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

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

+Fri Dec 23 20:13:27 UTC 2022 - Bjørn Lie <zaitor@opensuse.org>

+

+- Update to version 1.20.5

+

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

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

 - Update to version 1.20.4

​

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

 %define _version 1.20.0

 Name:           gstreamer-plugins-bad-codecs

-Version:        1.20.4

+Version:        1.20.5

 Release:        0

 Summary:        Codecs/plugins for gstreamer-plugins-bad

 License:        LGPL-2.1-or-later

​

@@ -1,7 +1,450 @@

+=== release 1.20.5 ===

+

+2022-12-19 23:34:42 +0000  Tim-Philipp Müller <tim@centricular.com>

+

+   * NEWS:

+   * RELEASE:

+   * gst-plugins-bad.doap:

+   * meson.build:

+     Release 1.20.5

+

+2022-12-19 23:34:34 +0000  Tim-Philipp Müller <tim@centricular.com>

+

+   * ChangeLog:

+     Update ChangeLogs for 1.20.5

+

+2022-12-16 15:03:12 +0100  Edward Hervey <edward@centricular.com>

+

+   * gst/mpegtsdemux/mpegtsbase.c:

+     mpegts: Always clear packetizer on DISCONT push mode

+     If a discontinuity is detected in push mode, we need to clear the cached section

+     observations since they might have potentially changed.

+     This was only done properly when operating with TIME segments (dvb, udp,

+     adaptive demuxers, ...) but not with BYTE segments (such as with custom app/fd

+     sources).

+     We still don't want to flush out the PCR observations, since this might be

+     needed for seeking in push-based BYTE sources.

+     Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1650

+     Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3600>

+

+2022-12-17 20:04:01 +0900  Seungha Yang <seungha@centricular.com>

+

+   * sys/d3d11/gstd3d11window_win32.cpp:

+     d3d11videosink: Fixing focus lost on desktop layout change

+     Watch all message on the window thread, instead of internal window only.

+     Otherwise, some global window messages, such as desktop layout change,

+     wouldn't be handled by our window.

+     Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3594>

+

+2022-12-17 01:01:00 +0530  Nirbheek Chauhan <nirbheek@centricular.com>

+

+   * sys/applemedia/avfvideosrc.m:

+     avfvideosrc: Report latency when doing screen capture

+     There is no `device` when doing screen capture, but there is always an

+     `input`, so use that to decide when we can reply to a latency query.

+     Without this, the latency query just fails.

+     Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3591>

+

+2022-12-16 21:42:50 +0900  Seungha Yang <seungha@centricular.com>

+

+   * sys/d3d11/gstd3d11window_win32.cpp:

+     d3d11videosink: Call ShowWindow() from window thread

+     ... when rendering on external HWND. ShowWindow() will cause

+     synchronous message passing to window thread and then can be blocked.

+     At the same time, window thread can wait for GStreamer thread.

+     Instead of the synchronous call, queue the task to window message

+     and performs from the window thread.

+     Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3588>

+

+2022-12-15 01:15:10 +0900  Seungha Yang <seungha@centricular.com>

+

+   * sys/d3d11/gstd3d11videosink.cpp:

+   * sys/d3d11/gstd3d11window.cpp:

+   * sys/d3d11/gstd3d11window.h:

+   * sys/d3d11/gstd3d11window_dummy.cpp:

+   * sys/d3d11/gstd3d11window_win32.cpp:

+     d3d11videosink: Fix deadlock when parent window is busy

+     Deadlock sequence:

+     * From a streaming thread, d3d11videosink sends synchronous message

+     to the parent window, so that internal (child) window can be

+     constructed on the parent window's thread

+     * App thread (parent window thread) is waiting for pipeline's

+     state change (to GST_STATE_NULL) but streaming thread is

+     blocked and waiting for app thread

+     To avoid the deadlock, GstD3D11WindowWin32 should send message

+     to the parent window asynchronously.

+     Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3573>

+

+2022-12-15 20:11:30 +0900  Seungha Yang <seungha@centricular.com>

+

+   * sys/d3d11/gstd3d11window.cpp:

+     d3d11videosink: Error out if RTV is unavailable

+     Add RTV pointer validation

+     Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3573>

+

+2022-07-03 01:18:19 +0900  Seungha Yang <seungha@centricular.com>

+

+   * sys/d3d11/gstd3d11videosink.cpp:

+   * sys/d3d11/gstd3d11videosink.h:

+     d3d11videosink: Protect window with lock at every place

+     Access to the object should be thread safe to support runtime

+     property update

+     Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3573>

+

+2022-10-21 22:46:16 -0500  A. Wilcox <AWilcox@Wilcox-Tech.com>

+

+   * gst-libs/gst/mpegts/gstmpegtsdescriptor.c:

+   * tests/check/libs/mpegts.c:

+     mpegts: Handle when iconv doesn't support ISO 6937

+     Systems like musl libc don't support ISO 6937 in iconv.  This ensures

+     that the MPEG-TS plugin can cope with that.  There is existing support

+     in the plugin for other methods, so it seems to have been the original

+     intent anyway.

+     Fixes: #1314

+     Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3563>

+

+2022-12-06 11:30:47 +0100  Edward Hervey <edward@centricular.com>

+

+   * gst/mpegtsdemux/mpegtspacketizer.c:

+     mpegts: Check continuity counter on section streams

+     This wasn't really done, and is needed in order to detect potential section

+     changes for sections that have got identical information (such as when switching

+     between streams that have the same PAT/PMT pid and subtable information).

+     Other checks exist in tsbase to detect if the "new" PAT/PMT really is an update or not.

+     Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3556>

+

+2022-12-01 19:35:07 +0200  Marek Olejnik <molejnik@fluendo.com>

+

+   * gst-libs/gst/codecs/gsth265decoder.c:

+     h265decoder: Do not abort when failed to prepare ref pic set

+     Currently the element calls abort when failed to prepare reference

+     picture set. This can happent when the input stream is somehow

+     corrupted, like a rtsp strem with lost packets. Now it will only

+     return with GST_FLOW_ERROR instead of terminating whole process.

+     Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3542>

+

+2022-12-07 10:21:52 +0200  Sebastian Dröge <sebastian@centricular.com>

+

+   * gst/dvbsubenc/gstdvbsubenc.c:

+     dvbsubenc: Forward GAP events as-is if we wouldn't produce an end packet and are not in the middle of an existing subtitle

+     An end packet is only produced once for the last subtitle, so multiple

+     GAP events between subtitles would result only in a single end packet

+     and nothing else otherwise. This would potentially starve downstream

+     then, so instead forward the GAP events in that case.

