Getting accurate sound levels in spectrum analysis

[Mxsmanic]

I installed a freeware plug-in on Sound Forge that provides a spectrum
analysis. When I analyze something, it seems like there's a lot of sound
energy at low frequencies, even though I don't seem to be hearing that much at
the low end. There's a "slope" adjustment in the analyzer, but I'm not sure
what it does--can someone explain it to me?

I tried generating some white noise and then adjusting the slope so that the
spectrum was relatively flat (since I presume that white noise contains equal
amounts of sound energy at all frequencies), but I'm not sure that this
accomplished what I want. I'd just like to see the actual sound levels for
each frequency.

If it makes a difference, the audio editing program is Sound Forge (the Audio
Studio version) and the plug-in is VOXengo SPAN.

[Les Cargill]

Here are the Fletcher-Munson curves:

<http://blogs.msdn.com/blogfiles/audiofool/WindowsLiveWriter/LouderSoundsBetter_12855/FletcherMunson_EqualLoudness2.jpg>

--
Les Cargill

[Mxsmanic]

Les Cargill writes:

> Here are the Fletcher-Munson curves:
>
> <http://blogs.msdn.com/blogfiles/audiofool/WindowsLiveWriter/LouderSoundsBetter_12855/FletcherMunson_EqualLoudness2.jpg>

But I don't want the loudness measurement, I want the actual sound pressure
level, since that is independent of anyone's hearing thresholds.

[Arny Krueger]

"Mxsmanic" <mxsm...@gmail.com> wrote in message
news:fc61j7hm60sp0ne8a...@4ax.com...

>I installed a freeware plug-in on Sound Forge that provides a spectrum
> analysis. When I analyze something, it seems like there's a lot of sound
> energy at low frequencies, even though I don't seem to be hearing that
> much at
> the low end.

What Les was trying to cue you into is the fact that your ears are
increasingly insensitive to frequencies as they drop below 1,000 Hz. Your
ears are 60 dB less sensitive at 20 Hz than 1 KHz. If it is less than 60
dB SPL you might not hear it at all!

> There's a "slope" adjustment in the analyzer, but I'm not sure
> what it does--can someone explain it to me?

It's got markings, right? If they are like pink, white, etc., then they
refer to whether the plot shows equal intensity per frequency or equal
intensity per band.

> I tried generating some white noise and then adjusting the slope so that
> the
> spectrum was relatively flat (since I presume that white noise contains
> equal
> amounts of sound energy at all frequencies), but I'm not sure that this
> accomplished what I want. I'd just like to see the actual sound levels for
> each frequency.

Then, don't apply any slope. That should make your FFT flat for a white
noise input.

> If it makes a difference, the audio editing program is Sound Forge (the
> Audio
> Studio version) and the plug-in is VOXengo SPAN.

I was kinda under the impression that SF had its own FFT facility. Most DAW
software does!

[Scott Dorsey]

Mxsmanic <mxsm...@gmail.com> wrote:
>I installed a freeware plug-in on Sound Forge that provides a spectrum
>analysis. When I analyze something, it seems like there's a lot of sound
>energy at low frequencies, even though I don't seem to be hearing that much at
>the low end. There's a "slope" adjustment in the analyzer, but I'm not sure
>what it does--can someone explain it to me?

There are two possibilities. First, the lowest bins on an FFT system may
not be accurate unless the window is made very, very large. You may want
to play with that.

Secondly, your speaker system probably just can't reproduce low end accurately.

>I tried generating some white noise and then adjusting the slope so that the
>spectrum was relatively flat (since I presume that white noise contains equal
>amounts of sound energy at all frequencies), but I'm not sure that this
>accomplished what I want. I'd just like to see the actual sound levels for
>each frequency.

For that you'll need a sound level meter with some accurate calibration.
--scott
--
"C'est un Nagra. C'est suisse, et tres, tres precis."

[Scott Dorsey]

In article <rni1j7df3ddk34j1j...@4ax.com>,

Then you will have to measure it.

However, you should be aware that it will change dramatically when you move
the listening position by a foot.

[Mark]

On Feb 7, 8:06 am, "Arny Krueger" <ar...@cocmast.net> wrote:
> "Mxsmanic" <mxsma...@gmail.com> wrote in message

It is also interesting to note that I have seen two styles of FFT
analyzers.

One style has a constant resolution BW (standard FFT ) and therefore
will display a flat spectral density with WHITE noise input and a
decreasing density with a PINK noise input.

The other type sometimes called a real time audio analyzer has a BW
that increases with frequency i.e. it has say 5 bands per octave for
example. This type will display a flat spectral density with a PINK
noise input and a rising density with a WHITE noise input.

You need to know what kind of ruler you are using.

Mark

[Mxsmanic]

Scott Dorsey writes:

> Then you will have to measure it.

That's what I'm trying to do, but it doesn't seem that my spectrum-analysis
plug-in provides a clear indication of actual intensity in each frequency
band.

What does the slope adjustment do?

[Mxsmanic]

Arny Krueger writes:

> What Les was trying to cue you into is the fact that your ears are
> increasingly insensitive to frequencies as they drop below 1,000 Hz. Your
> ears are 60 dB less sensitive at 20 Hz than 1 KHz. If it is less than 60
> dB SPL you might not hear it at all!

But there is a certain intensity recorded in the recorded material itself. I
want to see what that intensity is at each frequency, whether I can hear it or
not.

> It's got markings, right? If they are like pink, white, etc., then they
> refer to whether the plot shows equal intensity per frequency or equal
> intensity per band.

Not markings, just a number, from like -9 to +9 dB/octave.

> Then, don't apply any slope. That should make your FFT flat for a white
> noise input.

It seems to ... so does that mean that slope = 0 shows the actual distribution
of sound at different frequencies?

> I was kinda under the impression that SF had its own FFT facility. Most DAW
> software does!

This is the consumer version. Either it doesn't have it or I'm not sure where
to look for it, but someone pointed me to the VOXengo free plug-in, so I've
been trying to use that.

[Mxsmanic]

Scott Dorsey writes:

> There are two possibilities. First, the lowest bins on an FFT system may
> not be accurate unless the window is made very, very large.

Which window?

> Secondly, your speaker system probably just can't reproduce low end accurately.

I'm not listening to the sound so much as looking at it.

> For that you'll need a sound level meter with some accurate calibration.

Just to look at the recording? That shouldn't be necessary, since the sound
levels in the recording are fixed by definition.

I set the slope to zero, which seemed to flatten out the spectrum for white
noise. So far, so good. But then I looked at a recording I made on the street,
and it seems that the bulk of the sound in the recording is at very low
frequencies, below 100 Hz. Is this an accurate reflection of what the sound is
actually like outside, or is it an artifact of the microphone or the recording
process, or what? When I was actually recording it and monitoring it, the
sound in the headphones seemed virtually identical to what I heard without the
headphones, but it puzzles me that I see such an enormous proportion of the
sound at such low frequencies. Where could all that low-frequency sound be
coming from? There wasn't any wind, so it wasn't that.

[Scott Dorsey]

Mxsmanic <mxsm...@gmail.com> wrote:
>Scott Dorsey writes:
>
>> Then you will have to measure it.
>
>That's what I'm trying to do, but it doesn't seem that my spectrum-analysis
>plug-in provides a clear indication of actual intensity in each frequency
>band.

Then get one that does, like SpectraFOO. SpectraFOO will also allow you
to adjust the window width and shape, so you can see what this does to
the bottom bins.

>What does the slope adjustment do?

It's an EQ that lets you go from white to pink, etc?

[Scott Dorsey]

Mxsmanic <mxsm...@gmail.com> wrote:
>Scott Dorsey writes:
>
>> There are two possibilities. First, the lowest bins on an FFT system may
>> not be accurate unless the window is made very, very large.
>
>Which window?

The window over which you're collecting samples to put into the FFT.

>> Secondly, your speaker system probably just can't reproduce low end accurately.
>
>I'm not listening to the sound so much as looking at it.

So you're looking at the signal coming out of the noise generator, without
actually putting them through the speaker system and reference microphone?
In that case, what you see should be constant over whatever interval the
noise is constant over. Except the lowest FFT bins may contain trash because
that's how it is. The longer you set the window on the FFT, the lower
frequency you'll see accurately, and the longer it will take to update the
display.

>> For that you'll need a sound level meter with some accurate calibration.
>
>Just to look at the recording? That shouldn't be necessary, since the sound
>levels in the recording are fixed by definition.

I thought you said you wanted to measure what you were HEARING. The vast
majority of response aberrations in what you're hearing are due to the
speaker.

>I set the slope to zero, which seemed to flatten out the spectrum for white
>noise. So far, so good. But then I looked at a recording I made on the street,
>and it seems that the bulk of the sound in the recording is at very low
>frequencies, below 100 Hz. Is this an accurate reflection of what the sound is
>actually like outside, or is it an artifact of the microphone or the recording
>process, or what? When I was actually recording it and monitoring it, the
>sound in the headphones seemed virtually identical to what I heard without the
>headphones, but it puzzles me that I see such an enormous proportion of the
>sound at such low frequencies. Where could all that low-frequency sound be
>coming from? There wasn't any wind, so it wasn't that.

Set the window so it's as wide as the whole recording, and see what the low
end looks like. If you're trying to look at the spectrum in realtime, you
are taking periodic chunks of it, and how low the FFT is valid at will depend
on how long a chunk you are taking.

[Les Cargill]

Well, when I do a spec an on generated white noise in CoolEdit, the
result is ruler flat.

--
Les Cargill

[Mxsmanic]

Scott Dorsey writes:

> So you're looking at the signal coming out of the noise generator, without
> actually putting them through the speaker system and reference microphone?

Yes. I'm just trying to get the spectrum to match what the actual signal
should be like. For white noise, I presume it should be flat over the entire
frequency range.

> In that case, what you see should be constant over whatever interval the
> noise is constant over. Except the lowest FFT bins may contain trash because
> that's how it is.

That's how it is? Why is it that way?

> I thought you said you wanted to measure what you were HEARING.

Oh no, I'm just trying to get a visual image on the screen that accurately
reflects the frequency distribution of the sound in the recording. That way
I'll be able to correctly assess parts of the sound spectrum that I may not be
able to hear.

For example, if there's a great deal of high- or low-frequency noise in the
recording, beyond my range of hearing, I'd like to see it on the screen, so
that I can take steps to remove it if necessary.

> Set the window so it's as wide as the whole recording, and see what the low
> end looks like. If you're trying to look at the spectrum in realtime, you
> are taking periodic chunks of it, and how low the FFT is valid at will depend
> on how long a chunk you are taking.

OK, I will try that, if this plug-in allows it.

[William Sommerwerck]

"Mxsmanic" <mxsm...@gmail.com> wrote in message

news:fg94j7t1jepoad0k3...@4ax.com...

> Scott Dorsey writes:

>> So you're looking at the signal coming out of the noise generator,
without
>> actually putting them through the speaker system and reference
microphone?

> Yes. I'm just trying to get the spectrum to match what the actual signal
> should be like. For white noise, I presume it should be flat over the
entire
> frequency range.

Sorry, Charlie. The power spectrum of white noise rises at 3dB/8ve. This is
because it has equal energy "per frequency". You need pink noise for a flat
response. Pink noise has equal energy per octave.

[Mxsmanic]

William Sommerwerck writes:

> Sorry, Charlie. The power spectrum of white noise rises at 3dB/8ve. This is
> because it has equal energy "per frequency". You need pink noise for a flat
> response. Pink noise has equal energy per octave.

I'm looking for equal energy per linear increment in frequency, but I see your
point. I guess you have to integrate over some non-zero interval just to get
the spectrum in the first place.

I still don't understand what the "slope" control actually changes, though.

[Arny Krueger]

"Mxsmanic" <mxsm...@gmail.com> wrote in message

news:fqc3j71l198hc3k8q...@4ax.com...

