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[TenTec] Digital Speech Processing & Pegasus

To: <tentec@contesting.com>
Subject: [TenTec] Digital Speech Processing & Pegasus
From: mark@microenh.com (Mark Erbaugh)
Date: Tue, 19 Feb 2002 08:06:07 -0500

Thanks for your most informative post on speech processing.  Please see my
comments below:

> Mark, historically speaking, processing at audio baseband has not proved
> be all that effective in enhancing articulation and improving effective
> for SSB signals. True, it can be highly effective in increasing average
> power, which is alright is one is willing to accept that most of the
> increase comes from distortion products within the baseband

Are the distortion products inherent in the way the baseband signal is
processed? Could appropriate DSP 'shaping' of the baseband signal eliminate
the distortion while keeping the increased average power? For example, the
abrupt phase shift in a PSK31 signal could cause distortion of the audio
signal were it not superimposed on a sine wave. Or does increasing average
power really improve readability at the other end?
> Over the years a number of r-f clipping and filtering systems have been
> developed and marketed, most of which I have had an opportunity to work
> with. My pick of them all has been the Comdel processor which generated
> from baseband audio input an SSB signal at a 500 KHz suppressed carrier
> frequency.

Why does clipping at baseband introduce distortion, but not clipping at 500
kHz? Can you give me more details on the type of clipping and filtering
involved. I believe that this approach is feasible in my application.
Currently, PCs have no problem sampling at 88 kHz (44 kHz stereo). Assuming
that this is the max rate a PC could process data (and I'm sure it is not),
since the signal in my application doesn't need to be processed in real
time, if we expanding the processing by a factor of 11 or 12, we could
effectively sample at 1 MHz, which is fast enough to process a 500 kHz
signal. What does it matter if it takes a minute to process a 5 second .WAV
file? Once the processed .WAV file is created and stored on the PC, it can
be played in real time into the transceiver. Since this is just intended to
get the DX's attention, the same .WAV file or files can be used over and
over again.

> The reason is that most modern rigs now incorporate some form of
> "processing" if for no other reason that to be able to make the marketing
> claim in the specs. Usually this takes the form of mild compression of the
> signal somewhere along the IF chain.

Good point.

> The one completely audio processor that was both effective and
> low-distortion was the Vomax made by Maximillian Associates. I mention
> especially since it represents an approach that you perhaps might want to
> explore for implementing in the computer with .wav files as input.

I don't have the old Ham Radio magazines, but I will search on the internet.
If you have any more detailed information on the processing algorithm of the
Vomax, please let me know. As you suspected, this processing scheme may also
be of use, and it certainly is within the range of the capabilities of a PC
with soundcard.  However, assuming I could implement either the Comdel or
Vomax processing would the Vomax be preferable?

Thanks again,

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