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optimizing recorded audio

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Tony Estep

optimizing recorded audio

Some have the Elecraft audio recorder to send their call in a contest or
pileup, but others no doubt use their control software to send recorded
audio. I use MixW (I've plugged it before on here -- great program), but all
control software can send a .wav file to your K2 or K3. But don't just make
a .wav file and play it back; process it for maximum talk power.

I have been using Audacity to record and process these files. It's a
wonderful program, free and open source. Get it from:

http://audacity.sourceforge.net/

being sure you get the right version for your OS.

After you get it going (very simple and clearly explained by a quickstart
help file), record your message. Then you can optimize it to taste, and here
are the tricks I've found to work for me:

First, compress it. Use moderate or light compression and repeat until you
notice some compression, but stop before it's really dense.

Then use the effect called "leveller." This will selectively compress voice
peaks and is extremely effective at making the audio punchy without seeming
to change the natural sound.

Finally, go to "change tempo." This can make your recording play faster or
slower without changing pitch. Don't change the beats, but rather trim the
time length of your recording. A little bit goes a long way. For example,
when I recorded my call the clip was about 3 seconds long. Cutting it to 2.6
seconds got rid of inter-syllabic pauses etc and made it sound urgent but
didn't introduce any unnatural sound.

Then export it as .wav, and you're ready to rock n roll in the phone
pileups.

73,
Tony KT0NY
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Jim Brown-10

Re: optimizing recorded audio

On 4/21/2011 4:51 PM, Tony Estep wrote:
> I have been using Audacity

Yes, that's good software. I use Adobe Audition, which I bought because
I'm a pro sound engineer. Gold Wave is another one.

> First, compress it. Use moderate or light compression and repeat until you
> notice some compression, but stop before it's really dense.
>
> Then use the effect called "leveller." This will selectively compress voice
> peaks and is extremely effective at making the audio punchy without seeming
> to change the natural sound.

This is NOT a good idea if you're using compression in the K3 (and you
should be using about 10dB) -- if the K3 is set right for your live mic,
it will be quite excessive on your pre-recorded material. Excessive
compression and leveling brings up room noise, which I hear FAR TOO MUCH
of during contests. It IS very important though to get the recorded
level right, and to use the same mic with your computer that you use
with the K3.

> Finally, go to "change tempo." This can make your recording play faster or
> slower without changing pitch. Don't change the beats, but rather trim the
> time length of your recording. A little bit goes a long way. For example,
> when I recorded my call the clip was about 3 seconds long. Cutting it to 2.6
> seconds got rid of inter-syllabic pauses etc and made it sound urgent but
> didn't introduce any unnatural sound.

Time compression tends to make voices sound artificial. Time
compression, or talking too fast with poor articulation, can make it
difficult for others to copy your call. This happens FAR too often
during contests! A better way is to record each message, listen to it
carefully, and do it again until you like it. I've learned to record
each phrase or word quickly but with good articulation with pauses
between each, then edit out the pauses. To do that, get the waveform on
the screen and carefully remove parts of the longer pauses between words
by highlighting the dead space with your mouse and hitting delete.
Moderation is the key -- don't remove ALL of the blank space, only part
of it. Listen and repeat if necessary.

73, Jim Brown K9YC
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Tony Estep

Re: optimizing recorded audio

On Thu, Apr 21, 2011 at 7:20 PM, Jim Brown <[hidden email]>wrote:

> On 4/21/2011 4:51 PM, Tony Estep wrote:
> ....This is NOT a good idea...tends to make voices sound artificial. ...
>
>
All I can say is, try it and you'll like the results.

Tony KT0NY
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Gary Gregory

Re: optimizing recorded audio

Personally I find using the K3 DVR is simple and the recorded audio heard by
other stations is also the same as when you answer their call...hence, no
confusion.

I agree with Jim emphatically, too many times we are subjected to absolutely
terrible audio that can make it dreadfully difficult to understand what is
being said and therefore time is wasted asking for repeats.

