How to minimize delay in encoded audio streaming?

jeyp4 jaiforfriend at gmail.com
Wed Feb 19 11:20:14 UTC 2020


Hi

I am observing ~0.5 sec delay in speaking to mic and listening from speaker
in following pipeline.
gst-launch-1.0 pulsesrc ! audioconvert ! lamemp3enc ! decodebin !
audioconvert ! autoaudiosink 

How can I minimize this delay?


I noticed that below pipeline gives minimum delay.
gst-launch-1.0 -v pulsesrc buffer-time=20000 ! volume volume=1.0 ! pulsesink
buffer-time=20000



--
Sent from: http://gstreamer-devel.966125.n4.nabble.com/


More information about the gstreamer-devel mailing list