This bug was fixed in the package gstreamer-editing-services1.0 -
1.14.5-0ubuntu1~18.04.1

---------------
gstreamer-editing-services1.0 (1.14.5-0ubuntu1~18.04.1) bionic; urgency=medium

  * New upstream bugfix release (LP: #1832123)
  * Update references to 1.14.5

 -- Iain Lane <iain.l...@canonical.com>  Thu, 04 Jul 2019 13:34:01 +0100

** Changed in: gst-libav1.0 (Ubuntu Bionic)
       Status: Fix Committed => Fix Released

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https://bugs.launchpad.net/bugs/1832123

Title:
  [SRU] Bugfix release 1.14.5

Status in gst-libav1.0 package in Ubuntu:
  Fix Released
Status in gst-plugins-bad1.0 package in Ubuntu:
  Fix Released
Status in gst-plugins-base1.0 package in Ubuntu:
  Fix Released
Status in gst-plugins-good1.0 package in Ubuntu:
  Fix Released
Status in gst-plugins-ugly1.0 package in Ubuntu:
  Fix Released
Status in gst-python1.0 package in Ubuntu:
  Fix Released
Status in gst-rtsp-server1.0 package in Ubuntu:
  Fix Released
Status in gstreamer-editing-services1.0 package in Ubuntu:
  Fix Released
Status in gstreamer-vaapi package in Ubuntu:
  Fix Released
Status in gstreamer1.0 package in Ubuntu:
  Fix Released
Status in gst-libav1.0 source package in Bionic:
  Fix Released
Status in gst-plugins-bad1.0 source package in Bionic:
  Fix Released
Status in gst-plugins-base1.0 source package in Bionic:
  Fix Released
Status in gst-plugins-good1.0 source package in Bionic:
  Fix Committed
Status in gst-plugins-ugly1.0 source package in Bionic:
  Fix Committed
Status in gst-python1.0 source package in Bionic:
  Fix Released
Status in gst-rtsp-server1.0 source package in Bionic:
  Fix Released
Status in gstreamer-editing-services1.0 source package in Bionic:
  Fix Released
Status in gstreamer-vaapi source package in Bionic:
  Fix Released
Status in gstreamer1.0 source package in Bionic:
  Fix Released

Bug description:
  [ Description ]

  We should keep up with GStreamer's bugfix releases in the 1.14 series
  that 18.04 shipped with.

  [ QA and testing ]

  Play a range of videos in Totem. Play a range of audio tracks in
  Rhythmbox. Try to stream audio and/or video. Try to install a missing
  codec.

  In all cases, make sure that everything which worked before still
  works.

  [ Regression potential ]

  If one of the fixes is bad then it might break video or audio playback, or 
could crash any application that tries to use gstreamer.
  -------------------------------------------

  Highlighted bugfixes in 1.14.5

  GStreamer core

  -   aggregator: take the pad lock around queue gap event removal
  -   aggregator: don’t leak gap buffer when out of segment
  -   buffer: fix possible memory corruption in gst_buffer_foreach_meta() when 
removing metas
  -   bus: Make removing of signal/bus watches thread-safe
  -   bus: Don’t allow removing signal watches with gst_bus_remove_watch()
  -   controlbinding: Check if the weak pointer was cleared before explicitly 
removing it
  -   ptp clock: Wait for ANNOUNCE before selecting a master; increase 
tolerance for late follow-up and delay-resp
  -   segment: Allow stop == -1 in gst_segment_to_running_time() and negative 
rate
  -   g-i: annotations fixes

  gst-plugins-base

  -   audioconvert: fix endianness conversion for unpacked formats (e.g. 
S24_32BE)
  -   audioringbuffer: Fix wrong memcpy address when reordering channels
  -   decodebin2: Make sure to remove pad probes when freeing GstDecodeGroup
  -   glviewconvert: fix output when a transformation matrix is used
  -   glupload: prevent segfault when updating caps
  -   gl/egl: Determine correct format on dmabuf import
  -   glupload: dmabuf: be explicit about gl formats used
  -   id3tag: validate the year from v1 tags before passing to GstDateTime
  -   rtpbasepayload: fix sequence numbers when using buffer lists
  -   rtspconnection: fix security issue, potential heap overflow 
(CVE-2019-9928)
  -   rtspconnection: fix GError set over the top of a previous GError
  -   rtspconnection: do not duplicate authentication headers
  -   subparse: don’t assert when failing to parse subrip timestamp
  -   video: various convert sample frame fixes
  -   video-converter: fix conversion from I420_10LE/BE, I420_12LE/BE, 
A420_10LE/BE to BGRA/RGBA which created corrupted output
  -   video-format: Fix GBRA_10/12 alpha channel pixel strides