If operators spent as much time on their audio as they spend on 'other'
chores in their shack the results would be greatly appreciated by many.

Gosh, in the last contest we had to put up with the latest fad to annoy a
majority of folks...roger beeps...good grief...what next?...I can only
imagine.

73's
Gary

On 22 April 2011 10:57, Tony Estep <[hidden email]> wrote:

> On Thu, Apr 21, 2011 at 7:20 PM, Jim Brown <[hidden email]
> >wrote:
>
> > On 4/21/2011 4:51 PM, Tony Estep wrote:
> > ....This is NOT a good idea...tends to make voices sound artificial. ...
> >
> >
> All I can say is, try it and you'll like the results.
>
> Tony KT0NY
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>

--

VK4FD - Motorhome Mobile
Elecraft Equipment
K3 #679, KPA-500 #018
Living the dream!!!
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Guy, K2AV

Re: optimizing recorded audio

In reply to this post by Tony Estep

Tried compressing with the audio program pretty much per Tony's
procedure and got bad reports and too many requests for repeats. Now
I keep mic audio untouched in the PC except for level and trimmed for
time on the PC. Now I let the K3 do all the audio processing, and can
mix in live audio with no change in quality.

For me anyway, the reports with the K3 doing the processing are far better.

73, Guy.

On Thu, Apr 21, 2011 at 8:57 PM, Tony Estep <[hidden email]> wrote:

> On Thu, Apr 21, 2011 at 7:20 PM, Jim Brown <[hidden email]>wrote:
>
>> On 4/21/2011 4:51 PM, Tony Estep wrote:
>> ....This is NOT a good idea...tends to make voices sound artificial. ...
>>
>>
> All I can say is, try it and you'll like the results.
>
> Tony KT0NY
> ______________________________________________________________
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Gary Gregory

Re: optimizing recorded audio

Tony,

Whilst you may have found nirvana with your audio, I tried your idea and as
close as I came, two other K3 owners who know me very well agreed we could
not get the same quality from the PC.

There are many variables in play here also and I appreciate that aspect. I
hope you are blessed with a better 'natural' voice than I have...:-)

I went off topic somewhat, but, hey you gotta get on the soapbox
sometime....:-)

Happy Easter guys

15 and 10M has been open to the US from VK...hint hint?

73's
Gary

On 22 April 2011 11:47, Guy Olinger K2AV <[hidden email]> wrote:

> Tried compressing with the audio program pretty much per Tony's
> procedure and got bad reports and too many requests for repeats. Now
> I keep mic audio untouched in the PC except for level and trimmed for
> time on the PC. Now I let the K3 do all the audio processing, and can
> mix in live audio with no change in quality.
>
> For me anyway, the reports with the K3 doing the processing are far better.
>
> 73, Guy.
>
> On Thu, Apr 21, 2011 at 8:57 PM, Tony Estep <[hidden email]> wrote:
> > On Thu, Apr 21, 2011 at 7:20 PM, Jim Brown <[hidden email]
> >wrote:
> >
> >> On 4/21/2011 4:51 PM, Tony Estep wrote:
> >> ....This is NOT a good idea...tends to make voices sound artificial. ...
> >>
> >>
> > All I can say is, try it and you'll like the results.
> >
> > Tony KT0NY
> > ______________________________________________________________
> > 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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David Gilbert

Re: optimizing recorded audio

In reply to this post by Jim Brown-10

I fully agree, Jim. I don't know why anyone would want to run their
audio through two different compression schemes, or artificially alter
the timing. KT0NY's approach almost guarantees that his recorded clips
will sound different than his mic voice, which is a sure fire source of
potential confusion and possible error in a contest. It's a great way
to slow down your rates.

I record my voice clips to the computer, use Goldwave to make sure all
the separate files (CQ, report, etc) are at the same volume, and trim
dead spots. I keep a couple of reference recordings handy for
comparison any time I need to generate audio clips for a different
contest ... that way I am able to maintain the same gain and compression
settings on my K3 all of the time.

