K3 pwr out on digimodes

Hello,

I'm not a specialist, but this very interesting discussion remember some
points
during my past professional activity.

I was employed in a society involved in the design and fabrication of
RF link for broadcast TV cameras. ( My job was limited to camera remote
control
via separate UHF link(hard&firmware), PCB design and some help with RF
measurements).
The modulations used were QPSK, 16QAM and 64QAM (2K carriers, 8MHz BW)
with descending robustness and ascending bit rate.
According link conditions and/or production requirements, the optimum
"mode" was selected.
As i remember, we counted for a minimum of 10 dB crest factor and the
amplifier
plugged at the camera was designed to output over 40 dBm with 1 dB
compression,
but the max power allowed was (only) 28 dBm. We used also ultra-linear
class a PA
and this minimum 10 dB margin was always the rule. ( I remember a 50W
output COFDM PA
designed for 2.3 to 2.7GHz, a "monster"!)
We never spoke about ALC; the correct input level was adjusted with some
attenuation if necessary and the the signal quality analyzed with a
spectrum.
I had access to the "driver" (max 1W COFDM) schematic and no ALC circuit
is present.
 From FPGA generating I/Q numeric, trough DA converters, modulation,
mixers, etc
all levels were defined with some tolerances for fabrication, but no ALC
We only do power measurements with bolometer, because peak power
measurement
was not an easy task. The link quality was evaluated by BER before and after
Viterbi correction. Gain linearity was specified to be under 1dB for a
8MHz spectrum.
(and other parameters like group delay, etc)
On the RX side, the specifications, to get a "freeze" free MPEG2 video,
was a minimum input RX power level of -93dBm( QPSK mode), -88dBm 16QAM
and -83 dBm (64QAM)..

For the (my) K3, i must admit that the TX linearity i measured with 2.8k
or 2.7k filter is not so good; may be one
of the reason, with RX, i've difficulties to decode ham digital pictures
with good RF levels? but no problem with DRM
using K3 IF output, LP-Pan and DREAM.(i will look for this point....i
have measured the 250Hz
and 2.8kHz filters with the software proposed by KS7D and the 2.8kHz is
far to have a good ~ flat usable width...)

I can understand that the costs are not of the same order, but the
required bandwidth
and the video/audio broadcast requirements not the same too!

As you say, just my 2 cents!

73 QRO fr Rudolf, HB9ARI



Joe Subich, W4TV wrote:

