[Elecraft] W2IHY 8 band equalizer and EQ Plus

[Ron Gould]

Has anyone tried either of these products out on a K3 and care to share their results compared to using the built in 8 band equalizer built into the K3. I am not interested in any comments by others who have not actually tried these out. If you would like to take this discussion off list please contact me directly. If someone has these products and is willing to set up a schedule on the air all the better. Thanks

[Jim Brown]

Post by Ron Gould
Has anyone tried either of these products out on a K3 and care to share
their results compared to using the built in 8 band equalizer built into
the K3.

As a pro audio guy who has used hundreds of equalizers, I can tell you that
the TX and RX in the K3 are all you need, and that adding anything outboard
would be a waste of money.

73,

Jim K9YC

[Bob McGraw - K4TAX]

I too agree with Jim on this point. Myself, being a retired pro audio guy
and used lots of microphones and EQ's and processors, I've not seen the need
for EQ on any current production or model ham radios. Take something made
back in the 50's or 60's or 70's and they likley could stand some help.
Even today should one use a lousy mike on a new radio and then add
compression, gating and EQ and you'll have a lousy sounding signal with lots
of compression, gating and EQ. No amount of EQ will make the average hams'
voice sound like Don Pardo.

Microphone and microphone technique is the real secret to good sounding
audio. Of course one must have the radio adjusted properly and thus not
"all knobs to the right" to make the meter move higher.

73
Bob, K4TAX


----- Original Message -----
From: "Jim Brown" <jim at audiosystemsgroup.com>
To: <Elecraft at mailman.qth.net>
Sent: Sunday, April 25, 2010 8:28 PM
Subject: Re: [Elecraft] W2IHY 8 band equalizer and EQ Plus

Post by Jim Brown

Post by Ron Gould
Has anyone tried either of these products out on a K3 and care to share
their results compared to using the built in 8 band equalizer built into
the K3.

As a pro audio guy who has used hundreds of equalizers, I can tell you that
the TX and RX in the K3 are all you need, and that adding anything outboard
would be a waste of money.
73,
Jim K9YC
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[Gary Gregory]

The only EQ Plus and W2IHY equipment I have heard on air to date have been
little short of dreadful to my ears....But that's MY ears, and they are the
only ones I have and the brains DSP is most likely not that crash hot either
at my age.

Having said that, Jim has hit the nail on the head, and he should know, of
all people.

I listened to a guy with an old Collins and the W2IHY etc...yep it sounded
good..but it sure was wide let me tell you.

Oh well, YMMV but I agree with Jim and Hactor et al, save your money, if you
have the gear, then sell it off I guess.

73's
Gary - VK4FD

Post by Bob McGraw - K4TAX
I too agree with Jim on this point. Myself, being a retired pro audio guy
and used lots of microphones and EQ's and processors, I've not seen the need
for EQ on any current production or model ham radios. Take something made
back in the 50's or 60's or 70's and they likley could stand some help.
Even today should one use a lousy mike on a new radio and then add
compression, gating and EQ and you'll have a lousy sounding signal with lots
of compression, gating and EQ. No amount of EQ will make the average hams'
voice sound like Don Pardo.
Microphone and microphone technique is the real secret to good sounding
audio. Of course one must have the radio adjusted properly and thus not
"all knobs to the right" to make the meter move higher.
73
Bob, K4TAX
----- Original Message -----
From: "Jim Brown" <jim at audiosystemsgroup.com>
To: <Elecraft at mailman.qth.net>
Sent: Sunday, April 25, 2010 8:28 PM
Subject: Re: [Elecraft] W2IHY 8 band equalizer and EQ Plus

Post by Jim Brown

Post by Ron Gould
Has anyone tried either of these products out on a K3 and care to share
their results compared to using the built in 8 band equalizer built into
the K3.

As a pro audio guy who has used hundreds of equalizers, I can tell you that
the TX and RX in the K3 are all you need, and that adding anything outboard
would be a waste of money.
73,
Jim K9YC
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[Andy Wood]

I have to agree strongly with Jim - for me, the K3 does not need any form of
external processing or "extras". I receive an unusually high number of
unsolicited comments about the quality of my audio. In fact, here is the
latest email I have received about it after working a few NA contacts on the
75/80m DX window:

--------------------------------
Hi Andy,

I was talking to David, W6ANR today he informed me you are running K3
transceiver and you had the best audio he has ever heard from any K3. So I
am asking if you could give a run down on all your setting as you have them
on your radio that pertains to Audio so I can help my neighbor Bob, W9KNI
get his dialed in. His sounds lousy and he has always been a CW man but has
now been enjoying using the mic instead of the paddle. He has had comments
that he need to dial in his audio so I told him I would help. SO maybe with
all your setting etc I can have a good place to start as I am not to
familiar with that radio.

I hope your summer was a good for you? Here we are anxious for spring to
arrive so I can get out and do some real work out in the yard and get the
veggie garden started.


Best 73,


Rich K7ZV
--------------------------------

Here is my reply to Rich:

Hi Rich,

Great to hear from you - it has been a long time since we have worked on
80/75m. I haven't been very active lately, due to work and family
commitments.

I hope I am not going to disappoint anyone when I reveal my audio setup on
the K3 as it is far from spectacular..hi hi. The fist mic I was using is a
five dollar CB type that you could find at any hamfest (no kidding!). I have
not adjusted the TX equalizer so it is still at the factory default (all
"flat"). I do use the compressor, but only allow it to indicate between
about 5-10dB on the meter. I also always make sure the ALC bargraph
indication is well below the upper limit. For what it's worth, my mic gain
is set to 23 and the compression setting is at 19.

Not very fancy is it? I found the K3 to work well "out of the box". As the
old saying goes "if it ain't broke, don't fix it".
--------------------------------

I think this speaks for itself - nothing else required!

Andy VK4KY

[Bill NY9H]

ins to Audio so I can help my neighbor Bob, W9KNI get his dialed in.
His sounds lousy and he

if you want a handheld for a K3,,,or K2 or icom...
check out
:http://www.shure.com/ProAudio/Products/WiredMicrophones/us_pro_527C_content

you will notice that this mic has a rising hi end response,,,
popularized by Shure Bros...(not Heil)
for articulation,,,,and works great with ssb,,,, not flat ,,,

You could drive a car over the case and not hurt it ,,,
the PTT switch will last longer that all of us,,,,not a mini microswich
The cord, more rubber than plastic, isn't so stiff or thick it will
pull that radio off the shelf.... but close. The 527 requires some
dc, the 526 is straight dynamic,,,,

while it was made in US now it's made in america,,,,, mexico I think..

anyway , i've got three one wired for k3 , two for K2/ICOM......

sold at all the usual suspect places,,,, and cheaper than the piece
of junk from ...

As far as a great base mic.,,, just about anything will do,
especially if it's an electret , as you can eq it to sound like a
telefunken U-47 ribbon, if you wish...

I also have a sennheiser headset...HMD..280, with the d meaning the
boom mic is dynamic... a sennheiser HME would be an electret.
Found mine at dayton, as an AV lab castaway... ebay is another
source as new is too much$.
No longer have external eqs here.

I thought any noise gate issues were worked out by lyle.

bill

[Hector Padron]

"Has anyone tried either of these products out on a K3 and care to share
their results compared to using the built in 8 band equalizer built into
the K3".

Even I don't have those boxes anymore I can tell that about a year and a half ago when I sold my ProIII to replace it for my actual K3,I kept the W2IHY 8 bands EQ and the companion EQ Plus which I used together with the ProIII.
When I received the K3 having its builtin EQ flat I tried those two W2IHY boxes and had no sucess at all,only noise and distorsion was what I got,I ended up selling them up.
As Jim said,the K3 has an excellent 8 bands TX EQ that will allow you to EQ any mic,also it has a builtin noise gate that works great,save your money and don't buy them.
?
AD4C
?

"For a refined ham it is compulsory to own a k3"

--- On Mon, 4/26/10, Jim Brown <jim at audiosystemsgroup.com> wrote:


From: Jim Brown <jim at audiosystemsgroup.com>
Subject: Re: [Elecraft] W2IHY 8 band equalizer and EQ Plus
To: "Elecraft at mailman.qth.net" <Elecraft at mailman.qth.net>
Date: Monday, April 26, 2010, 1:28 AM

Has anyone tried either of these products out on a K3 and care to share
their results compared to using the built in 8 band equalizer built into
the K3.

As a pro audio guy who has used hundreds of equalizers, I can tell you that
the TX and RX in the K3 are all you need, and that adding anything outboard
would be a waste of money.

73,

Jim K9YC


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[Lance Collister]

Hello Ron,

I use a W2IHY 2 band equalizer and noise gate with my K3 and am VERY PLEASED with
the performance. I highly recommend it!

I don't use it for trying to customize my transmit audio fidelity or anything - I
think you could do that very successfully with the K3 adjustments, if you wanted
to. As a point in fact, I have never changed my K3 audio output settings at all.

The reason I use the W2IHY unit is for the excellent noise gate that it provides.
It very effectively cuts out the background blower noise, and I get excellent on
the air reports with it. When I acquired my K3, I tried to use the noise gate
built into the K3 so I could keep the W2IHY unit connected to my old IC746.
However, I could not seem to adjust the K3 noise gate to avoid getting reports of
audio distortion :-( So, reluctantly, I had to disable the noise gate in the K3
and return to the external W2IHY unit.

