Hi all,
I think I know what an RF speech processor is. First I'll describe
what I think it is:
You connect your microphone audio to a little AM or DSB generator,
working at some frequency such as 455 kHz. Then you compress or even
clip that signal to increase the average power output. Probably goes
though some bandwidth limiting filters. After that it gets demodulated
back to audio, and feeds the microphone input of the SSB rig. Since the
compression/clipping/whatever is done at some "intermediate frequency"
and then demodulated back down to audio, the IMD "splatter" and
harmonics that are generated, don't get transmitted. Only bandwidth
limited audio goes to the microphone input of the SSB transmitter, and
the SSB transmitter audio stages and RF stages are never driven into
non-linearity. The transmitted signal bandwidth is limited by the usual
crystal or mechanical filters. You get a really dense high average power
signal.
I'm sure I may be wrong in some of the details, yet I'm pretty sure
I've got the basic principle right.
What I don't get is, with DSP used in the transmit section of a
modern rig, can't you write an algorithm running in the DSP system to do
that, and do it better, without resorting to this old technology? I'm
not saying it's bad. When a tried and true method works, it works. Just
seems like this is one of those functions that DSP ought to be able to do.
DE N6KB
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