+     Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3536>

+

+2022-12-05 19:30:43 +0530  Nirbheek Chauhan <nirbheek@centricular.com>

+

+   * sys/dshowvideosink/dshowvideosink.cpp:

+     directshow: Fix build error with glib 2.75 and newer

+     Starting with glib 2.75, `NULL` is `nullptr`, which cannot be

+     implicitly coerced to `0`, unlike `NULL`. So explicitly pass `0`.

+     ```

+     3206/4524 Compiling C++ object subprojects/gst-plugins-bad/sys/directshow/gstdirectshow.dll.p/dshowvideosink.cpp.obj

+     FAILED: subprojects/gst-plugins-bad/sys/directshow/gstdirectshow.dll.p/dshowvideosink.cpp.obj

+     "cl" "-Isubprojects\gst-plugins-bad\sys\directshow\gstdirectshow.dll.p" "-Isubprojects\gst-plugins-bad\sys\directshow" "-I..\subprojects\gst-plugins-bad\sys\directshow" "-Isubprojects\gst-plugins-bad" "-I..\subprojects\gst-plugins-bad" "-Isubprojects\gst-plugins-base\gst-libs" "-I..\subprojects\gst-plugins-base\gst-libs" "-Isubprojects\gstreamer\libs" "-I..\subprojects\gstreamer\libs" "-Isubprojects\gstreamer" "-I..\subprojects\gstreamer" "-Isubprojects\orc" "-I..\subprojects\orc" "-I..\subprojects\gst-plugins-bad\sys\directshow\strmbase\baseclasses" "-Isubprojects\gst-plugins-base\gst-libs\gst\video" "-Isubprojects\gstreamer\gst" "-Isubprojects\gst-plugins-base\gst-libs\gst\audio" "-Isubprojects\gst-plugins-base\gst-libs\gst\tag" "-IC:/gst-install/include/glib-2.0" "-IC:/gst-install/lib/glib-2.0/include" "-IC:/gst-install/include" "/MD" "/nologo" "/showIncludes" "/utf-8" "/W2" "/EHsc" "/O2" "/Zi" "/wd4018" "/wd4146" "/wd4244" "/wd4305" "/utf-8" "/we4002" "/we4003" "/we4013" "/we4020" "/we4027" "/we4029" "/we4033" "/we4045" "/we4047" "/we4053" "/we4062" "/we4098" "/we4101" "/we4189" "/utf-8" "-D_MBCS" "/wd4189" "/wd4456" "/wd4701" "/wd4703" "/wd4706" "/wd4996" "-DHAVE_CONFIG_H" "/Fdsubprojects\gst-plugins-bad\sys\directshow\gstdirectshow.dll.p\dshowvideosink.cpp.pdb" /Fosubprojects/gst-plugins-bad/sys/directshow/gstdirectshow.dll.p/dshowvideosink.cpp.obj "/c" ../subprojects/gst-plugins-bad/sys/directshow/dshowvideosink.cpp

+     ../subprojects/gst-plugins-bad/sys/directshow/dshowvideosink.cpp(62): warning C5051: attribute 'noinline' requires at least '/std:c++20'; ignored

+     ../subprojects/gst-plugins-bad/sys/directshow/dshowvideosink.cpp(123): error C2664: 'LRESULT SendMessageA(HWND,UINT,WPARAM,LPARAM)': cannot convert argument 3 from 'nullptr' to 'WPARAM'

+     ../subprojects/gst-plugins-bad/sys/directshow/dshowvideosink.cpp(123): note: A native nullptr can only be converted to bool or, using reinterpret_cast, to an integral type

+     C:\Program Files (x86)\Windows Kits\10\include\10.0.19041.0\um\winuser.h(3690): note: see declaration of 'SendMessageA'

+     ../subprojects/gst-plugins-bad/sys/directshow/dshowvideosink.cpp(635): error C2664: 'BOOL SystemParametersInfoA(UINT,UINT,PVOID,UINT)': cannot convert argument 2 from 'nullptr' to 'UINT'

+     ../subprojects/gst-plugins-bad/sys/directshow/dshowvideosink.cpp(635): note: A native nullptr can only be converted to bool or, using reinterpret_cast, to an integral type

+     C:\Program Files (x86)\Windows Kits\10\include\10.0.19041.0\um\winuser.h(13153): note: see declaration of 'SystemParametersInfoA'

+     ../subprojects/gst-plugins-bad/sys/directshow/dshowvideosink.cpp(1593): error C2664: 'LRESULT SendMessageA(HWND,UINT,WPARAM,LPARAM)': cannot convert argument 3 from 'nullptr' to 'WPARAM'

+     ../subprojects/gst-plugins-bad/sys/directshow/dshowvideosink.cpp(1593): note: A native nullptr can only be converted to bool or, using reinterpret_cast, to an integral type

+     C:\Program Files (x86)\Windows Kits\10\include\10.0.19041.0\um\winuser.h(3690): note: see declaration of 'SendMessageA'

+     ```

+     Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3533>

+

+2022-11-29 02:22:50 +0900  Seungha Yang <seungha@centricular.com>

+

+   * sys/mediafoundation/gstmfvideosrc.c:

+     mfvideosrc: Fix buffer leak

+     The allocated buffer should be released

+     Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3496>

+

+2022-11-29 02:10:42 +0900  Seungha Yang <seungha@centricular.com>

+

+   * sys/mediafoundation/gstmfsourcereader.cpp:

+   * sys/mediafoundation/gstwin32devicewatcher.cpp:

+     mediafoundation: Chain up from GObject::constructed

+     ... so that GstTracer can trace it

+     Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3496>

+

+2022-11-29 02:06:11 +0900  Seungha Yang <seungha@centricular.com>

+

+   * sys/d3d11/gstd3d11videosink.cpp:

+   * sys/d3d11/gstd3d11window_corewindow.cpp:

+   * sys/d3d11/gstd3d11window_dummy.cpp:

+   * sys/d3d11/gstd3d11window_swapchainpanel.cpp:

+   * sys/d3d11/gstd3d11window_win32.cpp:

+     d3d11: Use correct ref/unref methods

+     Those objects are GstObject subclasses

+     Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3496>

+

+2022-11-25 18:38:27 +0800  He Junyan <junyan.he@intel.com>

+

+   * gst-libs/gst/codecparsers/gsth264parser.c:

+     h264parser: Fix a typo in pred_weight_table parsing.

+     When setting default values, the reference list number of l1 is wrong.

+     Fix: https://gitlab.freedesktop.org/gstreamer/gstreamer-vaapi/-/issues/336

+     Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3468>

+

+2022-11-21 19:40:35 +0200  Sebastian Dröge <sebastian@centricular.com>

+

+   * gst/dvbsubenc/gstdvbsubenc-util.c:

+   * gst/dvbsubenc/gstdvbsubenc.c:

+   * gst/dvbsubenc/gstdvbsubenc.h:

+     dvbsubenc: Write Display Definition Segment if a non-default width/height is used

+     Otherwise it can't be rendered by dvbsuboverlay or ffmpeg at least.

+     Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3459>

+

+2022-11-16 09:42:08 +0100  Edward Hervey <bilboed@bilboed.com>

+

+   * gst/mpegtsdemux/mpegtspacketizer.c:

+     Revert "mpegtspacketizer: memcmp potentially seen_before data"

+     This reverts commit fcad4cc646a23e4e621ec5e8485958ab78d98090.

+     This was wrong is so many ways.

