1) When you say out of phase do you mean 180 degrees out of phase?
2) What difference does it make? That is, under what situation would this be a problem?
Please email me your answer so I won't waste any more of other people's time.
David (email@example.com), N1DNA
I don't think this is an unusual problem when running AM with a lot of present day ham gear but would be very surprised if it was a widespread issue with the Flex 6000 line as Flex has a good history of being very much on top of that kind of detail.
Since I use the same external hardware audio chain with multiple transmitters (with some exhibiting inverted phase) I finally gave up and now use a 1:1 (600 ohm) transformer at the output of the audio chain with a DPDT switch on the secondary so I can flip the phase so that the waveform is correct on the scope. The Edcor Transformer company " WSM600/600" 20-20k flat response transformer for just over 10 bucks works beautifully.
Many dynamic elements used in ham-type hand mics don't even have a plus or minus label on the element so it's totally random whether or not they produce in-phase audio so that an inward excursion of the mic engine produces a positive voltage. Not a problem with SSB but as you have noted with AM it makes it impossible to reach 100% positive peaks before reaching 100% negative peaks.
In practice, this makes no difference at all with SSB but it does make a significant difference when running AM. When an AM transmitter is being modulated out-of-phase (100% negative reached before 100% positive) it's easily corrected if you are using a balanced audio source by flipping the balanced pair leads.
I'm still trying to wrap my head around why any of this matters.
Running less than 100% modulation is a way of buying yourself some margin against fading. I just checked the WWV pages and also find that they run most of their modulation at levels between 50% and 75%, presumably to help stay ahead of selective fading on the carrier.
This isn't a stereo system where running one channel out of phase with the other produces incorrect spatialization. So what is the phase reference here? And furthermore, the description about having more than 100% modulation would always produce distortion no matter what you think of the phasing.
I understand fully about the asymmetry of voiced speech waveforms. Any negative excursions in the modulating waveform would produce modulated levels near the 0 line, and should never drive below that, or you will induce distortion. Positive excursions produce the high amplitude levels of the modulated signal.
Phase rotators have been used in the past to help even out the positive and negative excursions in the modulating waveform and keep higher average modulation levels in the transmitted signal.
Not sure what you mean by stating that SDR modulation is different from plate modulation?
I'm not an AM'er, but maybe this kind of modulation+ is a commonly used method among the crowd? If so, it really is a form of overmodulation. And if rigs try to use average voice levels to set the reference DC level, that is a big mistake unless they take care ahead of time to even out the excursions, as with a phase rotator.
We need the DC reference level to be halfway between positive and negative excursions, and the carrier needs to be that same half p-p amplitude in order to guarantee distortion free transmission and reception. Carriers at less than that amplitude will produce distortion under some conditions.
just as an aside... how bad can things get? Here is an example of a waveform that sounds exactly like a square wave (or at least the first 9 partials):
It was generated by feeding a sinewave into a sum of Tchebyshev polynomials, acting as the modulator of the sinewave. of orders 1, 3, 5, 7, and 9, and with amplitudes 1, 1/3, 1/5, 1/7, and 1/9. But if you play this waveform on a synth it sounds exactly like a square wave. The ear can't tell the difference, all it needs are partials and amplitudes.
Hmm... okay, so I looked at my own voice recording that I use for tuneup. Use Audacity or any other tool. My recording shows almost no asymmetry. Why? Where does the asymmetry ascribed to vocal speech come from?
Turns out it seems to come from the bass frequencies. Harmonics exhibit almost no asymmetry. So a quick fix until Flex comes up with something better is to use a high pass filter to saw off your fundamental voice frequency. You don't need that anyway.
There is a psychoacoustic effect where the brain, on hearing a progression of harmonically related partials will reconstruct a fundamental for itself. That's the secret behind "bass boosters", and while not labeled as such, was well known at least back to the time of J.S.Bach for his grand pipe organ music. Just play a C and a descending 4th at G and lo and behold you hear a C an octave lower.
For radio communications you simply don't need those fundamentals. This isn't high-fi sound, it is communications.
As I’ve mentioned in other threads, voices vary widely in their asymmetric characteristics. For AM broadcasters, the classic Kahn Symmetra-Peak was a passive chain of all-pass filters, placed *before* all other audio processors. Its output yielded audio that was symmetric around the baseline, so that there were no lop-sided peaks to artificially force the compressor/limiter into unnecessary gain reduction.
At one small AM station, long ago, one announcer always seemed to have lower voice levels than the others. It turned out that his voice was strongly asymmetrical, forcing the limiter into more gain reduction than usual. Since the station could not afford a Symmetra-Peak, a phase reversal switch was installed in the main DJ mic circuit for that particular announcer to use. We went on to find the optimum phase settings for the other announcers.
Modern AM broadcast processors, of course, do all of this in DSP. Further, the processor “modifies” (distorts?) the waveform so the positive peaks consistently reach 125% modulation, while limiting the negative peaks to 100%. The FCC allows these limits for broadcasters.
Bottom line: Any good voice processor “scrambles” the voice phase into symmetry before any compression/limiting takes place. This should be easy with DSP (several all-pass filters in series), and I’d love to see it added to the Flex mic processor for both SSB and AM.