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SSB transmit audio - Where's the punch?

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Jim Brown-10

Re: SSB transmit audio - Where's the punch?

On 10/4/2010 11:13 AM, Rob May wrote:
> I'll turn the compression up to 25.

That is the WRONG WAY to set compression. Rather, compression should be
set using the COMPRESSION section of the K3's meter (the one that's to
the left of the ALC indicator.

Grant said:

>Just guessing here, but 65 watts of "average" power out with 100 watts
>peak must sound pretty awful ..

You're absolutely right, Grant. Any good audio (or broadcast) engineer
knows that speech has a peak-to-average ratio on the order of 24dB. That
ratio can be reduced (using peak limiting and compression) by 10-14dB
before it starts sounding bad. Broadcasters use VERY sophisticated signal
processing systems to reduce that ratio by another 3-4 dB. Many years ago,
I sold some of this gear to broadcasters. Cost was in the $5K range in the
early80s. One of the common techniques is to split the spectruminto
multiple frequency bands and process each band separately, then combine
them. This only works with full bandwidth audio (that is, 20-20,000 Hz);
the bandwidth we transmit would fit into one of those individually
processed bands.

73, Jim Brown K9YC

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Guy, K2AV

Re: SSB transmit audio - Where's the punch?

In reply to this post by Jim Brown-10

A lot of the "punch" on older rigs is because the low end rolloff on
TX for communications audio IS IN THE TX components and cannot be
changed or "optioned" out. The K3, to satisfy all the vocal desirers
of options, lets the user set everything, so that every body can have
it his own way, from "I want my beautiful deep bass voice to be heard
on ESSB", to contesters' "I just want my highs out there on the power
peaks, I ONLY care about maximum QSO's, fidelity be damned".

In the old analog rigs, all that stuff was wired in, one resistor and
capacitor at a time, and the choice implied by the discrete components
used was THE choice, PERIOD. I note that some of the rigs quoted
earlier are in that collection. What was done in those was to favor a
highs-emphasized TX audio, with as little distortion as possible.
SOME clipping helped with average power. ESSB advocates should note
that NONE of those emphasized the bass, unless someone went in and
monkeyed with the discrete components, or put a banded preamp between
the mic and the rig.

What that state of affairs did was ENFORCE a defacto communications
audio default, very soft on bass and hard on highs.

Now you have K3 users with no clue about how to set the TX eq and
clipping level OPTIONS to get the IDENTICAL shape to their voices on a
K3 as the other rigs. There actually is a clarity advantage to the
K3's clipping method IF you know where to set all the options.

The problem is that in the bright new digital world, with options to
satisfy every conceivable preference, ONE HAS TO KNOW HOW TO SET THE
OPTIONS AND LEVELS TO GET WHAT HE WANTS. Add that to NOBODY EVER
WANTS TO READ THE MANUAL. (I, personally am no better than anyone else
here, I HATE reading manuals.) With the combo you get complaints that
"My K3 is broken" because the user doesn't know that's an option and
he has to set it his way in that menu. Likely perceived awful and
confusing, because to understand the menu you have to read the manual.
And there are so many optional behaviors that keeping up the manual
is a real piece of work, and requires the most talented of technical
writers to explain it in a straight-forward effective manner.

This is not a peculiarity with a K3. N1MM has that problem because of
the huge number of options, as does all the MicroHam stuff, which
serve a very wide audience IF the users understand the options. I
pretty much suffered brain damage learning MM logger. Microham was
better because I had W4TV. K3 was easier yet because of the reflector.
Flexibility generates confused digital options newbies all over the
place in all kinds of pursuits. The universal curse of the age of
digital options freedom.

These days RTFM is really the only way out. Unless someone who has
the time, inclination, and the sharp knowledge of all the options,
sets up a utility which sets a spectrum of options based upon older
rigs.

Since the version D DSP board, this actually seems possible. Once you
learn all the diddles, the K3 is marvelous. I have its RX sounding
like my 75A3, IF I use a good speaker. ALL my computer speakers turn
out to be crap beside my old-time Acoustic Research bookshelf speaker.
If I run my 75A3 to the computer speakers, the 75A3 sounds like crap.
Of course the sonorous old AR 8 ohm needs a 10 watt audio stage or an
amp to drive it because it is so brutally inefficient.
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dj7mgq

Re: SSB transmit audio - Where's the punch?

In reply to this post by Nate Bargmann

> You're absolutely correct, David. Some people are just reluctant to
> get their diaphram into the act and speak with confidence, not just

Full ack.

Go to a news studio or most other TV studios, and listen to with which
volume and dynamic style is spoken. I think a lot of Hams would be very
surprised.

vy 73 de toby

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Joe Subich, W4TV-4

Re: SSB transmit audio - Where's the punch?

In reply to this post by Guy, K2AV

> Now you have K3 users with no clue about how to set the TX eq and
> clipping level OPTIONS to get the IDENTICAL shape to their voices on a
> K3 as the other rigs. There actually is a clarity advantage to the
> K3's clipping method IF you know where to set all the options.

