[TenTec] speech processing idea

[Martin AA6E]

Dr. Gerald N. Johnson wrote:
> On Fri, 2006-10-27 at 00:30 -0400, Gary Hoffman wrote:
>> They had pc's in 1958 ?  Interesting.  I was still using mainframes and
>> terminals in 72....shows how much I missed.
>>
>>
> Not really PCs. Tended to take up a whole room or building, and tended
> to have lots of noise from the card readers. I don't know of any
> analogue inputs in real time.
> 
> But speech clippers were tried and they caused much splatter on SSB.
> Even when filtered. The only effective clipping was RF clipping that was
> reformed by running the clipped IF signal through another mechanical or
> crystal filter. But fast acting ALC sometimes does as much. Though one
> rig's popular mod was to add 10 MFD to the ALC line for more punch. But
> then the PA was driven to clipping for a long time before the ALC could
> do anything. A case of slow AGC.
> 
> The tests on punch power that I've done hint that the relative phase
> responses of the microphone, audio stages, and SSB filter have a great
> effect on the average power. I've only gone through trying several
> microphones. I have a Turner made for shipboard steam boiler room
> service that weighs a ton (and its a hand microphone) that in one of my
> VHF rigs gives great punch. It may be that its a bit bassy and so keeps
> down the higher frequency parts of my voice to cut back on the peaks
> that can be inherent when mixing multiple tons of similar amplitudes.
> The peak to average power of a multiple channel cell phone amplifier
> goes down with each additional signal. RF clipping only raises the
> intermod and interference between those signals so the PA has to get
> along with the condition. That could well be a fundamental problem in
> SSB that average power is always going to be significantly lower than
> peak power and there is no solution that doesn't introduce excessive
> intermod which is usually detected as splatter.

Hey, the "solution" to equalize peak-to-average is to use CW, PSK31, 
RTTY, or if you must, digital voice!  I suppose you could also use NBFM 
if you want to keep your linear busy.

Seriously, the Orion v1 software has "real" RF signal compression 
(increased average talk power w/o much distortion), while v2 "SP" seems 
to be more like an audio compressor that does not change the fast peak 
to average ratio much.  It's easy to verify this by watching an output 
power meter.  That's one important reason I'm still on v1.

73 Martin AA6E