73,
Dave AB7E

On 4/21/2011 5:20 PM, Jim Brown wrote:

> On 4/21/2011 4:51 PM, Tony Estep wrote:
>> I have been using Audacity
> Yes, that's good software. I use Adobe Audition, which I bought because
> I'm a pro sound engineer. Gold Wave is another one.
>
>> First, compress it. Use moderate or light compression and repeat until you
>> notice some compression, but stop before it's really dense.
>>
>> Then use the effect called "leveller." This will selectively compress voice
>> peaks and is extremely effective at making the audio punchy without seeming
>> to change the natural sound.
> This is NOT a good idea if you're using compression in the K3 (and you
> should be using about 10dB) -- if the K3 is set right for your live mic,
> it will be quite excessive on your pre-recorded material. Excessive
> compression and leveling brings up room noise, which I hear FAR TOO MUCH
> of during contests. It IS very important though to get the recorded
> level right, and to use the same mic with your computer that you use
> with the K3.
>
>> Finally, go to "change tempo." This can make your recording play faster or
>> slower without changing pitch. Don't change the beats, but rather trim the
>> time length of your recording. A little bit goes a long way. For example,
>> when I recorded my call the clip was about 3 seconds long. Cutting it to 2.6
>> seconds got rid of inter-syllabic pauses etc and made it sound urgent but
>> didn't introduce any unnatural sound.
> Time compression tends to make voices sound artificial. Time
> compression, or talking too fast with poor articulation, can make it
> difficult for others to copy your call. This happens FAR too often
> during contests! A better way is to record each message, listen to it
> carefully, and do it again until you like it. I've learned to record
> each phrase or word quickly but with good articulation with pauses
> between each, then edit out the pauses. To do that, get the waveform on
> the screen and carefully remove parts of the longer pauses between words
> by highlighting the dead space with your mouse and hitting delete.
> Moderation is the key -- don't remove ALL of the blank space, only part
> of it. Listen and repeat if necessary.
>
> 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
>

gm3sek

Re: optimizing recorded audio

In reply to this post by Jim Brown-10

Jim Brown wrote:

>> Finally, go to "change tempo." This can make your recording play faster or
>> slower without changing pitch. Don't change the beats, but rather trim the
>> time length of your recording. A little bit goes a long way. For example,
>> when I recorded my call the clip was about 3 seconds long. Cutting it to 2.6
>> seconds got rid of inter-syllabic pauses etc and made it sound urgent but
>> didn't introduce any unnatural sound.
>
>Time compression tends to make voices sound artificial. Time
>compression, or talking too fast with poor articulation, can make it
>difficult for others to copy your call.

That is confusing a problem with its solution. The problem, everyone
agrees, is trying to talk too fast with poor articulation. But time
compression isn't part of that problem - applied correctly, it is part
of the CURE.

Try this: record your messages (CQ and callsign) focusing 100% of your
attention on speaking clearly. Don't worry at all about speed; speak as
slowly as you find necessary to get good articulation. Then you can use
the 'tempo' function to bring the recording up to a normal speed.

Most people can improve articulation dramatically by slowing down only
10-20%, so it only requires a modest increase in the tempo setting to
restore a normal brisk speed. Time compression is a re-sampling
technique and it does introduce some artefacts, but these are minor
compared with everything else that happens to a SSB voice signal.

At this point, you can bring in a third factor: pacing. If the gaps
between words or syllables don't sound quite right, you can experiment
by cutting out (or pasting in) small segments of 'quiet time'. This is a
very simple cut-copy-paste operation in Audacity, easy to follow on the
scope trace.

The golden rule is: listen to the results after every step in the
editing process. If it doesn't sound good, then Undo that step and try
something else.

If it's done well (which really isn't hard, and quite fun to learn),
you'll find that your voice sounds clearer, but still quite natural. In
fact, you'll probably sound *more* natural than if you were straining
for optimum articulation, speed and pacing, all at the same time. And
after 24-48 hours of SSB contesting, your recorded voice is *guaranteed*
to sound better than your natural self!