How very true ...

As K9YC (and others) has often advised, start by setting the first two
(LF) bands of the TX Eq to -16 and set the third (200 Hz) to - 9dB or
lower. Set the top three bands for at least 3 dB/octave (+3, +5, +6
dB) boost ... that works well with a mic that has some natural high
frequency boost like the HC-4. For flatter mics (like the new HC-6
or a CM-500) use more high frequency boost (between +6. +10, +12 dB
and +9, +16, +16 dB) to provide clarity. Since the human voice has
little energy in the 600 - 1200 Hz band, I like to add a bit of a
"notch" in the middle (-6dB at 800 Hz) ... cutting that band helps
to reduce background noise without impacting voice quality.

With reasonable adjustments to enhance the frequencies important for
communications (vs. some "golden ear" belief in a bandwidth more
appropriate to classical music), reducing the power wasted in the
lower octaves that do not contribute to enunciation, and 10 to 15 dB
of clipping, the K3 can more than hold it own while remaining very
clean compared to the Yaecomwood rigs that drive the PA into clipping
in order to generate ALC!

73,

... Joe, W4TV

On 10/5/2010 2:23 PM, Guy Olinger K2AV wrote:

> A lot of the "punch" on older rigs is because the low end rolloff on
> TX for communications audio IS IN THE TX components and cannot be
> changed or "optioned" out. The K3, to satisfy all the vocal desirers
> of options, lets the user set everything, so that every body can have
> it his own way, from "I want my beautiful deep bass voice to be heard
> on ESSB", to contesters' "I just want my highs out there on the power
> peaks, I ONLY care about maximum QSO's, fidelity be damned".
>
> In the old analog rigs, all that stuff was wired in, one resistor and
> capacitor at a time, and the choice implied by the discrete components
> used was THE choice, PERIOD. I note that some of the rigs quoted
> earlier are in that collection. What was done in those was to favor a
> highs-emphasized TX audio, with as little distortion as possible.
> SOME clipping helped with average power. ESSB advocates should note
> that NONE of those emphasized the bass, unless someone went in and
> monkeyed with the discrete components, or put a banded preamp between
> the mic and the rig.
>
> What that state of affairs did was ENFORCE a defacto communications
> audio default, very soft on bass and hard on highs.
>
> Now you have K3 users with no clue about how to set the TX eq and
> clipping level OPTIONS to get the IDENTICAL shape to their voices on a
> K3 as the other rigs. There actually is a clarity advantage to the
> K3's clipping method IF you know where to set all the options.
>
> The problem is that in the bright new digital world, with options to
> satisfy every conceivable preference, ONE HAS TO KNOW HOW TO SET THE
> OPTIONS AND LEVELS TO GET WHAT HE WANTS. Add that to NOBODY EVER
> WANTS TO READ THE MANUAL. (I, personally am no better than anyone else
> here, I HATE reading manuals.) With the combo you get complaints that
> "My K3 is broken" because the user doesn't know that's an option and
> he has to set it his way in that menu. Likely perceived awful and
> confusing, because to understand the menu you have to read the manual.
> And there are so many optional behaviors that keeping up the manual
> is a real piece of work, and requires the most talented of technical
> writers to explain it in a straight-forward effective manner.
>
> This is not a peculiarity with a K3. N1MM has that problem because of
> the huge number of options, as does all the MicroHam stuff, which
> serve a very wide audience IF the users understand the options. I
> pretty much suffered brain damage learning MM logger. Microham was
> better because I had W4TV. K3 was easier yet because of the reflector.
> Flexibility generates confused digital options newbies all over the
> place in all kinds of pursuits. The universal curse of the age of
> digital options freedom.
>
> These days RTFM is really the only way out. Unless someone who has
> the time, inclination, and the sharp knowledge of all the options,
> sets up a utility which sets a spectrum of options based upon older
> rigs.
>
> Since the version D DSP board, this actually seems possible. Once you
> learn all the diddles, the K3 is marvelous. I have its RX sounding
> like my 75A3, IF I use a good speaker. ALL my computer speakers turn
> out to be crap beside my old-time Acoustic Research bookshelf speaker.
> If I run my 75A3 to the computer speakers, the 75A3 sounds like crap.
> Of course the sonorous old AR 8 ohm needs a 10 watt audio stage or an
> amp to drive it because it is so brutally inefficient.
> ______________________________________________________________
> 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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rlindzen

Re: SSB transmit audio - Where's the punch?

Where does TXG VCE fit into this brew?

