vanc packets (without even the

+   * length). Treat them as gaps */

+  if (map.size == 0)

+    goto no_data;

+

   num_packets = gst_byte_reader_get_uint16_be_unchecked (&reader);

   if (num_packets == 0) {

     /* SMPTE 436-1:2013 5.5 The Number of VI Lines or ANC Packets Property

​

@@ -520,7 +520,7 @@

     case GST_EVENT_SEGMENT:{

       gdouble rate;

       GstFormat format;

-      gint64 start_value, stop_value, base;

+      gint64 start_value, stop_value;

       const GstSegment *segment;

       GstSegment seg;

@@ -530,7 +530,6 @@

       format = segment->format;

       start_value = segment->start;

       stop_value = segment->stop;

-      base = segment->base;

       gst_event_unref (event);

@@ -545,7 +544,6 @@

         start_value /= filter->speed;

       if (stop_value >= 0)

         stop_value /= filter->speed;

-      base /= filter->speed;

       /* this would only really be correct if we clipped incoming data */

       filter->timestamp = start_value;

​

@@ -1177,29 +1177,30 @@

   if (seq_header.color_config.color_description_present_flag) {

     GstVideoColorimetry cinfo;

-    gboolean have_cinfo = TRUE;

     gchar *colorimetry = NULL;

-    if (have_cinfo) {

-      if (seq_header.color_config.color_range)

-        cinfo.range = GST_VIDEO_COLOR_RANGE_16_235;

-      else

-        cinfo.range = GST_VIDEO_COLOR_RANGE_0_255;

-

-      cinfo.matrix = gst_video_color_matrix_from_iso

-          (seq_header.color_config.matrix_coefficients);

-      cinfo.transfer = gst_video_transfer_function_from_iso

-          (seq_header.color_config.transfer_characteristics);

-      cinfo.primaries = gst_video_color_primaries_from_iso

-          (seq_header.color_config.color_primaries);

-      colorimetry = gst_video_colorimetry_to_string (&cinfo);

-    }

+    if (seq_header.color_config.color_range)

+      cinfo.range = GST_VIDEO_COLOR_RANGE_0_255;

+    else

+      cinfo.range = GST_VIDEO_COLOR_RANGE_16_235;

+

+    cinfo.matrix = gst_video_color_matrix_from_iso

+        (seq_header.color_config.matrix_coefficients);

+    cinfo.transfer = gst_video_transfer_function_from_iso

+        (seq_header.color_config.transfer_characteristics);

+    cinfo.primaries = gst_video_color_primaries_from_iso

+        (seq_header.color_config.color_primaries);

+

+    colorimetry = gst_video_colorimetry_to_string (&cinfo);

-    if (g_strcmp0 (colorimetry, self->colorimetry)) {

-      g_clear_pointer (&self->colorimetry, g_free);

+    if (g_strcmp0 (colorimetry, self->colorimetry) != 0) {

+      g_free (self->colorimetry);

       self->colorimetry = colorimetry;

+      colorimetry = NULL;

       self->update_caps = TRUE;

     }

+

+    g_clear_pointer (&colorimetry, g_free);

   }

   if (self->subsampling_x != seq_header.color_config.subsampling_x) {

@@ -1544,7 +1545,7 @@

   GstMapInfo map_info;

   GstAV1OBU obu;

   GstFlowReturn ret = GST_FLOW_OK;

-  GstAV1ParserResult res;

+  GstAV1ParserResult res = GST_AV1_PARSER_INVALID_OPERATION;

   GstBuffer *buffer = gst_buffer_ref (frame->buffer);

   guint32 total_consumed, consumed;

   gboolean frame_complete;

@@ -1773,7 +1774,7 @@

   GstAV1Parse *self = GST_AV1_PARSE (parse);

   GstMapInfo map_info;

   GstAV1OBU obu;

-  GstAV1ParserResult res;

+  GstAV1ParserResult res = GST_AV1_PARSER_INVALID_OPERATION;

   GstBuffer *buffer = gst_buffer_ref (frame->buffer);

   gboolean got_seq, got_frame;

   gboolean frame_complete;

​

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

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

-  version : '1.20.1',

+  version : '1.20.2',

   meson_version : '>= 0.59',

   default_options :  'warning_level=1',

                       'buildtype=debugoptimized' )

​

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

 msgstr ""

 "Project-Id-Version: gst-plugins-bad-1.0\n"

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

-"POT-Creation-Date: 2022-03-14 11:40+0000\n"

+"POT-Creation-Date: 2022-05-02 23:30+0100\n"

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

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

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

@@ -89,7 +89,7 @@

 msgstr ""

 #: gst-libs/gst/adaptivedemux/gstadaptivedemux.c:4102

-#: gst/mpegtsdemux/mpegtsbase.c:1751

+#: gst/mpegtsdemux/mpegtsbase.c:1757

 msgid "Internal data stream error."

