Greg,
The Elecraft power control is done in the RF stages and not the audio.
After a band change or a change in the POWER knob, the power control
loop is reset.
The response time is quick enough that you do not notice it in normal
operation, although it does take a dot or two in CW or a couple of
syllables in voice to obtain enough RF output to be properly measured by
the wattmeter and fed back to the MCU.
In the case of digital modes, if the audio is not sufficient, the
measured power will be lower than what is requested by the power knob
and the radio will begin to ramp up power - but if one or more of the RF
stages has reached its limit, there will not be enough power, and the RF
output level will come up quite slowly or not at all - so the radio
continually "hunts" for the power the power requested, but never makes it.
So keep the audio up to that specified for data (4 bars solid, 5th
flashing) or voice modes (5 to 7 bars) and all will be well.
73,
Don W3FPR
On 6/29/2019 1:35 PM, Greg Troxel wrote:
I have always been a bit unclear on what the Elecraft ALC/power was
really doing.
You mention low power then then hunting, and the notion of a set power
level, measuring, and adjusting gain. That makes me think that there is
an initial gain that is computed/calibrated somehow, that if right will
not result in an adjustment right away. Presumably that's a known
relationship between an audio input of an appropriate power and the
desired RF level. Is that a fair characterization, or if not, could you
explain how it's wrong?
I am guessing also that there is some kind of silence detection or max
gain, as keying the mic without speaking doesn't lead to ramping up to
full-power transmitted noise. I could see the power loop having a
narrow range, enough to account for how much variance there should be,
but not 40 dB worth.
I have seen repeatedly about "4 bars solid and 5th flashing" as the
right place, and the "no ALC" point. Is there really an ALC circuit,
that reduces gain in the audio stage when a higher input level is
measured? Does this have a sufficiently long decay time constant so
that an FT8 or PSK signal would get adjusted acceptably and still have a
low-distortion signal? Or is there still a transfer slope where the
output ends up too high?
Overall, this seems like having two coupled control loops instead of
one, and I'm guessing that the point is to have different time constants
in each one, or to get the levels right at two points in the circuit
instead of one, by having audio gain and RF gain separately controlled.
Is that accurate?
Separately, I wonder about issues with different levels of different
audio frequencies resulting in a transmitted signal which while not
necessarily distorted is off plan and therefore less well decoded
(assuming there isn't adaptive equalization in the decoder after pilot
tones, and I don't have the impression there is). Does this end up
being an issue in practice? I would hope that with the relatively close
spacing within an FT8 signal that this is likely to be not a problem.
tnx es 73 de n1dam
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