Dick, WO1I, K3# 911

Quoting "Joe Subich, W4TV" <[hidden email]>:

>
> > Now you have K3 users with no clue about how to set the TX eq and
> > clipping level OPTIONS to get the IDENTICAL shape to their voices on a
> > K3 as the other rigs. There actually is a clarity advantage to the
> > K3's clipping method IF you know where to set all the options.
>
> How very true ...
>
> As K9YC (and others) has often advised, start by setting the first two
> (LF) bands of the TX Eq to -16 and set the third (200 Hz) to - 9dB or
> lower. Set the top three bands for at least 3 dB/octave (+3, +5, +6
> dB) boost ... that works well with a mic that has some natural high
> frequency boost like the HC-4. For flatter mics (like the new HC-6
> or a CM-500) use more high frequency boost (between +6. +10, +12 dB
> and +9, +16, +16 dB) to provide clarity. Since the human voice has
> little energy in the 600 - 1200 Hz band, I like to add a bit of a
> "notch" in the middle (-6dB at 800 Hz) ... cutting that band helps
> to reduce background noise without impacting voice quality.
>
> With reasonable adjustments to enhance the frequencies important for
> communications (vs. some "golden ear" belief in a bandwidth more
> appropriate to classical music), reducing the power wasted in the
> lower octaves that do not contribute to enunciation, and 10 to 15 dB
> of clipping, the K3 can more than hold it own while remaining very
> clean compared to the Yaecomwood rigs that drive the PA into clipping
> in order to generate ALC!
>
> 73,
>
> ... Joe, W4TV
>
>
> On 10/5/2010 2:23 PM, Guy Olinger K2AV wrote:
>> A lot of the "punch" on older rigs is because the low end rolloff on
>> TX for communications audio IS IN THE TX components and cannot be
>> changed or "optioned" out. The K3, to satisfy all the vocal desirers
>> of options, lets the user set everything, so that every body can have
>> it his own way, from "I want my beautiful deep bass voice to be heard
>> on ESSB", to contesters' "I just want my highs out there on the power
>> peaks, I ONLY care about maximum QSO's, fidelity be damned".
>>
>> In the old analog rigs, all that stuff was wired in, one resistor and
>> capacitor at a time, and the choice implied by the discrete components
>> used was THE choice, PERIOD. I note that some of the rigs quoted
>> earlier are in that collection. What was done in those was to favor a
>> highs-emphasized TX audio, with as little distortion as possible.
>> SOME clipping helped with average power. ESSB advocates should note
>> that NONE of those emphasized the bass, unless someone went in and
>> monkeyed with the discrete components, or put a banded preamp between
>> the mic and the rig.
>>
>> What that state of affairs did was ENFORCE a defacto communications
>> audio default, very soft on bass and hard on highs.
>>
>> Now you have K3 users with no clue about how to set the TX eq and
>> clipping level OPTIONS to get the IDENTICAL shape to their voices on a
>> K3 as the other rigs. There actually is a clarity advantage to the
>> K3's clipping method IF you know where to set all the options.
>>
>> The problem is that in the bright new digital world, with options to
>> satisfy every conceivable preference, ONE HAS TO KNOW HOW TO SET THE
>> OPTIONS AND LEVELS TO GET WHAT HE WANTS. Add that to NOBODY EVER
>> WANTS TO READ THE MANUAL. (I, personally am no better than anyone else
>> here, I HATE reading manuals.) With the combo you get complaints that
>> "My K3 is broken" because the user doesn't know that's an option and
>> he has to set it his way in that menu. Likely perceived awful and
>> confusing, because to understand the menu you have to read the manual.
>> And there are so many optional behaviors that keeping up the manual
>> is a real piece of work, and requires the most talented of technical
>> writers to explain it in a straight-forward effective manner.
>>
>> This is not a peculiarity with a K3. N1MM has that problem because of
>> the huge number of options, as does all the MicroHam stuff, which
>> serve a very wide audience IF the users understand the options. I
>> pretty much suffered brain damage learning MM logger. Microham was
>> better because I had W4TV. K3 was easier yet because of the reflector.
>> Flexibility generates confused digital options newbies all over the
>> place in all kinds of pursuits. The universal curse of the age of
>> digital options freedom.
>>
>> These days RTFM is really the only way out. Unless someone who has
>> the time, inclination, and the sharp knowledge of all the options,
>> sets up a utility which sets a spectrum of options based upon older
>> rigs.
>>
>> Since the version D DSP board, this actually seems possible. Once you
>> learn all the diddles, the K3 is marvelous. I have its RX sounding
>> like my 75A3, IF I use a good speaker. ALL my computer speakers turn
>> out to be crap beside my old-time Acoustic Research bookshelf speaker.
>> If I run my 75A3 to the computer speakers, the 75A3 sounds like crap.
>> Of course the sonorous old AR 8 ohm needs a 10 watt audio stage or an
>> amp to drive it because it is so brutally inefficient.
>> ______________________________________________________________
>> 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
>>
> ______________________________________________________________
> 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
>

Joe Subich, W4TV-4

Re: SSB transmit audio - Where's the punch?

In reply to this post by Jim Brown-10

Jim,

> This only works with full bandwidth audio (that is, 20-20,000 Hz);
> the bandwidth we transmit would fit into one of those individually
> processed bands.

