EV664

[CTMorton]

I have a classic Electro-Voice 664 microphone. Does anyone use one with
their K2 and if so, what did you have to do to get it to "play" properly.
73
Tom
K6CT

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[Don Wilhelm-3]

Tom,

I was able to find some info on the EV 664 mic and found it is capable of
being configured for 'high impedance' or 'low impedance', but I could find
no reference for how low is low nor how high is high, and I found no mention
of the output level.  You could just try the low impedance connection and
see how it performs.  If it produces peaks on the K2 LED display at (or
slightly below) the point where the RF output is indicated on CW (at the
same power level), it is producing sufficient output.

I found the info at http://bama.edebris.com/manuals/elvoice/ev-664 - the
quality is not the best, but it is readable - you will need to have a DjVu
Reader to open the file.

73,
Don W3FPR

> -----Original Message-----
>
> I have a classic Electro-Voice 664 microphone. Does anyone use one with
> their K2 and if so, what did you have to do to get it to "play" properly.
> 73
> Tom
> K6CT
>
>
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[Jim Brown-10]

On Wed, 24 Aug 2005 12:40:47 -0400, W3FPR - Don Wilhelm wrote:

>I was able to find some info on the EV 664 mic and found it is capable of
>being configured for 'high impedance' or 'low impedance', but I could find
>no reference for how low is low nor how high is high, and I found no mention
>of the output level.  You could just try the low impedance connection and
>see how it performs.  If it produces peaks on the K2 LED display at (or
>slightly below) the point where the RF output is indicated on CW (at the
>same power level), it is producing sufficient output.

I'm looking at an original data sheet for the 664! It does not tell me how high is
high, but it does say that low is 150 ohms. This is pretty much standard for
"pro"mics. I would guess that the "high impedance" output is probably on the order
of 10K-20K, but I'm only guessing.

Because I'm in the pro audio biz, I have a lot of good pro mics laying around, and
I use them in my ham station. As a result, I've taken the time to figure out how
to interface them well with my rigs.

I would do two things. First, I have modified the mic input circuit of my KSB2's
to raise the input Z and also raise the low frequency rolloff. This has the effect
of increasing the input senstivity of the KSB2 mic input, which is much lower than
is ideal. It also makes the audio a bit "punchier" by reducing the low frequency
content that would otherwise hit the peak limiter and reduce the audio level. The
low frequency rolloff will now be 250 Hz. When you're running QRP, every little
bit helps, and I would expect this to help audio punch by 3-4 dB (more than double
the power).

The mods are quite simple. First, change R14 from 1K to 2K. Second, change C34 to
0.22 uF. Third, add 2.2uF in series with R15.

If you can find some good audio input transformers (like the old UTC O-series or
A-series), you can pick up an additional 6dB of gain by using one that provides a
1:2 or 1:3 turns ratio. This will be most effective if you have done the above
mods to the input stage.

I've used pro mics with my K2, and these mods help a lot. Without the mods, the
K2's audio punch is a bit lacking. There are also some mods on the Elecraft
website that show how to increase the drive level quite a bit further through the
audio chain (after the peak limiter). I haven't done them, but they are VERY well
thought out, and should help even more. I probably will do them when I have time.  

BTW -- all of these mods are good no matter whether you're using pro mics or
communications mics.

Getting back to the original question. I suggest that you first do my mods, then
try the mic both high-Z and low-Z, and use whichever configuration provides the
best sounding (and hottest) audio.

When a mic is wired for a high-Z output, it has more output voltage, but through a
higher resistance source. If the mic is loaded by a low value resistor, the source
resistor and the load resistor form a voltage divider, and much of the higher
output voltage is lost.

And if anyone wants a copy of the data sheet, give me a FAX number and leave your
machine on. It's four pages.

73,

Jim Brown  K9YC
http://audiosystemsgroup.com



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[Jim Brown-10]

On Sat, 27 Aug 2005 10:26:57 -0500, Jim Brown wrote:

>The mods are quite simple. First, change R14 from 1K to 2K. Second,
>change C34 to 0.22 uF. Third, add 2.2uF in series with R15.

A correction to one of my suggested mods by Don, W3FPR, who understands the K2
in far more detail than I do, and emailed me to discuss it. The cap in series
with R15 is a bad idea, for several reasons, based on the control logic for
the compressor. If a cap were to be added there, it ought to be in parallel
with R15, not in series, and it ought to be smaller -- perhaps about 1 uF.

