George,
I use a tentec model 234 speech processor that works almost
exactly like the Comdel you described. It works great and
introduces very little distortion but increases average talk
power considerably. I think tentec stopped making it around
1985.
Carl Moreschi N4PY
Franklinton, NC
> ----- Original Message -----
> From: "George, W5YR" <w5yr@att.net>
> To: "Mark Erbaugh" <mark@microenh.com>
> Cc: <tentec@contesting.com>
> Sent: Tuesday, February 19, 2002 1:42 AM
> Subject: Re: [TenTec] Digital Speech Processing & Pegasus
>
>
> > Mark, historically speaking, processing at audio baseband has not proved
> to
> > be all that effective in enhancing articulation and improving effective
> S/N
> > 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.
> >
> > It is for that reason primarily that audio "compression" or clipping has
> > fallen into disrepute and is seldom used in linear systems such as an
SSB
> > transmitter. FM rigs can benefit from direct audio processing, as can AM
> > rigs to an extent, but not SSB.
> >
> > 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.
> >
> > This signal was then heavily clipped and filtered at 500 KHz to remove
all
> > the distortion products which theoretically should all lie outside the
> > original audio passband.
> > Next, the clipped and filtered signal is heterodyned back to baseband
with
> > the same oscillator that was used to generate the 500 KHz SSB signal,
thus
> > avoiding any frequency-shift distortion.
> >
> > The resulting audio signal is then amplified and low-pass filtered and
> > output to the mic input of the transmitter. The audio output when used
> with
> > speech is of notably higher average power with a minimum of distortion
if
> > the increase is held to about 6 db to 9 db. It is possible to obtain
10-15
> > db increase but the distortion by then is noticeable to the ear. At 6
db,
> a
> > sine wave input is indistinguishable from the Comdel output, the
> distortion
> > is that low.
> >
> > Several systems of that type have appeared over the years: DX
Engineering,
> > Waters, and others no longer around. TenTec used to make a similar
product
> > that I understand was quite effective - I do not know if it is still
being
> > made; I doubt it.
> >
> > 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.
> >
> > The one completely audio processor that was both effective and
> > low-distortion was the Vomax made by Maximillian Associates. I mention
> this
> > especially since it represents an approach that you perhaps might want
to
> > explore for implementing in the computer with .wav files as input.
> >
> > The Vomax was a "split-band" processor that functions altogether at
> > baseband. The incoming mic signal is split into four adjacent audio
> > frequency channels, and the signal in each channel is clipped and
filtered
> > as with r-f processing. Then, the magic comes when the clipped and
> filtered
> > multi-band signals are adroitly recombined with minimum distortion to
> > produce a baseband output with improved articulation and a modest
increase
> > in average power.
> > The limit to its capability, as with the others, lay in how much
> distortion
> > you were willing to accept for how much power increase.
> >
> > I have used all of these types on the air usually with outboard units. I
> > may be wrong here, but I have been told that the Icom PRO implements a
> form
> > of split-band audio processing within its DSP core. And I have heard
that
> > the PRO II has improved upon this to generate a true r-f clipped and
> > filtered signal in DSP. If you have access to old issues of Ham Radio
> > Magazine, the prototype of the Vomax was a feature article sometime in
the
> > early 70's as I recall. Later on, QST published an updated design using
> the
> > same approach but by a different author.
> >
> > So, perhaps all this trivia may encourage you to pursue some form of
> speech
> > processing capablity in your software. I would be leery of simple
"dynamic
> > limiting" however. Unless done with fairly long time constants, in which
> > case the effectiveness is impaired, then the effective clipping of the
> > audio signal generates distortion products that remain within the audio
> > baseband signal.
> >
> > I suspect that you will also find it advisable to afford the user with
> > means to tailor the audio chacteristics of the input and/or output
signal
> > in your processing. Standing out in a pileup is not done readily with a
> > broadband, but beautiful sounding, SSB signal with tons of bass content.
> >
> > Please keep us informed on your progress. Speech processing was a hobby
> > within a hobby for me back in the 70's when I designed and built an r-f
> > processor for my Heathkit SB-401 transmitter that was quite successful.
> >
> > 72/73/oo, George W5YR - the Yellow Rose of Texas
> > Fairview, TX 30 mi NE of Dallas in Collin county EM13qe
> > Amateur Radio W5YR, in the 56th year and it just keeps getting better!
> > QRP-L 1373 NETXQRP 6 SOC 262 COG 8 FPQRP 404 TEN-X 11771
> > Icom IC-756PRO #02121 Kachina #91900556 IC-765 #02437
> >
> > All outgoing email virus-checked by Norton Anti-Virus 2002
> >
> > Mark Erbaugh wrote:
> > >
> > > As I continue my work on my own custom Pegasus control software, I
have
> now
> > > incorporated the ability to play .WAV files through the aux input.
This
> will
> > > allow me to use the computer as a digital voice keyer.
> > >
> > > However, I think that I can take advantage of some sound processing
> software
> > > on the computer to customize the .WAV file to give it more punch.
Other
> than
> > > doing some dynamic limiting to increase the average power, are there
> things
> > > that can be done to increase the chances of standing out or at least
> being
> > > heard in a pileup? Is there a model of HF propagation that can be
used
> to
> > > experiment with the effects of various enahancements?
> >
> > _______________________________________________
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>
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