Wes Stewart, N7WS had a very successful split band approach for amateur
use years ago ... it was butchered by Alpha (the Vomax) but three bands
(300 - 600 Hz, 1200 - 2000 Hz and 2000 - 3500 Hz) is a very good choice
for "communications/talk." By keeping each band less than one octave
in width, IMD and harmonic products from clipping can be minimized even
with analog techniques.

With modern DSP technology one can add phase scrambling (rotation),
per band noise gating, pre- and post-clipping equalization and use
half octave bands with minimal hardware cost to achieve very clean
compression/clipping with very tight dynamic range even in noisy
environments.

73,

... Joe, W4TV

On 10/5/2010 1:11 PM, Jim Brown wrote:

> On 10/4/2010 11:13 AM, Rob May wrote:
>> I'll turn the compression up to 25.
>
> That is the WRONG WAY to set compression. Rather, compression should be
> set using the COMPRESSION section of the K3's meter (the one that's to
> the left of the ALC indicator.
>
> Grant said:
>
>> Just guessing here, but 65 watts of "average" power out with 100 watts
>> peak must sound pretty awful ..
>
> You're absolutely right, Grant. Any good audio (or broadcast) engineer
> knows that speech has a peak-to-average ratio on the order of 24dB. That
> ratio can be reduced (using peak limiting and compression) by 10-14dB
> before it starts sounding bad. Broadcasters use VERY sophisticated signal
> processing systems to reduce that ratio by another 3-4 dB. Many years ago,
> I sold some of this gear to broadcasters. Cost was in the $5K range in the
> early80s. One of the common techniques is to split the spectruminto
> multiple frequency bands and process each band separately, then combine
> them. This only works with full bandwidth audio (that is, 20-20,000 Hz);
> the bandwidth we transmit would fit into one of those individually
> processed bands.
>
> 73, Jim Brown K9YC
>
> ______________________________________________________________
> 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
>

Lu Romero - W4LT

Re: SSB transmit audio - Where's the punch?

In reply to this post by WN3R, Dick

All:

I have no issues with lack of "punch" with the K3, and I am
considered an audio "dweeb" by some of my contesting
contemporaries.

On my former TS-850, I had "lotsa punch". My audio chain
consisted of the following:

Heil DX-4 headset
Behringer EuroRack 802 mixer with three band "classic" EQ
Radio Design Labs ST-GCA2 limiter/leveler/AGC
Behringer MDX-2600 Compressor/Gate
Home made precision T-Pad line to mic level interface
microHAM microKEYER

It made nice, crisp "razor sharp" audio that cut through
piles, if a little "dirty" at the edges. "Thin" sounding
but penetrating.

Now, I have the following audio chain with my K3:

Yamaha CM-500 headset
microHAM microKEYER 2

And I feel I have the same punch, but with a little
difference... I how have some minor bass "presence" at
around 300-400hz that I couldnt get with the DX4, thanks to
the very nice EQ built in to the K3 and the very broad
CM-500 capsule. And I have none of the "grunge" around the
edges that I had with the 850. The K3 is much cleaner.

I have the same peak to average ratios, as witnessed by my
scope waveforms, the same "penetration" and the same
"punch", but with a mellower, cleaner, less "thin" sound.

It has taken me six months of tweaking and the Yamaha
headset to get to to where I am 98% happy with K3 transmit
audio.

I have a very noisy shack environment, so there are some
caveats:

First, the DX4 is a "close talking" microphone. You have to
almost eat it for it to work right. The CM-500'ds Electret
capsule has a pronounced proximity effect. You HAVE to keep
it AT LEAST an inch from your mouth or it sounds mushy and
bassy. This, combined with its omnidirectional response and
the need to run mic gain relatively high to command good
peak to average ratios from the K3 processor, cause "much"
room noise (not a lot really, but Im picky) to find its way
to the transmitter.

Second, I could fix this with the AF GATE function, but the
gate has and issue (to me) that it does not have enough
hysteresis, that is, it drops RIGHT NOW, and when it drops,
it drops IMMEDIATELY off the cliff with a straight as a
ruler response. WHAM! Audio's GONE!

I have worked with the IHY product at the Multi-Multi I work
with (we have standardized on it for all positions now using
TenTec Orions) and I find both my old Behringer and Julius'
Gate to have a logarithmic ramping (the gate ation starts
slow and speeds up over time) plus there is some hysterisis
in the gate's response to instantaneous noise spikes, so
these two are not as "nervous sounding" as the K3 gate is.
As it is, the K3 gate responds so very quickly that it adds
a "crackle" to the audio because it partially triggers with
small noise spikes (lacks hysteresis and immediately turns
on and off). The effect is rather annoying and spoils the
gate's effectiveness. But the processor works so well now
that I will put up with the gate till Lyle can get around to
it.

Other than those niggling comments, Im a happy camper with a
much simpler audio chain. Simple is better!