What both caps do is reduce modulation from breath pops and low frequency
noise in your shack. Popping and low frequency boost is a common problem with
directional mics used close up. Even in pro applications, we use a lot of low
cut on these mics, and it is even more important to do so in communications
circuits.

Jim


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[Stewart Baker]

Jim,

You might like to view this page of my site where I detail changes to KSB2
capacitor values to reduce the excessive LF response.

http://homepages.nildram.co.uk/~baker/K2_SSB_Mods_Overview.html

73
Stewart G3RXQ

On Sun, 28 Aug 2005 09:47:05 -0500, Jim Brown wrote:

> On Sat, 27 Aug 2005 10:26:57 -0500, Jim Brown wrote:
>
>> The mods are quite simple. First, change R14 from 1K to 2K. Second,
>> change C34 to 0.22 uF. Third, add 2.2uF in series with R15.
>>
> A correction to one of my suggested mods by Don, W3FPR, who understands the K2
> in far more detail than I do, and emailed me to discuss it. The cap in series
> with R15 is a bad idea, for several reasons, based on the control logic for
> the compressor. If a cap were to be added there, it ought to be in parallel
> with R15, not in series, and it ought to be smaller -- perhaps about 1 uF.
>
> What both caps do is reduce modulation from breath pops and low frequency
> noise in your shack. Popping and low frequency boost is a common problem with
> directional mics used close up. Even in pro applications, we use a lot of low
> cut on these mics, and it is even more important to do so in communications
> circuits.
>
> Jim
>
>
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[Francis Belliveau]

I am now confused by the existence of two different mod's with the same
goal.

It makes perfect sense to me that I should not be wasting Tx energy on audio
frequencies as low as 20Hz.  Beyond that I am somewhat in the dark.

Jim Brown and Stewart Baker have each suggested a modification to improve
this situation.

I am at a loss to know which, if either, I should implement.  I must admit
that I an very tempted to incorporate the indicated improvements.  However,
I would have no means of evaluating the results and therefore look to the
remainder of those on this list for further opinion.

Summary of the mod's are:

from Jim:
    Change R14 from 1K to 2K.
    Change C34 from 2.2hF to 0.22 uF.
    Add 1uF in parallel with R15

from Stewart:
    Change C34 from 2.2uF to 0.47uF
    Change C31 from 2.2uF to 1uF
    Change C20 from 0.33uF to 0.047uF

Is the situation severe enough to warrant an official mod for this?
Inquiring minds want to know.

Fran


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[Ron D'Eau Claire-2]

There seems to be two issues here. One is how the frequency response affects
the speech compressor, and the other is the frequency response that gets
transmitted.

The frequencies transmitted are controlled mainly by the SSB (OPT1) filter.
When properly aligned, the SSB filter (OPT1) bandpass is positioned to roll
off the low frequencies so they don't get transmitted. Properly positioned,
the filter also allows the high frequencies be transmitted for best
intelligibility. That is most important when using the stock, narrower
bandpass OPT1. If it's set for too much low frequency response, the audio
sounds "muddy" or "muffled". That's not so much because of the excess lows,
but because the filter is to narrow to pass the high audio frequencies that
are required to hear the sibilant sounds (such as the letter "s"). Those
highs get cut off by the OPT1 filter if it's position too close to the
carrier frequency.

In my experience, after roughing in the BFO setting with Spectrogram, final
adjustment requires listening to one's own signal and making minor tweaks to
the BFO frequency using CAL FIL as recommended in the SSB manual. My K2
sounds best with the OPT1 filter rolling off below 400 Hz and very steeply
below 300 Hz. That fits with "good Ham practice" as I've seen it applied for
many years. Even the old AM rigs rolled off frequencies below 200 or 300 Hz.
My voice has a lot of low-frequency drone in it and a little more aggressive
roll-off at the low-frequency end works best for me.

I tried my K2 using a "hi-fi" mic designed for music and found that it was
not as "crisp" or nice-sounding as my electret. I didn't pursue it further
because it wasn't an issue to me.  

I've been following this thread because there seems to be an issue with how
the speech amp in the K2 is reacting to different mics. I did all of my
testing with the compressor off. I'm wondering if the difference is just a
difference in impedances, or whether something else is going on.