> Arny Krueger writes:
>
>> What Les was trying to cue you into is the fact that your ears are
>> increasingly insensitive to frequencies as they drop below 1,000 Hz.
>> Your
>> ears are 60 dB less sensitive at 20 Hz than 1 KHz. If it is less than
>> 60
>> dB SPL you might not hear it at all!
>
> But there is a certain intensity recorded in the recorded material itself.
> I
> want to see what that intensity is at each frequency, whether I can hear
> it or
> not.
>
>> It's got markings, right? If they are like pink, white, etc., then they
>> refer to whether the plot shows equal intensity per frequency or equal
>> intensity per band.
>
> Not markings, just a number, from like -9 to +9 dB/octave.
>
>> Then, don't apply any slope. That should make your FFT flat for a white
>> noise input.
>
> It seems to ... so does that mean that slope = 0 shows the actual
> distribution
> of sound at different frequencies?

0 probably means no adjustment.

FFTs by default show a flat spectral response for equal energy per
frequency. White noise should show as being flat.

-3 probably corresponds to equal energy per octave, so +3 should make pink
noise show as being flat.

+6 should make red or brown noise show as being flat.

Kinda a neat option.

[Scott Dorsey]

Mxsmanic <mxsm...@gmail.com> wrote:
>Scott Dorsey writes:
>
>> So you're looking at the signal coming out of the noise generator, without
>> actually putting them through the speaker system and reference microphone?
>
>Yes. I'm just trying to get the spectrum to match what the actual signal
>should be like. For white noise, I presume it should be flat over the entire
>frequency range.

Depends on how your display is set up. White noise has equal power for a
given bandwidth (ie. equal power per hertz). Pink noise has equal power
in bands that are proportionally wide (ie. equal power per octave). Your
spectrum display will have settings that allow you to look at the frequency
plot in several different ways.

>> In that case, what you see should be constant over whatever interval the
>> noise is constant over. Except the lowest FFT bins may contain trash because
>> that's how it is.
>
>That's how it is? Why is it that way?

Because you're not going the FFT over the whole dataset, you're doing the
FFT on small chunks of the dataset so you can see the display in realtime.
How big a chunk you pick and what algorithm you use to overlap the chunks
will affect the low frequency display.

[Mxsmanic]

Arny Krueger writes:

> -3 probably corresponds to equal energy per octave, so +3 should make pink
> noise show as being flat.

So -3 means reduce the height of the curve by 3 dB per octave of increasing
frequency? Or am I still misunderstanding it?

[Mxsmanic]

Scott Dorsey writes:

> Because you're not going the FFT over the whole dataset, you're doing the
> FFT on small chunks of the dataset so you can see the display in realtime.
> How big a chunk you pick and what algorithm you use to overlap the chunks
> will affect the low frequency display.

Hmm. Well, if you do it for 10-millisecond chunks, then I suppose that would
distort the spectrum for sounds down around 100 Hz and below. But intuitively
I'd expect the low end of the spectrum to diminish, since low frequencies
would be harder and harder to discern with smaller and smaller sampling
intervals. I had not previously thought of the influence of chunk size.

My ultimate purpose is mainly to make sure that there are no weird sounds
intruding on the recording, particularly sounds that I can't hear. If there's
some sort of loud whistling at 30 kHz in the recording that I can't hear but
that could still cause a problem, I'd like to be able to see it represented in
correct proportions on the spectrum so that I can do something about it.

[Scott Dorsey]

Mxsmanic <mxsm...@gmail.com> wrote:
>Hmm. Well, if you do it for 10-millisecond chunks, then I suppose that would
>distort the spectrum for sounds down around 100 Hz and below. But intuitively
>I'd expect the low end of the spectrum to diminish, since low frequencies
>would be harder and harder to discern with smaller and smaller sampling
>intervals. I had not previously thought of the influence of chunk size.

This is very fundamental to using any FFT displays. There are a large number
of parameters that you can set which will totally change the way the signal
is displayed. You need to read the manuals.

>My ultimate purpose is mainly to make sure that there are no weird sounds
>intruding on the recording, particularly sounds that I can't hear. If there's
>some sort of loud whistling at 30 kHz in the recording that I can't hear but
>that could still cause a problem, I'd like to be able to see it represented in
>correct proportions on the spectrum so that I can do something about it.

I think you'll find that in the long run the spectrum analyzer is a poor
substitute for having accurate monitoring.

[Arny Krueger]

"Mxsmanic" <mxsm...@gmail.com> wrote in message

news:l5h5j797r0nmnticb...@4ax.com...

> Arny Krueger writes:
>
>> -3 probably corresponds to equal energy per octave, so +3 should make
>> pink
>> noise show as being flat.
>
> So -3 means reduce the height of the curve by 3 dB per octave of
> increasing
> frequency?

I would think so. What happens when you apply this adjustment to a source
that should display as flat, such as white noise?

[Arny Krueger]

"Mark" <mako...@yahoo.com> wrote in message
news:b5ffe2f3-d167-467c...@hb4g2000vbb.googlegroups.com...

> It is also interesting to note that I have seen two styles of FFT
analyzers.

> One style has a constant resolution BW (standard FFT ) and therefore
will display a flat spectral density with WHITE noise input and a
decreasing density with a PINK noise input.

> The other type sometimes called a real time audio analyzer has a BW
that increases with frequency i.e. it has say 5 bands per octave for
example. This type will display a flat spectral density with a PINK
noise input and a rising density with a WHITE noise input.

> You need to know what kind of ruler you are using.

Both could be the same FFT only with different ways of displaying the data.

The RTA display is also known as (fractional) octave display.

Here's an article that might help:

http://www.dspdimension.com/admin/dft-a-pied/

[dbd]

On Feb 7, 3:12 pm, Mark <makol...@yahoo.com> wrote:

> ...

> It is also interesting to note that I have seen two styles of FFT
> analyzers.
>
> One style has a constant resolution BW (standard FFT ) and therefore
> will display a flat spectral density with WHITE noise input and a
> decreasing density with a PINK noise input.
>
> The other type sometimes called a real time audio analyzer has a BW
> that increases with frequency i.e. it has say 5 bands per octave for
> example.  This type will display a flat spectral density with a PINK
> noise input and a rising density with a WHITE noise input.
>
> You need to know what kind of ruler you are using.
>
> Mark

B&K have made both kinds of analyzers and provided good ap-notes.

From Bruel&Kjear:
Basic Frequency Analysis of Sound
33pp
available at
http://nengvib.sydneyinstitute.wikispaces.net/file/view/BA767612.pdf/33225113/BA767612.pdf

The B&K site: www.bksv.com provides a library of manuals, papers, and
ap-notes but requests a sign-in.

Dale B. Dalrymple

[Mxsmanic]

Arny Krueger writes:

> I would think so. What happens when you apply this adjustment to a source
> that should display as flat, such as white noise?

If I set it to zero, the spectrum looks flat. If I set it to -9, the low end
rises and the high end falls, and vice versa for +9.

[Mike Rivers]

On 2/8/2012 1:59 PM, Mxsmanic wrote:

> My ultimate purpose is mainly to make sure that there are no weird sounds
> intruding on the recording, particularly sounds that I can't hear. If there's
> some sort of loud whistling at 30 kHz in the recording that I can't hear but
> that could still cause a problem, I'd like to be able to see it represented in
> correct proportions on the spectrum so that I can do something about it.

If it's there and you can't hear it, you probably shouldn't
be working in this field, or you should have better monitors
so you CAN hear it. I remember in the early days of home
recording, people were sending projects in for mastering and
replication with way too much low end because they couldn't
hear it on their home speakers. My local tape duplicator
would put a 50 Hz high pass filter between the tape deck and
console on everything that came in just to prevent damaging
their monitors when first playing through the tape.

As for 30 kHz, if you're working at or mastering for
standard sample rate, you won't record anything at that
frequency anyway.


--
"Today's production equipment is IT based and cannot be
operated without a passing knowledge of computing, although
it seems that it can be operated without a passing knowledge
of audio." - John Watkinson

http://mikeriversaudio.wordpress.com - useful and
interesting audio stuff

[Mark]

>
> > You need to know what kind of ruler you are using.
>
> Both could be the same FFT only with different ways of displaying the data.
>
> The RTA display is also known as (fractional) octave display.
>
> Here's an article that might help:
>
> http://www.dspdimension.com/admin/dft-a-pied/

thanks for quoting me Arnie, :-)

I was beginning to think I was invisible.. (which actually may be
true for some since I'm posting via Google)

Mark

[Mxsmanic]

Mike Rivers writes:

> If it's there and you can't hear it, you probably shouldn't
> be working in this field, or you should have better monitors
> so you CAN hear it.

I can't agree. A person might be an expert in audio systems and yet not have
perfect hearing. A recording might contain loud noise at 18 kHz that a sound
engineer might not be able to hear, but the intended audience might be able to
hear. I don't think the engineer should change careers just because he can't
hear it, but I also think that he should take steps to ensure that even things
he can't hear are at least noticed and handled appropriately, since there may
be other people who can hear those sounds.

Of course, if a person is entirely deaf, working with audio is going to be
difficult, but writing off a career simply because of a hearing impairment
seems extreme, insofar as many other skills are involved besides simply
hearing things alone.

> I remember in the early days of home
> recording, people were sending projects in for mastering and
> replication with way too much low end because they couldn't
> hear it on their home speakers. My local tape duplicator
> would put a 50 Hz high pass filter between the tape deck and
> console on everything that came in just to prevent damaging
> their monitors when first playing through the tape.

If those people had looked at the spectrum of the sound before sending in
their projects, perhaps they would have seen and removed the low end.

[0jun...@nomail.bellsloth.net]

On 2012-02-09 mxsm...@gmail.com said:
>Mike Rivers writes:
>> If it's there and you can't hear it, you probably shouldn't
>> be working in this field, or you should have better monitors
>> so you CAN hear it.
>I can't agree. A person might be an expert in audio systems and yet
>not have perfect hearing. A recording might contain loud noise at
>18 kHz that a sound engineer might not be able to hear, but the
>intended audience might be able to hear. I don't think the engineer
>should change careers just because he can't hear it,

<snip remainder>

Are you doing this for money? I doubt it from your reply
the other day to me. IF you are doing this for money then
you've got a bunch more learning to do before you're ready
to charge people.

Spectrum analysis tools and all this are nice, but the major
arbiter between the business end of the microphone and
product out the door should always be ears.



Richard webb,

replace anything before at with elspider

[Arny Krueger]

"Mxsmanic" <mxsm...@gmail.com> wrote in message

news:jp76j7h496rs80m0p...@4ax.com...

That's what I would expect.

[Arny Krueger]

"Mike Rivers" <mri...@d-and-d.com> wrote in message
news:jgv8rs$i5t$1...@dont-email.me...

> On 2/8/2012 1:59 PM, Mxsmanic wrote:
>
>> My ultimate purpose is mainly to make sure that there are no weird sounds
>> intruding on the recording, particularly sounds that I can't hear. If
>> there's
>> some sort of loud whistling at 30 kHz in the recording that I can't hear
>> but
>> that could still cause a problem, I'd like to be able to see it
>> represented in
>> correct proportions on the spectrum so that I can do something about it.
>
> If it's there and you can't hear it, you probably shouldn't be working in
> this field, or you should have better monitors so you CAN hear it.

I don't know about people can hear 30 KHz and monitors that can reproduce 30
KHz (particularly off axis).

I would never fault a person or a loudspeaker only because of inability to
respond to 30 KHz.

> I remember in the early days of home recording, people were sending
> projects in for mastering and replication with way too much low end
> because they couldn't hear it on their home speakers. My local tape
> duplicator would put a 50 Hz high pass filter between the tape deck and
> console on everything that came in just to prevent damaging their monitors
> when first playing through the tape.

I've been known to do that.

> As for 30 kHz, if you're working at or mastering for standard sample rate,
> you won't record anything at that frequency anyway.

Exactly.

[Arny Krueger]

"Mxsmanic" <mxsm...@gmail.com> wrote in message

news:ln37j79cpeooff0lt...@4ax.com...

> Mike Rivers writes:

>> If it's there and you can't hear it, you probably shouldn't
>> be working in this field, or you should have better monitors
>> so you CAN hear it.

> I can't agree. A person might be an expert in audio systems and yet not
> have
> perfect hearing.

In fact there are a lot of experts in audio who have far from perfect
hearing. At 65 years, my hearing isn't the best, but recently I was sitting
in a listening room next to a well-known audio expert who is a lot younger
than I am. I was having a hard time enjoying the music due to a very strong
60 Hz noise that was probably due to a ground loop. He didn't notice it.