The key to remember is that we are transmitting into a very
noisy medium (the HF bands during a contest). You need to
do two major things: Sacrifice wide "dynamic range" in your
transmit waveform for effective communications and
concentrate your available transmit power into the frequency
bands that will do you the most good for inteligibility.
Bass below 300hz is your enemy! If you do this while
maintaining the highest possible peak to average ratio with
the lowest distortion figures your transmitter can muster,
you will be heard and, most importantly, you will be
UNDERSTOOD. And it helps to have proper enounciation skills
(ask any ATC or pilot!)

This is not as simple to do as it sounds. It takes
discipline with mic technique and voice projection BEFORE
you apply any digital processor "magic". Garbage in -
Garbage out!

Then there's VooDoo... I always put a 3x5 card on the
monitor that says "I AM LOUD". If you think you are, you
are!

Lu Romero
W4LT - #3192

Message: 10
Date: Mon, 4 Oct 2010 20:52:41 -0500
From: "Jim Miller KG0KP" <[hidden email]>
Subject: Re: [Elecraft] SSB transmit audio - Where's the
punch?
To: <[hidden email]>
Message-ID: <D6C7D2F59E51421A9944095DE3F9D62D@HMJM1800>
Content-Type: text/plain; format=flowed;
charset="iso-8859-1";
reply-type=original

Don't get it too high. Higher is NOT necessarily better. I
hear station in
contests that are very strong and they are almost impossible
to understand
because they have their compression WAY to high. Turning it
down would make
them much better.

I just made 1752 contacts in a special event and got many
unsolicited
comments on my audio being very good, excellent, etc. I was
running the HC4
and the mic gain at 25 and the compression at 25. My voice
is relatively
low.

73, de Jim KG0KP

----- Original Message -----
From: <[hidden email]>
To: "Dick, WN3R" <[hidden email]>
Cc: <[hidden email]>
Sent: Monday, October 04, 2010 1:11 PM
Subject: Re: [Elecraft] SSB transmit audio - Where's the
punch?

> Dick,
>
> You use "some compression"? Crank it up!
>
> Dump the external EQ, and use either the HC-4 or HC-5 (no
bias) and crank
> the TX EQ bands up or down until it sounds nasty enough.
You'll generally
> reduce the lows and increase the highs.
>
> Crank up the compression and it'll sound as bad as the
Yaesu does.
>
> Your fellow contester's claim reveals his lack of
knowledge about the K3
> as it can be adjusted to sound any way you want, with
nearly any
> microphone.
>
> Generally speaking, if it doesn't sound right, crank it
up.
>
> 73,
>
> Bob W5OV
>
> P.S. I know that audio purists will not like the above
advice. But, you
> are looking to bust pileups, not get an audio quality
award.
>
>
>> I need some advice in how to get the same punch my fellow
contester gets
>> on
>> his FT-1000MP. He claims the K-3 audio will never be as
good as Yaesu,
>> Kenwood, or ICOM.
>>
>>
>>
>> Yesterday we spent several hours between ourselves and
several European
>> stations tuning and testing. I never got a bad audio
report, BUT I never
>> got a report I had the same PUNCH!!! He reduced power
and still
>> overtalked
>> me.
>>
>>
>>
>> I have tried HC-4, HC-5, and HC-6 using W2IHY eight band
EQ with the

>> EQPlus
>> with some compression from the K-3.
>>
>>
>>
>> I need some ideas before the next contest.
>>
>>
>>
>> 73, Dick, WN3R
>>
>>
>>

Jim Brown-10

Re: SSB transmit audio - Where's the punch?

On 10/5/2010 2:09 PM, Lu Romero wrote:
> First, the DX4 is a "close talking" microphone. You have to
> almost eat it for it to work right. The CM-500'ds Electret
> capsule has a pronounced proximity effect. You HAVE to keep
> it AT LEAST an inch from your mouth or it sounds mushy and
> bassy. This, combined with its omnidirectional response and
> the need to run mic gain relatively high to command good
> peak to average ratios from the K3 processor, cause "much"
> room noise (not a lot really, but Im picky) to find its way
> to the transmitter.

The CM500 is NOT an omni-directional mic -- you've noted that it has
proximity effect, and that's BECAUSE it is a cardioid (directional) mic.
As to the proximity effect -- you can compensate that the same way that
virtually ALL vocal mics are compensated within the mic -- with
additional low end rolloff.

73, Jim K9YC
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Ignacy

Re: SSB transmit audio - Where's the punch?

I used to have TS-850 and FT-1000MKF. The processor in TS-850 was very good and in MKF not as strong but audio was nice. If you read some reports, ALC in 1000MP and probably other radios in the series is too fast and produces lots of IMD at high processing. ALC in K3 is very slow and thus avoids distortion but also the extra (useless) power that goes with distortion.

K2 IMHO beats both radios. I am amazed that even cranking COMP to 30 does not produce distortion but brings extra contacts.

As mentioned before, set equalization first, MIC gain second, and crank up COMP. I use COMP of 15 in casual conditions, 20 as a norm, 25 for normal DXing or contesting, and 30 when things go tough. Never a bad report.

