Hi Jim:
You mean you use the QF1 on receive to make the wide, full audio sound like communication quality, penetrating, intelligble SSB?

What an idea!

Just kidding, but there is something to my madness....seems to be a lot of folks don't really realize what they sound like feeding wide-band audio to a 50W transmitter. Unless you are working down the street, It takes a lot of poop to sound like an FM Sleepy Time DJ!
I am thinking that a lot of the Apache's rep for sounding "scratchy" is due to the use of the D-104 without a correct input load R. I think they had it pretty much correct as far as the audio set-up internally. Cut-off the really low stuff below a couple hundred Hz, clip so you can run hi levels without overmod--essentially compression, then filter at 3KHz. That spectrum is pretty darned effective.
Using a different mic makes it sound much different--one on air report was that it sounded like a 32V collins rig----I thinked him!
I wasn't too concerned about the QF1 I/O impedance. I use the effects send/return path from the Euro Mixer. Lo Z going into Hi-Z usually works just fine and I think that is the situation on the QF output. Going in, I don't imagine that there is a 8 OHM input Z on the QF. It works because it gets a current driving speaker signal going into its moderate (hi) Z input. Just talking out my butt on that because I have not actually looked at the schematic but it is so common in audio stuff that I took the chance. It woks fine.
You must have the A version...mine can only do one of the functions at a time. It also has quite a lot of AC HUM that new caps did not fix. But it was kind of a proof of concept for me.
Of course, all is out the window if you are trying to change your voice qualities vs simply reproducing your natural voice at the receiving end.
Regarding the scope display....yes the PSDR has a time domain (scope) display. It is what is coming out of the SDR. I find that simple RF sampler with the typical 1N34 detector for monitoring works but leaves a lot to be desired in the true reproduction of the input signal is concerned.
My thinking is, there a multiple stages of the path that all have different effects on the output--we must understand where the big changes (distortions) take place and make changes we can control to create either a close facsimile of the input, or a copy that meets our ideas of how it should sound. Because the receiving end is out of our control, and the ears(hardware) and subjective nature of the person(software) is different than our own, the improvement process can be confusing.
In the popular music business, it was common practice to mix masters that provided exaggerated EQ just so the broadcast to cheap AM radios would sound "punchy"---pretty much the antithesis of the ORTF/European one sweet mic--no eq--right to tape methods for classical recordings.
Anyway, its all fun....I'm off to map my actually voice characteristics on the input side. I don't have anyone close enough and clear enough to do meaningful recording on air yet so will live with putting out what I think is good and wait for the reports to burn me!
BTW....'Splain to me what the mic phasing is all about. I understand those effects in a multiple mic multiple source environment but not sure I grasp why we need to know in a single mic set-up. If it is simply to insure that the larger voice peaks due to asymmetry are on the positive modulation part of the envelope, then I get that. Is there another reason or do I have it?
Cheers
Curt
KU8L