Best wishes for success! GL and VY 73, Lance

Post by Ron Gould
Has anyone tried either of these products out on a K3 and care to share their
results compared to using the built in 8 band equalizer built into the K3. I
am not interested in any comments by others who have not actually tried these
out. If you would like to take this discussion off list please contact me
directly. If someone has these products and is willing to set up a schedule on
the air all the better. Thanks
__________________________________

--
Lance Collister, W7GJ (ex: WN3GPL, WA3GPL, WA1JXN, WA1JXN/C6A, ZF2OC/ZF8, E51SIX)
P.O. Box 73
Frenchtown, MT 59834 USA
QTH: DN27UB
TEL: [phone] URL: http://www.bigskyspaces.com/w7gj
LIVE MESSENGER CHAT: w7gj at hotmail.com
2m DXCC #11, 6m DXCC #815

Interested in 6m EME? Ask me about subscribing to the Magic Band EME email!
http://6meme.com/mailman/listinfo/magic_6meme.com

[Jim Brown]

Post by Lance Collister
It very effectively cuts out the background blower noise

Some of the major causes of audible background noise are 1) working too
far from the mic; 2) running the mic gain too high; 3) using too much
compression/processing; and 4) not rolling off the low frequencies.

In a noisy environment, it always helps to work close to the mic. It is
ALWAYS good practice to use the minimum mic gain needed to get good
modulation, use no more than about 10dB of compression/processing, and
roll off the low frequency content. It's good engineering practice for the
highest quality broadcast stations, and it's good practice for ham radio.

Indeed, the only difference between what's right for broadcasting and for
ham radio is WHERE to cut the low end and HOW MUCH money to spend on
compression/processing. Many years ago, I sold processing systems for
broadcast stations that cost upwards of $10K in today's dollars, and I
helped the chief engineers of those stations adjust them. I suspect that
W8JI and K4TAX have similar experience. Before I spent ANY money on an
outboard box for a ham rig, I would first follow all of those elements of
good engineering practice.

73,

Jim Brown K9YC

[ab2tc]

On the K3 what would be the approximate setting for 10dB compression?

AB2TC - Knut

<snip>
In a noisy environment, it always helps to work close to the mic. It is
ALWAYS good practice to use the minimum mic gain needed to get good
modulation, use no more than about 10dB of compression/processing, and
roll off the low frequency content. It's good engineering practice for the
highest quality broadcast stations, and it's good practice for ham radio.
<snip again>

[Guy Olinger K2AV]

Page 26 of D6 owner's manual, text states that the number displayed is
the approximate compression in dB. Or 10 is 10.

73, Guy.

Post by ab2tc
On the K3 what would be the approximate setting for 10dB compression?
AB2TC - Knut

<snip>
In a noisy environment, it always helps to work close to the mic. It is
ALWAYS good practice to use the minimum mic gain needed to get good
modulation, use no more than about 10dB of compression/processing, and
roll off the low frequency content. It's good engineering practice for the
highest quality broadcast stations, and it's good practice for ham radio.
<snip again>

--
View this message in context: http://elecraft.365791.n2.nabble.com/W2IHY-8-band-equalizer-and-EQ-Plus-tp4959845p4964331.html
Sent from the [K3] mailing list archive at Nabble.com.
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[Robert Mcgraw]

And do remember, more compression brings up background noise between
words by the amount of the compression value. Hence with 10 dB of
compression, a background noise level that would normally produce and
output of 1 watt will then sound like 10 watts of noise.

73
Bob, K4TAX

Post by ab2tc
On the K3 what would be the approximate setting for 10dB compression?
AB2TC - Knut

<snip>
In a noisy environment, it always helps to work close to the mic. It is
ALWAYS good practice to use the minimum mic gain needed to get good
modulation, use no more than about 10dB of compression/processing, and
roll off the low frequency content. It's good engineering practice for the
highest quality broadcast stations, and it's good practice for ham radio.
<snip again>

--
http://elecraft.365791.n2.nabble.com/W2IHY-8-band-equalizer-and-EQ-Plus-tp4959845p4964331.html
Sent from the [K3] mailing list archive at Nabble.com.
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[Robert Mcgraw]

I said it before and written it before, good audio has to do with a good
mike and GOOD MIKE TECHNIQUE. The best mike money will buy and poor mike
technique will result in poor audio.

We use noise gates in the professional audio studio to create special
effects.

The noise gate used on ham radio to eliminate blower noise is a false
since of secuirty for when the gate opens, along comes the noise with the
voice. It only chops the noise in between words. It is still there with
the voice.

73
Bob, K4TAX

Post by Jim Brown

Post by Lance Collister
It very effectively cuts out the background blower noise

Some of the major causes of audible background noise are 1) working too
far from the mic; 2) running the mic gain too high; 3) using too much
compression/processing; and 4) not rolling off the low frequencies.
In a noisy environment, it always helps to work close to the mic. It is
ALWAYS good practice to use the minimum mic gain needed to get good
modulation, use no more than about 10dB of compression/processing, and
roll off the low frequency content. It's good engineering practice for the
highest quality broadcast stations, and it's good practice for ham radio.
Indeed, the only difference between what's right for broadcasting and for
ham radio is WHERE to cut the low end and HOW MUCH money to spend on
compression/processing. Many years ago, I sold processing systems for
broadcast stations that cost upwards of $10K in today's dollars, and I
helped the chief engineers of those stations adjust them. I suspect that
W8JI and K4TAX have similar experience. Before I spent ANY money on an
outboard box for a ham rig, I would first follow all of those elements of
good engineering practice.
73,
Jim Brown K9YC
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[Robert Mcgraw]

Very well said Jim.

73
Bob, K4TAX

Post by Jim Brown

Post by Lance Collister
It very effectively cuts out the background blower noise

Some of the major causes of audible background noise are 1) working too
far from the mic; 2) running the mic gain too high; 3) using too much
compression/processing; and 4) not rolling off the low frequencies.
In a noisy environment, it always helps to work close to the mic. It is
ALWAYS good practice to use the minimum mic gain needed to get good
modulation, use no more than about 10dB of compression/processing, and
roll off the low frequency content. It's good engineering practice for the
highest quality broadcast stations, and it's good practice for ham radio.
Indeed, the only difference between what's right for broadcasting and for
ham radio is WHERE to cut the low end and HOW MUCH money to spend on
compression/processing. Many years ago, I sold processing systems for
broadcast stations that cost upwards of $10K in today's dollars, and I
helped the chief engineers of those stations adjust them. I suspect that
W8JI and K4TAX have similar experience. Before I spent ANY money on an
outboard box for a ham rig, I would first follow all of those elements of
good engineering practice.
73,
Jim Brown K9YC
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[Jim Brown]

Post by ab2tc
On the K3 what would be the approximate setting for 10dB compression?

Follow instructions in the K3 manual to set the meter to read
compression. Then follow the instructions in the manual for first
setting the mic gain without any compression, and then set the
compression with the compression control. When you talk, you'll see the
compression bouncing around as you talk. It will start sounding bad when
it goes above 10dB on the hottest voice peaks.

73,

Jim Brown K9YC

[ab2tc]

I am using a Heil Proset, IC element, Mic gain to LO, setting 35 and
compression set at 20. With this I can not talk the CMP reading above 10 or
the ALC reading above 7. I get good audio reports and just as importantly I
get heard remarkably well by DX stations.

AB2TC - Knut

Post by Jim Brown

Post by ab2tc
On the K3 what would be the approximate setting for 10dB compression?

Follow instructions in the K3 manual to set the meter to read
compression. Then follow the instructions in the manual for first
setting the mic gain without any compression, and then set the
compression with the compression control. When you talk, you'll see the
compression bouncing around as you talk. It will start sounding bad when
it goes above 10dB on the hottest voice peaks.
73,
Jim Brown K9YC
<snip>

[ab2tc]

Case in point. Just broke a huge pileup for 8R1AK/P. I am barefoot with a
G5RV.

AB2TC - Knut

Post by ab2tc
I am using a Heil Proset, IC element, Mic gain to LO, setting 35 and
compression set at 20. With this I can not talk the CMP reading above 10
or the ALC reading above 7. I get good audio reports and just as
importantly I get heard remarkably well by DX stations.
AB2TC - Knut

[Mel Farrer]

Yes, and watch the ALC level while you are doing it.? I found that you can increase the compression and not trip the 5-7 bars of ALC, but it will sound terrible.? Stay below 10 dB of compression and under 5 bars of ALC for the smoothest sound and lack of splatter.? FWIW.

Mel, K6KBE

--- On Mon, 4/26/10, Jim Brown <jim at audiosystemsgroup.com> wrote:

From: Jim Brown <jim at audiosystemsgroup.com>
Subject: Re: [Elecraft] W2IHY 8 band equalizer and EQ Plus
To: "elecraft at mailman.qth.net" <elecraft at mailman.qth.net>
Date: Monday, April 26, 2010, 3:02 PM

Post by ab2tc
On the K3 what would be the approximate setting for 10dB compression?

Follow instructions in the K3 manual to set the meter to read
compression. Then follow the instructions in the manual for first
setting the mic gain without any compression, and then set the
compression with the compression control.? When you talk, you'll see the
compression bouncing around as you talk. It will start sounding bad when
it goes above 10dB on the hottest voice peaks.

73,

Jim Brown K9YC


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[John King]

Post by Robert Mcgraw
And do remember, more compression brings up background noise between
words by the amount of the compression value. Hence with 10 dB of
compression, a background noise level that would normally produce and
output of 1 watt will then sound like 10 watts of noise.
73
Bob, K4TAX

Bob's comment above is right on the money and is well worth re-reading.

Professional announcers use proper mic technique *and* they work in
sound booths or acoustically inert studios. Sports announcers who must
work in noisy environments use headsets with high quality noise-canceling
mics inches from their lips, followed by processors far more sophisticated
than any used by hams.

A tip to the phone contest fellows: prepare your CQ loops in a quiet
room well in advance of the event. I am always astounded at the poor
quality of the CQ loops in contests - many from the well known big
gun stations who should know better. I wonder about the impact on the
Q-rate when the listeners (especially those whose native language is
other than English) have difficulty understanding the CQ loop.

73,
john WA1ABI

[Jim Brown]

Post by John King
I wonder about the impact on the
Q-rate when the listeners (especially those whose native language is
other than English) have difficulty understanding the CQ loop.