--

73 from Ian GM3SEK
http://www.ifwtech.co.uk/g3sek
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Barry

Re: optimizing recorded audio

Is it possible to access and download the audio file of each stored M1-M4 memory, play with it in AUdicity, then upload it back to the radio?
Barry W2UP

Tony Estep

Re: optimizing recorded audio

In reply to this post by gm3sek

On Fri, Apr 22, 2011 at 2:40 AM, Ian White GM3SEK <[hidden email]>wrote:

> ...time compression isn't part of that problem - applied correctly, it is
> part
> of the CURE...
>

Yeah, it's sort of obvious that more control is better than less. The idea
that using software that can shape audio to what you want will automatically
screw things up is easily disproved by a few minutes of experimentation. Of
course it's possible to make a mess of things, just as it is with any power
tool.

Part of the confusion surrounding this topic comes from the fact that audio
software is generally so opaque. A lot of audio software is hellaciously
complex and has a gigantic learning curve (I'm thinking in particular of
Cubase, but Pro Tools and others are similarly hard to master). But Audacity
is easy to get into. It has a lot of pre-packaged goodies, written by some
of the world's great FFT engineers, and the "leveller" and tempo changers
are real jewels for our purposes; they give good results even if you're not
Jay-Z.

If you make a clip that's compressed to the max and then compress it again,
you'll get a mess, but that's cockpit error.

Guy's comment that he had tried it and didn't like it is fair enough,
although I think if he fiddled some more he'd like it. The rest of the
objections are pure conjecture and could easily be disproved by spending a
few minutes trying it. The assertion that time compression always creates a
bad sound results, as Ian sez, from confusion.

Tony KT0NY
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Don Wilhelm-4

Re: optimizing recorded audio

When doing audio processing, always keep a copy of the original file
until you are done.
Each step in the process does create some loss of quality. That means,
the more you mess with it, there is potential for the result to end up
bad. Keep notes on what is being done - how much leveling, how much
tempo change, etc. Then after your experimentation is complete, start
again with the original file and apply the full changes - the result
will be better than the result obtained by incremental changes during
your experimentation.

I would also recommend using only the K3 to apply compression. You
already have compression applied to the mic input, and that same
compression will be added to the computer audio stream. In general,
compressing an already compressed file will produce bad results.

I have done only a moderate amount of audio editing work, so I consider
the words of those experts (like Jim Brown) who have done a lot of it as
sage guidance for me.

73,
Don W3FPR

On 4/22/2011 10:18 AM, Tony Estep wrote:

> On Fri, Apr 22, 2011 at 2:40 AM, Ian White GM3SEK<[hidden email]>wrote:
>
>> ...time compression isn't part of that problem - applied correctly, it is
>> part
>> of the CURE...
>>
> Yeah, it's sort of obvious that more control is better than less. The idea
> that using software that can shape audio to what you want will automatically
> screw things up is easily disproved by a few minutes of experimentation. Of
> course it's possible to make a mess of things, just as it is with any power
> tool.
>
> Part of the confusion surrounding this topic comes from the fact that audio
> software is generally so opaque. A lot of audio software is hellaciously
> complex and has a gigantic learning curve (I'm thinking in particular of
> Cubase, but Pro Tools and others are similarly hard to master). But Audacity
> is easy to get into. It has a lot of pre-packaged goodies, written by some
> of the world's great FFT engineers, and the "leveller" and tempo changers
> are real jewels for our purposes; they give good results even if you're not
> Jay-Z.
>
> If you make a clip that's compressed to the max and then compress it again,
> you'll get a mess, but that's cockpit error.
>
> Guy's comment that he had tried it and didn't like it is fair enough,
> although I think if he fiddled some more he'd like it. The rest of the
> objections are pure conjecture and could easily be disproved by spending a
> few minutes trying it. The assertion that time compression always creates a
> bad sound results, as Ian sez, from confusion.
>
> Tony KT0NY
>

Jim Brown-10

Re: optimizing recorded audio

In reply to this post by gm3sek

On 4/22/2011 12:40 AM, Ian White GM3SEK wrote:
> Most people can improve articulation dramatically by slowing down only
> 10-20%, so it only requires a modest increase in the tempo setting to
> restore a normal brisk speed. Time compression is a re-sampling
> technique and it does introduce some artefacts, but these are minor
> compared with everything else that happens to a SSB voice signal.

