On Tue, 2007-10-30 at 17:26 -0400, Paul Christensen wrote:
> > "So you NEVER get excited and shout when chasing DX, so never drive the
> PA to clipping??? If that never happens you are one cool ham."
>
> Generally, when I make a mistake, it's during the initial tune-up procedure
> and not during an operating event. Once I have maximized drive level into
> the amp and accordingly adjusted a tube-based amp for maximum linearity,
> shouting into the mic during a moment of excitement is a technical
> "non-event." But, it depends on the quality of the exciter driving the amp.
> If the exciter is prone to leading-edge power spikes and ALC thermal drift,
> then all bets are off -- even if an ALC line is connected between the amp
> and exciter.
Though the amplifier detected ALC has a chance of reducing the splatter
from those spikes and since ALC is a feedback loop, and many PA
thresholds are set by grid current, not a driftable DC threshold, the PA
ALC has a chance of compensating for the ALC threshold thermal drift in
the exciter.
>
> >> A better means of proper amp operations includes the use of a scope in
> >> trapezoid X-Y mode between the input and output of the amp.
>
> > But it may not always inspire you to keep from a bit of peak
> compression, will it? A little bit of peak compression can show up as a
> lot of higher order intermod, otherwise known as splatter.
I was speaking of peak clipping in the PA, not intentional RF
compression, though ALC is most often used as RF compression.
>
> I have never used RF compression techniques on SSB. The transmitted
> bandwidth is generally narrow enough that I see little added value until TX
> bandwidth increases. Incidentally, there's been a lot of discussion
> concerning inadequate "talk power" from some new radios on the Yaesu and
> Icom user groups. Part of these complaints stem from the fact that as lower
> and upper frequency Tx bandwidth opens up, power is distributed into these
> spectrum areas. As such, lower TX duty-cycle should be expected in the
> traditional narrow SSB range of 300Hz-2.4 kHz when the transmitter is
> producing energy significantly above and below these values. My suspicion
> is that if these complaining ops narrow their Tx bandwidth that they would
> see their so-called "talk power" increase.
Talk power is not on perceived audio loudness, its perceived as how the
average reading power meter shows SSB compared to CW key down. Fact is
that in SSB a narrower audio tends to make a peakier output and since
the output is limited by the peaks the "talk power" is then smaller.
Talk power also depends on the relative phases of audio components that
are affected by the voice box, the throat, the mouth, the microphone,
the audio circuits, and the SSB filter. If the initial audio could have
weak harmonics, essentially a single tone, then there wouldn't
components to add up to make peaks. Using a microphone with a rising
audio characteristic tends to make the voice harmonics stronger and so
to add up to higher peaks at the output. Extending the bass response
while reducing the high frequency response can improve the metered
average power. I don't know if it improves intelligibility, I kind of
doubt it and I know when applied on local FM those users of handheld
radios with sub 1" speakers grumble about the distortion in their audio.
I figure that adding bass response puts more power outside the response
of the typical receiver with 300 Hz lower roll off so all that added
power showing on the wattmeter is rejected at the receiver. And so hurts
audio recovery in those receivers listening.
>
> >> Until the advent of DSP, an even better method consisted of applying mic
> >> audio to one set of scope plates while applying final RF to the opposite
> >> set. The benefit is that the entire system from mic to amp antenna port
> is
> >> used to measure total system linearity. However, since transmit DSP
> >> introduces latency, even if slight, I've found that method no longer
> gives
> >> an accurate display of linearity.
>
> > It didn't work too good in the analog AM transmitter either because of
> frequency
> sensitive phase shifts through the audio chain. And for SSB the fundamental
> is
> that the envelope of the SSB signal is not the same as the envelope of the
> audio
> from the microphone so it won't work at all for SSB.
>
> Not in my experience. There's generally not enough phase shift in
> traditional analog circuits to significantly affect the trapezoid display
> for this purpose.
>
> Paul, W9AC
73, Jerry, K0CQ
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