[Lowfer] TAG in PSKAM10 Mode

Stewart Nelson [email protected]
Thu, 13 Nov 2003 16:50:29 -0800


Hi Johan,

> Another alternative is a complex filter.
> It takes care of the 90 degree shift and does
> bandpass filtering at the same time. My favorite.

A very good idea!  That would get rid of any garbage
at the low and high ends of the audio spectrum,
which would otherwise appear in both sidebands.
I suspect that in a digital application, it would
also cause less ISI, but my math is too weak to
say for sure.

> A few weeks ago I decided to try Alberto di
> Bene's new software radio so I built an LF front
> end inspired by Gerald Youngblood's design "A
> Software Defined Radio for the Masses" from QEX
> (available somewhere on the web as PDF).
> ...
> It works very well with one exception...
> My cheap soundcard seems to have only one A/D 
> converter which is sampling the two channels
> with a 15-20 us time difference... The resulting
> phase shift destroys the suppression of the
> unwanted sideband at higher audio frequencies...
> The delay seems to be independent of sampling rate.

Most "modern" soundcards always sample at 48 kHz,
and convert to lower rates in the software driver,
or sometimes in firmware on the chip.  So a constant
delay is not surprising, but I would expect it to
be about 1/96 kHz = 10.4 us.

If this is a common problem, perhaps Alberto could
add a prefilter to correct for it.  Alternatively, I
would think that a simple analog delay line, e.g.
two inductors + three caps, or an op-amp equivalent,
should do the job.

73,

Stewart KK7KA