+     * The memcmp was badly used (it should use == 0 to check the data is identical,

+     and not != 0)

+     * There was no boundary checks on the present stream section_data when passing

+     it to memcmp.

+     * The return value should have been TRUE (i.e. we have done all checks, none of

+     them failed, therefore the section has been seen before)

+     * stream->section_data would *always* be NULL if the section had already been

+     processed

+     Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1559

+     Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3427>

+

+2022-11-16 09:29:59 +0100  Edward Hervey <edward@centricular.com>

+

+   * gst/mpegtsdemux/mpegtsbase.c:

+     mpegts: Check is program is identical before updating

+     There is no need to update the program if it's identical :)

+     Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3427>

+

+2022-11-08 19:26:39 +0900  Seungha Yang <seungha@centricular.com>

+

+   * sys/d3d11/gstd3d11screencapturedevice.cpp:

+   * sys/d3d11/gstd3d11screencapturesrc.cpp:

+     d3d11screencapturesrc: Specify PAR 1/1 to template caps

+     ... otherwise PAR can be wrongly signalled during the negotiation

+     Fixing below pipeline when desktop resolution is not 640x480

+     gst-launch-1.0.exe \

+     d3d11screencapturesrc ! videoscale !

+     video/x-raw,width=640,height=480,pixel-aspect-ratio=1/1 ! d3d11videosink

+     Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3372>

+

+2022-11-15 00:11:15 +1100  Jan Schmidt <jan@centricular.com>

+

+   * ext/aes/gstaesdec.c:

+     aesdec: Fix padding removal for per-buffer-padding=FALSE

+     When per-buffer-padding is FALSE, the OpenSSL context needs

+     to be told to remove any padding at the end of the ciphertext

+     Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1243

+     Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3411>

+

+2021-12-01 16:23:41 +0100  Jan Alexander Steffens (heftig) <jan.steffens@ltnglobal.com>

+

+   * gst/rtmp2/gstrtmp2sink.c:

+   * gst/rtmp2/gstrtmp2src.c:

+   * gst/rtmp2/rtmp/rtmpclient.c:

+     rtmp2: Improve error messages

+     Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3368>

+

+2021-12-01 16:04:24 +0100  Jan Alexander Steffens (heftig) <jan.steffens@ltnglobal.com>

+

+   * gst/rtmp2/gstrtmp2sink.c:

+   * gst/rtmp2/gstrtmp2src.c:

+   * gst/rtmp2/rtmp/rtmpclient.c:

+   * gst/rtmp2/rtmp/rtmpconnection.c:

+     rtmp2/connection: Pass triggering GError in 'error' signal

+     Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3368>

+

+2021-12-01 16:02:53 +0100  Jan Alexander Steffens (heftig) <jan.steffens@ltnglobal.com>

+

+   * gst/rtmp2/rtmp/rtmpconnection.c:

+     rtmp2/connection: Pass triggering GError to _emit_error

+     Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3368>

+

+2021-12-01 16:01:10 +0100  Jan Alexander Steffens (heftig) <jan.steffens@ltnglobal.com>

+

+   * gst/rtmp2/rtmp/rtmpconnection.c:

+     rtmp2/connection: Discern reasons for cancelling all commands

+     Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3368>

+

+2021-12-01 15:53:13 +0100  Jan Alexander Steffens (heftig) <jan.steffens@ltnglobal.com>

+

+   * gst/rtmp2/rtmp/rtmpconnection.c:

+     rtmp2/connection: Handle EOF like error

+     Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3368>

+

+2021-12-01 15:49:22 +0100  Jan Alexander Steffens (heftig) <jan.steffens@ltnglobal.com>

+

+   * gst/rtmp2/rtmp/rtmpclient.c:

+     rtmp2/client: Make sure 'salt' is not NULL

+     Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3368>

+

+2021-12-01 15:46:42 +0100  Jan Alexander Steffens (heftig) <jan.steffens@ltnglobal.com>

+

+   * gst/rtmp2/rtmp/rtmpclient.c:

+     rtmp2/client: Make sure 'reason' is not NULL

+     Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3368>

+

+2021-12-01 16:13:42 +0100  Jan Alexander Steffens (heftig) <jan.steffens@ltnglobal.com>

+

+   * gst/rtmp2/rtmp/rtmpclient.c:

+     rtmp2/client: Make sure 'desc' is not NULL

+     Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3368>

+

+2021-12-01 16:11:31 +0100  Jan Alexander Steffens (heftig) <jan.steffens@ltnglobal.com>

+

+   * gst/rtmp2/rtmp/rtmpclient.c:

+     rtmp2/client: Make sure 'code' is not NULL

+     Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3368>

+

+2022-11-06 03:26:31 +0900  Seungha Yang <seungha@centricular.com>

+

+   * sys/d3d11/gstd3d11window.cpp:

+     d3d11videosink: Always clear back buffer on resize

+     Swapchain may not need to be resized if the size of backbuffer

+     is equal to the previous size. Then previously rendered frame will be stay

+     on the screen. Do clear back buffer whenever resize() is called

+     Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3342>

+

+2022-07-27 15:59:35 +0200  Jan Alexander Steffens (heftig) <jan.steffens@ltnglobal.com>

+

+   * ext/srt/gstsrtobject.c:

+     srt: Remove callers for which srt_bstats fails

+     This keeps them from accumulating in the element and in the stats while

+     the sink is not being fed, as long as we at least periodically grab

+     stats.

+     Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3329>

+

+2022-07-29 11:53:18 +0200  Jan Alexander Steffens (heftig) <jan.steffens@ltnglobal.com>

+

+   * ext/srt/gstsrtobject.c:

+     srt: Use simpler list operations for callers

+     Avoid `g_list_append` and `g_list_remove` (which have to scan the list)

+     and replace them with `g_list_prepend` and `g_list_delete_link`.

+     Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3329>

+

+2020-11-10 10:06:37 +0100  Jan Alexander Steffens (heftig) <jan.steffens@ltnglobal.com>

+

+   * ext/srt/gstsrtobject.c:

+     srt: Clean up poll/sock lifecycle

+     Make sure `srtobject->poll_id` is never invalid as long as `srtobject`

+     exists. Only remove our caller socket from it when the socket becomes

+     invalid.

+     Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3329>

+

+2022-06-08 16:35:54 +0200  Jan Alexander Steffens (heftig) <jan.steffens@ltnglobal.com>

+

+   * ext/srt/gstsrtobject.c:

+     srt: Clean up error handling

+     - Make the srt_epoll_wait loops more uniform.

+     - Error only via GError when possible; let the element send the error

+     message. Avoids a second error message.

+     - Return 0 when cancelled. Avoids an error message from the element.

+     - Don't send an error message from send_headers when we're a server

+     sink.

+     Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3329>

+

+2022-10-11 11:14:59 +0200  Jan Alexander Steffens (heftig) <jan.steffens@ltnglobal.com>

+

+   * ext/srt/gstsrtobject.c:

+     srt: Simplify socket stats

+     Don't hide stats depending on whether we're a sending or receiving

+     socket. While we're here, add some more debug logs.

