Interaudio sink/src pair, seems to push last received buffer, instead of silence

Althaf K Backer k.althaf at ais.aliftech.co.in
Wed May 29 02:17:48 PDT 2013


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. You can even visualize the audio residue
in wave scope ie what is finally being pushed in stead of silence.

Gstreamer: 1.0.7

Issue can be replicated with pipelines below,

>From 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

>From Terminal-2

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


>From Terminal-3

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


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