Caveat: this expert does design, development and production of equipment,
not recordings. I don't know how someone could be that deaf and do live
sound or recording. IME live sound probably requires the best hearing
because the opportunities to work through live sound situations with FFTs
and the like are pretty limited. With track-per-mic recordings and good DAW
software, the right tools are more available and you have the time to use
them.

A lot of older audio workers have younger assistants that they rely on to
double check their work at least occasionally. Let's face it, most of what
really matters is in frequency ranges and at levels that are easy enough to
hear. A lot of audio production is about knowing which knobs to turn to get
things to sound a certain way which does not require hearing really subtle
details. A lot of people can hear those subtle details but never master
knowing which knobs to turn.

> A recording might contain loud noise at 18 kHz that a sound
> engineer might not be able to hear, but the intended audience might be
> able to
> hear.

To be heard by anybody, that sound at 18 KHz would have to be very large,
because of masking.

For example, the world is full of MP3s that are brick-walled at 16 KHz, and
still manage to sound crisp and even hot.

In general almost nobody hears brick wall filters very easily until their
corner frequency is 16 Khz or even somewhat lower.

> I don't think the engineer should change careers just because he can't
> hear it, but I also think that he should take steps to ensure that even
> things
> he can't hear are at least noticed and handled appropriately, since there
> may
> be other people who can hear those sounds.

Being able to read waveforms and use a FFT can be very useful for everybody,
but for folks whose hearing isn't the best, they can be survival tools. DAWs
make using these tools far easier.

> Of course, if a person is entirely deaf, working with audio is going to be
> difficult, but writing off a career simply because of a hearing impairment
> seems extreme, insofar as many other skills are involved besides simply
> hearing things alone.

>> I remember in the early days of home
>> recording, people were sending projects in for mastering and
>> replication with way too much low end because they couldn't
>> hear it on their home speakers. My local tape duplicator
>> would put a 50 Hz high pass filter between the tape deck and
>> console on everything that came in just to prevent damaging
>> their monitors when first playing through the tape.

> If those people had looked at the spectrum of the sound before sending in
> their projects, perhaps they would have seen and removed the low end.

If they'd looked at the cones of their woofers...

[Mxsmanic]

0jun...@nomail.bellsloth.net writes:

> Are you doing this for money?

No.

> I doubt it from your reply
> the other day to me. IF you are doing this for money then
> you've got a bunch more learning to do before you're ready
> to charge people.

I'm sure there are people who know less than I do who still charge money for
it. To charge money for something, you only need to know more about it than
your client. Or you need to be able to give the impression that you know more
about it.

> Spectrum analysis tools and all this are nice, but the major
> arbiter between the business end of the microphone and
> product out the door should always be ears.

Even for extremely high or low frequencies that may not be audible but might
still produce unwanted effects?

[Mxsmanic]

Arny Krueger writes:

> A lot of people can hear those subtle details but never master
> knowing which knobs to turn.

That would be me. My hearing is still pretty good, but I hear something wrong,
I don't necessarily know how to fix it.

I have the same problem with music. If someone is even very slightly out of
tune, I hear it, but I'm unable to describe what's wrong because I just don't
know enough about music theory.

Of course, I could learn these things, I guess. But there is so much to learn
in so many domains, and there are only 24 hours in a day.

> To be heard by anybody, that sound at 18 KHz would have to be very large,
> because of masking.

I've read about extremely low sounds that make people very irritable, anxious,
and restless, even though they might not hear the sounds. I've already
mentioned somewhere here the concert at which the audience was made very
restless by an extremely high whistle that (presumably) nobody could actually
here.

I would want to eliminate those from my recordings if I found them.

> For example, the world is full of MP3s that are brick-walled at 16 KHz, and
> still manage to sound crisp and even hot.

I've noticed some MP3s that seemed to stop at exactly 16 kHz, and was
wondering about that very thing. So it's a deliberate choice, not a limitation
of MP3?

> Being able to read waveforms and use a FFT can be very useful for everybody,
> but for folks whose hearing isn't the best, they can be survival tools. DAWs
> make using these tools far easier.

Since vision provides much more bandwidth than hearing, in theory it should be
possible to represent sounds in a completely visual format with no loss,
allowing even a deaf person to perceive sound accurately. It would take
training but it should work.

I thought up a system like this for computers and deaf people once, but nobody
was interested.

Unfortunately, the opposite is not true: there's no way to represent visual
information to blind people completely with a simple audio system. The
bandwidth difference is 1000 to 1.

> If they'd looked at the cones of their woofers...

I've seen commercials and movies where a close-up of a speaker is shown with
the cone bouncing in and out, and I've always wondered if that's just a
special effect for purposes of exaggeration, or if it's a real sound being
played over the speaker. Which in turn reminds me of a certain scene in _Back
to the Future_ where Marty McFly turns up a huge amplifier to play his guitar
through a huge speaker. Which in turn reminds me of a certain print
advertisement for a stereo system with a guy sitting in his chair, his wind
blown backwards by the sound blast from his stereo (but I can't remember the
brand!).

[Mxsmanic]

Arny Krueger writes:

> I don't know about people can hear 30 KHz and monitors that can reproduce 30
> KHz (particularly off axis).

A loud ultrasonic noise could still affect listeners, even if they couldn't
actually hear the sound.

[Frank Stearns]

Mxsmanic <mxsm...@gmail.com> writes:

>Arny Krueger writes:

>> A lot of people can hear those subtle details but never master
>> knowing which knobs to turn.

>That would be me. My hearing is still pretty good, but I hear something wrong,
>I don't necessarily know how to fix it.

A spectrum analyzer is handy tool, just like a VU or peak meter.

A VU and peak meter might be like your doctor's stethascope and blood pressure cuff,
where the analyzer might be more like an ultrasound of your heart (or maybe one of
those micro-cams snaking through your blood vessles!)

You can simply "see" certain things with one tool that the other tools might only
hint at. And ears can be limited at times; it's useful, IMO, to occasionally "look"
at our work in other ways.

Because I'm always curious, I listen to CDs in general with a spectrum analyzer
running on one of the side video monitors.

I'm amazed at the number of recordings showing a 15,750 hz spike from a CRT
monitor.

(Heck, even one of my own pipe organ recordings managed to acoustically pick up
15,200 hz or so from a closed-circuit 9" video monitor that was up on the organ
console. The organ was a tracker, which meant the console was part of the organ, and
the organ was way up in a loft. The monitor allowed the organist to watch a
conductor's position down on the main floor of the church. The monitor was probably
1970s vintage; seems high-volume flyback noise was pretty common in those days.

First video production room I walked into -- with all those various CRTs -- made my
eyes water and my teeth hurt because of all the flyback noise.)


But a few things to consider: I don't think very many folks hear a flyback spike
embedded in a recording -- they're typically 20-40 dB down, and usually well-masked
by lots of energy at surrounding frequencies.

Even with the narrowest Q you might have on an EQ, it's damn near impossible to
notch out a single frequency like that. A 1/12 octave notch at 15K is going to put
quite a dip in your spectrum up there -- and you *will* hear that, whereas the
single offending freq you well might not.

The analyzer can also help you find system problems. I had a noisy full-sine UPS
that was putting out spikes at 19.5K and 26K and sometimes (oh the horror) 10.5K.
Worse, it was injecting this crap into the building ground! Zooming in I saw
additional "sidebands" on these spikes at 60 and 120 Hz intervals. The spectrum
analyzer was a very useful tool to investigate this.

The manufacturer thought I was crazy, until I showed them short video screen caps
from the analyzer. "Oh, yeah. Well... Those are our oscillators starting up on the
UPS..."

To their discredit, the issue was never resolved and the UPS was abandoned in favor
of lighter, cheaper systems that were dead quiet, even when fully powering the
loads. (I've also seen that 19.5K spike in a few other commercial recordings; I'm
guessing they had the same defective UPS running during the recording!)

Bottom line is that while you can't (and shouldn't) "mix" on an analyzer, it's a
useful piece of kit to have. Nowadays we have fairly comprehensive ones that can be
done cheap or even "free" in software, whereas a few decades back such measurement
devices were bulky hardware, their measurements coarse by comparison, and they cost
a bloody fortune.

Frank
Mobile Audio

--
.

[Mike Rivers]

On 2/9/2012 4:20 AM, Mxsmanic wrote:

> Mike Rivers writes:
>> If it's there and you can't hear it, you probably shouldn't
>> be working in this field, or you should have better monitors
>> so you CAN hear it.

> I can't agree. A person might be an expert in audio systems and yet not have
> perfect hearing. A recording might contain loud noise at 18 kHz that a sound
> engineer might not be able to hear, but the intended audience might be able to
> hear.

I know many audio engineers, both live and studio, that have
hearing loss and still manage to do good work. They don't
look at a spectrum analyzer to see what they can't here,
though.

> I don't think the engineer should change careers just
because he can't
> hear it, but I also think that he should take steps to ensure that even things
> he can't hear are at least noticed and handled appropriately, since there may
> be other people who can hear those sounds.

That's true. He needs to recognize his limitations and
figure out how to deal with them. This might be as simple as
asking someone to listen to the program material and ask if
he hears any booms or whistles. But the best way to handle
problems like "loud noise at 18 kHz" is to avoid them in the
first place. Where might such a noise enter the recording
process? It isn't likely to come from anyone's vocal cords
or an instrument. It likely means something is broken.

I only suggested a career change because of a clear
misunderstanding of what spectrum analysis is good for.

>> I remember in the early days of home
>> recording, people were sending projects in for mastering and
>> replication with way too much low end because they couldn't
>> hear it on their home speakers.

> If those people had looked at the spectrum of the sound before sending in
> their projects, perhaps they would have seen and removed the low end.

In those days, a spectrum analyzer cost $25,000. Mastering
(which includes removing things that clearly shouldn't be
there, if possible) only cost $100, and you had a real human
to make the judgment, not a piece of hardware that you
probably shouldn't trust with a critical decision that
affects the sound of your production.

[Mike Rivers]

On 2/9/2012 10:02 AM, Mxsmanic wrote:

>> A lot of people can hear those subtle details but never master
>> knowing which knobs to turn.
>
> That would be me. My hearing is still pretty good, but I hear something wrong,
> I don't necessarily know how to fix it.

This is what engineers learn (or they get out of the
business). I

> I have the same problem with music. If someone is even very slightly out of
> tune, I hear it, but I'm unable to describe what's wrong because I just don't
> know enough about music theory.

You can't say "the guitar is out of tune?

> Of course, I could learn these things, I guess. But there is so much to learn
> in so many domains, and there are only 24 hours in a day.

Well, you see, there are tools for people like you who want
to make music but don't have the skills or experience. It's
easy to assemble a system that doesn't have any fundamental
flaws. And if you know what you're recording, you can be
reasonably safe that the only thing that will make your
recordings unpleasant for someone (or you) to listen to is
your own bad taste. But this is something that you have to
develop as an artistic skill, not by buying a tool.

> I've read about extremely low sounds that make people very irritable, anxious,
> and restless, even though they might not hear the sounds.

This is true. That principle is even used in warfare. Make
the enemy want to shit and he'll have something else on his
mind than fighting a war. But think: How is such a sound
going to creep into your recording?

> the concert at which the audience was made very
> restless by an extremely high whistle that (presumably) nobody could actually
> here.

Somebody could hear it, probably a lot of people could.
Maybe there was no engineer manning the controls. Maybe
there was and he couldn't hear it. Maybe he could hear it,
did everything he could to eliminate or reduce it and still
couldn't get rid of it.

I used to do shows in an auditorium in a Government building
where, every hour, something would be sent along the power
lines to correct all the clocks. It caused a whistle in the
PA system for about 30 seconds and then it was gone. I
noticed it, as did some people in the audience. They
probably thought I did something to cause it.

> I would want to eliminate those from my recordings if I found them.

And the way to do that is to know that your equipment is
working properly and simply not make noises like that.

> I've noticed some MP3s that seemed to stop at exactly 16 kHz, and was
> wondering about that very thing. So it's a deliberate choice, not a limitation
> of MP3?

It's a choice. There are a lot of options for creating an
MP3 file. Properly conducted tests have shown that it's
possible to make an MP3 file and a CD of the same material
and not be able to tell them apart. And it's not just for
selected material, it was with a wide range of material.
Thing is that MP3 files grew out of a lack of storage space
so the original goal was to provide something
distinguishable as music which could be stored in as little
space as possible. Now that iPods have 160 GB of storage, if
someone were to give you 320 kbps 44.1 kHz MP3 files to load
on to it, you wouldn't be able to tell that you weren't
listening to a CD. But then there are people who still think
that CDs sound unacceptable. Some do, some don't.