>> No, I do not misunderstand what ALC does. Perhaps I am just
>> not very good at explaining what I am trying to say. I
>> realize that ALC can only reduce the gain. That is why it is
>> necessary to set the audio drive level so that you get a
>> little bit of ALC across the whole passband range that you
>> want to use. Then, the ALC will be able to reduce the gain by
>> the necessary amount so that the RF output is constant across
>> that passband range.
>>    
>
> Absolutely not!  When conventional rigs are generating ALC
> their final amplifiers are ALREADY in compression and they
> are generating IMD in the PSK31 signal.  PSK31 can not have
> ANY compression or clipping without causing increased IMD.
> Increasing the drive to the point that a comventional rig
> generates some level of ALC at all points virtually assures
> the generation of unacceptable IMD at most frequencies in
> the IF passband.
>
>  
>> The FT-817 is not capable of 15W. But that is beside the
>> point.
>>    
>
> OK, do it at 5 watts ...
>
>  
>> What would this experiment prove? I know as well as you do
>> that it would not be identical. That's why I am saying that
>> some form of level control needs to be in operation across
>> the whole of the passband you wish to use.
>>    
>
> That's my point exactly.  The K3 does not provide level control
> across the passband it provides a constant output from the DSP
> modulator (15 KHz IF).  
>
>  
>> A small amount of ALC with a slow time constant will not harm
>> the IMD in PSK31 mode.
>>    
>
> Wrong! ANY COMPRESSION OR CLIPPING will effect the INTENTIONAL
> amplitude shaping in the PSK31 signal designed to prevent IMD
> and control the banwidth of the transmitted signal.  So called
> IMD is the result of a non-linear amplitued response in the
> audio/IF/RF chain.  PSK31 drive can be thought of as a signal
> that has been "predistorted" in such a way to cancel the
> modulation sidebands.  If that (amplitude) predistortion is
> lost due to compression or clipping at any point in the
> transmitter chain, the result is the appearance of the normal
> modulation sidebands (known as IMD).
>  
>
>
>
>
>  
>> -----Original Message-----
>> From: [hidden email]
>> [mailto:[hidden email]] On Behalf Of Julian, G4ILO
>> Sent: Friday, January 16, 2009 4:40 PM
>> To: [hidden email]
>> Subject: Re: [Elecraft] K3 pwr out on digimodes
>>
>>
>>
>>
>> Joe Subich, W4TV-4 wrote:
>>    
>>> This is your fundamental misunderstanding of ALC.  Traditional
>>> ALC does NOT (and can not) increase gain - it can only DECREASE
>>> gain to bring the grid/base/gate current down to the level at
>>> which the final amplifier is not being overdriven.  DSP based
>>> ALC is similar, it can only adjust the modulator (DSP) output
>>> to the level determined to be appropriate at the single point
>>> in the IF filter at which the power calibration (5W/50W) was
>>> made.  
>>>
>>> ALC (and power control) can not compensate for differences in
>>> gain across the passband - in other words, non-flat response.
>>> If you wanted to compensate for that non-flat response, it
>>> would be necessary to calibrate the power (5W/50W) at regular
>>> intervals across the passband and build a gain vs. frequency
>>> table.  The DSP would then need to measure the frequency of
>>> any narrow band modulation and adjust the assumed gain of the
>>> power control circuit based on the gain of the IF chain at
>>> that frequency.
>>>
>>>
>>>      
>> No, I do not misunderstand what ALC does. Perhaps I am just
>> not very good at explaining what I am trying to say. I
>> realize that ALC can only reduce the gain. That is why it is
>> necessary to set the audio drive level so that you get a
>> little bit of ALC across the whole passband range that you
>> want to use. Then, the ALC will be able to reduce the gain by
>> the necessary amount so that the RF output is constant across
>> that passband range.
>>
>>
>>
>>    
>>> The PSK31 operator must adjust his drive (mic gain or power
>>> control) with ANY transmitter if he changes his "subcarrier"
>>> frequency.  Even with conventional ALC if the drive is set
>>> to the ALC threshold when operating at the point of maximum
>>> gain, the output will be reduced when operating at other places
>>> in the passband (particularly when operating at the edges of
>>> the passand).
>>>
>>>      
>> No. This misses the whole point of the convenience of "click and call"
>> operating with programs like Digipan and HRD. I never
>> suggested that the
>> drive must be set to the ALC threshold when operating at the point of
>> maximum gain. It must be set so that you get some ALC across
>> the whole of
>> the passband you wish to use (probably corresponding to no
>> more than the
>> -3dB bandwidth points, beyond that and the steepness of the
>> filter slopes
>> would cause unacceptable distortion of even a narrowband
>> signal such as
>> PSK31.) Then you will achieve constant power across that
>> range, because the
>> ALC will be able to apply varying amounts of gain reduction
>> to achieve that
>> constant power level. The K2 can do this. Why can't the K3?
>>
>>
>>
>>    
>>> Try this experiment with your FT-817 ... set the power level
>>> to the maximum, turn off any compression, set your soundcard
>>> to 1700 Hz and the level for 15 Watts out of the FT-817 with
>>> NO ALC.  Now, without making any other changes measure the
>>> power output with a 200 Hz tone and a 3200 Hz tone.  Is it
>>> identical AT ALL THREE points?
>>>
>>>      
>> The FT-817 is not capable of 15W. But that is beside the
>> point. What would
>> this experiment prove? I know as well as you do that it would not be
>> identical. That's why I am saying that some form of level
>> control needs to
>> be in operation across the whole of the passband you wish to
>> use. A small
>> amount of ALC with a slow time constant will not harm the IMD
>> in PSK31 mode.
>> The K2 demonstrates this.
>>
>> In SSB mode the K3 ALC provides the level control I am
>> describing. If I turn
>> CMP off and switch the audio source to LINE IN it works
>> exactly as I expect
>> it to work in data modes (apart from not disabling EQ, which
>> is a different
>> issue.) In DATA A mode it does not do this, so you must
>> constantly tweak the
>> drive level whenever you change frequency if you wish to
>> transmit a specific
>> power level.
>>
>> If this is how the K3 is supposed to operate then I shall put
>> it up for sale
>> right away, because it is not something that people who use
>> data modes a lot
>> want to do. (Well, perhaps there are some, because if I have
>> learned nothing
>> else from my participation in ham forums like this, it is
>> that some people
>> actually like doing things the hard way.)
>>
>> -----
>> Julian, G4ILO. K2 #392  K3 #222.
>> http://www.g4ilo.com/ G4ILO's Shack  
>> http://www.ham-directory.com/ Ham
>> Directory    http://www.g4ilo.com/kcomm.html KComm for
>> Elecraft K2 and K3
>>    
>
>
>
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Peter,