Ron AC7AC


-----Original Message-----
From: [hidden email]
[mailto:[hidden email]] On Behalf Of Francis Belliveau
Sent: Sunday, August 28, 2005 10:19 AM
To: Elecraft List
Subject: [Elecraft] K2 SSB Mic Input Frequency response?


I am now confused by the existence of two different mod's with the same
goal.

It makes perfect sense to me that I should not be wasting Tx energy on audio
frequencies as low as 20Hz.  Beyond that I am somewhat in the dark.

Jim Brown and Stewart Baker have each suggested a modification to improve
this situation.

I am at a loss to know which, if either, I should implement.  I must admit
that I an very tempted to incorporate the indicated improvements.  However,
I would have no means of evaluating the results and therefore look to the
remainder of those on this list for further opinion.

Summary of the mod's are:

from Jim:
    Change R14 from 1K to 2K.
    Change C34 from 2.2hF to 0.22 uF.
    Add 1uF in parallel with R15

from Stewart:
    Change C34 from 2.2uF to 0.47uF
    Change C31 from 2.2uF to 1uF
    Change C20 from 0.33uF to 0.047uF

Is the situation severe enough to warrant an official mod for this?
Inquiring minds want to know.

Fran

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[JEAN-FRANCOIS MENARD-3]

I'm also agree with Fran, at least an explanation from Elecraft about  
this mod would be really appreciated.

Le 05-08-28 à 13:18, Francis Belliveau a écrit :

> I am now confused by the existence of two different mod's with the  
> same
> goal.
>
> It makes perfect sense to me that I should not be wasting Tx energy  
> on audio
> frequencies as low as 20Hz.  Beyond that I am somewhat in the dark.
>
> Jim Brown and Stewart Baker have each suggested a modification to  
> improve
> this situation.
>
> I am at a loss to know which, if either, I should implement.  I  
> must admit
> that I an very tempted to incorporate the indicated improvements.  
> However,
> I would have no means of evaluating the results and therefore look  
> to the
> remainder of those on this list for further opinion.
>
> Summary of the mod's are:
>
> from Jim:
>     Change R14 from 1K to 2K.
>     Change C34 from 2.2hF to 0.22 uF.
>     Add 1uF in parallel with R15
>
> from Stewart:
>     Change C34 from 2.2uF to 0.47uF
>     Change C31 from 2.2uF to 1uF
>     Change C20 from 0.33uF to 0.047uF
>
> Is the situation severe enough to warrant an official mod for this?
> Inquiring minds want to know.
>
> Fran
>
>
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>

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[Jim Brown-10]

On Sun, 28 Aug 2005 13:18:35 -0400, Francis Belliveau wrote:

>I am now confused by the existence of two different mod's with the same
>goal.

>It makes perfect sense to me that I should not be wasting Tx energy on audio
>frequencies as low as 20Hz.  Beyond that I am somewhat in the dark.

>Jim Brown and Stewart Baker have each suggested a modification to improve
>this situation.

>I am at a loss to know which, if either, I should implement.  I must admit
>that I an very tempted to incorporate the indicated improvements.  However,
>I would have no means of evaluating the results and therefore look to the
>remainder of those on this list for further opinion.

>Summary of the mod's are:

>from Jim:
>    Change R14 from 1K to 2K.
>    Change C34 from 2.2hF to 0.22 uF.
>    Add 1uF in parallel with R15

>from Stewart:
>    Change C34 from 2.2uF to 0.47uF
>    Change C31 from 2.2uF to 1uF
>    Change C20 from 0.33uF to 0.047uF

Differences:  I'm doing more low cut with C34 than Stewart is, but Stewart is
doing more low cut with C31, which I haven't changed. Both mods are good, and are
additive. At some point we go too far and the audio gets too thin.

Since I don't know anything about the details of U3 (the comp/limiter chip), I
haven't messed with C31. If it is reducing the LF compression, it is probably a
good thing. Perhaps Stewart can comment.

>Is the situation severe enough to warrant an official mod for this?

Severe is the wrong word. What Stewart and I and KI6WX have done is tweak a
design that is lacking in gain and has too much LF response. Yes, these mods
SHOULD be integrated into the K2 standard production.

On Sun, 28 Aug 2005 11:18:20 -0700, Ron D'Eau Claire wrote:

>I'm wondering if the difference is just a
>difference in impedances, or whether something else is going on.