And things seem to be getting worse with every contest. At least 10% of the
SSB signals I hear are so badly processed that the audio is so badly
distorted that I can't copy the call sign, even if I listen to it a dozen
times. And yes, some of these lids consider themselves big guns.

Several times in the last month or so, R1FJM has been spotted on 20M SSB. I'd
like to work him, and I can hear him, but his audio is so awful I can't
understand a word. If someone knows him, please tell him.

73,

Jim K9YC

[Lu Romero]

Here Here! (or Hear Hear):

Jim, are you a disciple of Frank Foti? :)

I like Julius' gear, and I have worked with the EQPlus
device at the NQ4I Multi Multi station on Rick's Orions.
This box does make those radios sound quite good, especially
with just a hint of "Delay" dialed in.

I do find that the noise gate on the K3 has a "crackling"
sound when muting and unmuting, making somewhat useless to
me (if you pay attention to that nuance... I do. Otherwise,
it works just fine) Lyle has done a great job with the TX
chain on the K3... I would like to see some handles on
Attack and Release as well as ratio, but then that could be
painful to use if you dont know what youre doing. RF
Clipper's attack and decay characteristics are rather
generalised.

The best way to fix audio ambient noise issues is through
your environment's acoustics instead of "fixing it in the
mix" with processing and gating.

Folks shouldnt forget that we are transmitting into a very
noisy medium. High dynamic range defeats intelligibility.
SENSIBLE "compression" (RF Clipping) settings are your
friends, as you then reduce the dynamic range (the
difference between the loudest and the softest sounds in a
given audio waveform) and have more "modulation density" to
rise above the ambient noise on the band.

Tailoring your frequency response to concentrate power in a
given voice range will go a long way to making your signal
"pop" out of the noise. Close talk the mic as much as
possible and reduce the mic gain as Jim describes.

A good example on how all these parameters work together to
make your signal stand out can be gleaned by downloading
VE3NEA's excellent "Voice Shaper" simulator program (its
free). Use your favorite air mic and play with it for a
while to get an understanding of how gates,
compressor/limiters and EQ affect your signal in QRM and QRN
conditions.

Try to pay attention to the natural acoustics in your
operating position, if you can. Curtains help, hard walls
hurt. Carpet helps, Terrazo floors hurt. Try to set your
operating position and/or microphone somewhat at an angle
between hard reflecting walls to reduce phase cancelling or
adding from the reflecting walls/surfaces.

Personally, I am not a believer in ESSB. But different
strokes for different folks, and I wont criticise folks who
practice this "voodoo" until they become 8kHz wide and QRM
me or I am able to understand them when listening to their
SSB signal in AM mode (all that bass often creates a "pseudo
carrier").

You would be surprised how well you can be heard using the
built in features provided by Elecraft in the K3. It takes
practice and a commitment to resist the temptation to "turn
it up to eleven".

-lu-W4LT-


Date: Mon, 26 Apr 2010 10:57:04 -0700
From: "Jim Brown" <jim at audiosystemsgroup.com>
Subject: Re: [Elecraft] W2IHY 8 band equalizer and EQ Plus
To: "Elecraft List" <elecraft at mailman.qth.net>
Message-ID: <20100426175705.D90C957C38 at gw1.nlenet.net>
Content-Type: text/plain; charset="us-ascii"

Post by Lance Collister
It very effectively cuts out the background blower noise

Some of the major causes of audible background noise are 1)
working too
far from the mic; 2) running the mic gain too high; 3) using
too much
compression/processing; and 4) not rolling off the low
frequencies.

In a noisy environment, it always helps to work close to the
mic. It is
ALWAYS good practice to use the minimum mic gain needed to
get good
modulation, use no more than about 10dB of
compression/processing, and
roll off the low frequency content. It's good engineering
practice for the
highest quality broadcast stations, and it's good practice
for ham radio.

Indeed, the only difference between what's right for
broadcasting and for
ham radio is WHERE to cut the low end and HOW MUCH money to
spend on
compression/processing. Many years ago, I sold processing
systems for
broadcast stations that cost upwards of $10K in today's
dollars, and I
helped the chief engineers of those stations adjust them. I
suspect that
W8JI and K4TAX have similar experience. Before I spent ANY
money on an
outboard box for a ham rig, I would first follow all of
those elements of
good engineering practice.

73,

Jim Brown K9YC

[Jim Brown]

Post by Lu Romero
Jim, are you a disciple of Frank Foti? :)

Never heard of him.

Post by Lu Romero
The best way to fix audio ambient noise issues is through
your environment's acoustics instead of "fixing it in the
mix" with processing and gating.

Yes. The noise sources in my shack are the fans in my Ten Tec Titan
power amps, and a few outboard fans I've added to cool their power
supplies for high duty-cycle contesting. The Titan fans can get
pretty noisy, and their case/chassis can amplify their vibrations,
making them louder. I keep the screws tightened so the lids don't
vibrate, and I've got both amps (I do SO2R) sitting on soft acoustic
foam to decouple them from the desktop that they're sitting on. I've
also placed some absorption on the walls behind the amps. Although I
haven't measured it, I'd estimate that I've attenuated it by 6dB or
so.

Post by Lu Romero
Folks shouldnt forget that we are transmitting into a very
noisy medium. High dynamic range defeats intelligibility.
SENSIBLE "compression" (RF Clipping) settings are your
friends, as you then reduce the dynamic range (the
difference between the loudest and the softest sounds in a
given audio waveform) and have more "modulation density" to
rise above the ambient noise on the band.

EXACTLY!

Post by Lu Romero
Try to pay attention to the natural acoustics in your
operating position, if you can. Curtains help, hard walls
hurt. Carpet helps, Terrazo floors hurt.

More very good advice. The key benefit of this is to reduce the wild
sound (both noise and wall reflections) that the mic hears. In
addition to funky carpet on the floor, more absorption is provided
by walls holding books on shelves.

73,

Jim K9YC

[Bob McGraw - K4TAX]

One of the things we often fail to implement is the transmit bandwidth
being adjusted much like the receiving bandwidth on the other end. In a
contest situation likely the receiving station is using a 1500 to 1800 Hz
receiver bandwidth with likely band pass tuning such that the lower 300 to
500 Hz of that is attenuated. This makes for an effective receive
bandwidth of some 1300 Hz or so.
Now then with our transmitter setting for a "full width" normal SSB
bandwidth of say 2.6 or 2.8 KHz and a low end roll off of say 120 Hz this
make for an effective bandwidth of some 2600 Hz. Yes our transmitter
power is spread over that bandwidth. Wouldn't it make more sense to
concentrate the transmitter power over say 1300 Hz rather than 2600 Hz?
In doing so one gains almost 3 dB of effective power increase with
actually no increase in PEP. Plus the other folks on the band will
appreciate the narrow signals.
Yes of course it will sound pinched up but in reality there is little
information in the male voice spoken range below 400 Hz and little above
1500 to 1800 Hz. But hey, some of the compressed and processed signals
only serve to occupy the full 2600 Hz of bandwidth, with what? It's not
pretty for sure. So if you want a screaming DX pileup busting signal,
squeeze in the bandwidth and don't worry about the EQ or the special
purpose mike. Your good sounding SSB mike into that transmitter bandwidth
will do the job just fine and the neighbors on either side of your
frequency will appreciate your efforts.
73
Bob, K4TAX
----- Original Message -----
From: "Lu Romero" <lromero at ij.net>
Sent: Tuesday, April 27, 2010 9:56 AM
Subject: Re: [Elecraft] W2IHY 8 band equalizer and EQ Plus

Post by Lu Romero
Jim, are you a disciple of Frank Foti? :)
I like Julius' gear, and I have worked with the EQPlus
device at the NQ4I Multi Multi station on Rick's Orions.
This box does make those radios sound quite good, especially
with just a hint of "Delay" dialed in.
I do find that the noise gate on the K3 has a "crackling"
sound when muting and unmuting, making somewhat useless to
me (if you pay attention to that nuance... I do. Otherwise,
it works just fine) Lyle has done a great job with the TX
chain on the K3... I would like to see some handles on
Attack and Release as well as ratio, but then that could be
painful to use if you dont know what youre doing. RF
Clipper's attack and decay characteristics are rather
generalised.
The best way to fix audio ambient noise issues is through
your environment's acoustics instead of "fixing it in the
mix" with processing and gating.
Folks shouldnt forget that we are transmitting into a very
noisy medium. High dynamic range defeats intelligibility.
SENSIBLE "compression" (RF Clipping) settings are your
friends, as you then reduce the dynamic range (the
difference between the loudest and the softest sounds in a
given audio waveform) and have more "modulation density" to
rise above the ambient noise on the band.
Tailoring your frequency response to concentrate power in a
given voice range will go a long way to making your signal
"pop" out of the noise. Close talk the mic as much as
possible and reduce the mic gain as Jim describes.
A good example on how all these parameters work together to
make your signal stand out can be gleaned by downloading
VE3NEA's excellent "Voice Shaper" simulator program (its
free). Use your favorite air mic and play with it for a
while to get an understanding of how gates,
compressor/limiters and EQ affect your signal in QRM and QRN
conditions.
Try to pay attention to the natural acoustics in your
operating position, if you can. Curtains help, hard walls
hurt. Carpet helps, Terrazo floors hurt. Try to set your
operating position and/or microphone somewhat at an angle
between hard reflecting walls to reduce phase cancelling or
adding from the reflecting walls/surfaces.
Personally, I am not a believer in ESSB. But different
strokes for different folks, and I wont criticise folks who
practice this "voodoo" until they become 8kHz wide and QRM
me or I am able to understand them when listening to their
SSB signal in AM mode (all that bass often creates a "pseudo
carrier").
You would be surprised how well you can be heard using the
built in features provided by Elecraft in the K3. It takes
practice and a commitment to resist the temptation to "turn
it up to eleven".
-lu-W4LT-
Date: Mon, 26 Apr 2010 10:57:04 -0700
From: "Jim Brown" <jim at audiosystemsgroup.com>
Subject: Re: [Elecraft] W2IHY 8 band equalizer and EQ Plus
To: "Elecraft List" <elecraft at mailman.qth.net>
Message-ID: <20100426175705.D90C957C38 at gw1.nlenet.net>
Content-Type: text/plain; charset="us-ascii"