+     Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3329>

+

+2020-10-05 19:50:13 +0200  Jan Alexander Steffens (heftig) <jan.steffens@ltnglobal.com>

+

+   * ext/srt/gstsrtobject.c:

+   * ext/srt/gstsrtobject.h:

+     srt: Replace stats accumulation with naive byte counting

+     srt_bstats cannot be used to get the stats of closed connections, so the

+     best we can do is keep the running count ourselves.

+     Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3329>

+

+2022-10-25 08:14:18 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

+

+   * gst/codecalpha/gstalphacombine.c:

+     alphacombine: Add missing query handler for gaps

+     The gap handling was in place, but there was no event handler to trigger it.

+     Implement the alpha sink event handler for the gaps. This fixes handling of

+     valid streams which may not refresh the alpha frames for every video frames.

+     It will also allow a clean error if the stream was missing the initial

+     alpha frame, at least until we find a better way to handle these

+     invalid frames.

+     Related to #1518

+     Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3279>

+

+2022-10-26 11:47:43 +0200  Ignacio Casal Quinteiro <qignacio@amazon.com>

+

+   * sys/applemedia/avfdeviceprovider.m:

+     avfdeviceprovider: do not leak the properties

+     Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3278>

+

+2022-10-21 21:50:01 +0530  Nirbheek Chauhan <nirbheek@centricular.com>

+

+   * gst-libs/gst/vulkan/cocoa/gstvkcocoa_utils.h:

+   * gst-libs/gst/vulkan/cocoa/gstvkwindow_cocoa.m:

+   * gst-libs/gst/vulkan/ios/gstvkios_utils.h:

+   * gst-libs/gst/vulkan/ios/gstvkwindow_ios.m:

+     gl/vulkan: Fix static linking on macOS

+     duplicate symbol '__invoke_on_main' in:

+     /Library/Frameworks/GStreamer.framework/Versions/1.0/lib/libgstvulkan-1.0.a(cocoa_gstvkwindow_cocoa.m.o)

+     /Library/Frameworks/GStreamer.framework/Versions/1.0/lib/libgstgl-1.0.a(cocoa_gstglwindow_cocoa.m.o)

+     ld: 1 duplicate symbol for architecture x86_64

+     clang: error: linker command failed with exit code 1 (use -v to see invocation)

+     Also make the same change in iOS for consistency.

+     Continuation of https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/1132

+     Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3263>

+

+2022-10-21 10:50:36 +1100  Matthew Waters <matthew@centricular.com>

+

+   * ext/webrtc/gstwebrtcbin.c:

+     webrtcbin: also add rtcp-fb ccm fir for video mlines by default

+     In addition to the 'nack pli' already added.  Both are supported by

+     rtpbin/rtpsession by default already.

+     Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3247>

+

+2022-10-14 18:51:43 +0200  Mathieu Duponchelle <mathieu@centricular.com>

+

+   * ext/webrtc/gstwebrtcice.c:

+     webrtc/nice: fix small leak of split strings

+     g_strfreev previously stopped at our manual NULL-termination. Fix by

+     restoring the pointer after joining.

+     Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3190>

+

+2022-10-14 16:21:07 -0400  Arun Raghavan <arun@asymptotic.io>

+

+   * gst/rtmp2/gstrtmp2sink.c:

+     rtmp2sink: Correctly return GST_FLOW_ERROR on error

+     If there is an error while connecting, the streaming task will be stopped, and

+     is_running() will be false, causing a GST_FLOW_FLUSHING to be returned. Instead,

+     we perform the error check (!self->connection) first, to return an error if

+     that's what occured.

+     Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3192>

+

+2022-10-07 08:30:05 +0200  Wojciech Kapsa <wojciech.kapsa@medvc.eu>

+

+   * sys/decklink/gstdecklinkvideosrc.cpp:

+   * sys/decklink/gstdecklinkvideosrc.h:

+     decklink: reset calculation of gst_decklink_video_src_update_time_mapping on no_signal. When the HDMI cable was disconnected for a long time, the calculation took too much time. SDI cable works fine.

+     Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3168>

+

+2022-10-12 18:40:25 +0100  Tim-Philipp Müller <tim@centricular.com>

+

+   * meson.build:

+     Back to development

+

 === release 1.20.4 ===

 2022-10-12 16:39:47 +0100  Tim-Philipp Müller <tim@centricular.com>

+   * ChangeLog:

    * NEWS:

    * RELEASE:

    * gst-plugins-bad.doap:

​

@@ -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 1.20.4 and was released

-on 12 October 2022.

+The latest bug-fix release in the 1.20 series is 1.20.5 and was released

+on 19 December 2022.

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

 version of this document.

-Last updated: Wednesday 12 October 2022, 15:00 UTC (log)

+Last updated: Monday 19 December 2022, 19:00 UTC (log)

 Introduction

@@ -1791,6 +1791,13 @@

 Known Issues

+-   GStreamer may fail to build the hotdoc documentation with the Meson

+    0.64.0 release owing to a Meson bug. This should only affect systems

+    where hotdoc is installed, and will be fixed in Meson 0.64.1 by this

+    Meson PR in combination with this GStreamer MR. In the meantime,

+    users can pass -Ddoc=disabledor downgrade to an older Meson version

+    (< 0.64.0).

+

 -   nothing in particular at this point (but also see possibly breaking

     changes section above)

@@ -2530,6 +2537,7 @@

     chroma (Y41B, YUV9, YVU9 and IYU9)

 -   macOS build and packaging fixes, in particular fix finding of gio

     modules on macOS for https/TLS support

+-   Fix consuming of the macOS package as a framework in XCode

 -   Performance improvements

 -   Miscellaneous bug fixes, memory leak fixes, and other stability and

     reliability improvements

@@ -2783,6 +2791,252 @@

 -   List of Merge Requests applied in 1.20.4

 -   List of Issues fixed in 1.20.4

+1.20.5

+

+The fifth 1.20 bug-fix release (1.20.5) was released on 19 December

+2022.

+

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

+from 1.20.x.

+

+Highlighted bugfixes in 1.20.5

+

+-   systemclock waiting fixes for certain 32-bit platforms/libcs

+-   alphacombine: robustness improvements for corner case scenarios

+-   avfvideosrc: Report latency when doing screen capture

+-   d3d11videosink: various thread-safety and stability fixes

+-   decklink: fix performance issue when HDMI signal has been lost for a

+    long time

+-   flacparse: Fix handling of headers advertising 32 bits per sample

+-   mpegts: Handle when iconv doesn’t support ISO 6937 (e.g. musl libc)

+-   opengl: fix automatic dispmanx detection for rpi4 and fix usage of

+    eglCreate/DestroyImage

+-   opusdec: Various channel-related fixes

+-   textrender: event handling fixes, esp. for GAP event

+-   subparse: Fix non-closed tag handling

+-   videoscale: fix handling of unknown buffer metas

+-   videosink: reverse playback handling fixes

+-   qtmux: Prefill mode fixes, especially for raw audio

+-   multiudpsink: allow binding to IPv6 address

+-   rtspsrc: Fix usage of IPv6 connections in SETUP

+-   rtspsrc: Only EOS on timeout if all streams are timed out/EOS

+-   splitmuxsrc: fix playback stall if there are unlinked pads

+-   v4l2: Fix SIGSEGV on state change during format changes

+-   wavparse robustness fixes

+-   Fix static linking on macOS (opengl, vulkan)