Both. The impedance of the mic input circuit is lower than it needs to be, and
the GAIN of the entire audio chain is a bit low. One of KI6WX's mods addresses
the gain issue AFTER the comp/limiter. The ones that Stewart and I are talking
about are BEFORE the comp/limiter. Both are good sets of mods.

In general, the Z of the input stage should NOT load the mic -- that is, Z in
should be 10X the Z of the mic. Pro mics are typically 150-250 ohms, and have
relatively low voltage output. The K2 input Z is roughly 1K, which is not bad for
pro mics, and the loading is minimal. BUT -- since the gain is low, you need a
transformer to step up the mic to get more voltage to the input stage. When you
do that, the turns ratio increases the voltage, BUT it also makes the mic look
like a much higher Z -- the mic Z is multiplied by the SQUARE of the turns ratio.
So a 1:3 step up gets you 3x the voltage (9.4 dB) but the 150 ohm mic now looks
like 1.35K, which forms a voltage divider with the input stage and causes much of
that 9.4 dB to be lost. Changing R14 to 2K gets most of that gain back. Going to
3.3K or 4.7K would give you another couple of dB.

The reason that good pro mics sound dull with ham transmitters is that for some
stupid reason, an international standards body long ago established an HF rolloff
for communications transmitters that is both too low and too severe -- -6dB at
2.6 kHz. A mic with a smooth, flat response will sound dull and muffled with that
kind of transmitter response. Rather than change the dumb standard, mic
manufacturers began building all their mics with a 6 dB peak between 3 and 4 kHz
to compensate. You can see this is their data sheets. The resulting response
approaches what it should have been in the first place -- flat to about 3 or 3.5
kHz, then rolled off sharply.

So when you use a communications mic, like a Shure 444 or the wildly overpriced
Heil mics, you're simply compensating for that standard that was wrong in the
first place!  All of these mics have some variation of that peaked response in
the 2-4 kHz region. Plug them into a good sound system and they sound awful.
Should that standard be changed?  Of course. But that's like trying to change the
course of a nation's foreign policy -- no one wants to admit they were wrong, and
too many people/companies have a vested interest in things remaining the same.

Yes, mis-alignment of the crystal filters and IF can screw up the overall
response too, but that is in addition to these purely audio issues.

Jim Brown  K9YC


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[Don Wilhelm-3]

Fran,

I will not say that one is better than the other, but Jim's change only
deals with the mic input itself, while Stewart's change also deals with the
internal behavior of the Speech Compressor chip by changing C31 and resricts
the low frequency fed to the Balanced Modulator by the Speech Compressor
chip by reducing the value of C20.

Both agree that the value of the input capacitor C34 should be reduced.

Jim's change in the R14 resistor value may provide a better match for the
pro-audio mics that he is using, and as such may be highly dependent on the
mic actually in use.  Changing R14 to a higher value has been a favored
change for those folks using low output mics (such as the Heil HC4/5 element
mics) for some long time now - R14 is usually changed to somewhere in the
range of 5k to 10k for use with the Heil elements and no additional
amplification.

Your mileage may vary, and is likely dependent on your particular mic.  In
the old 'spirit of ham radio', a bit of experimental work may be a worthy
endeavor, but then, if you don't have the tools to properly evaluate your
efforts, it may be better to leave the design 'stock' - not everyone's mods
belong on all K2s - special situations do exist, and they are discussed here
too.

73,
Don W3FPR

> -----Original Message-----
>
> I am now confused by the existence of two different mod's with the same
> goal.
>
> It makes perfect sense to me that I should not be wasting Tx
> energy on audio
> frequencies as low as 20Hz.  Beyond that I am somewhat in the dark.
>
> Jim Brown and Stewart Baker have each suggested a modification to improve
> this situation.
>
> I am at a loss to know which, if either, I should implement.  I must admit
> that I an very tempted to incorporate the indicated improvements.
>  However,
> I would have no means of evaluating the results and therefore look to the
> remainder of those on this list for further opinion.
>
> Summary of the mod's are:
>
> from Jim:
>     Change R14 from 1K to 2K.
>     Change C34 from 2.2hF to 0.22 uF.
>     Add 1uF in parallel with R15
>
> from Stewart:
>     Change C34 from 2.2uF to 0.47uF
>     Change C31 from 2.2uF to 1uF
>     Change C20 from 0.33uF to 0.047uF
>
> Is the situation severe enough to warrant an official mod for this?
> Inquiring minds want to know.
>
> Fran
>

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[John Magliacane]

--- Francis Belliveau <[hidden email]> wrote:

> I am now confused by the existence of two different mod's with the same
> goal.

