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

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

+Wed Oct 26 13:32:09 UTC 2022 - Bjørn Lie <zaitor@opensuse.org>

+

+- Update to version 1.20.4

+

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

 Sat Jun 25 13:49:39 UTC 2022 - Bjørn Lie <zaitor@opensuse.org>

 - Update to version 1.20.3

​

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

 %define _version 1.20.0

 Name:           gstreamer-plugins-bad-codecs

-Version:        1.20.3

+Version:        1.20.4

 Release:        0

 Summary:        Codecs/plugins for gstreamer-plugins-bad

 License:        LGPL-2.1-or-later

​

@@ -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

​

@@ -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

​

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

-This is GStreamer gst-plugins-bad 1.20.3.

+This is GStreamer gst-plugins-bad 1.20.4.

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

 of your favourite cross-platform multimedia framework!

​

@@ -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);

​

@@ -86,10 +86,7 @@

 /* class initialization */

-G_DEFINE_TYPE_WITH_CODE (GstOpenh264Dec, gst_openh264dec,

-    GST_TYPE_VIDEO_DECODER,

-    GST_DEBUG_CATEGORY_INIT (gst_openh264dec_debug_category, "openh264dec", 0,

-        "debug category for openh264dec element"));

+G_DEFINE_TYPE (GstOpenh264Dec, gst_openh264dec, GST_TYPE_VIDEO_DECODER);

 GST_ELEMENT_REGISTER_DEFINE_CUSTOM (openh264dec, openh264dec_element_init);

 static void

@@ -455,10 +452,12 @@

 static gboolean

 openh264dec_element_init (GstPlugin * plugin)

 {

+  GST_DEBUG_CATEGORY_INIT (gst_openh264dec_debug_category, "openh264dec", 0,

+      "debug category for openh264dec element");

   if (openh264_element_init (plugin))

     return gst_element_register (plugin, "openh264dec", GST_RANK_MARGINAL,

         GST_TYPE_OPENH264DEC);

- GST_ERROR ("Incorrect library version loaded, expecting %s", g_strCodecVer);

- return FALSE;

+  GST_ERROR ("Incorrect library version loaded, expecting %s", g_strCodecVer);

+  return FALSE;

 }

​

@@ -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;

 }

​

@@ -562,8 +562,14 @@

    * need to query it here, *before* any openmpt_module_select_subsong()

    * calls are done */

   {

+

+#if OPENMPT_API_VERSION_AT_LEAST(0,5,0)

+    gchar const *subsong_cstr =

+        openmpt_module_ctl_get_text (openmpt_dec->mod, "subsong");

+#else

     gchar const *subsong_cstr =

         openmpt_module_ctl_get (openmpt_dec->mod, "subsong");

+#endif

     gchar *endptr;

     if (subsong_cstr != NULL) {

​

@@ -26,6 +26,10 @@

 G_BEGIN_DECLS

+#define gst_opus_header_is_header gst_opusparse_header_is_header

+#define gst_opus_header_is_id_header gst_opusparse_header_is_id_header

+#define gst_opus_header_is_comment_header gst_opusparse_header_is_comment_header

+

 extern gboolean gst_opus_header_is_header (GstBuffer * buf,

     const char *magic, guint magic_size);

 extern gboolean gst_opus_header_is_id_header (GstBuffer * buf);

​

@@ -31,7 +31,6 @@

         '-Wno-missing-declarations',

         '-Wno-old-style-definition',

         '-Wno-redundant-decls',

-        '-Wno-error',

     )

 endif

@@ -170,6 +169,7 @@

     c_args: compile_args,

     dependencies: dependencies,

     include_directories: include_dirs,

+    override_options: 'werror=false',

     install: false)

 # Declare dependency

​

@@ -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 &&

​

@@ -928,12 +928,19 @@

     }

   }

-out:

   g_list_free (keys);

   g_free (user);

   g_free (pass);

   return uri;

+

+out:

+  g_list_free (keys);