Ignacy

n7ws

Re: SSB transmit audio - Where's the punch?

In reply to this post by Joe Subich, W4TV-4

With that introduction I guess I have to respond.

Actually, credit for the original concept belongs to Walt Schreuer, K1YZW. He developed the "SPB-3" sold by Maximilian Associates in about 1976. No schematics were provided, of course, so I and my late friend Jim Metzer, W7TKR, produced our own way to do it.

See: http://k6mhe.com/n7ws/S_PROC.pdf

Our simplified method for accommodating the phase shift through filters does lead to more passband ripple than more elegant phase correction, but operationally it's never been an issue.

IMHO, the split-band approach is better than rf clipping. Split-band clipping was the choice of the engineers of one of the arguably best sounding radios around, the Kenwood TS-870.

I personally know of several hundred of these that were built around the world, not counting however many kits were sold after the magazine editor founded a kit company on my design, the RADIOKIT DX-1.

Wes N7WS

--- On Tue, 10/5/10, Joe Subich, W4TV <[hidden email]> wrote:

Jim,

> This only works with full bandwidth audio (that is, 20-20,000 Hz);
> the bandwidth we transmit would fit into one of those individually
> processed bands.

Wes Stewart, N7WS had a very successful split band approach for amateur
use years ago ... it was butchered by Alpha (the Vomax) but three bands
(300 - 600 Hz, 1200 - 2000 Hz and 2000 - 3500 Hz) is a very good choice
for "communications/talk." By keeping each band less than one octave
in width, IMD and harmonic products from clipping can be minimized even
with analog techniques.

With modern DSP technology one can add phase scrambling (rotation),
per band noise gating, pre- and post-clipping equalization and use
half octave bands with minimal hardware cost to achieve very clean
compression/clipping with very tight dynamic range even in noisy
environments.

73,

... Joe, W4TV

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Lu Romero - W4LT

Re: SSB transmit audio - Where's the punch?

In reply to this post by WN3R, Dick

Jim:

Its very mildly cardioid then! :) This aint no shotgun, Jim!

I choose not to compensate for the boominess with EQ and not eat this mic
like an HC4 as it adds to the distortion I hear in the monitor and see on
the scope. I would rather run the mic gain a little hotter and put up with
the room tone than deal with the boominess electronically. I can always
gate the room tone. I have always tried not to "fix stuff in the mix" but
I'm severely crippled with a very poor operating environment for sound.
Then again, this is Ham Radio, not work!

And with 12 volt finals, I need all the signal cleanliness I can get :)

Of course, you are right, but I'm walking a fine line between distortion and
punch. I find the fatter foam windscreen from an Audio Technica Pro8HEx mic
helps me place the CM-500 capsule at just the right distance from my mouth
for me.

-lu-

Message: 27
Date: Tue, 05 Oct 2010 14:25:59 -0700
From: Jim Brown <[hidden email]>
Subject: Re: [Elecraft] SSB transmit audio - Where's the punch?
To: [hidden email]
Message-ID: <[hidden email]>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed

On 10/5/2010 2:09 PM, Lu Romero wrote:
> First, the DX4 is a "close talking" microphone. You have to
> almost eat it for it to work right. The CM-500'ds Electret
> capsule has a pronounced proximity effect. You HAVE to keep
> it AT LEAST an inch from your mouth or it sounds mushy and
> bassy. This, combined with its omnidirectional response and
> the need to run mic gain relatively high to command good
> peak to average ratios from the K3 processor, cause "much"
> room noise (not a lot really, but Im picky) to find its way
> to the transmitter.

The CM500 is NOT an omni-directional mic -- you've noted that it has
proximity effect, and that's BECAUSE it is a cardioid (directional) mic.
As to the proximity effect -- you can compensate that the same way that
virtually ALL vocal mics are compensated within the mic -- with
additional low end rolloff.

73, Jim K9YC

No virus found in this outgoing message
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Tim Tucker

Re: SSB transmit audio - Where's the punch?

I wanted to make sure and follow up my AVG power output comments that I made
earlier in this thread because I spent some more time testing. Several
private emails confirmed my suspicion that others were able to achieve
better results than I was seeing.

Tonight after playing around with the rig some more, I was actually able to
get the AVG power output real close to 60 watts on on specific syllables or
words by playing with the audio response, audio gain input levels, etc.
Typical voice response produced about 40-45 watts AVG. This is considerably
better than I reported earlier, all things considered. What's different
about the K3 is that it's AVG power output seems very sensitive to a lot of
variables, especially your EQ settings. The K3 CMP setting quickly reached
a point of diminishing returns for me, unlike what others are reporting, but
you probably have to with what works for your voice/mic combination. I just
can't run that much compression before the audio sounds like trash to me.

This definitely isn't a plug and play rig. It is probably easier to achieve
certain results with other rigs, but what is nice about the K3 is the
ability to tweak almost everything to your heart's content.

