Any tips for tuning streaming audio for minimum latency?

CarbonJoe klebau at apk.net
Tue Sep 24 09:03:12 PDT 2013


Does anyone have any useful tips for tuning streaming audio for minimum
latency?  I'm mixing 4 inbound mulaw streams using an adder, and the output
goes to alsasink. The audio is being streamed from 4 other netbooks. The
first requirement is clean, intelligible audio from the other 4 devices.
Next is minimum latency from the netbook out the mixed alsasink. Values I've
tried adjusting (under gstreamer 0.10) are:

  rtppcmupay min-ptime, max-ptime
  udpsink blocksize
  gstrtpjitterbuffer latency

  We're only able to get a latency down to around 285ms, or as high as
950ms. We've tried blocksizes of 160, 320, 640, 1280, 2560, 4096. latency
values were 20, 50, 100, 200, 500.  Min/max ptime were always 20000000. 
With the small jitterbuffer latency values and small blocksizes, the output
stops after around 90 seconds and never recovers. With jitterbuffer 500,
blocksize 4096, the audio is clear, but the latency is huge. And, after
about 30 minutes the audio gets choppy and eventually stops. 



--
View this message in context: http://gstreamer-devel.966125.n4.nabble.com/Any-tips-for-tuning-streaming-audio-for-minimum-latency-tp4662184.html
Sent from the GStreamer-devel mailing list archive at Nabble.com.


More information about the gstreamer-devel mailing list