Post by Lance Collister
It very effectively cuts out the background blower noise

Some of the major causes of audible background noise are 1)
working too
far from the mic; 2) running the mic gain too high; 3) using
too much
compression/processing; and 4) not rolling off the low
frequencies.
In a noisy environment, it always helps to work close to the
mic. It is
ALWAYS good practice to use the minimum mic gain needed to
get good
modulation, use no more than about 10dB of
compression/processing, and
roll off the low frequency content. It's good engineering
practice for the
highest quality broadcast stations, and it's good practice
for ham radio.
Indeed, the only difference between what's right for
broadcasting and for
ham radio is WHERE to cut the low end and HOW MUCH money to
spend on
compression/processing. Many years ago, I sold processing
systems for
broadcast stations that cost upwards of $10K in today's
dollars, and I
helped the chief engineers of those stations adjust them. I
suspect that
W8JI and K4TAX have similar experience. Before I spent ANY
money on an
outboard box for a ham rig, I would first follow all of
those elements of
good engineering practice.
73,
Jim Brown K9YC
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[Lu Romero]

Bob: This is becoming somewhat off topic...

While I agree in principle, in practice, there are some high
frequency sibilance "formants" that need bandwidth up to and
including 2.8kHz to almost 3kHz to be well understood.
There are also some harmonics, especially in male voices,
that go down below 300-400 Hz. This is all variable with
each person's individual voice, so YMMV.

Joe, W4TV, clued me in to a small nuance that I had
completely neglected a few months back. He listened to my
EQ settings and mentioned that my audio was punchy but
"thin" as he put it. I listened to a recording that AD4C
made of me and I realized that I was clipping off some low
harmonics by completely dropping everything below 400Hz.

I now add a little +3dB bump to my EQ settings around 200Hz.
It gives me some added "presence". That little hump sort
of "fattens" my voice and helps lift it over hissy band
noise a bit. Im setting up the NQ4I Orions the same way,
too, and its made quite a difference.

I also like a little hint of "Harmonic Distortion" in my
audio, especially in the DVK. Just my personal preference,
it adds what I call "edge" to the a CQ Message Loop. My
TS850 did this with "high boost" on by itself, and I can
duplicate it with the MicroKeyer's recording path. I record
ALL DVK through the MicroKeyer's recording facility now,
with consistently repeatable results. I play out through
N1MM's DVK through the MicroKeyer2's soundcard.

I have almost gotten used to the "different" EQ band center
frequencies in the K3 Equalizer, too :)

I have owned K3 #3192 for almost 6 months now. It has taken
me almost all that time to feel comfortable with the audio
chain to the extent that I was comfortable with my former
TS850S equipped with a slew of Behringer and Radio Design
Labs outboard processing toys.

The only things that I would add to the K3 transmitter
processing chain would be a bit more headroom, a Pre-DSP two
band audio leveler/AGC and handles for Attack, Decay and
Ratio on the RF Clipper. If Lyle could do this I would kiss
the ring!

They should hold a seminar on this at Elecraft and not let
anybody access the last three adjustments until they passed
a certification test, however! It would be like giving
Nuclear Missiles to Mohmar Khadaffi in the hands of the
uninitiated :)

-lu-w4lt-

----- Original Message Follows -----
From: "Bob McGraw - K4TAX" <RMcGraw at Blomand.net>
To: <lromero at ij.net>, <"Elecraft List
<elecraft"@mailman.qth.net>
Subject: Re: [Elecraft] W2IHY 8 band equalizer and EQ Plus
Date: Tue, 27 Apr 2010 20:40:25 -0500

One of the things we often fail to implement is the
transmit bandwidth being adjusted much like the receiving
bandwidth on the other end. In a contest situation likely
the receiving station is using a 1500 to 1800 Hz receiver
bandwidth with likely band pass tuning such that the lower
300 to 500 Hz of that is attenuated. This makes for an
effective receive bandwidth of some 1300 Hz or so.
Now then with our transmitter setting for a "full width"
normal SSB bandwidth of say 2.6 or 2.8 KHz and a low end
roll off of say 120 Hz this make for an effective
bandwidth of some 2600 Hz. Yes our transmitter power is
spread over that bandwidth. Wouldn't it make more sense to
concentrate the transmitter power over say 1300 Hz rather
than 2600 Hz? In doing so one gains almost 3 dB of
effective power increase with actually no increase in PEP.
Plus the other folks on the band will appreciate the
narrow signals.
Yes of course it will sound pinched up but in reality there
is little information in the male voice spoken range below
400 Hz and little above 1500 to 1800 Hz. But hey, some of
the compressed and processed signals only serve to occupy
the full 2600 Hz of bandwidth, with what? It's not pretty
for sure. So if you want a screaming DX pileup busting
signal, squeeze in the bandwidth and don't worry about the
EQ or the special purpose mike. Your good sounding SSB
mike into that transmitter bandwidth will do the job just
fine and the neighbors on either side of your frequency
will appreciate your efforts.
73
Bob, K4TAX

[Guy Olinger K2AV]

Why pre-DSP? He can do that in firmware. You can forget about a
hardware mod. Why do we keep trying to remake a digital radio into an
analog radio? A DSP version can have options. Just has to get in
line with all the other things taking resources in a small business.

How do you define "headroom"?

73, Guy.

Post by Lu Romero
The only things that I would add to the K3 transmitter
processing chain would be a bit more headroom, a Pre-DSP two
band audio leveler/AGC and handles for Attack, Decay and
Ratio on the RF Clipper. ?If Lyle could do this I would kiss
the ring!

[Don Wilhelm]

Guy,

You are absolutely right. While much of the pro-audio equipment is
going digital to be able to more precisely control parameters without
the expensive of many analog stages, we are seeing requests for adding
an analog front end to the K3 digital audio processing as though that
would accomplish some sort of magic.
By the same reasoning, we might obtain improvement to our analog solid
state transceivers by adding some front end vacuum tube gear! :-)

73,
Don W3FPR

Post by Guy Olinger K2AV
Why pre-DSP? He can do that in firmware. You can forget about a
hardware mod. Why do we keep trying to remake a digital radio into an
analog radio? A DSP version can have options. Just has to get in
line with all the other things taking resources in a small business.
How do you define "headroom"?
73, Guy.

Post by Lu Romero
The only things that I would add to the K3 transmitter
processing chain would be a bit more headroom, a Pre-DSP two
band audio leveler/AGC and handles for Attack, Decay and
Ratio on the RF Clipper. If Lyle could do this I would kiss
the ring!

[Jim Brown]

Post by Lu Romero
While I agree in principle, in practice, there are some high
frequency sibilance "formants" that need bandwidth up to and
including 2.8kHz to almost 3kHz to be well understood.
There are also some harmonics, especially in male voices,
that go down below 300-400 Hz.

YES. Because I've made my living designing sound systems for
highly reverberant churches, this is something that I've had to
carefully study. It is VERY well known that the most important
octave bands for speech intelligibilty are the octave bands
centered on 1,000 Hz and 2,000 Hz. Those bands range means from
about 720 Hz to about 2.8 kHz. The 500 Hz is next most important
(extending down to about 350 Hz), followed by the 4000 Hz octave
band. Human knowledge about this is VERY well established, and
dates back to the earliest days of telephony. That's why it's so
important to boost the mic response to compensate for the rolloff
by crystal filters between 2 and 3kHz!

73,

Jim Brown K9YC

[Joe Subich, W4TV]

It is VERY well known that the most important octave bands for
speech intelligibilty are the octave bands centered on 1,000 Hz
and 2,000 Hz. Those bands range means from about 720 Hz to about
2.8 kHz. The 500 Hz is next most important (extending down to
about 350 Hz), followed by the 4000 Hz octave band.

And in the 1000 Hz band, the important part is almost entirely
the upper third. Human voice has almost no energy and nothing
that contributes to intelligibility between approximately 700
and 1100 Hz (with some variation in the beginning/end of that
dead band). One can do extremely effective communications audio
with 300 - 600 Hz and 1200 Hz to 2400 Hz only. Extend that just
slightly (200 - 600 Hz and 1200 Hz - 2800 Hz) and one gets audio
with very good "presence" (the low format) and excellent
articulation (the high format).

It is also interesting that human hearing is most sensitive in
the very area that the human voice has no energy. Some have
speculated that to be an evolutionary "defense" which allowed
early man to hear danger in the middle of a crowd of voices.

73,

... Joe, W4TV

Post by Lu Romero
While I agree in principle, in practice, there are some high
frequency sibilance "formants" that need bandwidth up to and
including 2.8kHz to almost 3kHz to be well understood.
There are also some harmonics, especially in male voices,
that go down below 300-400 Hz.

YES. Because I've made my living designing sound systems for
highly reverberant churches, this is something that I've had to
carefully study. It is VERY well known that the most important
octave bands for speech intelligibilty are the octave bands
centered on 1,000 Hz and 2,000 Hz. Those bands range means from
about 720 Hz to about 2.8 kHz. The 500 Hz is next most important
(extending down to about 350 Hz), followed by the 4000 Hz octave
band. Human knowledge about this is VERY well established, and
dates back to the earliest days of telephony. That's why it's so
important to boost the mic response to compensate for the rolloff
by crystal filters between 2 and 3kHz!
73,
Jim Brown K9YC

[Tom W8JI]

It is VERY well known that the most important octave

bands for

speech intelligibilty are the octave bands centered on

1,000 Hz

and 2,000 Hz. Those bands range means from about 720 Hz

to about

2.8 kHz. The 500 Hz is next most important (extending

down to

about 350 Hz), followed by the 4000 Hz octave band.

