Fixed buffer sizes are used. Rx use s 3456 x 1st downsample rate x 5
audio frames of buffer space. On Windows this means that each
chunk (periodSize()) delivered from the audio stream is our initial
DSP processing chunk size, thus matching audio buffer latency exactly
with WSJT-X's own front end latency. This should result in optimal
resilience to high system loads that might starve the soundcard ADC of
buffers to fill and case dropped audio frames.
For Tx a buffer sufficient for 1 s of audio is used at present, on
Windows the period size will be set to 1/40 of that which gives
reasonably low latency and plenty of resilience to high system loads
that might starve the soundcard DAC of audio frames to render. Note
that a 1 s buffer will make the "Pwr" slider slow to respond, we may
have to reduce the Tx audio buffer size if this is seen as a problem.