I don't read your message before sending my "very long" mine but i
totally agree with you!
I can imagine that digimodes software generate  ± complex audio signals,
then we can
adjust this audio level to allow TX chain, including PA, to work in a
linear zone.
(I admit that we should be able to "anticipate" the maximum "peak"
level...)
Many years ago, i had a friend who was a fan in live music recording;
when possible,
he asked to the musicians to play just the loudest (fortissimo) part
before the public representation;
he adjust then his recorder, an old Revox, just before clipping; with
this procedure,
he was able to get the best dynamic with his "non professional" material.

2 cents more...

73 QRO fr Rudolf, HB9ARI

Peter Connors, F5VNB wrote:

> Joe Subich, W4TV wrote:
>  
>> Absolutely not!  When conventional rigs are generating ALC
>> their final amplifiers are ALREADY in compression and they
>> are generating IMD in the PSK31 signal.  
>>    
>
> A bit sweeping, Joe. Surely any sane engineer would put the onset of
> system ALC at a level *before* the PA goes into compression? (Such
> systems would not include grid current derived ALC from some class AB2
> PA of yesteryear.)
>
> To avoid the well-known problems of selecting suitable ALC time
> constants and loop gain, the K3 performs ALC at the DSP level: for any
> signal, SSB or PSK31, doesn't this demand a flat transmit passband in
> the system following the processing?
>
> Good thread!
>
> 73, Pete F5VNB/G4PLZ
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>  

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[Geoffrey Mackenzie-Kennedy]

Pete F5VNB wrote:

> Joe Subich, W4TV wrote:
>>
>> Absolutely not!  When conventional rigs are generating ALC
>> their final amplifiers are ALREADY in compression and they
>> are generating IMD in the PSK31 signal.
>
> A bit sweeping, Joe. Surely any sane engineer would put the onset of
> system ALC at a level *before* the PA goes into compression? (Such
> systems would not include grid current derived ALC from some class AB2
> PA of yesteryear.)

The problem with conventional closed loop ALC systems which sample at the PA
is of course that the PA, or the PA and the Driver, can be driven into
compression before the ALC system "catches up" and reduces drive, the point
Joe is making. While it is tempting to set the ALC threshold well below the
point at which PA compression takes place, the PA could still be driven into
compression during the ALC "catch up" period. Reducing the attack time
constant of the ALC system below some minimum can breed its own set of
problems.

Because this type of ALC system was first intended to be used as part of a
transmitter's protection system and not as a processor, one method of drive
level control frequently used samples at some point in the transmitter chain
and adjusts the gain of some stage AFTER the sample point. This method
ensures that there is no "catch up" problem. If used in a rig such as the K3
a suitable sample point would probably be just after the bandpass filters.

A parallel exists in receiver AGC systems.

73,
Geoff
GM4ESD



Peter Connors, F5VNB wrote:

> Joe Subich, W4TV wrote:
>>
>> Absolutely not!  When conventional rigs are generating ALC
>> their final amplifiers are ALREADY in compression and they
>> are generating IMD in the PSK31 signal.
>
> A bit sweeping, Joe. Surely any sane engineer would put the onset of
> system ALC at a level *before* the PA goes into compression? (Such
> systems would not include grid current derived ALC from some class AB2
> PA of yesteryear.)
>
> To avoid the well-known problems of selecting suitable ALC time
> constants and loop gain, the K3 performs ALC at the DSP level: for any
> signal, SSB or PSK31, doesn't this demand a flat transmit passband in
> the system following the processing?
>
> Good thread!
>
> 73, Pete F5VNB/G4PLZ
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[gm3sek]

Peter Connors, F5VNB wrote:

>>
>> Absolutely not!  When conventional rigs are generating ALC
>> their final amplifiers are ALREADY in compression and they
>> are generating IMD in the PSK31 signal.
>
>A bit sweeping, Joe. Surely any sane engineer would put the onset of
>system ALC at a level *before* the PA goes into compression? (Such
>systems would not include grid current derived ALC from some class AB2
>PA of yesteryear.)
>

A sane engineer surely would; but the huge majority of transceivers are
trying to use the ALC loop for too many different functions including
gain levelling between bands, peak limiting and also power control.

