Minimizing Transmit Audio Latency
These comments pertain to a virtual audio connection from your audio processing software or interface to PowerSDR
  • In PowerSDR - Setup - Audio - VAC, use Windows WDM-KS drivers, a buffer size of 128 and a sample rate of 96KHz together with a buffer latency of 20mS (you can only do this with WDM-KS).
  • Set your Virtual Audio Cable and your audio interface for 24bit and 96KHz in the driver software and/or in Windows Sound.
  • In your DAW software (e.g. Audiomulch), set a sample rate of 96KHz and use a buffer size of 1024 with 6 buffers.
  • Experiment and see if dropping down to 48KHz on everything will result in lower latency.

(thanks to WU2O)

However, on 23Mar15, John KF5SAB posted this:

On setting the VAC samplerate to 96khz: this is entirely counterproductive; the native DSP output is 48khz, so setting it to anything else introduces an additional resampler. While the 48->96 case is ideally lossless and has a single-sample delay I do not know if WDSP makes the optimizations, and I would suggest avoiding it anyway. I can see no advantage to running 96khz audio in any amateur radio application save for wideband I/Q transport.

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