But what you're talking about is a problem that you can
eliminate at the source. Using a spectrum analyzer to find
it after the fact is only helping you to put a Band Aid on
it, not fix the problem.

> Since vision provides much more bandwidth than hearing, in theory it should be
> possible to represent sounds in a completely visual format with no loss,
> allowing even a deaf person to perceive sound accurately. It would take
> training but it should work.

Yeah, but can you dance to it? For a realistic perspective,
check out Flanders & Swann's "A Song of Reproduction."

http://youtu.be/7fJmmDkvQyc

[Arny Krueger]

"Mxsmanic" <mxsm...@gmail.com> wrote in message

news:v0o7j79n1q2updt8e...@4ax.com...

Been there done that, but we're talking *really* loud, like over 120 dB SPL.
That doesn't happen with recordings, as a rule. For openers, they rarely get
played that loud, and in many cases if they were played that loud, the 30
KHz tone that created 120 dB SPL would quickly fry the tweeter, which has
happened in the real world.

[0jun...@nomail.bellsloth.net]

Mike Rivers writes:
<snip>

>But the best way to handle
>problems like "loud noise at 18 kHz" is to avoid them in the
>first place. Where might such a noise enter the recording
>process? It isn't likely to come from anyone's vocal cords
>or an instrument. It likely means something is broken.

Indeed, and one can figure out those things before that
happens, if one has some knowledge.

>I only suggested a career change because of a clear
>misunderstanding of what spectrum analysis is good for.

A post I made which didn't propagate stated much the same
thing to mxmanic. Spectrum analysis is a valuable tool, but
not the panacea he thinks it is.

<snip>

>> If those people had looked at the spectrum of the sound before
>>sending in their projects, perhaps they would have seen and
>removed the low end.
>In those days, a spectrum analyzer cost $25,000. Mastering
>(which includes removing things that clearly shouldn't be
>there, if possible) only cost $100, and you had a real human
>to make the judgment, not a piece of hardware that you
>probably shouldn't trust with a critical decision that
>affects the sound of your production.

As I also stated, and for some of us, who had to do
production for right here right now, the trick was resolving
such issues by applying some knowledge to make sure they
didn't creep in. Often those problems can be audible as
harmonics of the fundamental.

Mxmanic is doing this as a hobby, no doubt because he found
low cost software which permits him to dabble, and now
thinks it qualifies him to tell professionals how they
should be doing their jobs. It's good that he's interested
in learning, but I still think the faq for this newsgroup
and other reading material would help him gain a much better
understanding than stumbling around with a daw then coming
into a production oriented newsgroup and telling those of us
who actually do this to earn a dollar we're full of whatever
it may be, that he has all the answers.

Just for mxmanic, I record using digital not analog these
days, adn still don't bother to deal with spectrum analysis,
as old blind man uses a recorder that acts like the analog
recorders he built his skills using, and these don't offer
spectrum analysis tools. During later phases of production
I've had folks look at something with such tools as spectrum
analysis, but i endeavor to find other fixes for such
problems before they're necessary in mastering. But, if i
miss, that's why tehre's mastering <g>.



Finally for Mxmanic: Folks such as MIke and i aren't just
trying to urniate in your cornflakes just to have something
to do. Neophytes lurk here for useful information, and
they're endeavoring to put their product out before the
public, even if self produced and self engineered. Hence,
we have to do whatever we can to dispel myths and
misinformation. I'm not the gentlest at doing so. Sorry
'bout that.

[Richard Webb]

On Thu 2012-Feb-09 04:20, Mxsmanic writes:
>> If it's there and you can't hear it, you probably shouldn't
>> be working in this field, or you should have better monitors
>> so you CAN hear it.

>> I remember in the early days of home
>> recording, people were sending projects in for mastering and
>> replication with way too much low end because they couldn't
>> hear it on their home speakers. My local tape duplicator
>> would put a 50 Hz high pass filter between the tape deck and
>> console on everything that came in just to prevent damaging
>> their monitors when first playing through the tape.

> If those people had looked at the spectrum of the sound before
> sending in their projects, perhaps they would have seen and removed
> the low end.

Whether you agree or not, you don't have the experience to
make that judgment, as is obvious by that remark. Many
folks didn't have those tools readily available in real time before the advent of the daw. Have you ever mixed live for
broadcast or for paying customers? Did you just decide to
dabble in audio once you found low cost software? I'd guess the later.

IN the era that Mike's referencing most folks didn't have
spectrum analysis tools readily available as I said, ears
and the monitoring chain were what was used to make those
decisions. Mike is one of us in this group who's made his
daily bread working in audio, and we got some neophyte
basement tinkerer going to tell him he's full of shit?

Those are indeed beneficial tools, but but folks have been
making production decisions without them for the better part of a century. Such tools are especially useful as a sanity
check, or to make quicker decisions. Still, if you're going to cause money to change hands you need a better monitoring
chain it sounds like. if you're playing and tinkering for
your own good time, have at it, but expect the professionals in this group to challenge such assertions when you make
them here. After all, the neophytes know no better, and
they're trying to do work that is intended for public
consumption.


Regards,
Richard
... 10% of everything isn't crap, watch closely or you'll miss it!
--
| Remove .my.foot for email
| via Waldo's Place USA Fidonet<->Internet Gateway Site
| Standard disclaimer: The views of this user are strictly his own.

[Arkansan Raider]

Richard, to use a cue from pop culture, I'd rather have a Simon Cowell tell
me I'm doing it wrong and how to fix it than a Paula Abdul telling me I
sound great.

For the record (heh), I greatly appreciate the fact that there are
professionals who are willing to explain things to those of us who truly
want to know and want to learn.

I know a thing or two on the other side of the mic--and I want to know more
about your side of it to improve on what I do, and to improve my
communications with the folks who record me.

I'd also like to do my own demos so's to save more to put into the actual
album production.

Once again, thanks to you, Richard, and all of the rest of you.

--
---Jeff

[0jun...@nomail.bellsloth.net]

JEff writes:
>> Finally for Mxmanic: Folks such as MIke and i aren't just

>> trying to urinate in your cornflakes just to have something

>> to do. Neophytes lurk here for useful information, and
>> they're endeavoring to put their product out before the
>> public, even if self produced and self engineered. Hence,
>> we have to do whatever we can to dispel myths and
>> misinformation. I'm not the gentlest at doing so. Sorry
>> 'bout that.

>Richard, to use a cue from pop culture, I'd rather have a Simon
>Cowell tell me I'm doing it wrong and how to fix it than a Paula
>Abdul telling me I sound great.

Understood. mxmanic is lucky Fletcher isn't still lurking
around these here parts <g>.

>For the record (heh), I greatly appreciate the fact that there are
>professionals who are willing to explain things to those of us who
>truly want to know and want to learn.
>I know a thing or two on the other side of the mic--and I want to
>know more about your side of it to improve on what I do, and to
>improve my communications with the folks who record me.
>I'd also like to do my own demos so's to save more to put into the
>actual album production.
>Once again, thanks to you, Richard, and all of the rest of you.
>--

Understood, and often I lurk and let folks with better
skills impart the info and chime in where I can be of
assistance. I'm lucky enough to have been on both sides of
the glass as it were, and to have some actual musical chops,
which got me into this in the first place. Back in the day
when most stuff still had tubes I got the task of managing
the audio production live and recorded for the various
musical ensembles I worked with because of my electronics
interests as well, whereas these folks just wanted to play
and/or sing <g>.

When some guy waltzes into this group because he's got some
toys to play with and begins a whole bunch of threads that
add more to noise than signal and ends up telling some of
the folks from both sides of the glass that he knows more
about the art and science of capturing a good recording than
they do it rubs my fur the wrong direction.
There's enough bad product out there on homebrewed audio
that people want money for, bad audio on self produced
videos, etc. etc. Especially this visual representation
means no need to listen rubs me the wrong way.

[Mike Rivers]

On 2/9/2012 12:54 PM, 0jun...@nomail.bellsloth.net wrote:

> Mxmanic is doing this as a hobby, no doubt because he found
> low cost software which permits him to dabble, and now
> thinks it qualifies him to tell professionals how they
> should be doing their jobs.

Not really. He's telling us how he'd solve a problem if only
his tools would work the way he expects them to work . . .
which they don't, quite.

[Mxsmanic]

Frank Stearns writes:

> I'm amazed at the number of recordings showing a 15,750 hz spike from a CRT
> monitor.

Really? That's pretty interesting. I used to be able to hear that sound from
CRTs. Nowadays they have much higher scan rates, so I don't hear it very often
(but I can still hear it on headphones, so it's not my ears). Surprisingly,
even when I was a teenager, most people couldn't hear it--I'd walk into a
classroom and ask why the TV was still on, even though there was no picture,
and people thought I was imagining things. Which makes it hard to understand
how a so-called Mosquito is really supposed to work.

Anyway, it's interesting that the sound of nearby CRTs is getting recorded.
Surely that has declined somewhat with the advent of flat panels, though,
right?

> Heck, even one of my own pipe organ recordings managed to acoustically pick up
> 15,200 hz or so from a closed-circuit 9" video monitor that was up on the organ
> console. The organ was a tracker, which meant the console was part of the organ, and
> the organ was way up in a loft. The monitor allowed the organist to watch a
> conductor's position down on the main floor of the church. The monitor was probably
> 1970s vintage; seems high-volume flyback noise was pretty common in those days.

And easier to hear when the CRT had no signal, because the free-running
horizontal scan rate was somewhat lower than the synchronized rate.

> First video production room I walked into -- with all those various CRTs -- made my
> eyes water and my teeth hurt because of all the flyback noise.)

I've had similar experiences. I've been in equipment rooms filled with modems
that had similar high-frequency noise, although I don't know where it came
from. Mainframe computers used to make a lot of noise like this, too.

> But a few things to consider: I don't think very many folks hear a flyback spike
> embedded in a recording -- they're typically 20-40 dB down, and usually well-masked
> by lots of energy at surrounding frequencies.

I don't recall ever hearing it on music. I seem to remember hearing it on
other types of recordings.

> Even with the narrowest Q you might have on an EQ, it's damn near impossible to
> notch out a single frequency like that. A 1/12 octave notch at 15K is going to put
> quite a dip in your spectrum up there -- and you *will* hear that, whereas the
> single offending freq you well might not.

It's something that I will look for. And it's the sort of thing that I'd want
to remove, which is what might make a spectrum analysis handy, since 15 kHz
can be hard to hear.

> The analyzer can also help you find system problems. I had a noisy full-sine UPS
> that was putting out spikes at 19.5K and 26K and sometimes (oh the horror) 10.5K.
> Worse, it was injecting this crap into the building ground! Zooming in I saw
> additional "sidebands" on these spikes at 60 and 120 Hz intervals. The spectrum
> analyzer was a very useful tool to investigate this.

What do you do when you have a recording with that kind of noise?

> To their discredit, the issue was never resolved and the UPS was abandoned in favor
> of lighter, cheaper systems that were dead quiet, even when fully powering the
> loads. (I've also seen that 19.5K spike in a few other commercial recordings; I'm
> guessing they had the same defective UPS running during the recording!)

I didn't think that audio systems had a need for UPS systems, unless it's
really, really important to hear the entire song.

[Mxsmanic]

Mike Rivers writes:

> You can't say "the guitar is out of tune?

I guess I could, but I don't know how to tune it.

Also, sometimes the errors are more complex, and I'm not sure who is making
the mistake. In some cases, the "mistakes" are deliberate, as when the
Manhattan Transfer sings "A Nightingale Sang in Berkeley Square," which
contains a lot of unusual intervals.

> Well, you see, there are tools for people like you who want
> to make music but don't have the skills or experience.

I'm a music listener, rather than a music maker. I record ambient sounds or
speech, mostly, for my little touristy videos. Indeed, I avoid recording any
kind of music, because copyright trolls on YouTube will latch onto it to make
fraudulent infringement claims.

> This is true. That principle is even used in warfare. Make
> the enemy want to shit and he'll have something else on his
> mind than fighting a war. But think: How is such a sound
> going to creep into your recording?

