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[TenTec] Type of Speech Processing in Jupiter/Pegasus?

To: <tentec@contesting.com>
Subject: [TenTec] Type of Speech Processing in Jupiter/Pegasus?
From: mark@microenh.com (Mark Erbaugh)
Date: Sat, 23 Feb 2002 16:13:57 -0500

I'd be interested in the details of your processing algorithm. I'm looking
into using the PC as a DSP to process some pre-recorded .WAV files to punch
them up and use as a DVK with a soundcard. I'm a programmer (business
software) whose trying to get my feet wet with DSP. For this project, I plan
on processing the .WAV files once and won't have to do the processing in
real time, but I suspect that that would be doable with a moderately
powerful PC.


----- Original Message -----
From: "Tom Rauch" <w8ji@akorn.net>
To: "Joe Word" <joe_word@yahoo.com>; "George, W5YR" <w5yr@att.net>
Cc: <tentec@contesting.com>
Sent: Saturday, February 23, 2002 15:37 PM
Subject: Re: [TenTec] Type of Speech Processing in Jupiter/Pegasus?

> I haven't been following this closely, so forgive me if I repeat
> something that has already been said but......
> > The r-f clipping and filtering approach which prevents the distortion
> > products from appearing in the output passband is still regarded by
> > most folks as the most effective process to both increase average
> > power and increase articulation.
> The most effective processor is a split-band audio processor,
> assuming you can keep the phase constant between channels. It
> is much better than RF processing, which eliminates harmonic
> distortion but not the in-band odd-order IM distortion associated
> with mixing of inband signals with harmonics and harmonics with
> harmonics.
> Is that what the Jupiter does???
> The 1870's vintage VOMAX was a somewhat frail attempt at this,
> and it would be much better to do it with DSP now. I was actually
> working on a DSP processor which was the equivalent of gain
> compression (AGC), splitting audio into multiple channels with
> individual level controls, "clipping" them with  adjustable clipping for
> each channel, converting the clipped waveforms back to sinewaves
> (low pass filtering), and putting them back in phase with adjustable
> levels. But I don't care about SSB enough to finish the work I guess.
> If you bandpass-split audio into multiple channels, preventing
> clipping harmonics from falling inside cutoff of each clipper's low-
> pass output filter, you wind up without clipping-generated IMD or
> harmonic distortion no matter how much clipping you use.
> 73, Tom W8JI
> W8JI@contesting.com
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