There may be more than just two...

        http://www.qsl.net/kd2bd/ksb2.html

It looks like many of us have been down the same road.


73, de John, KD2BD


Visit John on the Web at:

        http://kd2bd.ham.org/
.
.
.
.

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[Ron D'Eau Claire-2]

Jim wrote:
So when you use a communications mic, like a Shure 444 or the wildly
overpriced
Heil mics, you're simply compensating for that standard that was wrong in
the
first place!  All of these mics have some variation of that peaked response
in
the 2-4 kHz region.

----------------------
As far as the audio bandpass range is concerned, Ham practice followed
commercial practice. The use of an upper frequency of about 2.5 kHz for
communications goes way back to WWII at least when AM transmitters were
designed with a bandwidth of 5 KHz (2.5 KHz per sideband). In the 1960's we
carefully rolled off telephone bandwidths for conventional lines at 2.5 KHz.
I haven't worked in the telephone industry since, but I believe that's still
the upper limit due to the use of frequencies right above 2.5 KHz for
signaling. (Wideband lines were used for broadcasters and other services of
course, and AM broadcasters were allowed exactly twice the bandwidth - 10
KHz - for music.)

Yes, I'm familiar with the standard "communications" mics. A classic example
is the D-104. It has a characteristic that puts the output at 500 Hz a full
12 dB below the output at 2.5 kHz. It continues to slope to more than 20 dB
below the peak at 100 Hz. I have never heard the claim that response was
intended to overcome the upper frequency limit of the transmitter, but that
extensive research had shown that a rising characteristic within the normal
communications bandwidth of about 300 to 2500 Hz produced the highest level
of intelligibility.

That rising characteristic (high-frequency emphasis) is usually provided by
the mic, as you pointed out, while the low frequency cut was traditionally
provided by the choice of capacitors in the audio amplifiers.

In the K2, the low frequencies are suppressed very effectively by the OPT1
filter if it's properly aligned.

Are you suggesting that those low frequencies are somehow being distorted or
causing other problems in the audio I.C. in the K2 before they get to the
filter?

Ron AC7AC

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[Ron D'Eau Claire-2]

John, KD2BD wrote:

There may be more than just two...

        http://www.qsl.net/kd2bd/ksb2.html

It looks like many of us have been down the same road.

----------------------------------------------------

As I understand the evolution (having followed it since I built S/N 1289 in
2000):

1) The K2 was designed with a minimum-bandwidth SSB filter. The Filter
provided an audio bandwidth of barely 2 KHz. The idea was that narrower
bandwidths provided the best signal/noise ratio - a very important issue for
QRP operation especially.

2) It was discovered that a number of early SSB filters were significantly
narrower than 2kHz, severely impacting their performance on SSB. The problem
was traced to too much variation in the characteristics of the crystals
being received from the Elecraft vendor. Elecraft changed their
specifications and matching process to ensure adequate consistency. They
also replaced crystals in anyone's K2 that exhibited poor bandwidth.

3) John, KI6WX, developed several different mods for the OPT1 filter which
allowed widening the bandwidth of the filters to more closely approximate
what other Ham SSB rigs use. This was especially popular after the KPA100
was released, allowing higher power for the non-QRP operators. These mod
kits consist of different sets of coupling capacitors that are used with the
original Elecraft crystals (it's important to use Elecraft crystals that
were shipped after the specifications were tightened - see 2 above). The
information is on the Elecraft web site. Kits of capacitors are also
available from Elecraft if the builder doesn't have the right ones handy.

4) The issue of audio gain has come up from time to time since the K2 was
released. The gain is actually a composite of audio and RF gain to produce
10 watts or 100 watts PEP (it is not recommended to exceed those power
levels on SSB in any event because the harmonic distortion levels increase
quickly). That's why a QRP K2 often shows just barely a full 10 watts output
on 10 meters, while it easily produces more than 10 watts on, say, 40
meters. Monitoring the transmit ALC action one sees that often only one bar
barely flickers on 10 meters while several bars flash on 40. What's
happening is that the transmitter RF strip has lower gain on 10, so more
output from the balanced modulator is needed there. Some mics don't drive
the audio chip hard enough to produce full output on the bands were the
transmitter strip RF gain is lower. Again, John, KI6WX, and others came up
with mods to improve the levels for full output with mics that were marginal
when used with the K2. Other operators used external amplifiers with
lower-level mics. Those mods are also documented on the Elecraft web site.