+-   gstreamer-vaapi: fix headless build against mesa >= 22.3.0

+-   GStreamer Editing Services library: Fix build with tools disabled

+-   webrtc example/demo fixes

+-   unit test fixes for aesdec and rtpjitterbuffer

+-   Cerbero: Fix ios cross-compile with cmake on M1; some recipe updates

+    and other build fixes

+-   Binary packages: pkg-config file fixes for various recipes (ffmpeg,

+    taglib, gstreamer)

+-   Binary packages: Enable high bitdepth support for libvpx (VP8/VP9

+    encoding/decoding)

+-   Binary packages: ship aes plugin

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

+    reliability improvements

+-   Performance improvements

+

+gstreamer

+

+-   allocator: Copy allocator name in gst_allocator_register()

+-   miniobject: support higher refcount values

+-   pads: Fix non-serialized sticky event push, e.g. instant change rate

+    events

+-   padtemplate: Fix annotations

+-   systemclock: Use futex_time64 syscall on x32 and other platforms

+    that always…

+-   Fix build of 1.20 branch with Meson 0.64.1 for those who have hotdoc

+    installed on their system.

+-   meson: fix check for pthread_setname_np()

+-   -Wimplicit-function-declaration in pthread_setname_np check (missing

+    _GNU_SOURCE)

+-   gst-inspect: Don’t leak list

+

+Core Elements

+

+-   concat: Properly propagate EOS seqnum

+-   fakesrc: avoid time overflow with datarate

+

+gst-plugins-base

+

+-   audioconvert, audioresample, audiofilter: fix divide by 0 for input

+    buffer without caps

+-   cdparanoia: Ignore compiler warning coming from the cdparanoia

+    header

+-   oggdemux, parsebin: More leak fixes

+-   opengl: fix automatic dispmanx detection for rpi4

+-   opengl: Fix usage of eglCreate/DestroyImage

+-   opengl: Fix static linking on macOS

+-   opusdec: Various channel-related fixes

+-   textrender: Negotiate caps on a GAP event if none were negotiated

+    yet

+-   textrender: Don’t blindly forward all events and don’t blindly

+    forward all events

+-   timeoverlay: fix pad leak

+-   oggdemux: Don’t leak incoming EOS event

+-   subparse: Fix non-closed tag handling.

+-   videodecoder: Only post latency message if it changed

+-   videoscale: buffer meta handling fixes (NULL-terminate array of

+    valid meta tags)

+-   videosink: Don’t return unknown end-time from get_times()

+-   Bump core requirement in 1.20 branch to 1.20.4

+

+Tools

+

+-   gst-play: Don’t leak the stream collection

+

+gst-plugins-good

+

+-   flacparse: Fix handling of headers advertising 32bps

+-   qt5: deactivate context if fill_info fails

+-   qt5: initialize GError properly in gst_qt_get_gl_wrapcontext()

+-   qtdemux: check return value from gst_structure_get in PIFF box

+-   qtdemux: use unsigned int types to store result of QT_UINT32

+-   qtmux: Prefill mode fixes

+-   oss4: Fix debug category initialization

+-   multiudpsink: allow binding to IPv6 address

+-   rtpjitterbuffer tests: Cast drop-messages-interval type properly

+    (fixing it on 32-bit architectures)

+-   rtspsrc: fix seek event leaks

+-   rtspsrc: Don’t replace 404 errors with “no auth protocol found”

+-   rtspsrc: Only EOS on timeout if all streams are timed out/EOS

+-   rtspsrc: Fix usage of IPv6 connections in SETUP

+-   splitmuxsrc: don’t queue data on unlinked pads

+-   v4l2: Fix SIGSEGV on ‘change state’ during ‘format change’

+-   v4l2videodec: Fix activation of internal pool

+-   wavparse: Avoid occasional crash due to referencing freed buffer.

+-   wavparse: Fix crash that occurs in push mode when header chunks are

+    corrupted in certain ways.

+

+gst-plugins-bad

+

+-   aesdec: Fix padding removal for per-buffer-padding=FALSE

+-   aesdec test failing in gst-plugins-bad

+-   alphacombine: Add missing query handler for gaps

+-   avfdeviceprovider: do not leak the properties

+-   avfvideosrc: Report latency when doing screen capture

+-   d3d11screencapturesrc: Specify PAR 1/1 to template caps

+-   d3d11videosink: Fixing focus lost on desktop layout change

+-   d3d11videosink: Call ShowWindow() from window thread

+-   d3d11videosink: Fix deadlock when parent window is busy

+-   d3d11videosink: Always clear back buffer on resize

+-   decklink: reset calculation of time_mapping to fix clipping HDMI

+    video

+-   directshow: Fix build error with glib 2.75 and newer

+-   dvbsubenc: Forward GAP events as-is if we wouldn’t produce an end

+    packet and…

+-   dvbsubenc: Write Display Definition Segment if a non-default

+    width/height is used

+-   h265decoder: Do not abort when failed to prepare ref pic set

+-   h264parser: Fix a typo in pred_weight_table parsing.

+-   mediafoundation, d3d11: Fix memory leak and make leak tracer happy

+-   mpegts: Handle when iconv doesn’t support ISO 6937 (e.g. musl libc)

+-   mpegts: Check continuity counter on section streams

+-   mpegts: Revert “mpegtspacketizer: memcmp potentially seen_before

+    data”

+-   mpegtspacketizer: memcmp potentially seen_before data

+-   mpegtsdemux: Always clear packetizer on DISCONT push mode

+-   srt: various fixes - improve stats and error handling

+-   rtmp2: Improve error messages

+-   rtmp2sink: Correctly return GST_FLOW_ERROR on error

+-   vulkan: Fix static linking on macOS

+-   webrtcbin: also add rtcp-fb ccm fir for video mlines by default

+-   webrtc/nice: fix small leak of split strings

+

+gst-plugins-ugly

+

+-   No changes

+

+gst-libav

+

+-   avdec_h265: Fix endless renegotiation with alternate interlacing

+-   avviddec: Avoid flushing on framerate changes

+

+gst-rtsp-server

+

+-   rtsp-server: Free client if no connection could be created

+

+gstreamer-vaapi

+

+-   vaapi: prefix internal USE_X11 define to fix build with mesa 22.3.0

+-   vaapi: libs: context: use queried value for attrib

+-   gstreamer-vaapi cannot be built without X11 with recent mesa

+

+gstreamer-sharp

+

+-   No changes

+

+gst-omx

+

+-   No changes

+

+gst-python

+

+-   No changes

+

+gst-editing-services

+

+-   Fix building ges with tools disabled

+-   Fix leaks and minor races in GES

+

+gst-examples:

+

+-   webrtc: Fix double free in webrtc-recvonly-h264 demo

+-   webrtc: Fix critical in webrtc-recvonly-h264 example

+-   webrtc/signalling examples: Fix compatibility with Python 3.10

+

+Development build environment + gst-full build

+

+-   No major changes

+

+Cerbero build tool and packaging changes in 1.20.5

+

+-   oven: output status line at least every minute

+-   Unconditionally set CMAKE_SYSTEM_NAME on Windows

+-   Fix ios cross-compile with cmake on M1

+-   Speed up downloads on Windows drastically

+-   Fix tar usage on bsdtar and print progress while compressing

+-   Actually print the sha for which the cache was not found

+

+Recipes

+

+-   ffmpeg: add patch to generate the pc files properly

+-   taglib: add patch to generate the pc files properly

+-   fontconfig: update to 2.14.1

+-   Windows: Crash on GStreamer 1.20.x x86_64 MSVC + MS-Windows due to

+    libfontconfig fonts.conf file invalid.