On Tue, Oct 5, 2010 at 7:36 PM, Luis V. Romero <[hidden email]> wrote:

> Jim:
>
> Its very mildly cardioid then! :) This aint no shotgun, Jim!
>
> I choose not to compensate for the boominess with EQ and not eat this mic
> like an HC4 as it adds to the distortion I hear in the monitor and see on
> the scope. I would rather run the mic gain a little hotter and put up with
> the room tone than deal with the boominess electronically. I can always
> gate the room tone. I have always tried not to "fix stuff in the mix" but
> I'm severely crippled with a very poor operating environment for sound.
> Then again, this is Ham Radio, not work!
>
> And with 12 volt finals, I need all the signal cleanliness I can get :)
>
> Of course, you are right, but I'm walking a fine line between distortion
> and
> punch. I find the fatter foam windscreen from an Audio Technica Pro8HEx
> mic
> helps me place the CM-500 capsule at just the right distance from my mouth
> for me.
>
> -lu-
>
>
> Message: 27
> Date: Tue, 05 Oct 2010 14:25:59 -0700
> From: Jim Brown <[hidden email]>
> Subject: Re: [Elecraft] SSB transmit audio - Where's the punch?
> To: [hidden email]
> Message-ID: <[hidden email]>
> Content-Type: text/plain; charset=ISO-8859-1; format=flowed
>
> On 10/5/2010 2:09 PM, Lu Romero wrote:
> > First, the DX4 is a "close talking" microphone. You have to
> > almost eat it for it to work right. The CM-500'ds Electret
> > capsule has a pronounced proximity effect. You HAVE to keep
> > it AT LEAST an inch from your mouth or it sounds mushy and
> > bassy. This, combined with its omnidirectional response and
> > the need to run mic gain relatively high to command good
> > peak to average ratios from the K3 processor, cause "much"
> > room noise (not a lot really, but Im picky) to find its way
> > to the transmitter.
>
> The CM500 is NOT an omni-directional mic -- you've noted that it has
> proximity effect, and that's BECAUSE it is a cardioid (directional) mic.
> As to the proximity effect -- you can compensate that the same way that
> virtually ALL vocal mics are compensated within the mic -- with
> additional low end rolloff.
>
> 73, Jim K9YC
>
> No virus found in this outgoing message
> Checked by PC Tools AntiVirus (6.1.0.25 - 6.14880).
> http://www.pctools.com/free-antivirus/
>
>
> ______________________________________________________________
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>
> This list hosted by: http://www.qsl.net
> Please help support this email list: http://www.qsl.net/donate.html
>

Jim Brown-10

Re: SSB transmit audio - Where's the punch?

On 10/5/2010 10:39 PM, Tim Tucker wrote:
> Typical voice response produced about 40-45 watts AVG. This is considerably
> better than I reported earlier, all things considered.

I would be very cautious about saying it's BETTER until you get some
critical audio reports on the air. You could easily have great meter
readings and lousy sounding audio.

73, Jim K9YC
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Don Wilhelm-4

Re: SSB transmit audio - Where's the punch?

In reply to this post by Tim Tucker

Tim,

As far as the response of the K3 is concerned, I refer you to Guy
Olinger's (K2AV) recent posts. The K3 is a highly configurable
transceiver, and "Read The Fine Manual" is an important step. If after
reading, you have problems understanding the implications, definitely
ask here on the reflector, and you will get an informed response.

When the questions are more like "tell me how to do ----", I often feel
like just replying RTFM, because it is obvious that the poster has not
even cracked the manual and just tried to do special things without
trying to understand how things work.

The K3 with its default settings is good for the average types of
operation, but for those who want to "do something special", it is
important to RTFM first and then ask questions - failure to do that will
likely lead to not understanding the responses to the questions, and
further confusion when what is suggested is not what the poster expects.

In other words, vary from the default setting only with a valid reason
to make that change. Many changes from the defaults are good for
certain situations, but will not produce the desired results for other
situations.

The K3 is highly configurable, but modify the default configuration with
an informed eye, and see if the results do what you desire rather than
accepting someone else's changes without question. I repeat, RTFM is
important for any modern transceiver with configuration menus - the K3
is no exception.

73,
Don W3FPR

On 10/6/2010 1:39 AM, Tim Tucker wrote:
>
> This definitely isn't a plug and play rig. It is probably easier to achieve
> certain results with other rigs, but what is nice about the K3 is the
> ability to tweak almost everything to your heart's content.
>
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David Woolley (E.L)

Re: SSB transmit audio - Where's the punch?

In reply to this post by Jim Brown-10

This (multiband compression) is essentially what digital hearing aids
do. In their case, the aim is to get the dynamic range into the gap
between the threshold of hearing and threshold of pain, both of which
tend to converge towards each other in hearing loss.

One figure of merit for such aids is the number of compression channels.

Beyond this, modern hearing aids are essentially sophisticated DSP NR
devices, in ham radio terms.

Top quote through list policy only.