+  g_free (user);

+  g_free (pass);

+  gst_uri_unref (uri);

+

+  return NULL;

 }

 void

​

@@ -215,6 +215,10 @@

 typedef struct _GstPlayMediaInfo GstPlayMediaInfo;

 typedef struct _GstPlayMediaInfoClass GstPlayMediaInfoClass;

+#ifdef G_DEFINE_AUTOPTR_CLEANUP_FUNC

+G_DEFINE_AUTOPTR_CLEANUP_FUNC(GstPlayMediaInfo, g_object_unref)

+#endif

+

 GST_PLAY_API

 GType         gst_play_media_info_get_type (void);

​

@@ -39,6 +39,10 @@

  */

 #define GST_PLAY_SIGNAL_ADAPTER_CAST(obj)        ((GstPlaySignalAdapter*)(obj))

+#ifdef G_DEFINE_AUTOPTR_CLEANUP_FUNC

+G_DEFINE_AUTOPTR_CLEANUP_FUNC(GstPlaySignalAdapter, g_object_unref)

+#endif

+

 GST_PLAY_API

 GType                  gst_play_signal_adapter_get_type               (void);

​

@@ -48,6 +48,10 @@

  */

 #define GST_PLAY_VIDEO_OVERLAY_VIDEO_RENDERER_CAST(obj)        ((GstPlayVideoOverlayVideoRenderer*)(obj))

+#ifdef G_DEFINE_AUTOPTR_CLEANUP_FUNC

+G_DEFINE_AUTOPTR_CLEANUP_FUNC(GstPlayVideoOverlayVideoRenderer, g_object_unref)

+#endif

+

 GST_PLAY_API

 GType gst_play_video_overlay_video_renderer_get_type (void);

​

@@ -49,6 +49,10 @@

   GstElement * (*create_video_sink) (GstPlayVideoRenderer * self, GstPlay * play);

 };

+#ifdef G_DEFINE_AUTOPTR_CLEANUP_FUNC

+G_DEFINE_AUTOPTR_CLEANUP_FUNC(GstPlayVideoRenderer, g_object_unref)

+#endif

+

 GST_PLAY_API

 GType        gst_play_video_renderer_get_type       (void);

​

@@ -56,6 +56,10 @@

 GST_PLAY_API

 void                      gst_play_visualizations_free (GstPlayVisualization **viss);

+#ifdef G_DEFINE_AUTOPTR_CLEANUP_FUNC

+G_DEFINE_AUTOPTR_CLEANUP_FUNC(GstPlayVisualization, gst_play_visualization_free)

+#endif

+

 G_END_DECLS

 #endif /* __GST_PLAY_VISUALIZATION_H__ */

​

@@ -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)

​

@@ -149,9 +149,21 @@

 }

 static void

+gst_player_video_info_finalize (GObject * object)

+{

+  GstPlayerVideoInfo *info = GST_PLAYER_VIDEO_INFO (object);

+

+  g_clear_object (&info->info);

+

+  G_OBJECT_CLASS (gst_player_video_info_parent_class)->finalize (object);

+}

+

+static void

 gst_player_video_info_class_init (G_GNUC_UNUSED GstPlayerVideoInfoClass * klass)

 {

-  /* nothing to do here */

+  GObjectClass *gobject_class = (GObjectClass *) klass;

+

+  gobject_class->finalize = gst_player_video_info_finalize;

 }

 /**

​

@@ -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);

​

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

  <release>

   <Version>

+   <revision>1.20.4</revision>

+   <branch>1.20</branch>

+   <name></name>

+   <created>2022-10-12</created>

+   <file-release rdf:resource="https://gstreamer.freedesktop.org/src/gst-plugins-bad/gst-plugins-bad-1.20.4.tar.xz" />

+  </Version>

+ </release>

+

+ <release>

+  <Version>

    <revision>1.20.3</revision>

    <branch>1.20</branch>

    <name></name>

​

@@ -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);

+