I use the built DSP processor. I have it set to 70 and have checked it out
with many station including a VE3 who knows my voice. I leave it on all the
time especially now that Carl's software turns it off for me when I go to AM
(thanks Carl). The VE3 says it adds more punch to my signal than my homebrew
300W amp. In fact I have put the amp up for sale on the local swapshop.
73, Dave, VE1ADH
----- Original Message -----
From: "Carl Moreschi" <n4py@earthlink.net>
To: "tentec" <tentec@contesting.com>
Sent: Tuesday, February 19, 2002 12:12 AM
Subject: Fw: [TenTec] Digital Speech Processing & Pegasus
>
>
> 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?
> > >
> > > _______________________________________________
> > > TenTec mailing list
> > > TenTec@contesting.com
> > > http://lists.contesting.com/mailman/listinfo/tentec
> >
>
> _______________________________________________
> TenTec mailing list
> TenTec@contesting.com
> http://lists.contesting.com/mailman/listinfo/tentec
>
>
|