Remove user adjustable audio buffer sizes from Settings

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.

Detector/Detector.cpp

@@ -56,7 +56,7 @@ void Detector::clear ()
 
 qint64 Detector::writeData (char const * data, qint64 maxSize)
 {
-  qDebug () << "Detector::writeData: size:" << maxSize;
+  //qDebug () << "Detector::writeData: size:" << maxSize;
   static unsigned mstr0=999999;
   qint64 ms0 = QDateTime::currentMSecsSinceEpoch() % 86400000;
   unsigned mstr = ms0 % int(1000.0*m_period); // ms into the nominal Tx start time