[Bug 701238] New: Interaudio sink/src pair, seems to push last received buffer, instead of silence

[GStreamer (bugzilla.gnome.org)]

https://bugzilla.gnome.org/show_bug.cgi?id=701238
  GStreamer | gst-plugins-bad | 1.0.7

           Summary: Interaudio sink/src pair, seems to push last received
                    buffer, instead of silence
    Classification: Platform
           Product: GStreamer
           Version: 1.0.7
        OS/Version: Linux
            Status: UNCONFIRMED
          Severity: normal
          Priority: Normal
         Component: gst-plugins-bad
        AssignedTo: gstreamer-bugs at lists.freedesktop.org
        ReportedBy: k.althaf at ais.aliftech.co.in
         QAContact: gstreamer-bugs at lists.freedesktop.org
     GNOME version: ---


Seems to push last received buffer or part of it, instead of silence when audio
streaming stops, in order to replicate after starting the pipeline 2 and
pipeline 3, start pipeline 1, after few seconds terminate both the 2 and 3.

Terminal-1 
----------

AUDIO_CAPS="application/x-rtp,media=(string)audio,clock-rate=(int)48000,encoding-name=(string)X-GST-OPUS-DRAFT-SPITTKA-00"

gst-launch-1.0 udpsrc caps=$AUDIO_CAPS port=5555 ! rtpbin ! \
rtpopusdepay ! opusdec !  audio/x-raw, rate=48000, channels=1,\
format=S16LE ! audioconvert   ! interaudiosink channel=s2 \
enable-last-sample=false interaudiosrc channel=s2 ! tee name=asrc \
asrc. ! queue !  wavescope ! videoconvert ! xvimagesink sync=false \
asrc. ! queue ! amix.   udpsrc caps=$AUDIO_CAPS port=6666 ! rtpbin ! \
rtpopusdepay ! opusdec !  audio/x-raw, rate=48000, channels=1, \
format=S16LE ! audioconvert   ! interaudiosink \
enable-last-sample=false channel=s1  interaudiosrc do-timestamp=true \
channel=s1 ! amix.  adder name=amix !  alsasink sync=false 

Terminal-2
----------

gst-launch-1.0  -vvv  audiotestsrc wave=8 ! opusenc ! rtpopuspay pt=96
! udpsink port=5555


Terminal-3
---------------

gst-launch-1.0  -vvv  audiotestsrc wave=3 ! opusenc ! rtpopuspay pt=96
! udpsink port=6666

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