Case in point? Just yesterday on 40 meters an S9 station
with all that wonderful bass called me, and I had to adjust
the shift lower (to cut bass response with the K3) and
narrow the filter to be able to understand him through
background noise that was several S units weaker than him. A
friend in Australia with normal communications audio was a
bit weaker, but much better copy. All that bass power
wasted, being transmitted just to be filtered and discarded
at the receiver. :-)

[Paul Christensen]

Post by Joe Subich, W4TV
It is also interesting that human hearing is most sensitive in
the very area that the human voice has no energy. Some have
speculated that to be an evolutionary "defense" which allowed
early man to hear danger in the middle of a crowd of voices.

There's actually quite a bit of short duration energy of the human voice at
or near 3 kHz. Interestingly, the classic Fletcher-Munson (FM) family of
curves shows peak sensitivity to sound pressure level also at or near 3 kHz.
The speculation by many speech pathologists, audiologists and researchers is
that the presence band near 3 kHz evolved over time, matching an important
part of human speech for maximum intelligibility near the 3 kHz octave-band
region.

The 3 kHz peak is a function of the ear canal length. As I recall from my
psychacoustics studies at NIU, the ear canal length does not change
significantly from the time birth to adulthood. The resonance point on the
FM curves is calculated as anyone would calculate the resonance point of a
closed pipe. The tympanic membrane forms the closure on one end of the
pipe.

I would say that maximum articulation occurs at an upper audio cut-off near
3 kHz, with diminishing returns above that point.

On the low end, it's easy to plot the lowest frequency generated by any
voice. Using FFT software and a sound card, I've measured the fundamental
point of most male voices between 80-95 Hz. That 15 Hz of difference may
not seem like much, but the difference is significant. I've measured only a
few voices on the air that reach slightly below 80 Hz and when they do,
they're the ones that could do well with commercial voice-over work.

Certainly any attempt to achieve a response in low-end audio below about 90
Hz is a wasted effort. Folks who adjust their low-end EQ to compensate for
a lack of deep bass in their voices can do nothing to sound the way they
really want to. Either we were born with the gift or we weren't and no
amount of EQ will change that -- excessive boost just makes it sound like
we're trying to compensate for something we don't have and can easily create
a phantom carrier when speaking in ESSB mode.

Paul, W9AC

[Lu Romero]

Guys... Hold on...

I have my reasons... Hear me out...

Post by Guy Olinger K2AV
Why pre-DSP?

Maybe I should have said "Before the input of the active
audio processing area in the DSP section of the radio".
That is, a way to control analog "dynamic range" at the
input of the RF Clipper section of the DSP audio chain.

What I was suggesting is an AGC funtion before the RF
Clipping section touches the audio waveform. Something to
smooth out the dynamic range of the audio input so that the
DSP processing engine would not have to work so hard dealing
with peaks and valleys and can be "let loose" some.

The K3's DSP audio management software understands the
Nyquist Limit of its ADC, and it plays a very conservative
game :) The RF Clipper does a good job of controlling
these, always keeping them within the ADC's non distorting
range, but nothing is perfect. If you hit the input stage
too hard (ever get excited calling a DX Station or a
Multiplier?) the clipper grabs the peak, which then reduces
the overall audio level. Since the decay value is somewhat
slower than the attack, you end up "punching a hole" in the
audio. Im really nitpicking here... Its really minor, but I
notice it and have learned to live with it. But it *IS*
there.

Post by Guy Olinger K2AV
He can do that in firmware.

Thats just fine with me. I dont care if he does it with
Squirrels on a treadmill, if it can be done, Im all for it!

Post by Guy Olinger K2AV
You can forget about a hardware mod.

I was not aware that I was requesting a hardware mod. Where
did you get that idea? I dont care how its done. I just
hope that it *CAN* be done.

Post by Guy Olinger K2AV
Why do we keep trying to remake a digital radio into an

analog radio?

I was not aware that only *analog* radios had an AGC ahead
of a limiter. I know many broadcast digital processors that
have these features incorporated in software. You can
always go to the nearest pawn shop and buy a Aphex Compellor
and a mic preamp and do the same thing. Just would be nice
to have it all in one box.

Post by Guy Olinger K2AV
A DSP version can have options.

Exactly! This is just one more option!

Post by Guy Olinger K2AV
Just has to get in line with all the other things taking

resources in a small business.

I have no problem with that. K3 is not broken, Im not
trying to fix it. Just in my opinion this would be a very
useful feature. Its up to Wayne and his staff to make the
ultimate decision. If I want to fix it, I have to dust off
my Compellor and add some complexity to my station. I dont
want to do that.

Post by Guy Olinger K2AV
How do you define "headroom"?

The difference between when the input to the DSP is at its
highest point (100% signal saturation) and the point where
the input waveform clips appreciably (usually 10% above full
modulation is a standard measurement) in an RMS waveform.
Remember we are using a ADC here. It QUANTIZES the analog
waveform in finite steps (254? 1024? Only Lyle knows what
he has implemented). If its, for example, 0 to 254, then
255 is distortion because the value is undefined and is
truncated in the quantization matrix. You have hit the
Nyquist wall!

For instance, in professional digital videotape machines,
maximum level is defined as
-20dB (-20dB = 0VU in analog) and "saturation" is defined as
0dB (the "clip point" in analog). The difference between
these levels is defined as "headroom" (the "Nyquist limit")
for those RMS peaks that happen in wide dynamic range
material.

Back in the analog days we would use 0VU as the RMS
reference level, but the peaks could go all the way into the
red without distorting... But then analog systems went into
distortion "gracefully" a lot higher than 0VU; they never
run out of numbers (reach the "Nyquist" limit), but they do
run out of the linear curve of the active devices in the
circuit. Digital systems just clip when they run out of
numbers, so there is no margin for error, hence the "digital
overhead to account for analog dynamic range" built in to
digital audio recorders.

What Im saying is that since the difference in the K3 is
somewhat narrow, because we are playing conservatively, in
my opinion (I have not done quantitative testing, Its just
my "by ear" opinion), if there were some "dynamic range
leveling" before the ADC, it would be helpful to the overall
system efficiency.

I will shut up now, learn not to yell into the mic or go
plug in my Compellor :)

-lu-w4lt-

[Tom W8JI]

Maybe I should have said "Before the input of the active
audio processing area in the DSP section of the radio".
That is, a way to control analog "dynamic range" at the
input of the RF Clipper section of the DSP audio chain>>>

We should not be clipping at RF anyway. RF clipping, or
clipping an entire band, is an old method that should have
been retired years ago.

<<What I was suggesting is an AGC funtion before the RF
Clipping section touches the audio waveform. Something to
smooth out the dynamic range of the audio input so that the
DSP processing engine would not have to work so hard dealing
with peaks and valleys and can be "let loose" some.>>

The proper way to process speech is to split the speech into
bands that are less than one octave wide. Then we clip and
process each frequency band.
The output is filtered in a filter to clean it up, and the
results are remixed in the ratio the user wants.

Take 300-500 and clip it, the closest harmonic is 600.
Filter it at 300-500.
500-900 and clip it, the closest harmonic is 1000. Filter it
at 500-900.
900-1700 and clip it, the closest harmonic is 1800. Filter
it at 900-1700.
1700-3300 and clip. Filter it at 1700-3300.

In a DSP algorithm this idea would be fantastic, instead of
emulating something that was never that good to start with.

Vomax did something like this in the 70's when op-amps first
came out. I had a homebrew system with slow input AGC,
gating, and split processing.

Why turn back the clock to a compressor preceding an RF
processor? Contest stations already waste too much energy in
distortion and by transmitting useless frequencies.

73 Tom

[Lu Romero]

Don, I never suggested that this be implemented by an analog
stage or in an analog fashion. I dont know where you guys
are getting that idea!

AGC can be perfectly implemented in a digital environment.
It would quantize the waveform and bring the valleys up to
the peaks, then hand them off to the clipper where the real
fun begins.

That's all Im describing! It has nothing to do with a 12AX7
at all!

We have beat this to death enough, so as not to rase the ire
of Eric the Mighty Moderator, I now terminate my comments on
this thread.

If we want to kibbitz, we can do it off the list.

-lu-w4lt-

----- Original Message Follows -----
From: Don Wilhelm <w3fpr at embarqmail.com>
To: Guy Olinger K2AV <olinger at bellsouth.net>
Cc: lromero at ij.net, elecraft at mailman.qth.net
Subject: Re: [Elecraft] W2IHY 8 band equalizer and EQ Plus
Date: Wed, 28 Apr 2010 14:14:44 -0400

Post by Don Wilhelm
Guy,
You are absolutely right. While much of the pro-audio
equipment is going digital to be able to more precisely
control parameters without the expensive of many analog
stages, we are seeing requests for adding an analog front
end to the K3 digital audio processing as though that
would accomplish some sort of magic. By the same reasoning,
we might obtain improvement to our analog solid state
transceivers by adding some front end vacuum tube gear! :-)
73,
Don W3FPR

Post by Guy Olinger K2AV
Why pre-DSP? He can do that in firmware. You can forget
about a hardware mod. Why do we keep trying to remake a
digital radio into an analog radio? A DSP version can
have options. Just has to get in line with all the other
things taking resources in a small business.
How do you define "headroom"?
73, Guy.

Post by Lu Romero
The only things that I would add to the K3 transmitter
processing chain would be a bit more headroom, a Pre-DSP
two band audio leveler/AGC and handles for Attack, Decay
and Ratio on the RF Clipper. If Lyle could do this I
would kiss the ring!

[Guy Olinger K2AV]

I personally have just as much trouble as anyone else remembering to
think of the K3 as digital.

