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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