That's the mystery. When I record street scenes, I see an awful lot of
low-frequency noise, but I have no idea where most of it is coming from. In
real life, I also hear it (based on comparisons I've done), but for whatever
reason, I don't notice it as much. Really low frequencies seem to be something
that you notice unconsciously, even when you can hear them.

> Somebody could hear it, probably a lot of people could.

I recall it being around 30 kHz, so the number of people actually hearing it
would be too small to explain a general restlessness. But maybe it affected
them in some other way than through hearing alone. Apparently it was very
loud.

> Maybe there was no engineer manning the controls. Maybe
> there was and he couldn't hear it. Maybe he could hear it,
> did everything he could to eliminate or reduce it and still
> couldn't get rid of it.

In the story I read, someone looked at a graphic equalizer or something and
noticed a huge spike at a very high frequency.

> I used to do shows in an auditorium in a Government building
> where, every hour, something would be sent along the power
> lines to correct all the clocks. It caused a whistle in the
> PA system for about 30 seconds and then it was gone. I
> noticed it, as did some people in the audience. They
> probably thought I did something to cause it.

Thank goodness we don't need systems like that now, although they are probably
still in use. There are other gadgets that send junk over power lines, too. I
guess nobody worries about interference.

> And the way to do that is to know that your equipment is
> working properly and simply not make noises like that.

But sometimes the noise is coming from something that isn't yours, and it's a
surprise.

> But what you're talking about is a problem that you can
> eliminate at the source. Using a spectrum analyzer to find
> it after the fact is only helping you to put a Band Aid on
> it, not fix the problem.

For field recordings, I don't have a spectrum analyzer handy.

> Yeah, but can you dance to it? For a realistic perspective,
> check out Flanders & Swann's "A Song of Reproduction."
>
> http://youtu.be/7fJmmDkvQyc

Well, places like discos already use similar concepts. But you could design
something much more detailed, like a spectrum analyzer, that would provide
enough information to understand things like speech, for a trained observer.

[Mxsmanic]

Mike Rivers writes:

> I know many audio engineers, both live and studio, that have
> hearing loss and still manage to do good work. They don't
> look at a spectrum analyzer to see what they can't here,
> though.

So what do they do? And did they damage their hearing from their work, or did
they lose it for other reasons?

> But the best way to handle
> problems like "loud noise at 18 kHz" is to avoid them in the
> first place. Where might such a noise enter the recording
> process? It isn't likely to come from anyone's vocal cords
> or an instrument. It likely means something is broken.

Well, if the room is filled with old-style CRTs, what can you do?

[Mxsmanic]

0jun...@nomail.bellsloth.net writes:

> Mxmanic is doing this as a hobby, no doubt because he found

> low cost software which permits him to dabble ...

> ... and now thinks it qualifies him to tell professionals how

> they should be doing their jobs.

No. But I do notice that some professionals who lack confidence in their own
abilities feel threatened any time someone they consider external to the
profession starts to ask questions or make any assertions at all. Those who
are not insecure remain undisturbed.

This is true in many, many professions, not just professional audio. There are
always a lot of people who aren't as competent as they'd like people to think,
and apparently they worry about being "exposed."

> Finally for Mxmanic: Folks such as MIke and i aren't just
> trying to urniate in your cornflakes just to have something
> to do. Neophytes lurk here for useful information, and
> they're endeavoring to put their product out before the
> public, even if self produced and self engineered. Hence,
> we have to do whatever we can to dispel myths and
> misinformation. I'm not the gentlest at doing so. Sorry
> 'bout that.

I am impervious to taking offense, so you need not worry. I know what I know,
and I also have an excellent idea of what I don't know.

[Mxsmanic]

Richard Webb writes:

> Have you ever mixed live for broadcast or for paying customers?

No.

> Did you just decide to dabble in audio once you found low cost
> software? I'd guess the later.

I don't know that dabble is the right word, but I do take an interest in many
technologies, including audio recording, and the availability of low-cost
equipment of good quality has allowed me to investigate audio recording to a
much greater extent than I would have been able to forty years ago, without
having to dedicate my professional life to it to do so.

> IN the era that Mike's referencing most folks didn't have
> spectrum analysis tools readily available as I said, ears
> and the monitoring chain were what was used to make those
> decisions. Mike is one of us in this group who's made his
> daily bread working in audio, and we got some neophyte
> basement tinkerer going to tell him he's full of shit?

If my posts bother you, don't read them. There are plenty of other people who
know what they are talking about and don't mind discussing their work with
neophytes.

> Still, if you're going to cause money to change hands you need a better
> monitoring chain it sounds like.

The only money that has changed hands has gone from my hands into the hands of
audio equipment dealers, and my budget is very small.

> ... if you're playing and tinkering for your own good time, have at it,

> but expect the professionals in this group to challenge such assertions
> when you make them here.

That's fine with me, and it would certainly be more productive than your
sophomoric invective, which wastes both my time and yours.

[Mxsmanic]

Arny Krueger writes:

> Been there done that, but we're talking *really* loud, like over 120 dB SPL.
> That doesn't happen with recordings, as a rule. For openers, they rarely get
> played that loud, and in many cases if they were played that loud, the 30
> KHz tone that created 120 dB SPL would quickly fry the tweeter, which has
> happened in the real world.

I don't recall the specifics of the (apocryphal) story I read, but it did
mention that the very-high-frequency noise was also very loud. The one thing
it didn't explain is where a noise like that could have been coming from, as I
recall.

[0jun...@nomail.bellsloth.net]

Mike Rivers writes:
>On 2/9/2012 12:54 PM, 0jun...@nomail.bellsloth.net wrote:
>> Mxmanic is doing this as a hobby, no doubt because he found
>> low cost software which permits him to dabble, and now
>> thinks it qualifies him to tell professionals how they
>> should be doing their jobs.
>Not really. He's telling us how he'd solve a problem if only
>his tools would work the way he expects them to work . . .
>which they don't, quite.

Yep, that too, and in a group frequented by folks in
production, especially that might be visited by those who
are learning it's important to dispel such misconceptions.
Note the recent thread on the pro-audio list re audio snake
oil.

What we have here is a man who has only one tool, and that's
a hammer, so everything appears to require a sharp blow.

Can't recall where I first heard it but, "when the only tool
you have is a hammer everything begins to look like a nail."

He does bring up an interesting point, older crt devices. I
was thinking more along the lines of misadjusted tape
machine as was Frank in one post.
Regards,

[Frank Stearns]

Mxsmanic <mxsm...@gmail.com> writes:

>Frank Stearns writes:

>> I'm amazed at the number of recordings showing a 15,750 hz spike from a CRT
>> monitor.

>Really? That's pretty interesting. I used to be able to hear that sound from
>CRTs. Nowadays they have much higher scan rates, so I don't hear it very often

That, and better construction of the deflection yokes. But CRTs are so last-century.
Not that many around any more, but you still find them, like that pipe organ
monitor.

>Anyway, it's interesting that the sound of nearby CRTs is getting recorded.
>Surely that has declined somewhat with the advent of flat panels, though,
>right?

There are a few different paths where flyback can get into a recording:

1. acoustically, through microphones. That's fairly rare, as the monitor will have
to be in pretty crappy shape to put out that much acoustical volume, the mics have
to be pretty good and in just the right spot. Move a small diaphram mic 1/4" to
1/2" and you'd see the 15.75K level swing probably some 30 dB. A large diaphragm
would have less variance, though probably a good 3-6 dB. (15.75K has a wavelength of
0.86 inches). A lot depends on how the sound is radiated from the monitor.

2. Inductively or electrostatically by way of unbalanced (or poorly balanced)
circuits going near a monitor.

3. Infiltration due to bad grounds or other poor practices in the facility. Older
wiring can introduce a series of potential "gotchas".

How can you tell how the spike got in? Generally, if acoustically, you'll typically
see the spike bouncing around in amplitude. If electronically, typically the spike
is rock solid and does not vary in amplitude. This is what I've typically seen when
commercial recordings have it. There were one or more CRT monitors or TVs in the
studio or machine room or some place, dumping crap into a susceptable audio circuit.
(And you can believe that a lot of even famous studios have some pretty
"interesting" wiring practices.)

>> But a few things to consider: I don't think very many folks hear a flyback spike
>> embedded in a recording -- they're typically 20-40 dB down, and usually well-masked
>> by lots of energy at surrounding frequencies.

>I don't recall ever hearing it on music. I seem to remember hearing it on
>other types of recordings.

Music will usually mask the flyback, unless it's really, really bad. Spoken word or
solo instruments with not a lot of HF content would be less able to mask such noise.

>> Even with the narrowest Q you might have on an EQ, it's damn near impossible to
>> notch out a single frequency like that. A 1/12 octave notch at 15K is going to put
>> quite a dip in your spectrum up there -- and you *will* hear that, whereas the
>> single offending freq you well might not.

>It's something that I will look for. And it's the sort of thing that I'd want
>to remove, which is what might make a spectrum analysis handy, since 15 kHz
>can be hard to hear.

Well, you might find it very difficult to remove. To minimize sonic damage you'd
need a notch filter with something like a 1/1000 (or smaller) octave width. I am not
a DSP expert; not even sure if you can do that.

>> The analyzer can also help you find system problems. I had a noisy full-sine UPS
>> that was putting out spikes at 19.5K and 26K and sometimes (oh the horror) 10.5K.
>> Worse, it was injecting this crap into the building ground! Zooming in I saw
>> additional "sidebands" on these spikes at 60 and 120 Hz intervals. The spectrum
>> analyzer was a very useful tool to investigate this.

>What do you do when you have a recording with that kind of noise?

At 19.5K, we lived with it. It was actually discovered by a friendly competitor with
his S.A.; darn decent of him to let me know. The 15.75 was never audible.

>> To their discredit, the issue was never resolved and the UPS was abandoned in favor
>> of lighter, cheaper systems that were dead quiet, even when fully powering the
>> loads. (I've also seen that 19.5K spike in a few other commercial recordings; I'm
>> guessing they had the same defective UPS running during the recording!)

>I didn't think that audio systems had a need for UPS systems, unless it's
>really, really important to hear the entire song.

That's not the issue. The UPS is used during location recording. If you only have
one shot at something, you don't want a recording hardware glitch due to a momentary
power company hiccup (or outright outage), nor some idiot accidently switching off
the AC outlet used for your gear or disconnecting your AC cord. After hundreds and
hundreds of location gigs, all of the above have happened at one time or another --
not often but they did. The USP saved the day each time. Without it there would have
either been a hole in the recording or a complete loss. Oh, and as a side
benefit, most UPS provide basic power filtering and protection. THAT'S why I carry a
UPS!

[Trevor]

"Arny Krueger" <ar...@cocmast.net> wrote in message
news:h-Sdnfz5xeMDnKnS...@giganews.com...

>> A loud ultrasonic noise could still affect listeners, even if they
>> couldn't actually hear the sound.
>
> Been there done that, but we're talking *really* loud, like over 120 dB
> SPL. That doesn't happen with recordings, as a rule. For openers, they
> rarely get played that loud, and in many cases if they were played that
> loud, the 30 KHz tone that created 120 dB SPL would quickly fry the
> tweeter,

Live sound speakers too at 120dB 30kHz!!! May very well be possible with a
jet engine whine I guess.

Trevor.

[Trevor]

"Arny Krueger" <ar...@cocmast.net> wrote in message

news:XJKdnZZ6RvqtW67S...@giganews.com...

> because the opportunities to work through live sound situations with FFTs
> and the like are pretty limited.

What makes you say that? Been doing live gigs with a laptop recorder and FFT
analyser for many years, as have many sound engineers I know. Sure you don't
want to have to look at them much during the gig, but invaluable for set up
IMO.

Trevor.

[Mxsmanic]

Frank Stearns writes:

> Well, you might find it very difficult to remove. To minimize sonic damage you'd
> need a notch filter with something like a 1/1000 (or smaller) octave width. I am not
> a DSP expert; not even sure if you can do that.

If you know it's actually from a CRT, couldn't you superimpose a signal of
exactly the same frequency and opposite phase and remove it, without affecting
anything else? Like astromomers do when they use extremely narrow filters to
completely remove the yellow light from low-pressure sodium-vapor streetlights
(which have an extremely monochromatic light).