You're right, Ian. My advice is really directed at users who are not
skilled in audio editing, and is part of a KISS (Keep It Simple, Stupid)
philosophy. I do a few things with editing that I wouldn't dream of
recommending that others try (and I won't even mention them) because
they are so complex and easy to screw up. My experience with time
compression goes back to the Lexicon D224 hardware product, a pro
product that sold for about $7,000 in the early 1980s. I demoed and sold
them to studios for the purpose of shortening radio spots (commercials).
I heard them on a lot of material, all with very expensive voices. 5%
compression sounded great, 10% was OK, but more than that was artificial
sounding.

Don suggested keeping a copy of the original file. Yes, a good idea, but
all of the editing software mentioned has an undo function, so if you
listen to each step as you go along, you can get away without that. And,
of course, you can always re-record the message, which I do occasionally
because I don't like the first attempt. In fact, I often record a
message a half dozen time (or more) before I start editing it.

A few other suggestions.

When recording, make sure your shack is quiet -- close the door, turn
off all the fans and air conditioners.

Work with the mic not too close to your mouth so that you don't get
breath pops and low end boost, and make sure that the audio levels are
right as shown on the editing software's meter and waveform display.
You should NEVER see any overload, and it's best to keep the peaks of
the waveform at least 3dB below max (0dB on the display). If you do,
throw out that recording and start over. You CANNOT fix it by turning
it down after it's been recorded.

After you've finished editing, use the EQ function to roll off the low
end at about 100 Hz, and to roll off the high end at about 6 kHz.

If you like to use VOX (I do), record a click at the beginning of each
CQ to activate the VOX a few milliseconds before the message starts.
This prevents losing the first syllable of the recording. Adjust the
peak level of the click to be 15-20 dB below the peak level of the
message. Use this click only on messages that will transmitted alone,
like your call, a CQ, and the Thanks message at the end of QSO. Do NOT
use it on an exchange -- you should activate the VOX with the live mic
when you say the other guy's call. When all this is working well the
click should not be transmitted. To get the timing and level right,
play the track through the rig and listen to the result.

Setting levels is VERY important. I like to keep the highest peaks of
the final recording between about -6dB and -3dB as indicated on the
Audacity waveform display. It's also important not to set the output
gain of the computer too high. Most sound cards have greatly increased
distortion when they get close to full output, so it's best to run their
output a bit lower to keep that distortion low. You don't need to
reduce it a lot -- 3-6 dB is enough.

When setting levels at the K3, remember that you want to match the level
of the live mic going straight into the K3 with the level of the
playback from the computer. We use the Line Input control to set the
level of the playback audio, and to do that, we must temporarily set the
K3 for Line Input and use the front panel Mic Gain. I usually set the
Line In gain so that I get the same indicated ALC and COMP indications
on the K3 meter display with playback as I do with the live mic (about
10dB of COMP on the hottest voice peaks).

73, Jim K9YC
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Lu Romero - W4LT

Re: optimizing recorded audio

In reply to this post by Tony Estep

This is a subject close to my heart...

I have a lot of tools at my disposal to "mangle" audio both
here at work and also at home. Soundtrack Pro is one of my
favorite weapons of mass destruction, and I have lots of
plug ins to modify stuff with.

Also, I have a lot of hardware
compressor/limiter/preamp/leveler/phase rotator/exciters too
to create audio mayhem with.

Having said all that, I will tell you what I do for
recording audio... I simply record my mic (almost
exclusively the Yamaha CM500'ds electret capsule) completely
flat through my MicroHam MicroKeyer 2's sound card into the
N1MM computer DVK via the MicroHam record facility. Then, I
apply all compression, equalization and gating within the K3
using its built in processing tools.