Post by Lu Romero
Don, I never suggested that this be implemented by an analog
stage or in an analog fashion. ?I dont know where you guys
are getting that idea!

Other than taking what you said as what you meant? Tough reading your
mind at this distance. You said:

"...a Pre-DSP two band audio leveler/AGC and handles for Attack, Decay
and Ratio on the RF Clipper."

That is, prior to Digital Signal Processing, or before analog is
converted to digital.

In your headroom explanation, is there a common instance of microphone
ADC saturation that needs to be reported? That's a pretty
seventh-grade mistake on Wayne's part if it's true. (Yeah, I know,
the first Hubble lenses, and the unit snafu on the Mars landers, also
very seventh-grade.)

You also said:

"What I was suggesting is an AGC funtion before the RF
Clipping section touches the audio waveform. Something to
smooth out the dynamic range of the audio input so that the
DSP processing engine would not have to work so hard dealing
with peaks and valleys and can be "let loose" some."

Elecraft is already accomplishing envelope leveling and shaping with
digital functions that don't appear to resemble the sledge and wedge
of RF clipping and AF variable band amplifiers. SOME folks get
excellent results using the K3's leveling and shaping processes for TX
audio. I would hate to bring up RTFM on setting up K3 mic gain and
compression, but the manual procedure does seem to work.

AND, since there is NO analog audio band circuitry in there anywhere,
BUT there ARE banded TX and RX equalizer functions being done in the
number soup, whose gains are being set by NUMBERS we enter in the
menu, what makes us think he hasn't already done something proprietary
about "banded gain" which he is developing further and is not about to
reveal so the competition can't copy it for free?

My grandchildren are growing up digital. My having my brain trained
on analog is my problem, certainly not theirs. Grandkids think number
soup and audio that turns analog as close as possible to the speakers
is the good stuff, especially if the last stage is a big tube followed
by a transformer (go figure).

You should have seen the look I got from the oldest one when I asked
him if the audio actually went through all the slide pots on one of
those big digital mixers he was running, the look that says "Please
don't talk like that when my friends are around."

Just a couple of the plentiful opportunities for mental disconnects,
there is no K3 RX AF analog circuit controlled by AF gain. There is
no K3 RX RF analog circuit controlled by RF gain. The AF and RF pot
settings are immediately turned into "advice" numbers and passed along
to the MCU.

There is a lot of misadventure to be had thinking of the K3 in analog terms.

73, Guy.

[Luis V. Romero]

Some quick questions (yeah, I know I said I would stop, but I cant help it,
I want to understand!)

"...a Pre-DSP two band audio leveler/AGC and handles for Attack, Decay and

Ratio on the RF Clipper."

That is, prior to Digital Signal Processing, or before analog is converted

to digital.

Where exactly does the input to the TX DSP section (OK, IF section) hit an
ADC? Where do we move from the Analog domain to the Digital domain? Where
is the "quantizer" for the mic input? Would make sense if it was before the
Clipper, right?

In your headroom explanation, is there a common instance of microphone ADC

saturation that needs to be reported? >That's a pretty seventh-grade
mistake on Wayne's part if it's true. (Yeah, I know, the first Hubble
lenses, and >the unit snafu on the Mars landers, also very seventh-grade.)

I don't believe that's it at all. I can punch holes into the audio if I
speak softly then speak loudly. Sounds to me like there is some kind of
compression happening (that is why I first incorrectly assumed that this
thing used a compressor not a kind of IF/RF Clipper). What happens is that
the radio grabs the loud syllable and holds off from releasing the gain
reduction for a bit, then recovers. If you hit it with something loud, then
something soft, you hear the hole. The decay of whatever is grabbing the
peak is slower than the attack. There is no overshoot that I can discern
with my ears, as the firmware must be dropping audio into a buffer and
setting the response in kind. I once heard people complain about delay in
the monitor, I do hear a very slight one. So there must be a look ahead
buffer that computes the response to the peak. All I'm saying is that it
would be nice to have a handle on at least the decay, so it can more match
the attack. Probably cant do that as it would create more delay in the
monitor. It's a fine line.
"What I was suggesting is an AGC funtion before the RF Clipping section
touches the audio waveform. Something to smooth out the dynamic range of
the audio input so that the DSP processing engine would not have to work so
hard dealing with peaks and valleys and can be "let loose" some."

Elecraft is already accomplishing envelope leveling and shaping with digital
functions that don't appear to >resemble the sledge and wedge of RF clipping
and AF variable band amplifiers. SOME folks get excellent results using the
K3's leveling and shaping processes for TX audio. I would hate to bring up
RTFM on setting up K3 mic gain and compression, but the manual procedure
does seem to work.
I would like to get a better understanding of the process that is used here.
I do use the manual settings, It does work well, and does sound good, but I
would like to understand what is happening under the hood better so I can
better adapt to what this rig likes to hear, which, to me, is consistent
levels. My old admittedly analog rig was less picky and had much more room
to play with (when I let it). Hopefully that is not Elecraft Secret Sauce.
AND, since there is NO analog audio band circuitry in there anywhere, BUT
there ARE banded TX and RX equalizer functions being done in the number
soup, whose gains are being set by NUMBERS we enter in the menu, what makes
us think he hasn't already done something proprietary about "banded gain"
which he is developing further and is not about to reveal so the competition
can't copy it for free?
This is probably the Secret Sauce I'm talking about. Are you implying
8-band digital split band processing?
You should have seen the look I got from the oldest one when I asked him if
the audio actually went through all the slide pots on one of those big
digital mixers he was running, the look that says "Please don't talk like
that when my friends are around."
That's pretty funny. On my side, the video side, I was explaining to a
pretty hot non linear editor how we did non-b-roll match frame editing with
timecode on helical composite VTR's and the importance of the color frame
sequence across 4 fields (360 degrees of subcarrier and matching subcarrier
to horizontal sync) so that the picture wouldn't jump at the match frame in
NTSC. Couldn't understand the process! Couldn't even begin to understand
true A/B rolls either. So I know what you mean.
there is no K3 RX AF analog circuit controlled by AF gain. There is no K3
RX RF analog circuit controlled by RF gain. The AF and RF pot settings are
immediately turned into "advice" numbers and passed along to the MCU.
Advice in the way of a VCA setting? Wonder what the granularity is. It is
an analog pot, so logically we read a voltage and digitize it, then report
it to the MCU.

Thanks for pointing me in the right direction, Guy. Now I know the probable
true reason for the annoying "cant talk to the RX controls while in TX"
behavior. A hybrid Parallel/Serial signal bus.

I'm afraid I pushed paper from the left side to the right side of a desk for
way too long.

-lu-




No virus found in this outgoing message
Checked by PC Tools AntiVirus (6.1.0.25 - 6.14830).
http://www.pctools.com/free-antivirus/

[Don Wilhelm]

Lu,

I don't know if it is my email client or your responses, but I cannot
make much sense of your in-line questions/comments.
Please ask your questions in plain English in a single coherent
statement. combing through a bunch of things trunchated by blue bars on
my email client is totally confusing.

Sorry if I am being an old 'fuddy-duddy', but I simply cannot readily
see which are your questions and which are the things you are referring
to. Concise questions please.

73,
Don W3FPR

Post by Luis V. Romero
Some quick questions (yeah, I know I said I would stop, but I cant help it,
I want to understand!)

"...a Pre-DSP two band audio leveler/AGC and handles for Attack, Decay and

Ratio on the RF Clipper."

That is, prior to Digital Signal Processing, or before analog is converted

to digital.
Where exactly does the input to the TX DSP section (OK, IF section) hit an
ADC? Where do we move from the Analog domain to the Digital domain? Where
is the "quantizer" for the mic input? Would make sense if it was before the
Clipper, right?

In your headroom explanation, is there a common instance of microphone ADC

saturation that needs to be reported? >That's a pretty seventh-grade
mistake on Wayne's part if it's true. (Yeah, I know, the first Hubble
lenses, and >the unit snafu on the Mars landers, also very seventh-grade.)
I don't believe that's it at all. I can punch holes into the audio if I
speak softly then speak loudly. Sounds to me like there is some kind of
compression happening (that is why I first incorrectly assumed that this
thing used a compressor not a kind of IF/RF Clipper). What happens is that
the radio grabs the loud syllable and holds off from releasing the gain
reduction for a bit, then recovers. If you hit it with something loud, then
something soft, you hear the hole. The decay of whatever is grabbing the
peak is slower than the attack. There is no overshoot that I can discern
with my ears, as the firmware must be dropping audio into a buffer and
setting the response in kind. I once heard people complain about delay in
the monitor, I do hear a very slight one. So there must be a look ahead
buffer that computes the response to the peak. All I'm saying is that it
would be nice to have a handle on at least the decay, so it can more match
the attack. Probably cant do that as it would create more delay in the
monitor. It's a fine line.
"What I was suggesting is an AGC funtion before the RF Clipping section
touches the audio waveform. Something to smooth out the dynamic range of
the audio input so that the DSP processing engine would not have to work so
hard dealing with peaks and valleys and can be "let loose" some."
Elecraft is already accomplishing envelope leveling and shaping with digital
functions that don't appear to >resemble the sledge and wedge of RF clipping
and AF variable band amplifiers. SOME folks get excellent results using the
K3's leveling and shaping processes for TX audio. I would hate to bring up
RTFM on setting up K3 mic gain and compression, but the manual procedure
does seem to work.
I would like to get a better understanding of the process that is used here.
I do use the manual settings, It does work well, and does sound good, but I
would like to understand what is happening under the hood better so I can
better adapt to what this rig likes to hear, which, to me, is consistent
levels. My old admittedly analog rig was less picky and had much more room
to play with (when I let it). Hopefully that is not Elecraft Secret Sauce.
AND, since there is NO analog audio band circuitry in there anywhere, BUT
there ARE banded TX and RX equalizer functions being done in the number
soup, whose gains are being set by NUMBERS we enter in the menu, what makes
us think he hasn't already done something proprietary about "banded gain"
which he is developing further and is not about to reveal so the competition
can't copy it for free?
This is probably the Secret Sauce I'm talking about. Are you implying
8-band digital split band processing?
You should have seen the look I got from the oldest one when I asked him if
the audio actually went through all the slide pots on one of those big
digital mixers he was running, the look that says "Please don't talk like
that when my friends are around."
That's pretty funny. On my side, the video side, I was explaining to a
pretty hot non linear editor how we did non-b-roll match frame editing with
timecode on helical composite VTR's and the importance of the color frame
sequence across 4 fields (360 degrees of subcarrier and matching subcarrier
to horizontal sync) so that the picture wouldn't jump at the match frame in
NTSC. Couldn't understand the process! Couldn't even begin to understand
true A/B rolls either. So I know what you mean.
there is no K3 RX AF analog circuit controlled by AF gain. There is no K3
RX RF analog circuit controlled by RF gain. The AF and RF pot settings are
immediately turned into "advice" numbers and passed along to the MCU.
Advice in the way of a VCA setting? Wonder what the granularity is. It is
an analog pot, so logically we read a voltage and digitize it, then report
it to the MCU.
Thanks for pointing me in the right direction, Guy. Now I know the probable
true reason for the annoying "cant talk to the RX controls while in TX"
behavior. A hybrid Parallel/Serial signal bus.
I'm afraid I pushed paper from the left side to the right side of a desk for
way too long.
-lu-
No virus found in this outgoing message
Checked by PC Tools AntiVirus (6.1.0.25 - 6.14830).
http://www.pctools.com/free-antivirus/
______________________________________________________________
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[Guy Olinger K2AV]