> Oh, and as a side benefit, most UPS provide basic power filtering and protection.
> THAT'S why I carry a UPS!

That's the main reason why I put my computers on a UPS, although the battery
back-up is nice, too. But they weigh a ton--they must be a hassle to lug
around on location.

[an...@ao1.com]

>> some guy waltzes into this group because he's got some
toys to play with and begins a whole bunch of threads that
add more to noise than signal

+1 That's the way I see it.

[0jun...@nomail.bellsloth.net]

On 2012-02-09 franks.pa...@pacifier.net said:
<snip>

>There are a few different paths where flyback can get into a
>recording:
>1. acoustically, through microphones. That's fairly rare, as the
>monitor will have
>to be in pretty crappy shape to put out that much acoustical volume,
>the mics have
>to be pretty good and in just the right spot. Move a small diaphram
>mic 1/4" to 1/2" and you'd see the 15.75K level swing probably some
>30 dB. A large diaphragm
>would have less variance, though probably a good 3-6 dB. (15.75K
>has a wavelength of
>0.86 inches). A lot depends on how the sound is radiated from the
>monitor.

Right, and as you note, rare.

>2. Inductively or electrostatically by way of unbalanced (or poorly
>balanced) circuits going near a monitor.
>3. Infiltration due to bad grounds or other poor practices in the
>facility. Older
>wiring can introduce a series of potential "gotchas".
>How can you tell how the spike got in? Generally, if acoustically,
>you'll typically
>see the spike bouncing around in amplitude. If electronically,
>typically the spike
>is rock solid and does not vary in amplitude. This is what I've
>typically seen when
>commercial recordings have it. There were one or more CRT monitors
>or TVs in the
>studio or machine room or some place, dumping crap into a
>susceptable audio circuit.

Right, and if your inputs and outputs are few you may be
able to eliminate this with some ferrite chokes. When I've
found something like this in two-way radio systems'
transmitters I've used the snap on ferrites effectively.

>(And you can believe that a lot of even famous studios have some
>pretty "interesting" wiring practices.)

<snip>

>>I didn't think that audio systems had a need for UPS systems,
>>unless it's really, really important to hear the entire song.
>That's not the issue. The UPS is used during location recording. If
>you only have
>one shot at something, you don't want a recording hardware glitch
>due to a momentary
>power company hiccup (or outright outage), nor some idiot
>accidently switching off

Something truncated your post, but I use one on location,
it's a don't leave home without it piece of gear for on site
recording, especially when recording to hard disk. Even if
the other gear isn't protected by it the recorder *must* be.
This way at least we can stop, and save our work to that
point. IN the case of folks like Frank and myself, folks
are paying good money to be recorded.

[Frank Stearns]

Mxsmanic <mxsm...@gmail.com> writes:

>Frank Stearns writes:

>> Well, you might find it very difficult to remove. To minimize sonic damage you'd
>> need a notch filter with something like a 1/1000 (or smaller) octave width. I am not
>> a DSP expert; not even sure if you can do that.

>If you know it's actually from a CRT, couldn't you superimpose a signal of
>exactly the same frequency and opposite phase and remove it, without affecting
>anything else? Like astromomers do when they use extremely narrow filters to
>completely remove the yellow light from low-pressure sodium-vapor streetlights
>(which have an extremely monochromatic light).

Interesting idea, and perhaps worth a try, but here are the difficulties I see:

- exact waveform of the interference... it's likely not going to be perfectly
symmetrical, as your inverted "cancel" signal will be. As a result, the residual
might be worse. That is, you might wind up with some sharply peaked and spikey
waveforms hanging around. And remember, it'd be tough to go look at just this one hz
because it's buried in program.

- the easiest way to set this up would be another mix channel with an oscillator.
You'd precisely match this to the interference hz which, as I've seen, is rarely
exactly at 15.75K. So you'd have nudge that around as you look for a null, or have a
*very* high-res spectrum analyzer (higher than I've ever seen).

You're also hoping that the hz of the source interference isn't drifting a hz or two
back and forth. When it drifts away from your cancel frequency, kiss your
null goodbye.

You'd also have a delay plug-in on that channel to get the signal exactly at the
perfect null point. Pressing the "polarity" button on the channel isn't going to
mean a thing -- you have no idea as to the timing of the original interference.

Problem is, the delay plug-in will be stepped in samples, which might be far too
coarse of a timing step to get the waveform nudged to perfect nulling, especially if
the data is at 44.1Khz.

- if the spike entered multiple channels at various delays and levels of a
multitrack recording, you might very well have to pair up a nuller channel with
each program channel.

Attempting to null just the two track might prove impossible. If indeed the spike
entered from multiple points at slightly different delays, not to mention what
reverb, pitch correction, or other effects applied to the original mix channels
would do to the noise signal, you're done before you begin. You'd really need
access to the original multi-tracks to make this feasible.

All this tedium assumes you HEAR the problem in the first place.

Now, if you couldn't do this in DSP with a super narrow notch nulling is worth a
try. You might get lucky. You'd certainly get some good exercise with your spectrum
analyzer.

(A super narrow notch would allow for some hz drift of the interference and wouldn't
care at all about timing.)

>> Oh, and as a side benefit, most UPS provide basic power filtering and protection.
>> THAT'S why I carry a UPS!

>That's the main reason why I put my computers on a UPS, although the battery
>back-up is nice, too. But they weigh a ton--they must be a hassle to lug
>around on location.

I miswrote that -- filtering was NOT the main reason I carried a UPS, it was ALL
things considered regarding the UPS.

My original noisy double-sine unit was 55 pounds in its own box. Yes, a pain. But
happily, its replacement was 15 pounds and fits in a 1U rack space.

My full-blown kit, 24-track primary and 24-track backup, plus microphone preamps,
goes into 3 cases (two 4U and one 6U) and weighs about 60 pounds per case. Add
another 250 pounds in snakes, cabling, microphones, and stands. 15 pounds out of all
that weight is a *very small* safety tax to pay. :)

[Mxsmanic]

Frank Stearns writes:

> Interesting idea, and perhaps worth a try, but here are the difficulties I see:
>
> - exact waveform of the interference... it's likely not going to be perfectly
> symmetrical, as your inverted "cancel" signal will be. As a result, the residual
> might be worse. That is, you might wind up with some sharply peaked and spikey
> waveforms hanging around. And remember, it'd be tough to go look at just this one hz
> because it's buried in program.

Hmm ... yes. But perhaps the sound of horizontal sync in CRTs follows a
predictable pattern. A sawtooth wave, perhaps, since that's what the sync
signal looks like internally.

If it were a big problem, it might be worth having a tool specifically to deal
with it, although with the declining use of CRTs (particularly those with low
scan rates), it's probably too late to care.

> - the easiest way to set this up would be another mix channel with an oscillator.
> You'd precisely match this to the interference hz which, as I've seen, is rarely
> exactly at 15.75K. So you'd have nudge that around as you look for a null, or have a
> *very* high-res spectrum analyzer (higher than I've ever seen).

Yeah, it would be somewhat specialized. And you'd probably need NTSC and PAL
versions. And the free-running sync of the CRTs will indeed be slightly
different (lower in frequency) than the synchronized rate.

> You're also hoping that the hz of the source interference isn't drifting a hz or two
> back and forth. When it drifts away from your cancel frequency, kiss your
> null goodbye.

If it's receiving a signal, the signal would have to be pretty stable in
frequency, as it is piloted by the signal source. If it's not receiving a
signal, it could drift a lot.

But other things have just occurred to me. If they are computer monitors, and
not just television sets, you have a new problem, because the noise could be
at any one of many different frequencies. Most latter-day computer CRTs have
scan rates so high that they are well out of the audio range, but older
interlaced monitors do indeed produce noise comparable to that of TV sets. But
you'd have to be able to adjust the frequency a lot.

> You'd also have a delay plug-in on that channel to get the signal exactly at the
> perfect null point. Pressing the "polarity" button on the channel isn't going to
> mean a thing -- you have no idea as to the timing of the original interference.
>
> Problem is, the delay plug-in will be stepped in samples, which might be far too
> coarse of a timing step to get the waveform nudged to perfect nulling, especially if
> the data is at 44.1Khz.

You could compute and sample a waveform at a very high rate and then compute
the samples that would be needed at a lower rate to cancel the noise with a
given phase, I think (?).

> - if the spike entered multiple channels at various delays and levels of a
> multitrack recording, you might very well have to pair up a nuller channel with
> each program channel.

This is looking more and more expensive, even in software.

> All this tedium assumes you HEAR the problem in the first place.

Well, you might see it in the spectrum.

> My original noisy double-sine unit was 55 pounds in its own box. Yes, a pain. But
> happily, its replacement was 15 pounds and fits in a 1U rack space.
>
> My full-blown kit, 24-track primary and 24-track backup, plus microphone preamps,
> goes into 3 cases (two 4U and one 6U) and weighs about 60 pounds per case. Add
> another 250 pounds in snakes, cabling, microphones, and stands. 15 pounds out of all
> that weight is a *very small* safety tax to pay. :)

Agreed. I'm limited to what I can carry on my person, which is just a few
pounds.

[Anahata]

On Fri, 10 Feb 2012 02:41:34 +0100, Mxsmanic wrote:

> If you know it's actually from a CRT, couldn't you superimpose a signal
> of exactly the same frequency and opposite phase and remove it, without
> affecting anything else? Like astromomers do when they use extremely
> narrow filters to completely remove the yellow light from low-pressure
> sodium-vapor streetlights (which have an extremely monochromatic light).

They use a filter (which is what Frank was suggesting in the first
place); they don't add an equal and opposite waveform, which would be
impossible as the original sodium light is not coherent.

You could do it the way you suggest for CRT line frequency. You'd need a
phased locked loop to generate a tone that matched the interference. You
only need a sine wave to cancel out the 18kHz fundamental. Harmonics at
36 kHz and above can be more conventially filtered.

Better to keep the interference out in the first place though, as
everyone has said.

--
Anahata
ana...@treewind.co.uk -+- http://www.treewind.co.uk
Home: 01638 720444 Mob: 07976 263827

[Anahata]

On Fri, 10 Feb 2012 12:33:58 +0100, Mxsmanic wrote:


> Hmm ... yes. But perhaps the sound of horizontal sync in CRTs follows a
> predictable pattern. A sawtooth wave, perhaps, since that's what the
> sync signal looks like internally.

You've no idea what it will look like when it's been picked up by your
audio system. But (as mentioned in my previous post) the waveform is
determined by harmonics outside the audible range which can be knocked
out with a low pass filter.

> If it were a big problem, it might be worth having a tool specifically
> to deal with it

Like some tape recorders that used to have a "MPX filter" to remove the
19kHz pilot tone in case you were recording from FM radio?

[Mike Rivers]

On 2/9/2012 6:37 PM, Mxsmanic wrote:

>> I know many audio engineers, both live and studio, that have
>> hearing loss and still manage to do good work.

> So what do they do? And did they damage their hearing from their work, or did
> they lose it for other reasons?

People can suffer hearing loss from many reasons including
just getting old. Some do have accelerated hearing loss due
to exposure to loud music as part of their work. Some were
in loud bands, lost their hearing there, and moved into
engineering and production. Lots of reasons.

As to what they do - they depend on experience, knowing how
they hear "good" recordings, understand the kind of music
that they're producing, and they are aware of what might be
unexpected. They ask for other opinions - few engineers,
even those with golden ears, work entirely without
consulting others. Maybe it's the band or musician they're
recording (the customer must be satisfied), maybe it's
another engineer, assistant, or intern working on the
session, maybe it's the mastering house. And some simply
move into roles where they don't need golden ears.

Now if you're making your own music and producing yourself
on a desert island and you have no idea what it actually
sounds like, well, I really don't have a good answer for you
other than to just play for fun and don't worry about what
others think. Or send it to someone else to polish it up.

> Well, if the room is filled with old-style CRTs, what can you do?

Turn them off or find another place to work. You wouldn't
try to record a lead vocal track on a busy street corner,
would you? Unless you wanted that for an effect, of course.