 msgstr ""

​

@@ -1011,18 +1011,15 @@

 {

   CMSampleBufferRef cmbuf;

   AVAssetReaderTrackOutput *areader = NULL;

-  GstCaps *caps;

   GstBuffer *buf;

   CMTime dur, ts;

   GST_LOG ("Reading %s next buffer", MEDIA_TYPE_TO_STR (type));

   if (type == GST_AVF_ASSET_READER_MEDIA_TYPE_AUDIO && audio_track != NULL) {

     areader = audio_track;

-    caps = audio_caps;

   } else if (type == GST_AVF_ASSET_READER_MEDIA_TYPE_VIDEO &&

       video_track != NULL) {

     areader = video_track;

-    caps = video_caps;

   }

   if (areader == NULL) {

​

@@ -77,13 +77,11 @@

 gst_av_sample_video_sink_class_init (GstAVSampleVideoSinkClass * klass)

 {

   GObjectClass *gobject_class;

-  GstElementClass *gstelement_class;

   GstBaseSinkClass *gstbasesink_class;

   GstVideoSinkClass *gstvideosink_class;

   GstElementClass *element_class;

   gobject_class = (GObjectClass *) klass;

-  gstelement_class = (GstElementClass *) klass;

   gstbasesink_class = (GstBaseSinkClass *) klass;

   gstvideosink_class = (GstVideoSinkClass *) klass;

   element_class = GST_ELEMENT_CLASS (klass);

​

@@ -206,10 +206,7 @@

   gst_apple_core_video_pixel_buffer_unref (gpixbuf);

   if (info) {

-    GstVideoMeta *video_meta;

-

-    video_meta =

-        gst_buffer_add_video_meta_full (buf, GST_VIDEO_FRAME_FLAG_NONE,

+    gst_buffer_add_video_meta_full (buf, GST_VIDEO_FRAME_FLAG_NONE,

         GST_VIDEO_INFO_FORMAT (info), info->width, info->height, n_planes,

         offset, stride);

   }

​

@@ -216,7 +216,7 @@

     IOSurfaceDecrementUseCount (memory->surface);

   memory->surface = surface;

   if (surface) {

-    GLuint tex_id, tex_target, texifmt, texfmt;

+    GLuint tex_id, tex_target, texifmt;

     guint plane;

     CGLError cglError;

@@ -224,9 +224,6 @@

     tex_id = gl_mem->tex_id;

     tex_target = gst_gl_texture_target_to_gl (gl_mem->tex_target);

     texifmt = gst_gl_format_from_video_info (context, &gl_mem->info, plane);

-    texfmt =

-        gst_gl_sized_gl_format_from_gl_format_type (context, texifmt,

-        GL_UNSIGNED_BYTE);

     gl->BindTexture (tex_target, tex_id);

     cglError = CGLTexImageIOSurface2D ((CGLContextObj)

         gst_gl_context_get_gl_context (context), tex_target, texifmt,

​

@@ -68,18 +68,15 @@

     GstVideoFormat in_format, GstCaps * out_caps)

 {

   GstCaps *in_caps;

-  GstCapsFeatures *features;

   g_return_if_fail (gst_caps_is_fixed (out_caps));

   out_caps = gst_caps_copy (out_caps);

-  features = gst_caps_get_features (out_caps, 0);

   gst_video_info_from_caps (&cache->output_info, out_caps);

   in_caps = gst_caps_copy (out_caps);

   gst_caps_set_simple (in_caps, "format",

           G_TYPE_STRING, gst_video_format_to_string (in_format), NULL);

-  features = gst_caps_get_features (in_caps, 0);

   gst_video_info_from_caps (&cache->input_info, in_caps);

   gst_caps_take (&cache->in_caps, in_caps);

​

@@ -608,7 +608,7 @@

 {

   OSStatus status;

   CMSampleBufferRef cm_sample_buffer = NULL;

-  VTDecodeFrameFlags input_flags, output_flags;

+  VTDecodeFrameFlags input_flags;

   GstVtdec *vtdec = GST_VTDEC (decoder);

   GstFlowReturn ret = GST_FLOW_OK;

   int decode_frame_number = frame->decode_frame_number;

@@ -629,7 +629,6 @@

    * reordering ourselves.

    */

   input_flags = kVTDecodeFrame_EnableAsynchronousDecompression;

-  output_flags = 0;

   cm_sample_buffer =

       cm_sample_buffer_from_gst_buffer (vtdec, frame->input_buffer);

​

@@ -58,7 +58,7 @@

 #include <string>

 #ifdef HAVE_WINMM

-#include <timeapi.h>

+#include <mmsystem.h>