Hi Lu.

Schematics for the K3 are downloadable PDF on the Elecraft website.
Many questions can be answered right off just digging into the
drawings. I find the ability in Adobe Reader to search in the PDF
text very useful for finding stuff. Device specifics are in the
schematic, so I guess you could go looking for a manufacturers
technical writeup and figure out the granularity from that.

I think that some of what you are calling "punching" really is just
the K3 dealing with sudden power spikes. Professional microphone
technique would not include very soft followed by very loud. The K3
is interpreting the loudest audio as being the intentional "top" of
your speaking pattern and is setting power management to NOT punch out
the amp and cause ALC spikes coming back from the amp. This effect
would be further exacerbated if all of the various TX power
calibrations had not been done correctly, as it will be further
exacerbated if the users manual MIC/CMP/PWR setting procedure is not
used.

The terms you are using to describe the K3's internal functioning will
remain speculative unless Wayne publishes stuff, and our thought
patterns still have have that analog, sequential function sound to
them : >)

K3 has a digital transmit envelope management function that should be
preceded by proper TX gain calibrations, and the user manual
MIC/CMP/PWR setting.

73, Guy.

Post by Luis V. Romero
Some quick questions (yeah, I know I said I would stop, but I cant help it,
I want to understand!)

"...a Pre-DSP two band audio leveler/AGC and handles for Attack, Decay and

Ratio on the RF Clipper."

That is, prior to Digital Signal Processing, or before analog is converted

to digital.
Where exactly does the input to the TX DSP section (OK, IF section) hit an
ADC? ?Where do we move from the Analog domain to the Digital domain? ?Where
is the "quantizer" for the mic input? ?Would make sense if it was before the
Clipper, right?

In your headroom explanation, is there a common instance of microphone ADC

saturation that needs to be reported? ?>That's a pretty seventh-grade
mistake on Wayne's part if it's true. ?(Yeah, I know, the first Hubble
lenses, and >the unit snafu on the Mars landers, also very seventh-grade.)
I don't believe that's it at all. ?I can punch holes into the audio if I
speak softly then speak loudly. ?Sounds to me like there is some kind of
compression happening (that is why I first incorrectly assumed that this
thing used a compressor not a kind of IF/RF Clipper). ?What happens is that
the radio grabs the loud syllable and holds off from releasing the gain
reduction for a bit, then recovers. ?If you hit it with something loud, then
something soft, you hear the hole. ?The decay of whatever is grabbing the
peak is slower than the attack. ?There is no overshoot that I can discern
with my ears, as the firmware must be dropping audio into a buffer and
setting the response in kind. ?I once heard people complain about delay in
the monitor, I do hear a very slight one. ?So there must be a look ahead
buffer that computes the response to the peak. ?All I'm saying is that it
would be nice to have a handle on at least the decay, so it can more match
the attack. ?Probably cant do that as it would create more delay in the
monitor. ?It's a fine line.
"What I was suggesting is an AGC funtion before the RF Clipping section
touches the audio waveform. ?Something to smooth out the dynamic range of
the audio input so that the DSP processing engine would not have to work so
hard dealing with peaks and valleys and can be "let loose" some."
Elecraft is already accomplishing envelope leveling and shaping with digital
functions that don't appear to >resemble the sledge and wedge of RF clipping
and AF variable band amplifiers. SOME folks get excellent results using the
K3's leveling and shaping processes for TX audio. ?I would hate to bring up
RTFM on setting up K3 mic gain and compression, but the manual procedure
does seem to work.
I would like to get a better understanding of the process that is used here.
I do use the manual settings, It does work well, and does sound good, but I
would like to understand what is happening under the hood better so I can
better adapt to what this rig likes to hear, which, to me, is consistent
levels. ?My old admittedly analog rig was less picky and had much more room
to play with (when I let it). ?Hopefully that is not Elecraft Secret Sauce.
AND, since there is NO analog audio band circuitry in there anywhere, BUT
there ARE banded TX and RX equalizer functions being done in the number
soup, whose gains are being set by NUMBERS we enter in the menu, what makes
us think he hasn't already done something proprietary about "banded gain"
which he is developing further and is not about to reveal so the competition
can't copy it for free?
This is probably the Secret Sauce I'm talking about. ?Are you implying
8-band digital split band processing?
You should have seen the look I got from the oldest one when I asked him if
the audio actually went through all the slide pots on one of those big
digital mixers he was running, the look that says "Please don't talk like
that when my friends are around."
That's pretty funny. ?On my side, the video side, I was explaining to a
pretty hot non linear editor how we did non-b-roll match frame editing with
timecode on helical composite VTR's and the importance of the color frame
sequence across 4 fields (360 degrees of subcarrier and matching subcarrier
to horizontal sync) so that the picture wouldn't jump at the match frame in
NTSC. ?Couldn't understand the process! ?Couldn't even begin to understand
true A/B rolls either. ?So I know what you mean.
there is no K3 RX AF analog circuit controlled by AF gain. ?There is no K3
RX RF analog circuit controlled by RF gain. The AF and RF pot settings are
immediately turned into "advice" numbers and passed along to the MCU.
Advice in the way of a VCA setting? ?Wonder what the granularity is. ?It is
an analog pot, so logically we read a voltage and digitize it, then report
it to the MCU.
Thanks for pointing me in the right direction, Guy. ?Now I know the probable
true reason for the annoying "cant talk to the RX controls while in TX"
behavior. ?A hybrid Parallel/Serial signal bus.
I'm afraid I pushed paper from the left side to the right side of a desk for
way too long.
-lu-
No virus found in this outgoing message
Checked by PC Tools AntiVirus (6.1.0.25 - 6.14830).
http://www.pctools.com/free-antivirus/

[Lyle Johnson]

Post by Guy Olinger K2AV
The terms you are using to describe the K3's internal functioning will
remain speculative unless Wayne publishes stuff,...

The K3 Mic algorithm applies a fast attack, slow decay gain control loop
after the Tx Equalizer. The peak detector in this loop is displayed on
the ALC bargraph. The fifth bar indicates the threshold beyond which the
loop reduces gain. Currently, the 6th bar shows about 3 dB of loop gain
reduction and the 7th bar's threshold is about 6 dB of loop gain reduction.

"Fast" and "slow" are relative terms, and not user adjustable. Consider
them part of the rig's personality...

73,

Lyle KK7P

[Joe Subich, W4TV]

Lyle,

Post by Lyle Johnson
The K3 Mic algorithm applies a fast attack, slow decay gain control
loop after the Tx Equalizer. The peak detector in this loop is
displayed on the ALC bargraph.

Does this gain control occur before or after the compressor if it
is engaged?

Post by Lyle Johnson
"Fast" and "slow" are relative terms, and not user adjustable.
Consider them part of the rig's personality...

If the gain control is prior to the compressor, a fast decay might
be more appropriate and reduce the "punch out" effect.

73,

... Joe, W4TV

Post by Lyle Johnson

Post by Guy Olinger K2AV
The terms you are using to describe the K3's internal functioning will
remain speculative unless Wayne publishes stuff,...

The K3 Mic algorithm applies a fast attack, slow decay gain control loop
after the Tx Equalizer. The peak detector in this loop is displayed on
the ALC bargraph. The fifth bar indicates the threshold beyond which the
loop reduces gain. Currently, the 6th bar shows about 3 dB of loop gain
reduction and the 7th bar's threshold is about 6 dB of loop gain reduction.
"Fast" and "slow" are relative terms, and not user adjustable. Consider
them part of the rig's personality...
73,
Lyle KK7P

[Lyle Johnson]

Post by Joe Subich, W4TV

Post by Lyle Johnson
The K3 Mic algorithm applies a fast attack, slow decay gain control
loop after the Tx Equalizer. The peak detector in this loop is
displayed on the ALC bargraph.

Does this gain control occur before or after the compressor if it
is engaged?

Before.

Lyle KK7P

[Joe Subich, W4TV]

Post by Lyle Johnson

Post by Joe Subich, W4TV
Does this gain control occur before or after the compressor if it
is engaged?

Before.

Might be more appropriate for a long decay gain control loop to
FOLLOW the compressor (clipper). IIRC, that is the order in most
broadcast (talk formatted) use ... the clipper prevents unusual
peaks from causing distortion and the longer recovery compressor
smooths out the level (rides gain).