This is a perfect example of knowing what you're doing.
Actually, the problem you're more likely to have with CRTs
in a recording environment isn't with the horizontal sweep
frequency, which is really too high to be picked up by most
microphones unless they're just inches away. You're more
likely to get very noticeable low-mid buzz from the vertical
sweep being picked up by a guitar pickup. If you're really
worried about the horizontal sweep frequency, you could
simply filter everything above 15 kHz. What else would you
be recording that has usable musical content up there? You
probably could live without the CRTs while recording the
triangle solo.

[Mike Rivers]

On 2/9/2012 6:41 PM, Mxsmanic wrote:

> I do notice that some professionals who lack confidence in their own
> abilities feel threatened any time someone they consider external to the
> profession starts to ask questions or make any assertions at all. Those who
> are not insecure remain undisturbed.

And some are just naive. But you're right - there are a lot
of people who are insecure about their work, and live in
fear that someone will hear something that they missed. This
goes away with experience, and if it doesn't, then you're
REALLY in the wrong business. This mostly tends to be a
"gear" thing rather than a "stray noises" thing, though.

Novices are frequently asking if they should get a better
mic preamp, when what they really need is better monitoring
so they can make that decision themselves. Mostly, if there
are real problems with their work, it's with how they're
working and not what they're working with.

[Mike Rivers]

On 2/9/2012 6:35 PM, Mxsmanic wrote:

> I'm a music listener, rather than a music maker. I record ambient sounds or
> speech, mostly, for my little touristy videos.

This explains a lot. You should have mentioned this earlier.

> When I record street scenes, I see an awful lot of
> low-frequency noise, but I have no idea where most of it is coming from. In
> real life, I also hear it (based on comparisons I've done), but for whatever
> reason, I don't notice it as much. Really low frequencies seem to be something
> that you notice unconsciously, even when you can hear them.

Well, this is what's really there. Why not record them and
make use of them? Or, alternatively, be aware that they're
present in your recordings and, if inappropriate, simply use
a different recording or filter so that you get the effect
that you want.

> I recall it being around 30 kHz, so the number of people actually hearing it
> would be too small to explain a general restlessness. But maybe it affected
> them in some other way than through hearing alone. Apparently it was very
> loud.

Are you talking about a recording? Or about something in
real life? If you live adjacent to an artillery test range,
you most likely will be exposed to high amplitude low
frequency energy now and then. People who live adjacent to
Camp Pendelton get used to it. But recording something like
this (which isn't easy to do well) might be effective in one
of your videos.

> In the story I read, someone looked at a graphic equalizer or something and
> noticed a huge spike at a very high frequency.

Movies are better than ever. ;)

> But sometimes the noise is coming from something that isn't yours, and it's a
> surprise.

Some recordings just turn out to be unusable. So you do it
again, or make the best of what you have.

> For field recordings, I don't have a spectrum analyzer handy.

Are you also blind? Do you have no friends to work with who
can help you out with your hearing impairment, or confidence
impairment?

> Well, places like discos already use similar concepts. But you could design
> something much more detailed, like a spectrum analyzer, that would provide
> enough information to understand things like speech, for a trained observer.

People who study speech and hearing indeed do use tools like
spectrum analysis. But that's science, not recording
technology. Different tools for different purposes.

[Mike Rivers]

On 2/9/2012 8:41 PM, Mxsmanic wrote:

> If you know it's actually from a CRT, couldn't you superimpose a signal of
> exactly the same frequency and opposite phase and remove it, without affecting
> anything else?

No two are alike. It's a good theory that doesn't work in
practice.

[Mike Rivers]

On 2/9/2012 8:41 PM, Mxsmanic wrote:

> If you know it's actually from a CRT, couldn't you superimpose a signal of
> exactly the same frequency and opposite phase and remove it

Actually, I have a better answer than what I just posted. If
you have a sample of the noise that's fairly well isolated,
there are "noise cancellation" programs that, in a more
sophisticated version of what you propose, do what you're
dreaming about. They work by analyzing the spectrum of the
sample of the noise that you want to remove and subtracting
that from the program material. It can work fairly well for
things like line frequency hum or the noise of a fan (but
not the wind) from an air conditioner. But the more
frequencies that are involved in the noise source (and CRT
sweep noise isn't just a single frequency) the less
effective the noise reduction can be without damaging the
primary program material.

[Frank Stearns]

Mike Rivers <mri...@d-and-d.com> writes:

>sweep being picked up by a guitar pickup. If you're really
>worried about the horizontal sweep frequency, you could
>simply filter everything above 15 kHz. What else would you
>be recording that has usable musical content up there? You

Gawk! Mike! Are you sure about that???

With many program sources, I agree with you (including a run-of-the-mill pickup).

But others, say piano, acoustic guitar (recorded with good microphones or some of
the newer super-good-sounding pickups), drums, choral/orchestral, et al, it'd hurt
to lop off the upper half of the last octave. Up there you have some sweet
harmonics, some useful pick and attack noise, and so on.

I have done the narrow notch thing and wasn't keen on the results (you could
definitely hear that something was gone), so going to even more removal makes me
hesitate, unless I knew that as you say, no usable musical content was up there on
that particular track.

YMMV.

[Frank Stearns]

Mike Rivers <mri...@d-and-d.com> writes:

>On 2/9/2012 8:41 PM, Mxsmanic wrote:

>> If you know it's actually from a CRT, couldn't you superimpose a signal of
>> exactly the same frequency and opposite phase and remove it

>Actually, I have a better answer than what I just posted. If
>you have a sample of the noise that's fairly well isolated,
>there are "noise cancellation" programs that, in a more
>sophisticated version of what you propose, do what you're
>dreaming about. They work by analyzing the spectrum of the
>sample of the noise that you want to remove and subtracting
>that from the program material. It can work fairly well for

Mike, have you run across a particular package that does this really well? I've got
a couple different venues that have some specific noise spectra between 90 and 110
hz. (I know; weird frequency range. Seems to be some sort of "blow across the coke
bottle" resonance in the air returns which appear to be too small for the volume of
the halls. In one hall you can stand about four feet from one of the two returns,
release a piece of paper from chest height, and it quickly sails to the grill and
smack, becomes a prisoner.)

I've messed with the periodic noise removal tool in sound forge and it seems
completely useless for this. (And I have recorded some nice samples of this noise
when it was just me in the hall. Even though it sounds like periodic noise, perhaps
it's too chaotic? The analyzer shows a fair amount of random bouncing at that hz
range; turbulance in the resonate cavity of the return shaft, no doubt.)

Be curious to hear any product recommendations you might have. Maybe there's a
really smart one that could keep adjusting itself to variations in the noise
spectra.

Thanks in advance,

[Arny Krueger]

"Frank Stearns" <franks.pa...@pacifier.net> wrote in message
news:6c2dnROApvXkuqjS...@posted.palinacquisition...

Cool Edit Pro and Audition have a DTMF tone removal tool that is very
effective for removing narrowband noises and tones if you manually change
its center frequencies, which is fully supported.

If you can find a DTMF tone removal plug in with similar ease of use, it
might be just as good.

[Arny Krueger]

"Frank Stearns" <franks.pa...@pacifier.net> wrote in message

news:JY2dnfXxVYLCv6jS...@posted.palinacquisition...

> Mike Rivers <mri...@d-and-d.com> writes:
>
>>sweep being picked up by a guitar pickup. If you're really
>>worried about the horizontal sweep frequency, you could
>>simply filter everything above 15 kHz. What else would you
>>be recording that has usable musical content up there? You
>
> Gawk! Mike! Are you sure about that???
>
> With many program sources, I agree with you (including a run-of-the-mill
> pickup).
>
> But others, say piano, acoustic guitar (recorded with good microphones or
> some of
> the newer super-good-sounding pickups), drums, choral/orchestral, et al,
> it'd hurt
> to lop off the upper half of the last octave. Up there you have some sweet
> harmonics, some useful pick and attack noise, and so on.

Yes, but through the magic of the ear's masking, you may never notice their
loss in a DBT.

I pretty well predict that you will notice their loss in a sighted
evaluation, until you get really comfortable with the easily regrettable
results in the DBT.

It is pretty well known that brick walling at 16 KHz is usually benign.

You can test things like this using any of the software DBT test
controllers, such as the one that is a plug-in for the Foobar music player
freeware.

[Scott Dorsey]

Mxsmanic <mxsm...@gmail.com> wrote:
>I'm a music listener, rather than a music maker. I record ambient sounds or
>speech, mostly, for my little touristy videos. Indeed, I avoid recording any
>kind of music, because copyright trolls on YouTube will latch onto it to make
>fraudulent infringement claims.

Then, out of curiosity, what are you doing in an audio production newsgroup?
And what makes you think you can tell people in an audio production newsgroup
how to do their jobs?

>That's the mystery. When I record street scenes, I see an awful lot of
>low-frequency noise, but I have no idea where most of it is coming from. In
>real life, I also hear it (based on comparisons I've done), but for whatever
>reason, I don't notice it as much. Really low frequencies seem to be something
>that you notice unconsciously, even when you can hear them.

I can take NYC for about two days because of the constant rumble everywhere.
I can't help but notice it. I suspect people who live there all the time get
used to it. I can stop in a quiet office building and feel the subway going
by. And yes, it goes into everything, which is why you need good monitoring.
--scott
--
"C'est un Nagra. C'est suisse, et tres, tres precis."

[Frank Stearns]

"Arny Krueger" <ar...@cocmast.net> writes:

snips

>It is pretty well known that brick walling at 16 KHz is usually benign.

Eeeeekkkk!!!! Arny, my man! "Well known" to what people smoking what kind of rope?!

Seems to me someone going for that would have to have limited monitoring, a
poor room, or have perhaps been conditioned to the 16K brick wall used in most
MP3 encodes, even the higher bit-rate ones, to make a pronouncement like that.

As a classical and acoustic music engineer, I can assure you that 16-20K **IS**
significant, and I'd take a long walk of a short pier before lopping it off.

>You can test things like this using any of the software DBT test
>controllers, such as the one that is a plug-in for the Foobar music player
>freeware.

Something that huge is bloody obvious in my room. And with one exception, all of my
clients would immediately hear it as well (they wouldn't have to see a damn thing,
either).

My favorite mastering engineer would bring me the white jacket with the really long
sleeves if I did something like that. He doesn't use a spectrum analyzer, and
sometimes chides me when I ask for certain changes numerically. He's totally an
"ear" guy and is well aware of the suggestive nature of the visual component.

I can just see him if I brought in a master like that. He'd swing around in his
chair, lean forward, glare and me, and say, "WTF happened to this?"

No, thanks. I'll maintain to 20K. Not much point in going a whole lot higher, I
agree, but you're cutting pretty deep into bone at 16K -- at least for the kind of
work I do. With a lot of pop/rock even 14K is probably fine, perhaps even preferable
so as to remove nasty artifacts from over-processing, clipping, etc., but not with
acoustic sources that are to be maintained in hi-fi.

[Richard Webb]

On Fri 2012-Feb-10 07:15, Mike Rivers writes:
>> I do notice that some professionals who lack confidence in their own
>> abilities feel threatened any time someone they consider external to the
>> profession starts to ask questions or make any assertions at all. Those who
>> are not insecure remain undisturbed.

Sometimes asking the right questions will help the neophyte
learn however. As an old instructor once taught me, asking
questions is good, asking the right question can mvoe you
toward understanding <g>.

>And some are just naive. But you're right - there are a lot of
> people who are insecure about their work, and live in
> fear that someone will hear something that they missed. This goes
> away with experience, and if it doesn't, then you're
> REALLY in the wrong business. This mostly tends to be a
> "gear" thing rather than a "stray noises" thing, though.

INdeed, and one thing that may help mxmanic with the type of thing he does is to consider how he intends to use the
captured sound, then decide how much massaging to do with
it.


As a simple example, he can think of that dense arrangement
he hears where the acoustic guitar is more apparent from
just the sound of strumming. The track in isolation
probably doesn't sound that great, or that much like an
acoustic guitar, but when blended with the otehr parts of
the arrangement in the mix does just what one wants the
acoustic guitar to do.

That's where the monitoring environment ears and knowledge
can all come together. Often the trick is knowing what
should be worried about and what can be safely ignored
because in the final production it's going to be a nonissue. A good example of that was the discussion of crt scan noise. As Frank noted, in most musical arrangements it's a
nonissue, but if the piece is a single instrument or spoken
word only it becomes one.

What it often boils down to is the old adage "don't sweat
the small stuff."