That is NOT how ALC was originally intended to work. The basic idea was
invented and refined by the Collins company more than 30 years ago, but
was intended only to handle *small, dynamic* variations in SSB output
power (mostly due to filtering a clipped SSB signal, which re-introduces
a  few dB of amplitude variation).

But ALC is also being used to achieve much larger output variations,
notably for the front-panel RF PWR control. The ability to vary the
output power all the way from 100W down to say 1W requires a very high
gain in the ALC loop. As well as producing instability and overshoots,
such a high loop gain ensures that any activation of the ALC will cause
hard limiting - with obvious consequences for IMD.

SM5BSZ has published a  strong critique of the way that most rigs MISuse
the ALC  loop. Instead of ALC helping to prevent IMD, it has become a
major potential *source* of IMD. SM5BSZ's criticisms are backed up  by
measurements on a range of different rigs:

Real life Dynamic Range of Modern Amateur Transceivers
http://www.sm5bsz.com/dynrange/dubus205/dubus205.htm

Speech Processing for SSB Transmitters
http://www.sm5bsz.com/dynrange/dubus405/dubus405.htm

The conclusion of SM5BSZ's papers is that rig designers need to stop
demanding the impossible from the dynamic ALC loop, and go back to the
original Collins concept. The big variations required for power control
are "static" settings that do not require a dynamic control loop. They
should be achieved by calibrating and then pre-setting the TX exciter
gain to give the required power level for each band. Then the ALC can
return to its original role of handling small dynamic variations, which
only requires a low-gain ALC loop with a much lower potential for
re-introducing IMD.

Does this sound familiar? It should, because the K3's power control and
ALC system has been very much influenced by the ideas from SM5BSZ and
originally from Collins.

Only Elecraft know the intimate details of how these concepts have been
implemented in DSP code, and the extent to which the K3's ALC differs
from traditional rigs. The bars on the K3's ALC display have been
programmed to show something very different from the ALC meter in a
traditional rig.

Bottom line: don't make assumptions about ALC behavior in the K3, based
on the widespread misuse of ALC by other rigs.



--

73 from Ian GM3SEK
http://www.ifwtech.co.uk/g3sek
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[Julian, G4ILO]

Ian White GM3SEK wrote

A sane engineer surely would; but the huge majority of transceivers are
trying to use the ALC loop for too many different functions including
gain levelling between bands, peak limiting and also power control.

[snip]

I seem to have read this before - perhaps in one of your RadCom columns?

Certainly I am making the assumption that ALC is used to control the power to the level the operator has set, and that a responsible PSK operator has chosen a level of perhaps half the rated maximum SSB output or even less. I think Joe may be thinking of how older radios used to work, where the power control varied the gain of the TX stages and ALC did only start to operate once the rated output of 100W or whatever had been reached.

Since we are talking about digimodes here, and since the K3 like most other modern radios has a separate mode setting for data modes, it is surely possible for the ALC to act more slowly for data than for SSB, because the amplitude of a data signal does not vary much, if at all, and the amount of gain reduction needed over the course of a transmission will probably be more or less constant once the initial level has been set.

Again, I must reiterate that this is what the K2 appears to do. At the start of the first data transmission of the day you would see the RF output level climb or fall until the selected value was reached, after which the radio appeared to remember the setting used so that subsequent transmissions always started at whatever power had been set.

[Joe Subich, W4TV-4]

Peter,

> Surely any sane engineer would put the onset of system ALC
> at a level *before* the PA goes into compression?

Not necessarily.  Some incidental compression in the final
amplifier is not necessarily a bad thing for CW or SSB
systems.  If the ALC is derived from something like grid
current that does not begin to appear until the amplifier
enters the compression range and includes a suitable time
constant the compression will be minimal and only occur
on an occasional peak.  

This form of ALC is suitable for CW, SSB and FSK operation
in that compression is of no consequence for CW and FSK
which suffer no IMD issues in even saturated operation
and of minimal consequence for SSB since the peaks are of
sufficiently brief duration and the compression is quite
mild (one or two dB).  