+-   openssl: Fix compile errors on upgrades

+-   moltenvk: Also ship the static library on macOS

+-   gstreamer: Add some missing pkgconfig files

+-   gst-plugins-good: Fix post_install failure when qt5 is enabled

+-   gst-plugins-bad: Ship AES plugin

+-   libvpx: Enable high bitdepth support

+-   openssl: update to 1.1.1s

+-   glib: Update patch to auto-detect modules on macOS

+

+Contributors to 1.20.5

+

+Aleksandr Slobodeniuk, Arun Raghavan, A. Wilcox, Bo Elmgreen, Boyuan

+Zhang, Bunio FH, Célestin Marot, Devin Anderson, Edward Hervey, He

+Junyan, Ignacio Casal Quinteiro, Jacek Skiba, Jan Alexander Steffens

+(heftig), Jan Schmidt, Jonas Bonn, Jordan Petridis, Justin Chadwell,

+Linus Svensson, Marek Olejnik, Mathieu Duponchelle, Matthew Waters,

+Nicolas Dufresne, Nirbheek Chauhan, Patrick Griffis, Pawel Stawicki,

+Philippe Normand, Ruben Gonzalez, Sam Van Den Berge, Sebastian Dröge,

+Seungha Yang, Stéphane Cerveau, Tim-Philipp Müller, Vivia Nikolaidou,

+Wojciech Kapsa, Xavier Claessens,

+

+… 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 1.20.5

+

+-   List of Merge Requests applied in 1.20.5

+-   List of Issues fixed in 1.20.5

+

 Schedule for 1.22

 Our next major feature release will be 1.22, and 1.21 will be the

@@ -2790,7 +3044,7 @@

 development of 1.21/1.22 will happen in the git main branch.

 The plan for the 1.22 development cycle is now confirmed, and we aim for

-a 1.22.0 release in December 2022.

+a 1.22.0 release in December 2022 or early January 2023.

 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.

​

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

-This is GStreamer gst-plugins-bad 1.20.4.

+This is GStreamer gst-plugins-bad 1.20.5.

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

 of your favourite cross-platform multimedia framework!

​

@@ -583,11 +583,10 @@

     GST_ERROR_OBJECT (filter, "Could not initialize openssl cipher");

     return FALSE;

   }

-  if (filter->per_buffer_padding) {

-    if (!EVP_CIPHER_CTX_set_padding (filter->evp_ctx, 0)) {

-      GST_ERROR_OBJECT (filter, "Could not set padding");

-      return FALSE;

-    }

+  if (!EVP_CIPHER_CTX_set_padding (filter->evp_ctx,

+          filter->per_buffer_padding ? 0 : 1)) {

+    GST_ERROR_OBJECT (filter, "Could not set padding");

+    return FALSE;

   }

   return TRUE;

​

@@ -38,11 +38,13 @@

 #define GST_CAT_DEFAULT gst_debug_srtobject

 #if SRT_VERSION_VALUE > 0x10402

-#define SRTSOCK_ERROR_DEBUG ("libsrt reported: %s", srt_rejectreason_str (reason))

+#define REASON_FORMAT "s"

+#define REASON_ARGS(reason) srt_rejectreason_str (reason)

 #else

 /* srt_rejectreason_str() is unavailable in libsrt 1.4.2 and prior due to

  * unexported symbol. See https://github.com/Haivision/srt/pull/1728. */

-#define SRTSOCK_ERROR_DEBUG ("libsrt reported reject reason code %d", reason)

+#define REASON_FORMAT "s %d"

+#define REASON_ARGS(reason) "reject reason code", (reason)

 #endif

 /* Define options added in later revisions */

@@ -52,10 +54,6 @@

 #define SRTO_RETRANSMITALGO 61

 #endif

-#define ELEMENT_WARNING_SRTSOCK_ERROR(code, reason) \

-  GST_ELEMENT_WARNING (srtobject->element, RESOURCE, code, \

-  ("Error on SRT socket. Trying to reconnect."), SRTSOCK_ERROR_DEBUG)

-

 enum

 {

   PROP_URI = 1,

@@ -82,9 +80,6 @@

   gboolean sent_headers;

 } SRTCaller;

-static GstStructure *gst_srt_object_accumulate_stats (GstSRTObject * srtobject,

-    SRTSOCKET srtsock);

-

 static SRTCaller *

 srt_caller_new (void)

 {

@@ -118,14 +113,6 @@

 static void

 srt_caller_signal_removed (SRTCaller * caller, GstSRTObject * srtobject)

 {

-  GstStructure *stats;

-

-  stats = gst_srt_object_accumulate_stats (srtobject, caller->sock);

-

-  /* FIXME: These are the final statistics for the caller before we close its

-   * socket. Deliver the stats to the app before we throw them away. */

-  gst_structure_free (stats);

-

   g_signal_emit_by_name (srtobject->element, "caller-removed", 0,

       caller->sockaddr);

 }

@@ -376,11 +363,12 @@

 {

   g_return_if_fail (srtobject != NULL);

-  if (srtobject->poll_id != SRT_ERROR) {

-    srt_epoll_release (srtobject->poll_id);

-    srtobject->poll_id = SRT_ERROR;

+  if (srtobject->sock != SRT_INVALID_SOCK) {

+    srt_close (srtobject->sock);

   }

+  srt_epoll_release (srtobject->poll_id);

+

   g_cond_clear (&srtobject->sock_cond);

   GST_DEBUG_OBJECT (srtobject->element, "Destroying srtobject");

@@ -935,19 +923,19 @@

     GST_DEBUG_OBJECT (srtobject->element, "Waiting a request from caller");

-    if (srt_epoll_wait (srtobject->listener_poll_id, &rsock,

-            &rsocklen, 0, 0, poll_timeout, NULL, 0, NULL, 0) < 0) {

+    if (srt_epoll_wait (srtobject->listener_poll_id, &rsock, &rsocklen, 0, 0,

+            poll_timeout, NULL, 0, NULL, 0) < 0) {

       gint srt_errno = srt_getlasterror (NULL);

       if (srtobject->listener_poll_id == SRT_ERROR)

         return NULL;

