W2IHY 8 band equalizer and EQ Plus

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W2IHY 8 band equalizer and EQ Plus

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" <[hidden email]>
> To: <"Elecraft List <elecraft"@mailman.qth.net>
> Sent: Tuesday, April 27, 2010 9:56 AM
> Subject: Re: [Elecraft] W2IHY 8 band equalizer and EQ Plus
>
>
>> 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" <[hidden email]>
>> Subject: Re: [Elecraft] W2IHY 8 band equalizer and EQ Plus
>> To: "Elecraft List" <[hidden email]>
>> Message-ID: <[hidden email]>
>> Content-Type: text/plain; charset="us-ascii"
>>
>> On Mon, 26 Apr 2010 16:18:01 +0000, Lance Collister wrote:
>>
>>> 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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>>
>


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Re: W2IHY 8 band equalizer and EQ Plus

Lu Romero - W4LT
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" <[hidden email]>
To: <[hidden email]>, <"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
>
>



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Re: W2IHY 8 band equalizer and EQ Plus

Jim Brown-10
On Tue, 27 Apr 2010 22:49:14 -0400, Lu Romero wrote:

>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


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Re: W2IHY 8 band equalizer and EQ Plus

Joe Subich, W4TV-4


 > 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

On 4/28/2010 12:29 AM, Jim Brown wrote:

> On Tue, 27 Apr 2010 22:49:14 -0400, Lu Romero wrote:
>
>> 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
>

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Re: W2IHY 8 band equalizer and EQ Plus

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. :-)

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Re: W2IHY 8 band equalizer and EQ Plus

P.B. Christensen
In reply to this post by Joe Subich, W4TV-4
> 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
 

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Re: W2IHY 8 band equalizer and EQ Plus

Guy, K2AV
In reply to this post by Lu Romero - W4LT
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.

> 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!
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Re: W2IHY 8 band equalizer and EQ Plus

Don Wilhelm-4
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

Guy Olinger K2AV wrote:

> 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.
>
>  
>> 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!
>>    
>
>  
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Re: W2IHY 8 band equalizer and EQ Plus

Lu Romero - W4LT
In reply to this post by Bob McGraw - K4TAX
Guys... Hold on...

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

>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.

>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!

>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.

>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.

>A DSP version can have options.  

Exactly!  This is just one more option!

>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.

>
>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-


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Re: W2IHY 8 band equalizer and EQ Plus

Lu Romero - W4LT
In reply to this post by Bob McGraw - K4TAX
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 <[hidden email]>
To: Guy Olinger K2AV <[hidden email]>
Cc: [hidden email], [hidden email]
Subject: Re: [Elecraft] W2IHY 8 band equalizer and EQ Plus
Date: Wed, 28 Apr 2010 14:14:44 -0400

>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
>
>Guy Olinger K2AV wrote:
>> 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.
>>
>>  
>>> 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!
>>>    
>>
>>    
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Re: W2IHY 8 band equalizer and EQ Plus

W8JI
In reply to this post by Lu Romero - W4LT
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

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Re: W2IHY 8 band equalizer and EQ Plus

Guy, K2AV
In reply to this post by Lu Romero - W4LT
I personally have just as much trouble as anyone else remembering to
think of the K3 as digital.

On Wed, Apr 28, 2010 at 4:03 PM, Lu Romero <[hidden email]> wrote:
> 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.
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Re: W2IHY 8 band equalizer and EQ Plus

Lu Romero - W4LT
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-




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Re: W2IHY 8 band equalizer and EQ Plus

Don Wilhelm-4
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

Luis V. Romero wrote:

> 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/
> ______________________________________________________________
> Elecraft mailing list
> Home: http://mailman.qth.net/mailman/listinfo/elecraft
> Help: http://mailman.qth.net/mmfaq.htm
> Post: mailto:[hidden email]
>
> This list hosted by: http://www.qsl.net
> Please help support this email list: http://www.qsl.net/donate.html
>
>  
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Re: W2IHY 8 band equalizer and EQ Plus

Mike Scott-7
In reply to this post by Bob McGraw - K4TAX
>>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


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Re: W2IHY 8 band equalizer and EQ Plus

Guy, K2AV
In reply to this post by Lu Romero - W4LT
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.


On Wed, Apr 28, 2010 at 11:03 PM, Luis V. Romero <[hidden email]> wrote:

> 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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Re: W2IHY 8 band equalizer and EQ Plus

Lu Romero - W4LT
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: [hidden email] [mailto:[hidden email]] On Behalf Of Guy Olinger
K2AV
Sent: Thursday, April 29, 2010 9:30 PM
To: [hidden email]
Cc: [hidden email]; [hidden email]
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.


On Wed, Apr 28, 2010 at 11:03 PM, Luis V. Romero <[hidden email]> wrote:

> 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-
>
>
>
>
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> (6.1.0.25 - 6.14830).
> http://www.pctools.com/free-antivirus/
>



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Re: W2IHY 8 band equalizer and EQ Plus

Jim Brown-10
On Fri, 30 Apr 2010 02:22:40 -0400, Luis V. Romero wrote:

>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


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Re: W2IHY 8 band equalizer and EQ Plus

KK7P
In reply to this post by Guy, 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


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Re: W2IHY 8 band equalizer and EQ Plus

Jim Brown-10
On Fri, 30 Apr 2010 05:57:26 -0700, Lyle Johnson wrote:

>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



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