5) Now it appears that there are some questions about the audio QUALITY
produced by the audio portion of the K2 that Jim, K9YC, is exploring.

Elecraft has always encouraged builders to tinker and improve on the K2.

Many of the improvements to the K2 design that have been made over the years
have come from just such efforts, such as the filter mods by John KI6WX.
(Actually John has been responsible for several other significant mods as
well.)

What Jim, K9YC, and others are doing pursuing better audio quality is in the
best tradition of Elecraft builders and tinkerers.

Ron AC7AC

 

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[John Magliacane]

Hi Ron.

> Are you suggesting that those low frequencies are somehow being distorted or
> causing other problems in the audio I.C. in the K2 before they get to the
> filter?

I would say: YES.

Ideally, the audio frequency response prior to the compressor should
closely match the IF bandwidth of the crystal filter.   Here's why:

Excessive low-frequency response prior to the compressor forces the
compressor to react to audio energy that has no chance of ever making
it through the crystal filter.  If the audio response is too wide, then
the compressor will react to the extraneous audio energy, essentially
"desensing" the balanced modulator and the stages that follow.

I would hazard a guess that this "desense" might even produce a
perception of low or "muffled" audio in some cases when sufficient
gain actually exists.

Since we are working with such a small audio bandwidth, the tighter
the control of the audio response prior to the compressor, the less
variation there will be between different microphones.


73, de John, KD2BD


Visit John on the Web at:

        http://kd2bd.ham.org/
.
.
.
.


               
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[Jim Brown-10]

On Sun, 28 Aug 2005 12:59:40 -0700, Ron D'Eau Claire wrote:

>Are you suggesting that those low frequencies are somehow being distorted
>or causing other problems in the audio I.C. in the K2 before they get to
>the filter?

Distorted no. Causing problems -- well, sort of. We don't transmit that LF
audio (below the bandwidth of the xtal filters), but it does hit the
comp/limiter and cause it to reduce the gain on LF peaks (which also reduces
total modulation). So the result is less than ideal operation of the
comp/limiter, which in turn results in lower average modulation and less
audio punch. So the purpose of moving the low-cut up in frequency is simply
to remove that LF energy so that it doesn't hit the comp/lim chip.

Think of it this way. Speech energy is distributed between roughly 100 Hz
and 8 kHz, but the 1 kHz, 2 kHz, and 4 kHz octave bands provide the greatest
speech intelligibility. 500 Hz provides a bit, so does 8 kHz. the 250 Hz
octave band and below provides virtually none -- BUT they consume a lot of
power if we transmit them. We cannot transmit any of the 4 kHz or 8 kHz
octave bands. So it makes sense not to waste any of our limited transmitter
power on that energy below about 300 Hz -- it doesn't help us land that DX,
or get through noise. All it does is give more body to our voices. So it is
simply a matter of wasted RF power!

>AM broadcasters were allowed exactly twice the bandwidth - 10 KHz - for
music.)

Actually, AM broadcasting in the old days (before about 15 years ago) was
permitted to go to 15 kHz AUDIO bandwidth, which is 30 kHz occupied
bandwidth. Current practice is 10 kHz audio, 20 kHz occupied bandwidth, but
I don't recall if that was an actual rules change.

The 2.6 kHz upper limit on SSB audio is just plain stupid. Yes, ham radio
followed commercial practice. But that doesn't make it GOOD practice. :)  
Even POTS (plain old telco lines) goes above 3 kHz before it drops like a
brick wall!  Telco bandwidth is roughly 2.6-2.8 kHz, but it's 500 Hz to over
3 kHz, and it's not 6 dB down at those points!

73,

Jim K9YC


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[k6mr]

I know I've read this somewhere, but searching the archives didn't bring
anything to light.

Both of my K2-100s have birdies around 14.068.  They pulse on/off, and
change frequency slightly when you turn the power knob to activate the
KPA-100.  I'm sure I read somewhere what to do to move these out of the
band, but no luck in my search.  They are some sort of mixing harmonic
since they change frequency about 4 times faster than the tuning knob
actually tunes.

Anyone remember what to do?

tnx,

Ken  K6MR

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