> To avoid the well-known problems of selecting suitable ALC
> time constants and loop gain, the K3 performs ALC at the DSP
> level: for any signal, SSB or PSK31, doesn't this demand a
> flat transmit passband in the system following the processing?  

Again, an absolutely flat passband is not necessary for a
CW, FSK (including AFSK) or SSB system.  With CW and FSK
the only concern is the system loss at a single frequency
(or a very narrow range of frequencies for AFSK).  The power
calibration process compensates for the loss at the single
frequency for CW and the rate of change is sufficiently low
that the compensation will be correct +/- a few Hertz of the
compensated frequency needed for narrow FSK.

With SSB, one or two dB of passband ripple is not an issue.
The passband of the human voice has far more "ripple" than
any moderate quality filter ... ten dB or more of high
frequency boost is often applied through equalization or
designed into microphone responses to increase intelligibility.

Again, these issues are only important when transceivers
designed for CW, FSK and SSB are used for modes (e.g.,
multiple tone digital modes) they were not designed to
handle or operated in a way (e.g., "click on the waterfall")
they were not designed to be operated.  

Multiple tone digital modes are essentially a full-time
multi-tone signal (e.g., two-, three- or more-tone IMD
test).  Successful operation of multi-tone and multi-
carrier digital systems requires that the passband be
absolutely flat across the operating range and that any
amplifier have no compression (amplifiers are generally
specified for the 1 dB compression level).  These are
criteria are significantly more stringent than normal
amateur equipment.  

There are two approaches for successful amateur digital
operation.  First: use digital modes that are narrow
compared to any "ripple" in the passband, operate the
power amplifiers at least 3 dB plus the peak to average
(crest factor) of the particular digital mode below their
SSB PEP rating and reset the power level for any change
in "subcarrier frequency" relative to the SSB dial
frequency.  Otherwise: utilize on frequency RF generation
(do not employ any intermediate frequency stages, mixers
and filters in generating the RF) and use a power amplifier
chain that is capable of much higher saturated power output.

73,

   ... Joe, W4TV
 

> -----Original Message-----
> From: [hidden email]
> [mailto:[hidden email]] On Behalf Of Peter
> Connors, F5VNB
> Sent: Saturday, January 17, 2009 4:34 AM
> To: [hidden email]
> Subject: Re: [Elecraft] K3 pwr out on digimodes
>
>
> Joe Subich, W4TV wrote:
> >
> > Absolutely not!  When conventional rigs are generating ALC
> > their final amplifiers are ALREADY in compression and they
> > are generating IMD in the PSK31 signal.  
>
> A bit sweeping, Joe. Surely any sane engineer would put the onset of
> system ALC at a level *before* the PA goes into compression? (Such
> systems would not include grid current derived ALC from some
> class AB2
> PA of yesteryear.)
>
> To avoid the well-known problems of selecting suitable ALC time
> constants and loop gain, the K3 performs ALC at the DSP
> level: for any
> signal, SSB or PSK31, doesn't this demand a flat transmit passband in
> the system following the processing?
>
> Good thread!
>
> 73, Pete F5VNB/G4PLZ _______________________________________________
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[Kok Chen]

On Jan 17, 2009, at 1/17    3:00 AM, hb9ari wrote:

> I can imagine that digimodes software generate  ± complex audio  
> signals,
> then we can adjust this audio level to allow TX chain, including  
> PA, to work in a
> linear zone.

Out of the common HF digital modes, I believe that BPSK31 and QPSK31  
are the most sensitive to nonlinearities.  RTTY, MFSK16, Olivia and  
Domino are M-ary FSK (M=2 for RTTY) and are less sensitive to  
nonlinearities since their crest factor is 1.  However, with poor  
linearity, you can no longer use envelope waveshaping to help reduce  
the FSK keying sidebands.

BPSK31 is differential phase shift modulation, with either a phase  
change of 180 degrees or no phase change, between successive bits.  
The envelope is also waveshaped so that during a non-zero phase  
transition, the amplitude is modulated in a sinusoidal manner.   The  
envelope modulation, what W4TV had called "predistortion," it is  
simply the waveshaping of the keyed signal with a sinudoidal window,  
easily achieved in the I and Q components of the PSK31 generator.

