[sdiy] Weaver frequency shifter

Roman Sowa modular at go2.pl
Thu Apr 28 10:56:40 CEST 2022


We all do analog, digital is just analog amplifier working in overdrive

:)

W dniu 2022-04-28 o 09:40, Mike Bryant pisze:
> We all do digital, analogue is just an infinite sum of squarewaves after 
> all  ;-)
> 
> On 28 Apr 2022 08:38, David G Dixon via Synth-diy 
> <synth-diy at synth-diy.org> wrote:
> Yes.  I don't do digital.
> 
> ------------------------------------------------------------------------
> *From:* Synth-diy [mailto:synth-diy-bounces at synth-diy.org] *On Behalf Of 
> *ben gebhardt via Synth-diy
> *Sent:* Wednesday, April 27, 2022 8:01 PM
> *To:* synth-diy at synth-diy.org
> *Subject:* Re: [sdiy] Weaver frequency shifter
> 
> [*CAUTION:* Non-UBC Email]
> 
> At David Dixon:
> 
> Was your weaver frequency shifter attempted in the analog domain?
> 
> 
> 
> On Wednesday, April 27, 2022, 10:39:48 PM EDT, Eric Brombaugh via 
> Synth-diy <synth-diy at synth-diy.org> wrote:
> 
> 
> On 4/27/22 17:41, Brian Willoughby wrote:
>  > Do these tuner stages at the beginning and end of the Weaver signal 
> chain end up shifting the frequency band of the content?
> 
> Yes. The first tuner stage is used to create an analytic signal by doing
> a quadrature shift of the center of the audio band, resulting in a
> two-sided signal centered on DC. The quadrature LPF removes the unwanted
> image that's created in the process.
> 
> The tuner at the output shifts the analytic signal back up from DC the
> difference between the input shift and output shift is the total shift
> you end up hearing.
> 
>  > Would this help keep all signals within the sample rate? Are these 
> questions moot because of the potentials of under sampling?
> 
> This depends on how you plan out the bandwidths and tuner frequencies.
> In DSP systems the initial shift is often chosen to be Fs/4 which
> simplifies a lot of the math (the quadrature LO becomes a sequence of
> [1+j, -1+j, -1-j, 1-j] which is very easy to do). That also allows the
> quadrature LPFs to be implemented with half-band filters that can also
> simplify the math. Regardless, you have to keep the bandwidths within
> that allowed by Nyquist or suffer aliasing when the final real signal is
> created.
> 
> 
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