-      if (srt_errno == SRT_ETIMEOUT) {

+

+      if (srt_errno == SRT_ETIMEOUT)

         continue;

-      } else {

-        GST_ELEMENT_ERROR (srtobject->element, RESOURCE, FAILED,

-            ("abort polling: %s", srt_getlasterror_str ()), (NULL));

-        return NULL;

-      }

+

+      GST_ELEMENT_ERROR (srtobject->element, RESOURCE, FAILED,

+          ("abort polling: %s", srt_getlasterror_str ()), (NULL));

+      return NULL;

     }

     caller_sock =

@@ -985,7 +973,7 @@

           caller->sock);

       g_mutex_lock (&srtobject->sock_lock);

-      srtobject->callers = g_list_append (srtobject->callers, caller);

+      srtobject->callers = g_list_prepend (srtobject->callers, caller);

       g_cond_signal (&srtobject->sock_cond);

       g_mutex_unlock (&srtobject->sock_lock);

@@ -1183,7 +1171,7 @@

   if (sock == SRT_INVALID_SOCK) {

     g_set_error (error, GST_LIBRARY_ERROR, GST_LIBRARY_ERROR_INIT, "%s",

         srt_getlasterror_str ());

-    goto failed;

+    return FALSE;

   }

   if (!gst_srt_object_set_common_params (sock, srtobject, error)) {

@@ -1267,6 +1255,7 @@

   if (srt_connect (sock, sa, sa_len) == SRT_ERROR) {

     g_set_error (error, GST_RESOURCE_ERROR, GST_RESOURCE_ERROR_OPEN_READ, "%s",

         srt_getlasterror_str ());

+    srt_epoll_remove_usock (srtobject->poll_id, sock);

     goto failed;

   }

@@ -1275,18 +1264,7 @@

   return TRUE;

 failed:

-

-  if (srtobject->poll_id != SRT_ERROR) {

-    srt_epoll_release (srtobject->poll_id);

-  }

-

-  if (sock != SRT_INVALID_SOCK) {

-    srt_close (sock);

-  }

-

-  srtobject->poll_id = SRT_ERROR;

-  srtobject->sock = SRT_INVALID_SOCK;

-

+  srt_close (sock);

   return FALSE;

 }

@@ -1383,7 +1361,7 @@

 gst_srt_object_open (GstSRTObject * srtobject, GCancellable * cancellable,

     GError ** error)

 {

-  srtobject->previous_bytes = 0;

+  srtobject->bytes = 0;

   return gst_srt_object_open_internal (srtobject, cancellable, error);

 }

@@ -1394,17 +1372,7 @@

   g_mutex_lock (&srtobject->sock_lock);

   if (srtobject->sock != SRT_INVALID_SOCK) {

-    GstStructure *stats;

-

-    if (srtobject->poll_id != SRT_ERROR) {

-      srt_epoll_remove_usock (srtobject->poll_id, srtobject->sock);

-    }

-

-    stats = gst_srt_object_accumulate_stats (srtobject, srtobject->sock);

-

-    /* FIXME: These are the final statistics for the socket before we close it.

-     * Deliver the stats to the app before we throw them away. */

-    gst_structure_free (stats);

+    srt_epoll_remove_usock (srtobject->poll_id, srtobject->sock);

     GST_DEBUG_OBJECT (srtobject->element, "Closing SRT socket (0x%x)",

         srtobject->sock);

@@ -1452,33 +1420,25 @@

 static gboolean

 gst_srt_object_wait_caller (GstSRTObject * srtobject,

-    GCancellable * cancellable, GError ** error)

+    GCancellable * cancellable)

 {

   gboolean ret;

   g_mutex_lock (&srtobject->sock_lock);

-  if (srtobject->callers == NULL) {

+  ret = (srtobject->callers != NULL);

+  if (!ret) {

     GST_INFO_OBJECT (srtobject->element, "Waiting for connection");

-

-    while (!g_cancellable_is_cancelled (cancellable)) {

-      ret = (srtobject->callers != NULL);

-      if (ret) {

-        GST_DEBUG_OBJECT (srtobject->element, "Got a connection");

-        break;

-      }

-

+    while (!ret && !g_cancellable_is_cancelled (cancellable)) {

       g_cond_wait (&srtobject->sock_cond, &srtobject->sock_lock);

+      ret = (srtobject->callers != NULL);

     }

-  } else {

-    ret = TRUE;

   }

   g_mutex_unlock (&srtobject->sock_lock);

-  if (!ret) {

-    g_set_error (error, GST_RESOURCE_ERROR, GST_RESOURCE_ERROR_FAILED,

-        "Canceled waiting for a connection.");

+  if (ret) {

+    GST_DEBUG_OBJECT (srtobject->element, "Got a connection");

   }

   return ret;

@@ -1511,8 +1471,8 @@

   GST_OBJECT_UNLOCK (srtobject->element);

   if (connection_mode == GST_SRT_CONNECTION_MODE_LISTENER) {

-    if (!gst_srt_object_wait_caller (srtobject, cancellable, error))

-      return -1;

+    if (!gst_srt_object_wait_caller (srtobject, cancellable))

+      return 0;

     g_mutex_lock (&srtobject->sock_lock);

     if (srtobject->callers) {

@@ -1538,34 +1498,48 @@

             poll_timeout, NULL, 0, NULL, 0) < 0) {

       gint srt_errno = srt_getlasterror (NULL);

-      if (srt_errno != SRT_ETIMEOUT) {

+#if SRT_VERSION_VALUE >= 0x010402

+      if (srt_errno == SRT_EPOLLEMPTY)

         return 0;

-      }

-      continue;

+#endif

+

+      if (srt_errno == SRT_ETIMEOUT)

+        continue;

+

+      g_set_error (error, GST_RESOURCE_ERROR, GST_RESOURCE_ERROR_WRITE,

+          "Failed to poll socket: %s", srt_getlasterror_str ());

+      return -1;

     }

     if (wsocklen == 1 && rsocklen == 1) {

       /* Socket reported in wsock AND rsock signifies an error. */

       gint reason = srt_getrejectreason (wsock);

-      gboolean is_auth_error = (reason == SRT_REJ_BADSECRET

-          || reason == SRT_REJ_UNSECURE);

-      if (is_auth_error) {

-        ELEMENT_WARNING_SRTSOCK_ERROR (NOT_AUTHORIZED, reason);

-      }

+      if (reason == SRT_REJ_BADSECRET || reason == SRT_REJ_UNSECURE) {

+        if (connection_mode == GST_SRT_CONNECTION_MODE_LISTENER) {

+          GST_ELEMENT_WARNING (srtobject->element, RESOURCE, NOT_AUTHORIZED,

+              ("Caller failed to authenticate: %" REASON_FORMAT,

+                  REASON_ARGS (reason)), (NULL));

+          return 0;

+        }

-      if (connection_mode == GST_SRT_CONNECTION_MODE_LISTENER) {

-        /* Caller has disappeared. */

-        return 0;