That's it. Nothing else. Nada. Bupkis.

Some of you have heard me on the air. I believe my "canned"
sound is identical (discounting voice box fatigue) to my
recorded sound. I cannot tell them apart off air.

It has taken me a little while to get used to the tools
available in K3 to have what I consider competitive contest
audio punch. The "digitalness" of the radio is very
different from my venerable TS850S that, with lots of
outboard junk, provided me with 11 years worth of crackly
punch that sliced and diced through piles like the Ginsu
knife it was. You just have to keep in mind the "Spinal
Tap" rule and not get carried away. A little goes a long
way. Mic technique and placement is important.
Equalization is important. Room acoustics is important.
Voice technique is important.

One thing I would love to have in the K3 audio chain is a
fast attack medium decay "AGC/Leveler" of some sort, pre-RF
compression/clipper, post gate. This would make the rig
perfect from the audio perspective and with careful,
judicious use, would help those with "thin" voices or poor
mic placement, however, you could get into serious trouble
if too much AGC was applied in a noisy environment, so maybe
we should leave well enough alone...

Never forget that we are transmitting into an extremely
noisy medium where transmitted audio dynamic range and
wideband frequency response is your enemy. If this was
broadcast FM or TV, our priorities would be very different.
But its not, its "communications" audio. The point is to be
clearly understood and get the message through, not to sound
like Orson Wells or Ernie Anastos.

I dont know about time processing either. KT0NY mentions he
time compressed his clips from 3 seconds to 2.6 seconds. 4
tenths of a second total shortening. Does that really buy
you a lot? Yes, I can do the math, but does it REALLY buy
you *THAT* much to risk inteligibility?

Im in K9YC and W3FPR's corner on this discussion. And if
anybody wants to make me a good offer on 6 rack units of
Behringer, Symetrix, Aphex and London processing gear that
the K3's processor/gate has obsoleted, let me know.

-Lu-W4LT-
K3 # 3192

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

Message: 26
Date: Fri, 22 Apr 2011 10:39:04 -0400
From: Don Wilhelm <[hidden email]>
Subject: Re: [Elecraft] optimizing recorded audio
To: Tony Estep <[hidden email]>
Cc: [hidden email]
Message-ID: <[hidden email]>
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When doing audio processing, always keep a copy of the
original file
until you are done.
Each step in the process does create some loss of quality.
That means,
the more you mess with it, there is potential for the result
to end up
bad. Keep notes on what is being done - how much leveling,
how much
tempo change, etc. Then after your experimentation is
complete, start
again with the original file and apply the full changes -
the result
will be better than the result obtained by incremental
changes during
your experimentation.

I would also recommend using only the K3 to apply
compression. You
already have compression applied to the mic input, and that
same
compression will be added to the computer audio stream. In
general,
compressing an already compressed file will produce bad
results.

I have done only a moderate amount of audio editing work, so
I consider
the words of those experts (like Jim Brown) who have done a
lot of it as
sage guidance for me.

73,
Don W3FPR

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Gary Gregory

Re: optimizing recorded audio

In reply to this post by Jim Brown-10

Jim,

If only all users followed your advice. Sadly the last contest showed they
don't.

I was amazed at the number of stations who happily went on and on with
distorted, over driven or clipped audio. One such station called us on every
3 hour segment and the audio was nothing short of a joke. (I am being
polite)

I have tried several times over the past couple of years and I confess I was
not happy with any of the attempts. The K3 M1-M4 does the job for me and
this is reflected (I feel) in the low number of repeats I have to make.