(A digression: a PSK31 signal can be easily generated by starting  
with baseband I and Q signals that swing between +1 and -1.  
Depending on whether you want 0 or 180 degrees (and the addition of  
90 and 270 degrees for QPSK31), you either keep I or Q, or both,  
unchanged and slew any component that changes between +1 and -1 with  
a half a period of a cosine function).

When no information is being transmitted in PSK31, it puts out the  
"idle" Varicode, which in PSK31 simply causes a 180 degree phase  
transition at every bit period.  The dibit representations in QPSK31  
are chosen so that an idle Varicode also puts out a 180-degree phase  
shift per dibit period.

When a PSK31 signals goes into idle, it therefore creates the  
familiar two tone signal, with only two sidebands (the center  
frequency is suppressed) that are 31.25 Hz apart. This causes a clean  
PSK31 to look like two parallel "rails" on a waterfall display.

Any deviation from a perfect sinusoidal modulating envelope will  
cause an idling PSK31 signal to develop higher order keying  
sidebands, just like what a two tone test signal does with voice mode  
SSB.  The keying sidebands at 2.f0-f1 and 2.f1-f0 are measured,  
compared with the "fundamentals" at f0 and f1 and this ratio is  
reported as the IMD number in PSK31.

An IMD somewhere around -28 dB is considered decent, with many good  
stations putting out signals that are cleaner than that.  People do  
differ in what they consider to be "good enough" until they get  
QRM'ed by a loud, but not so clean, signal :-).

With the 100 watts PA turned off, I have measured an IMD of -34 dB on  
my K3, running at 10 watts peak (average idle signal of 5 watts).

However, with the PA turned on and running above 25 watts average on  
idle (50 watts peak), the IMD from my K3 is only around a -24 dB.  
(IMHO, very mediocre.)  With the PA turned off, the K3's IMD also  
starts to rise slowly when power is decreased below 500 mW.

Please note that this was measured on my K3 that was built from a kit  
(SN 1432).  Your K3's performance may be different from mine.

For rough comparisons, my FT-1000MP running below 5 watts and with no  
ALC action whatsoever, is capable of putting out an IMD that is  
cleaner than -40 dB IMD both on the bench and with on-the-air tests.

I have also copied a local who was using QRP levels on his TS-2000  
with a tad better than -40 dB IMD.  The IMD of my FT-1000MP does  
degrade to -30 dB dB when I raise the average PSK31 power to 25  
watts, but still adequately clean.

A clipped modulating signal will create something like -11 dB IMD,  
and I have seen even be worse than that on the air -- I suspect that  
the signal was not just clipping but had overshoots and undershoots  
in the envelope.

73
Chen, W7AY



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[Joe Subich, W4TV-4]

> I think Joe may be thinking of how older radios used to work,
> where the power control varied the gain of the TX stages and
> ALC did only start to operate once the rated output of 100W
> or whatever had been reached.

Older (tube) radios did not have "power level controls."  They
were operated at their design power and ALC reduced the gain
to prevent overdriving the final amplifiers.  

Later radios added additional comparators to the ALC system
to sense absolute power in an earlier stage and adjust for a
relative power level but that is not ALC it is power control.

The K3 operates much like the older radios except that the
level is controlled at the output of the DSP (before the IF
filter) and not the grid of the final amplifier.  The ALC
adjusts the DSP so that when the power control is set for 5.0
or 50 Watts the output power is 5.0 or 50 Watts at one specific
frequency point in the SSB filter.  The DSP then adjusts its
(peak) output as needed to maintain the correct peak level
based on the assumption that the gain of the RF chain does
not change.  

The 5.0 and 50 watt calibration levels are significant ...
they are far enough below the compression point of the
power amplifier to allow for slight gain changes within
the IF passband.  

Again, you are simply asking the ALC and power control circuits
of the K3 to do something they can not, and were not designed
to do.  The RF chain is not flat across the IF passband; as
long as the technology uses mixers and narrow filters it will
never be flat.  If you want to be able to "click to tune"
without manually adjusting power levels get a Flex-5000 or
a transmitter that uses "on frequency" RF generation.  