Jim Brown wrote:

> early80s. One of the common techniques is to split the spectruminto
> multiple frequency bands and process each band separately, then combine
> them. This only works with full bandwidth audio (that is, 20-20,000 Hz);
> the bandwidth we transmit would fit into one of those individually
> processed bands.

--
David Woolley
"we do not overly restrict the subject matter on the list, and we
encourage postings on a wide range of amateur radio related topics"
List Guidelines <http://www.elecraft.com/elecraft_list_guidelines.htm>
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Terry Schieler

Re: SSB transmit audio - Where's the punch?

In reply to this post by dj7mgq

Which is further acknowledged by the fact that singers seldom record while
sitting down. More diaphragm punch when standing (especially after lunch!)

Terry, W0FM

-----Original Message-----
From: Toby Deinhardt [mailto:[hidden email]]
Sent: Tuesday, October 05, 2010 1:31 PM
To: [hidden email]
Subject: Re: [Elecraft] SSB transmit audio - Where's the punch?

> You're absolutely correct, David. Some people are just reluctant to
> get their diaphram into the act and speak with confidence, not just

Full ack.

Go to a news studio or most other TV studios, and listen to with which
volume and dynamic style is spoken. I think a lot of Hams would be very
surprised.

vy 73 de toby

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Guy, K2AV

Re: SSB transmit audio - Where's the punch?

In reply to this post by Joe Subich, W4TV-4

On Tue, Oct 5, 2010 at 3:43 PM, Joe Subich, W4TV <[hidden email]> wrote:
>
....snip... For flatter mics (like the new HC-6
> or a CM-500) use more high frequency boost (between +6. +10, +12 dB
> and +9, +16, +16 dB) to provide clarity. Since the human voice has
> little energy in the 600 - 1200 Hz band,

Except for the lady folks who seem to START in that range. What
happens in so many female voices I hear (including broadcast stations)
is that high central tonality dominates to the point of overwhelming
the silibant range and 600-1200 NEEDS to be seriously suppressed to
let the silibants emerge. I'm not sure 6 dB is enough for them.

73, Guy.

> I like to add a bit of a
> "notch" in the middle (-6dB at 800 Hz) ... cutting that band helps
> to reduce background noise without impacting voice quality.
>
> With reasonable adjustments to enhance the frequencies important for
> communications (vs. some "golden ear" belief in a bandwidth more
> appropriate to classical music), reducing the power wasted in the
> lower octaves that do not contribute to enunciation, and 10 to 15 dB
> of clipping, the K3 can more than hold it own while remaining very
> clean compared to the Yaecomwood rigs that drive the PA into clipping
> in order to generate ALC!
>
> 73,
>
> ... Joe, W4TV7

Guy, K2AV

Re: SSB transmit audio - Where's the punch?

In reply to this post by Lu Romero - W4LT

On Tue, Oct 5, 2010 at 10:36 PM, Luis V. Romero <[hidden email]> wrote:
> Jim:

> Of course, you are right, but I'm walking a fine line between distortion and
> punch. I find the fatter foam windscreen from an Audio Technica Pro8HEx mic
> helps me place the CM-500 capsule at just the right distance from my mouth
> for me.

And drool or dripping spittle in the keyboard is not an issue? :>)

73, Guy.

>
> -lu-
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Eric Swartz - WA6HHQ

Re: SSB transmit audio - Where's the punch? [END of Thread]

Administrator

Time to end this thread as we are now hitting the point of diminishing
returns ;-)

73, Eric
Elecraft List modulator
---

On 10/6/2010 8:45 AM, Guy Olinger K2AV wrote:
> On Tue, Oct 5, 2010 at 10:36 PM, Luis V. Romero<[hidden email]> wrote:
>> Of course, you are right, but I'm walking a fine line between distortion and
>> punch. I find the fatter foam windscreen from an Audio Technica Pro8HEx mic
>> helps me place the CM-500 capsule at just the right distance from my mouth
>> for me.
> And drool or dripping spittle in the keyboard is not an issue? :>)
> 73, Guy.
>
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Lu Romero - W4LT

Re: SSB transmit audio - Where's the punch?

In reply to this post by WN3R, Dick

Ha!

That's another reason for the bigger foam windscreen, Guy,
its more absorbent! :)

-lu-

----- Original Message Follows -----
From: Guy Olinger K2AV <[hidden email]>
To: [hidden email]
Cc: [hidden email]
Subject: Re: [Elecraft] SSB transmit audio - Where's the
punch?
Date: Wed, 6 Oct 2010 11:45:38 -0400

>On Tue, Oct 5, 2010 at 10:36 PM, Luis V. Romero
>> <[hidden email]> wrote: Jim:
>
>> Of course, you are right, but I'm walking a fine line
>> between distortion and punch. I find the fatter foam
>> windscreen from an Audio Technica Pro8HEx mic helps me
>place the CM-500 capsule at just the right distance from my
>> mouth for me.
>
>And drool or dripping spittle in the keyboard is not an
>issue? :>)
>
>73, Guy.
>
>>
>> -lu-

123