  gst-plugins-good

  -   flv: Use 8kHz sample rate for alaw/mulaw audio
  -   flvdemux: Do not error out if the first added and chained pad is not 
linked
  -   flvmux: try harder to make sure timestamps are always increasing
  -   gdkpixbufdec: output a TIME segment which is what’s expected for raw video
  -   matroskademux: fix handling of MS ACM audio
  -   matroska: fix handling of FlagInterlaced
  -   pulsesink: Deal with not being able to convert a format to caps
  -   rtph265depay, rtph264depay; aggregation packet marker handling fixes
  -   rtpmp4gdepay: detect broken senders who send AAC with ADTS frames
  -   rtprawdepay: keep buffer pool around when flushing/seeking
  -   rtpssrcdemux: Forward serialized events to all pads
  -   qmlglsink: Handle OPENGL header guard changes
  -   qtdemux: fix track language code parsing; ignore corrupted CTTS box
  -   qtmux: Correctly set tkhd width/height to the display size
  -   splitmuxsink: various timecode meta handling fixes
  -   splitmuxsink: make work with audio-only encoders as muxers, e.g. wavenc
  -   v4l2sink: fix pool-less allocation query handling
  -   v4l2dec/enc: fix use after free when handling events
  -   vpx: Fix build against libvpx 1.8
  -   webmmux: allow resolutions above 4096

  gst-plugins-ugly

  -   sid: Fix cross-compilation by using AC_TRY_LINK instead of AC_TRY_RUN
  -   x264: Only enable dynamic loading code for x264 before v253

  gst-plugins-bad

  -   assrender: fix disappearing subtitles when seeking back in time
  -   decklinkvideosink: fix segfault when audiosink is closed before videosink
  -   decklinkvideosrc: respect pixel format property even if mode is set to 
auto
  -   d3dvideosink: Fix calculating buffer size of packed format; don’t leak 
thread object
  -   dtls: Don’t abort on non-fatal issues, make work with newer OpenSSL 
versions
  -   msdk: more robust error handling; fix intel sdk libdir path
  -   nvenc: Ensure drain all frames on finish; fix element reuse and clean up 
properly
  -   openh264dec: Fix handling of errors when doing EOS
  -   shmsrc: fixes a crash when is-live is true due a race condition
  -   shmsink: fix possible (racy) deadlock on shutdown
  -   siren: Fix invalid floating point operation
  -   tsdemux: Skew correction improvements: use upstream DTS if set
  -   wasapi: number of segments was always 2 (the absolute minimum) by accident
  -   wasapi: Fix infinite loop when the device disappears

  gst-libav

  -   libav: Update internal snapshot to ffmpeg n3.4.6
  -   avdemux: fix negative pts if start_time is bigger than the ts

  gst-rtsp-server

  -   rtsp-client: Fix crash in close handler and remove timeout GSource on 
cleanup
  -   rtsp-stream: Use cached address when allocating sockets
  -   rtsp-media: Handle set state when preparing
  -   rtsp-media: Fix race condition in finish_unprepare
  -   rtsp-stream: Use seqnum-offset for rtpinfo
  -   rtsp-stream: add source elements to the pipeline before activation for 
stream-status create message

  gst-editing-services

  -   Fix compilation with latest GLib
  -   layer: Resort clips before syncing priorities
  -   timeline: Better handle loading inconsistent timelines

  gstreamer-vaapi

  -   thread-safety and memory leak fixes
  -   improve caps negotiation if downstream takes ANY caps
  -   fix build with -DG_DISABLE_ASSERT

  gst-omx

  -   fix caps leak

  cerbero

  -   Add support for MacOSX 10.14, iOS 12.1, Fedora 29/30, Linux Mint Tara (19)
  -   Miscellaneous tarball download / error handling improvements
  -   disable parallel builds by default on Windows

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