Conditions are improving on 15 and 10M so I guess I can expect to hear some
great audio coming out of W land...:-)

73's
Gary

On 23 April 2011 02:33, Jim Brown <[hidden email]> wrote:

> On 4/22/2011 12:40 AM, Ian White GM3SEK wrote:
> > Most people can improve articulation dramatically by slowing down only
> > 10-20%, so it only requires a modest increase in the tempo setting to
> > restore a normal brisk speed. Time compression is a re-sampling
> > technique and it does introduce some artefacts, but these are minor
> > compared with everything else that happens to a SSB voice signal.
>
> You're right, Ian. My advice is really directed at users who are not
> skilled in audio editing, and is part of a KISS (Keep It Simple, Stupid)
> philosophy. I do a few things with editing that I wouldn't dream of
> recommending that others try (and I won't even mention them) because
> they are so complex and easy to screw up. My experience with time
> compression goes back to the Lexicon D224 hardware product, a pro
> product that sold for about $7,000 in the early 1980s. I demoed and sold
> them to studios for the purpose of shortening radio spots (commercials).
> I heard them on a lot of material, all with very expensive voices. 5%
> compression sounded great, 10% was OK, but more than that was artificial
> sounding.
>
> Don suggested keeping a copy of the original file. Yes, a good idea, but
> all of the editing software mentioned has an undo function, so if you
> listen to each step as you go along, you can get away without that. And,
> of course, you can always re-record the message, which I do occasionally
> because I don't like the first attempt. In fact, I often record a
> message a half dozen time (or more) before I start editing it.
>
> A few other suggestions.
>
> When recording, make sure your shack is quiet -- close the door, turn
> off all the fans and air conditioners.
>
> Work with the mic not too close to your mouth so that you don't get
> breath pops and low end boost, and make sure that the audio levels are
> right as shown on the editing software's meter and waveform display.
> You should NEVER see any overload, and it's best to keep the peaks of
> the waveform at least 3dB below max (0dB on the display). If you do,
> throw out that recording and start over. You CANNOT fix it by turning
> it down after it's been recorded.
>
> After you've finished editing, use the EQ function to roll off the low
> end at about 100 Hz, and to roll off the high end at about 6 kHz.
>
> If you like to use VOX (I do), record a click at the beginning of each
> CQ to activate the VOX a few milliseconds before the message starts.
> This prevents losing the first syllable of the recording. Adjust the
> peak level of the click to be 15-20 dB below the peak level of the
> message. Use this click only on messages that will transmitted alone,
> like your call, a CQ, and the Thanks message at the end of QSO. Do NOT
> use it on an exchange -- you should activate the VOX with the live mic
> when you say the other guy's call. When all this is working well the
> click should not be transmitted. To get the timing and level right,
> play the track through the rig and listen to the result.
>
> Setting levels is VERY important. I like to keep the highest peaks of
> the final recording between about -6dB and -3dB as indicated on the
> Audacity waveform display. It's also important not to set the output
> gain of the computer too high. Most sound cards have greatly increased
> distortion when they get close to full output, so it's best to run their
> output a bit lower to keep that distortion low. You don't need to
> reduce it a lot -- 3-6 dB is enough.
>
> When setting levels at the K3, remember that you want to match the level
> of the live mic going straight into the K3 with the level of the
> playback from the computer. We use the Line Input control to set the
> level of the playback audio, and to do that, we must temporarily set the
> K3 for Line Input and use the front panel Mic Gain. I usually set the
> Line In gain so that I get the same indicated ALC and COMP indications
> on the K3 meter display with playback as I do with the live mic (about
> 10dB of COMP on the hottest voice peaks).
>
> 73, Jim K9YC
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Gary Gregory

Re: optimizing recorded audio

In reply to this post by Lu Romero - W4LT

Lu,

I sincerely hope you don't get any offers...this from the receiving
end...:-)

I must add that some of the worse audio we hear in VK is poorly adjusted
W1HY audio processed signals...aaagh!...some are so bad a lot of us simply
QSY and pretend we didn't hear the call....:-)

73's
Gary

On 23 April 2011 04:38, Lu Romero <[hidden email]> wrote:

> This is a subject close to my heart...
>
> I have a lot of tools at my disposal to "mangle" audio both
> here at work and also at home. Soundtrack Pro is one of my
> favorite weapons of mass destruction, and I have lots of
> plug ins to modify stuff with.
>
> Also, I have a lot of hardware
> compressor/limiter/preamp/leveler/phase rotator/exciters too
> to create audio mayhem with.
>
> Having said all that, I will tell you what I do for
> recording audio... I simply record my mic (almost
> exclusively the Yamaha CM500'ds electret capsule) completely
> flat through my MicroHam MicroKeyer 2's sound card into the
> N1MM computer DVK via the MicroHam record facility. Then, I
> apply all compression, equalization and gating within the K3
> using its built in processing tools.
>
> That's it. Nothing else. Nada. Bupkis.
>
> Some of you have heard me on the air. I believe my "canned"
> sound is identical (discounting voice box fatigue) to my
> recorded sound. I cannot tell them apart off air.
>
> It has taken me a little while to get used to the tools
> available in K3 to have what I consider competitive contest
> audio punch. The "digitalness" of the radio is very
> different from my venerable TS850S that, with lots of
> outboard junk, provided me with 11 years worth of crackly
> punch that sliced and diced through piles like the Ginsu
> knife it was. You just have to keep in mind the "Spinal
> Tap" rule and not get carried away. A little goes a long
> way. Mic technique and placement is important.
> Equalization is important. Room acoustics is important.
> Voice technique is important.
>
> One thing I would love to have in the K3 audio chain is a
> fast attack medium decay "AGC/Leveler" of some sort, pre-RF
> compression/clipper, post gate. This would make the rig
> perfect from the audio perspective and with careful,
> judicious use, would help those with "thin" voices or poor
> mic placement, however, you could get into serious trouble
> if too much AGC was applied in a noisy environment, so maybe
> we should leave well enough alone...
>
> Never forget that we are transmitting into an extremely
> noisy medium where transmitted audio dynamic range and
> wideband frequency response is your enemy. If this was
> broadcast FM or TV, our priorities would be very different.
> But its not, its "communications" audio. The point is to be
> clearly understood and get the message through, not to sound
> like Orson Wells or Ernie Anastos.
>
> I dont know about time processing either. KT0NY mentions he
> time compressed his clips from 3 seconds to 2.6 seconds. 4
> tenths of a second total shortening. Does that really buy
> you a lot? Yes, I can do the math, but does it REALLY buy
> you *THAT* much to risk inteligibility?
>
> Im in K9YC and W3FPR's corner on this discussion. And if
> anybody wants to make me a good offer on 6 rack units of
> Behringer, Symetrix, Aphex and London processing gear that
> the K3's processor/gate has obsoleted, let me know.
>
> -Lu-W4LT-
> K3 # 3192
>
> ---------------------------
>
> Message: 26
> Date: Fri, 22 Apr 2011 10:39:04 -0400
> From: Don Wilhelm <[hidden email]>
> Subject: Re: [Elecraft] optimizing recorded audio
> To: Tony Estep <[hidden email]>
> Cc: [hidden email]
> Message-ID: <[hidden email]>
> Content-Type: text/plain; charset=ISO-8859-1; format=flowed
>
> When doing audio processing, always keep a copy of the
> original file
> until you are done.
> Each step in the process does create some loss of quality.
> That means,
> the more you mess with it, there is potential for the result
> to end up
> bad. Keep notes on what is being done - how much leveling,
> how much
> tempo change, etc. Then after your experimentation is
> complete, start
> again with the original file and apply the full changes -
> the result
> will be better than the result obtained by incremental
> changes during
> your experimentation.
>
> I would also recommend using only the K3 to apply
> compression. You
> already have compression applied to the mic input, and that
> same
> compression will be added to the computer audio stream. In
> general,
> compressing an already compressed file will produce bad
> results.
>
> I have done only a moderate amount of audio editing work, so
> I consider
> the words of those experts (like Jim Brown) who have done a
> lot of it as
> sage guidance for me.
>
> 73,
> Don W3FPR
>
>
> ______________________________________________________________
> 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
>