73,

   ... Joe, W4TV
 

   

> -----Original Message-----
> From: [hidden email]
> [mailto:[hidden email]] On Behalf Of Julian, G4ILO
> Sent: Saturday, January 17, 2009 10:27 AM
> To: [hidden email]
> Subject: Re: [Elecraft] K3 pwr out on digimodes
>
>
>
>
>
> Ian White GM3SEK wrote:
> >
> >
> > A sane engineer surely would; but the huge majority of transceivers
> > are
> > trying to use the ALC loop for too many different functions
> including
> > gain levelling between bands, peak limiting and also power control.
> >
> > [snip]
> >
>
> I seem to have read this before - perhaps in one of your
> RadCom columns?
>
> Certainly I am making the assumption that ALC is used to
> control the power
> to the level the operator has set, and that a responsible PSK
> operator has
> chosen a level of perhaps half the rated maximum SSB output
> or even less. I
> think Joe may be thinking of how older radios used to work,
> where the power
> control varied the gain of the TX stages and ALC did only
> start to operate
> once the rated output of 100W or whatever had been reached.
>
> Since we are talking about digimodes here, and since the K3
> like most other
> modern radios has a separate mode setting for data modes, it is surely
> possible for the ALC to act more slowly for data than for
> SSB, because the
> amplitude of a data signal does not vary much, if at all, and
> the amount of
> gain reduction needed over the course of a transmission will
> probably be
> more or less constant once the initial level has been set.
>
> Again, I must reiterate that this is what the K2 appears to
> do. At the start
> of the first data transmission of the day you would see the
> RF output level
> climb or fall until the selected value was reached, after
> which the radio
> appeared to remember the setting used so that subsequent transmissions
> always started at whatever power had been set.
>
> -----
> Julian, G4ILO. K2 #392  K3 #222.
> http://www.g4ilo.com/ G4ILO's Shack  
> http://www.ham-directory.com/ Ham
> Directory    http://www.g4ilo.com/kcomm.html KComm for
> Elecraft K2 and K3
> --
> View this message in context:
> http://n2.nabble.com/K3-pwr-out-on-digimodes-tp2163956p2173365.html
> Sent from the Elecraft mailing list archive at Nabble.com.
>
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[Pete Connors F5VNB]

CONTENTS DELETED

The author has deleted this message.

[gm3sek]

Julian, G4ILO wrote:
>
>I seem to have read this before - perhaps in one of your RadCom
>columns?
>
Yes;  the first discussions were here, sometime around March 2008.

Then you saw it again in my May 2008 column in Radcom, although that
article was specifically about the deficiencies of "normal"
hard-limiting ALC (the K3 wasn't mentioned).

That would probably explain  this morning's feelings of deja vu twice
over.


--

73 from Ian GM3SEK         'In Practice' columnist for RadCom (RSGB)
http://www.ifwtech.co.uk/g3sek
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[Julian, G4ILO]

Joe Subich, W4TV-4 wrote

Older (tube) radios did not have "power level controls."  They
were operated at their design power and ALC reduced the gain
to prevent overdriving the final amplifiers.  

Later radios added additional comparators to the ALC system
to sense absolute power in an earlier stage and adjust for a
relative power level but that is not ALC it is power control.

OK. I'm not going to get into an argument over terminology. I can't even remember what the controls were called on radios I owned 35 years ago. In these days of SDR nothing works the way it did then anyway. But we still tend to use the same names for things that appear to provide similar functionality.

The K3 operates much like the older radios except that the
level is controlled at the output of the DSP (before the IF
filter) and not the grid of the final amplifier.  The ALC
adjusts the DSP so that when the power control is set for 5.0
or 50 Watts the output power is 5.0 or 50 Watts at one specific
frequency point in the SSB filter.  The DSP then adjusts its
(peak) output as needed to maintain the correct peak level
based on the assumption that the gain of the RF chain does
not change.

That does appear to be the case when DATA mode is selected, which is why I observed a 3dB output power variation over the range of passband that a PSK operator might operate in. But when I transmitted tones over a range of frequencies in SSB mode with a bit of ALC active, the output power measured level, which suggests to me that in SSB some gain control is being applied from after the IF filter.

Again, you are simply asking the ALC and power control circuits
of the K3 to do something they can not, and were not designed
to do.  The RF chain is not flat across the IF passband; as
long as the technology uses mixers and narrow filters it will
never be flat.  If you want to be able to "click to tune"
without manually adjusting power levels get a Flex-5000 or
a transmitter that uses "on frequency" RF generation.  

That may be the eventual outcome. However the K3 appears to behave as I would expect in SSB mode. So I am not convinced that I am expecting the impossible.