+        GST_ELEMENT_WARNING (srtobject->element, RESOURCE, NOT_AUTHORIZED,

+            ("Failed to authenticate: %" REASON_FORMAT ". Trying to reconnect",

+                REASON_ARGS (reason)), (NULL));

       } else {

-        if (!is_auth_error) {

-          ELEMENT_WARNING_SRTSOCK_ERROR (READ, reason);

+        if (connection_mode == GST_SRT_CONNECTION_MODE_LISTENER) {

+          /* Caller has disappeared. */

+          return 0;

         }

-        gst_srt_object_close (srtobject);

-        if (!gst_srt_object_open_internal (srtobject, cancellable, error)) {

-          return -1;

-        }

+        GST_ELEMENT_WARNING (srtobject->element, RESOURCE, READ,

+            ("Error on SRT socket: %" REASON_FORMAT ". Trying to reconnect",

+                REASON_ARGS (reason)), (NULL));

+      }

+

+      gst_srt_object_close (srtobject);

+      if (!gst_srt_object_open_internal (srtobject, cancellable, error)) {

+        return -1;

       }

       continue;

     }

@@ -1584,6 +1558,8 @@

         return -1;

       }

     }

+

+    srtobject->bytes += len;

     break;

   }

@@ -1597,7 +1573,7 @@

   /* Removing all socket descriptors from the monitoring list

    * wakes up SRT's threads. We only have one to remove. */

-  if (srtobject->sock != SRT_INVALID_SOCK && srtobject->poll_id != SRT_ERROR) {

+  if (srtobject->sock != SRT_INVALID_SOCK) {

     srt_epoll_remove_usock (srtobject->poll_id, srtobject->sock);

   }

@@ -1615,7 +1591,7 @@

 static gboolean

 gst_srt_object_send_headers (GstSRTObject * srtobject, SRTSOCKET sock,

     gint poll_id, gint poll_timeout, GstBufferList * headers,

-    GCancellable * cancellable)

+    GCancellable * cancellable, GError ** error)

 {

   guint size, i;

@@ -1629,36 +1605,51 @@

   for (i = 0; i < size; i++) {

     SRTSOCKET wsock = sock;

     gint wsocklen = 1;

+    gint sent;

     GstBuffer *buffer = gst_buffer_list_get (headers, i);

     GstMapInfo mapinfo;

     if (g_cancellable_is_cancelled (cancellable)) {

-      return FALSE;

+      return TRUE;

     }

-    if (poll_id > 0 && srt_epoll_wait (poll_id, 0, 0, &wsock,

-            &wsocklen, poll_timeout, NULL, 0, NULL, 0) < 0) {

-      continue;

+    if (poll_id >= 0 && srt_epoll_wait (poll_id, 0, 0, &wsock, &wsocklen,

+            poll_timeout, NULL, 0, NULL, 0) < 0) {

+      gint srt_errno = srt_getlasterror (NULL);

+

+#if SRT_VERSION_VALUE >= 0x010402

+      if (srt_errno == SRT_EPOLLEMPTY)

+        return TRUE;

+#endif

+

+      if (srt_errno == SRT_ETIMEOUT)

+        continue;

+

+      g_set_error (error, GST_RESOURCE_ERROR, GST_RESOURCE_ERROR_WRITE,

+          "Failed to poll socket: %s", srt_getlasterror_str ());

+      return FALSE;

     }

     GST_TRACE_OBJECT (srtobject->element, "sending header %u %" GST_PTR_FORMAT,

         i, buffer);

     if (!gst_buffer_map (buffer, &mapinfo, GST_MAP_READ)) {

-      GST_ELEMENT_ERROR (srtobject->element, RESOURCE, READ,

-          ("Could not map the input stream"), (NULL));

+      g_set_error (error, GST_RESOURCE_ERROR, GST_RESOURCE_ERROR_WRITE,

+          "Failed to map header buffer");

       return FALSE;

     }

-    if (srt_sendmsg2 (wsock, (char *) mapinfo.data, mapinfo.size,

-            0) == SRT_ERROR) {

-      GST_ELEMENT_ERROR (srtobject->element, RESOURCE, WRITE, NULL,

-          ("%s", srt_getlasterror_str ()));

+    sent = srt_sendmsg2 (wsock, (char *) mapinfo.data, mapinfo.size, 0);

+    if (sent == SRT_ERROR) {

+      g_set_error (error, GST_RESOURCE_ERROR, GST_RESOURCE_ERROR_WRITE, "%s",

+          srt_getlasterror_str ());

       gst_buffer_unmap (buffer, &mapinfo);

       return FALSE;

     }

+    srtobject->bytes += sent;

+

     gst_buffer_unmap (buffer, &mapinfo);

   }

@@ -1668,30 +1659,36 @@

 static gssize

 gst_srt_object_write_to_callers (GstSRTObject * srtobject,

     GstBufferList * headers,

-    const GstMapInfo * mapinfo, GCancellable * cancellable, GError ** error)

+    const GstMapInfo * mapinfo, GCancellable * cancellable)

 {

-  GList *callers;

+  GList *item, *next;

   g_mutex_lock (&srtobject->sock_lock);

-  callers = srtobject->callers;

-  while (callers != NULL) {

+  for (item = srtobject->callers, next = NULL; item; item = next) {

+    SRTCaller *caller = item->data;

     gssize len = 0;

     const guint8 *msg = mapinfo->data;

     gint sent;

     gint payload_size, optlen = sizeof (payload_size);

-    SRTCaller *caller = callers->data;

-    callers = callers->next;

+    next = item->next;

     if (g_cancellable_is_cancelled (cancellable)) {

       goto cancelled;

     }

     if (!caller->sent_headers) {

-      if (!gst_srt_object_send_headers (srtobject, caller->sock, -1,

-              -1, headers, cancellable)) {

+      GError *error = NULL;

+

+      if (!gst_srt_object_send_headers (srtobject, caller->sock, -1, 0,

+              headers, cancellable, &error)) {

+        GST_WARNING_OBJECT (srtobject->element,

+            "Failed to send headers to caller %d: %s",

+            caller->sock, error->message);

+        g_error_free (error);

         goto err;

       }

+

       caller->sent_headers = TRUE;

     }

@@ -1710,12 +1707,13 @@

         goto err;

       }

       len += sent;

+      srtobject->bytes += sent;

     }

     continue;

   err:

-    srtobject->callers = g_list_remove (srtobject->callers, caller);

+    srtobject->callers = g_list_delete_link (srtobject->callers, item);

     srt_caller_signal_removed (caller, srtobject);

     srt_caller_free (caller);

   }

@@ -1725,7 +1723,7 @@

 cancelled:

   g_mutex_unlock (&srtobject->sock_lock);

-  return -1;

+  return 0;

 }

 static gssize

@@ -1749,9 +1747,10 @@

   if (!srtobject->sent_headers) {

     if (!gst_srt_object_send_headers (srtobject, srtobject->sock,

-            srtobject->poll_id, poll_timeout, headers, cancellable)) {

+            srtobject->poll_id, poll_timeout, headers, cancellable, error)) {

       return -1;

     }

+