73,

... Joe, W4TV

Post by Lyle Johnson

Post by Joe Subich, W4TV

Post by Lyle Johnson
The K3 Mic algorithm applies a fast attack, slow decay gain control
loop after the Tx Equalizer. The peak detector in this loop is
displayed on the ALC bargraph.

Does this gain control occur before or after the compressor if it
is engaged?

Before.
Lyle KK7P

[Lyle Johnson]

Post by Joe Subich, W4TV

Post by Lyle Johnson

Post by Joe Subich, W4TV
Does this gain control occur before or after the compressor if it
is engaged?

Before.

Might be more appropriate for a long decay gain control loop to
FOLLOW the compressor (clipper).

That loop is handled by the ALC circuitry/logic in the K3.

The K3's Tx gain system in voice modes is the mic gain loop I described,
then the clipper if engaged by the operator, then the power control/ALC
loop.

73,

Lyle KK7P

[Guy Olinger K2AV]

Sounds sweet to me, even if it is just number soup. As some have said,
try it, you might like it.

73, Guy.

Post by Lyle Johnson

Post by Joe Subich, W4TV

Post by Lyle Johnson

Post by Joe Subich, W4TV
Does this gain control occur before or after the compressor if it
is engaged?

Before.

Might be more appropriate for a long decay gain control loop to
FOLLOW the compressor (clipper).

That loop is handled by the ALC circuitry/logic in the K3.
The K3's Tx gain system in voice modes is the mic gain loop I described,
then the clipper if engaged by the operator, then the power control/ALC
loop.
73,
Lyle KK7P
______________________________________________________________
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[Mike Scott]

Post by Don Wilhelm
By the same reasoning, we might obtain improvement to our analog solid

state transceivers by adding some front end vacuum tube gear! :-)

Sweet! Now we are talking about real radios!


Mike Scott - AE6WA
Tarzana, CA (DM04 / near LA)
NAQCC 3535
K3-100 #508 / KX1 #1311

[Luis V. Romero]

Thanks Guy and Don. Don, I apologize for my muddled email. My fault, not
yours.

I will look closer at the system schematics so as to discover the flow of
the process when I have some time. Thanks for pointing out some of the
nuances from your perspectives.

What I call "punching holes in the audio" is exactly that... If you have a
short loud sound followed by a soft one, part of the soft one goes away due
to the attack/decay ratio of whatever magic waveform modification process is
at work in this rig. Analog monolithic compressors do this when the attack
is set faster than the decay on short duration spikes.

I have adapted to the way this process "thinks". I was just curious to the
way it was programmed to "think" and looking for a way, other than hang a
AGC at the mic input, to manage the process better.

However it works, it works quite well. But the smoother the waveform going
in, the smoother the waveform is going out.

May we live in interesting digital times indeed!

-lu-w4lt-

-----Original Message-----
From: guyk2av at gmail.com [mailto:guyk2av at gmail.com] On Behalf Of Guy Olinger
K2AV
Sent: Thursday, April 29, 2010 9:30 PM
To: lromero at ij.net
Cc: don at w3fpr.com; elecraft at mailman.qth.net
Subject: Re: [Elecraft] W2IHY 8 band equalizer and EQ Plus

Hi Lu.

Schematics for the K3 are downloadable PDF on the Elecraft website.
Many questions can be answered right off just digging into the drawings. I
find the ability in Adobe Reader to search in the PDF text very useful for
finding stuff. Device specifics are in the schematic, so I guess you could
go looking for a manufacturers technical writeup and figure out the
granularity from that.

I think that some of what you are calling "punching" really is just the K3
dealing with sudden power spikes. Professional microphone technique would
not include very soft followed by very loud. The K3 is interpreting the
loudest audio as being the intentional "top" of your speaking pattern and is
setting power management to NOT punch out the amp and cause ALC spikes
coming back from the amp. This effect would be further exacerbated if all
of the various TX power calibrations had not been done correctly, as it will
be further exacerbated if the users manual MIC/CMP/PWR setting procedure is
not used.

The terms you are using to describe the K3's internal functioning will
remain speculative unless Wayne publishes stuff, and our thought patterns
still have have that analog, sequential function sound to them : >)

K3 has a digital transmit envelope management function that should be
preceded by proper TX gain calibrations, and the user manual MIC/CMP/PWR
setting.

73, Guy.

Post by Luis V. Romero
Some quick questions (yeah, I know I said I would stop, but I cant
help it, I want to understand!)

Post by Guy Olinger K2AV
"...a Pre-DSP two band audio leveler/AGC and handles for Attack, Decay
and

Ratio on the RF Clipper."

Post by Guy Olinger K2AV
That is, prior to Digital Signal Processing, or before analog is
converted

to digital.
Where exactly does the input to the TX DSP section (OK, IF section)
hit an ADC? ?Where do we move from the Analog domain to the Digital
domain? ?Where is the "quantizer" for the mic input? ?Would make sense
if it was before the Clipper, right?

Post by Guy Olinger K2AV
In your headroom explanation, is there a common instance of microphone
ADC

saturation that needs to be reported? ?>That's a pretty seventh-grade
mistake on Wayne's part if it's true. ?(Yeah, I know, the first Hubble
lenses, and >the unit snafu on the Mars landers, also very
seventh-grade.)
I don't believe that's it at all. ?I can punch holes into the audio if
I speak softly then speak loudly. ?Sounds to me like there is some
kind of compression happening (that is why I first incorrectly assumed
that this thing used a compressor not a kind of IF/RF Clipper). ?What
happens is that the radio grabs the loud syllable and holds off from
releasing the gain reduction for a bit, then recovers. ?If you hit it
with something loud, then something soft, you hear the hole. ?The
decay of whatever is grabbing the peak is slower than the attack. ?
There is no overshoot that I can discern with my ears, as the firmware
must be dropping audio into a buffer and setting the response in kind. ?
I once heard people complain about delay in the monitor, I do hear a
very slight one. ?So there must be a look ahead buffer that computes
the response to the peak. ?All I'm saying is that it would be nice to
have a handle on at least the decay, so it can more match the attack. ?
Probably cant do that as it would create more delay in the monitor. ?It's

a fine line.

Post by Luis V. Romero
"What I was suggesting is an AGC funtion before the RF Clipping
section touches the audio waveform. ?Something to smooth out the
dynamic range of the audio input so that the DSP processing engine
would not have to work so hard dealing with peaks and valleys and can be

"let loose" some."

Post by Luis V. Romero
Elecraft is already accomplishing envelope leveling and shaping with
digital functions that don't appear to >resemble the sledge and wedge
of RF clipping and AF variable band amplifiers. SOME folks get
excellent results using the K3's leveling and shaping processes for TX
audio. ?I would hate to bring up RTFM on setting up K3 mic gain and
compression, but the manual procedure does seem to work.
I would like to get a better understanding of the process that is used

here.

Post by Luis V. Romero
I do use the manual settings, It does work well, and does sound good,
but I would like to understand what is happening under the hood better
so I can better adapt to what this rig likes to hear, which, to me, is
consistent levels. ?My old admittedly analog rig was less picky and
had much more room to play with (when I let it). ?Hopefully that is not

Elecraft Secret Sauce.

Post by Luis V. Romero
AND, since there is NO analog audio band circuitry in there anywhere,
BUT there ARE banded TX and RX equalizer functions being done in the
number soup, whose gains are being set by NUMBERS we enter in the
menu, what makes us think he hasn't already done something proprietary

about "banded gain"

Post by Luis V. Romero
which he is developing further and is not about to reveal so the
competition can't copy it for free?
This is probably the Secret Sauce I'm talking about. ?Are you implying
8-band digital split band processing?
You should have seen the look I got from the oldest one when I asked
him if the audio actually went through all the slide pots on one of
those big digital mixers he was running, the look that says "Please
don't talk like that when my friends are around."
That's pretty funny. ?On my side, the video side, I was explaining to
a pretty hot non linear editor how we did non-b-roll match frame
editing with timecode on helical composite VTR's and the importance of
the color frame sequence across 4 fields (360 degrees of subcarrier
and matching subcarrier to horizontal sync) so that the picture
wouldn't jump at the match frame in NTSC. ?Couldn't understand the
process! ?Couldn't even begin to understand true A/B rolls either. ?So I

know what you mean.

Post by Luis V. Romero
there is no K3 RX AF analog circuit controlled by AF gain. ?There is
no K3 RX RF analog circuit controlled by RF gain. The AF and RF pot
settings are immediately turned into "advice" numbers and passed along to

the MCU.

Post by Luis V. Romero
Advice in the way of a VCA setting? ?Wonder what the granularity is. ?
It is an analog pot, so logically we read a voltage and digitize it,
then report it to the MCU.
Thanks for pointing me in the right direction, Guy. ?Now I know the
probable true reason for the annoying "cant talk to the RX controls while

in TX"

Post by Luis V. Romero
behavior. ?A hybrid Parallel/Serial signal bus.
I'm afraid I pushed paper from the left side to the right side of a
desk for way too long.
-lu-
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[Jim Brown]

Post by Luis V. Romero
What I call "punching holes in the audio" is exactly that... If you have a
short loud sound followed by a soft one, part of the soft one goes away due
to the attack/decay ratio

YES! This is a very common problem generated by low frequency sounds like
P-popping (which is really a blast of air from your mouth hitting mic
diaphram when you pronounce a P-sound), and is a major reason why it's
important to roll off the low end before it hits the compressor. It's also a
big reason why multi-band compression is such a good idea, as Tom was
talking about.

73,

Jim K9YC

[Jim Brown]

Post by Lyle Johnson
The K3 Mic algorithm applies a fast attack, slow decay gain control loop
after the Tx Equalizer.

AFTER the TX Eq is critical, and you've done it right.

73,

Jim K9YC