HEre's an example for him from a project I did about a
decade ago.

Gospel group, the keyboard player had a little bit of a
ground loop problem with one of his keyboards he used on the session. For most of their album project it was a nonissue, that hummmmmm buried in the mix. But, there was one
selection where that keyboard was all alone naked for the
first eight bars. When we went to mixing I bugged him more
than once about bringing in that keyboard sans the rest of
his rig and retracking the intro, and even though we weren't doing the customary billing by the hour he refused. hE was
quite happy with it, but those eight bars drove me crazy
every time I listened to the intro.

I knew i was going to be present during the mastering
however, so I let it ride, mixed the album, and went to the
mastering session. When we came to that track we did the
transfer, then applied a filter to the intro which wasn't
applied once the rest of the ensemble started to play. I
captured enough of the noise in isolation even as I did the
mix so that we could tailor the filter to the hum's exact
characteristics. My only concern then was whether the
filter would negatively impact the tone of the keyboard.

Yeah I know, only eight bars, and he was happy as a clam
with the whole project, but I never would have been happy
with that one going out the door with my name on it and that hum present ,g>.


Regards,
Richard
--
| Remove .my.foot for email
| via Waldo's Place USA Fidonet<->Internet Gateway Site
| Standard disclaimer: The views of this user are strictly his own.

[Mike Rivers]

On 2/10/2012 9:19 AM, Frank Stearns wrote:

> Mike, have you run across a particular package that does this really well? I've got
> a couple different venues that have some specific noise spectra between 90 and 110
> hz.

If you can find it in a spectrogram, you might be able to
reduce it with a "spectral editor" tool. WaveLab has one, so
does newer versions of Audition, so I expect nearly everyone
has it in their latest DAW version. Izotope has a plug-in
that's supposed to be really good, and they have a free
trial demo. You might check it out.

The only thing I've used this technique for is with
transient noise, thoug, like a cough or a chair squeak. I
suspect that Izotope might be better with the kind of
continuous noise you have than Sound Forge. Give it a shot.

[Mike Rivers]

On 2/10/2012 8:57 AM, Frank Stearns wrote:

> piano, acoustic guitar (recorded with good microphones or some of
> the newer super-good-sounding pickups), drums, choral/orchestral, et al, it'd hurt
> to lop off the upper half of the last octave. Up there you have some sweet
> harmonics, some useful pick and attack noise, and so on.

This is where the experience comes in. You have to make a
value judgment as to which is worse - losing the top half
octave that most listeners probably won't hear, or losing a
constant noise that some might find annoying because it's
always there. Though it may be difficult to hear, it's not
musical so it will stand out more than the loss of some high
frequency sounds related to the music.

> I have done the narrow notch thing and wasn't keen on the results (you could
> definitely hear that something was gone)

That's because narrow notches are never narrow enough. You
set up a steep rolloff and start moving it down until you
either improve it or decide you can't.

> unless I knew that as you say, no usable musical content was up there on
> that particular track.

Best way to tell that is to listen to it as you're working.
If you can't hear a change, there's probably nothing that
you can change (with the filter you choose).

[Mike Rivers]

On 2/10/2012 9:52 AM, Arny Krueger wrote:

> Yes, but through the magic of the ear's masking, you may never notice their
> loss in a DBT.

I wouldn't bother with a double blind test. I'd listen and
ask myself "Is it better or worse this way." I would expect
to hear a difference, but I don't want to depend on a
spectrum analyzer to tell me if the difference is good or
bad. I want to rely on my judgment.

I might be wrong sometimes. So send back the record and I'll
give yo ua refund.

[ChrisCoaster]

On Feb 6, 10:28 pm, Mxsmanic <mxsma...@gmail.com> wrote:
> I installed a freeware plug-in on Sound Forge that provides a spectrum
> analysis. When I analyze something, it seems like there's a lot of sound
> energy at low frequencies, even though I don't seem to be hearing that much at
> the low end. There's a "slope" adjustment in the analyzer, but I'm not sure
> what it does--can someone explain it to me?
>
> I tried generating some white noise and then adjusting the slope so that the
> spectrum was relatively flat (since I presume that white noise contains equal
> amounts of sound energy at all frequencies), but I'm not sure that this
> accomplished what I want. I'd just like to see the actual sound levels for
> each frequency.
>
> If it makes a difference, the audio editing program is Sound Forge (the Audio
> Studio version) and the plug-in is VOXengo SPAN.
___________________
I have Audacity - can analyze the same way. I believe it analyzes
based on the "presence" or energy level at certain frequencies vs the
volume of what we actually hear in a given song. Bass contains more
energy and thus the mountains skewed to the left in most of my cases.
If you see a really tall HUMP or peak between 50 & 100Hz then you're
probably analyzing a rap song; the lows were purposely pumped up in
post.

I purposely EQd a "normal" song - without the hip-hop hump(!) just to
see - and hear - what it would sound like "flat", but the result was
tinny and thin sounding, even if the spectro was less skewed and
flatter than before. Why? Our ears are not programmed to hear that
way. If you are EQing a project in post and it has any outstanding
peaks or dips across its spectro there is probably something wrong or
too much EQ is being used in that location.

-CC

[Mxsmanic]

Mike Rivers writes:

> No two are alike. It's a good theory that doesn't work in
> practice.

The horizontal sync is locked on all receivers that are capturing the same
broadcast signal.

[Mxsmanic]

Mike Rivers writes:

> Well, this is what's really there. Why not record them and
> make use of them?

If it's really there, I don't have a problem with it. I just want to make sure
that the equipment is not exaggerating it. It seems less obvious in real life
than it does on the recording, but perhaps that's just because I'm listening
to the recording away from the original environment. When I monitor it during
recording, I don't seem to notice it.

> Are you talking about a recording? Or about something in
> real life?

It was a concert, so I presume the sound was live.

> If you live adjacent to an artillery test range,
> you most likely will be exposed to high amplitude low
> frequency energy now and then. People who live adjacent to
> Camp Pendelton get used to it. But recording something like
> this (which isn't easy to do well) might be effective in one
> of your videos.

I think it would be interesting to analyze a recording of a sonic boom, but
those are hard to come by these days (I did hear them a lot when I was
little).

> Are you also blind? Do you have no friends to work with who
> can help you out with your hearing impairment, or confidence
> impairment?

I can see. I have no one to work with. I lack confidence in domains about
which I don't have a high level of knowledge, which seems very logical to me.

[Mxsmanic]

Scott Dorsey writes:

> Then, out of curiosity, what are you doing in an audio production
> newsgroup?

It seemed like a good source of information.

> And what makes you think you can tell people in an audio production newsgroup
> how to do their jobs?

I don't recall saying anything either way.

> I can take NYC for about two days because of the constant rumble everywhere.
> I can't help but notice it. I suspect people who live there all the time get
> used to it. I can stop in a quiet office building and feel the subway going
> by. And yes, it goes into everything, which is why you need good monitoring.

A subway line runs under my apartment building, and sometimes I can hear the
subway (faintly) passing below. Maybe I'll try recording it sometime, and see
how it turns out.

[Mxsmanic]

Mike Rivers writes:

> Turn them off or find another place to work. You wouldn't
> try to record a lead vocal track on a busy street corner,
> would you? Unless you wanted that for an effect, of course.

But you could be shooting a documentary in a TV studio, and then you'd have
CRTs all around, and they probably could not be shut off. Of course, that's a
rather contrived example.

[Scott Dorsey]

Mxsmanic <mxsm...@gmail.com> wrote:
>
>I think it would be interesting to analyze a recording of a sonic boom, but
>those are hard to come by these days (I did hear them a lot when I was
>little).

http://ntrs.nasa.gov and also the NTIA database will have plenty of papers
with pictures of N-wave waveforms. Since you can't actually reproduce them,
they _are_ best just viewed as time domain plots.

[Scott Dorsey]

That's what engineers do every day.

And unfortunately as people move around in the studio, the noise waveforms
from the monitors change.

That's why we have notch filters.

[Frank Stearns]

True, but you might well get some waveform distortions (based on how the sync signal
got into your audio) that would make cancellation much less than what the
theoretical ideal.

Over the years I've been down the cancellation route on a number of different issues
and indeed, theory is way better than practice.

The problem is, to get an effective cancellation for any application, the target and
inverted cancel signal must be exactly in step -- amplitude, timing,
frequency/waveform -- to do any good. And in the real world, it's durn difficult to
get all three. Usually you're very lucky to get two right, or even one.

Assuming you can take the spectral hit and get a narrow enough notch, the notch will
likely be the better practical solution -- assuming, of course, you can't go back
and correct the initial problem.

[Mike Rivers]

On 2/10/2012 2:31 PM, Mxsmanic wrote:

> The horizontal sync is locked on all receivers that are capturing the same
> broadcast signal.

Kid's got an answer for everything, it seems.

The frequency may be the same, but not all transformers are
the same. It's not the sweep that makes the noise, it's all
the things rattle around that aren't supposed to be transducer.

[Mike Rivers]

On 2/10/2012 2:36 PM, Mxsmanic wrote:

> If it's really there, I don't have a problem with it. I just want to make sure
> that the equipment is not exaggerating it.

Why would your equipment be exaggerating a particular
frequency range? Sure, mics aren't perfectly flat, but you
should know their characteristics before you go to work.

> It seems less obvious in real life
> than it does on the recording, but perhaps that's just because I'm listening
> to the recording away from the original environment.

This is true with just about every recording. You capture
one sound field and play it back in a different environment.
Accurate monitors in a room with some care toward acoustic
accuracy at the listening position goes a long way toward
making things sound like they did when you were there, but
you'll never get an exact match.

> I think it would be interesting to analyze a recording of a sonic boom, but
> those are hard to come by these days (I did hear them a lot when I was
> little).

Bob Katz used to have a recording of a Cape Canaveral
reocket launch that he recorded (with permission) at fairly
close range. www.digido.com

> I can see. I have no one to work with. I lack confidence in domains about
> which I don't have a high level of knowledge, which seems very logical to me.

In that case, you need to get some friends and get some
confidence. Your friends don't need golden ears, just play
them the recording and ask them what they hear.

[Mxsmanic]

Mike Rivers writes:

> The frequency may be the same, but not all transformers are
> the same. It's not the sweep that makes the noise, it's all
> the things rattle around that aren't supposed to be transducer.

But they should all rattle at the same frequency, since the sync frequency is
locked.

[Mxsmanic]

Mike Rivers writes:

> Why would your equipment be exaggerating a particular
> frequency range?

I don't know. Maybe it's not perfect.

> In that case, you need to get some friends and get some
> confidence. Your friends don't need golden ears, just play
> them the recording and ask them what they hear.

Well, I upload things to YouTube, but I don't get much feedback, especially on
the audio (but the audio is usually just ambient noise).

[Scott Dorsey]

Sadly, the fundamental is, but nothing else is.

[hank alrich]

iZotope RX is alleged to be quite powerful for noise removal.

www.izotope.com/rx/

--
shut up and play your guitar * http://hankalrich.com/
http://www.youtube.com/walkinaymusic
http://www.sonicbids.com/HankandShaidri

[Mark]

true, but...

In order to "cancel" it out, you need to match both the amplitude and
phase.

by the time the acoustical signal travels to your mic and gets into
your recorder after bouncing around in the room, the amplitude and
phase will be essentially random. If there is any movement in the
room, the amplitude and phase will change. Some burglar systems use
this principal in fact.

If you are looking for something educational to do, take a mic and
hook it up to your computer and run a real time analyzer and observe
various sounds. It is surprising how well you can see the H sync from
a TV even in the next room, but it is far from stable.

A Notch filter will work well. Cancellation is all but impossible
without some kind of real time phase and amplitude tracking loop.

This program is very educational...

http://www.qsl.net/dl4yhf/spectra1.html

have fun

Mark

[Mike Rivers]

On 2/10/2012 7:21 PM, Mxsmanic wrote:

> Mike Rivers writes:
>> Why would your equipment be exaggerating a particular frequency range?

> I don't know. Maybe it's not perfect.

Well, of course it's not perfect, but if it's exaggerating a
particular frequency range enough to be bothersome to a
reasonable listener, or even to a critical listener, it's
either unsuitable for the job or it's broken. You shouldn't
be using it.

> Well, I upload things to YouTube, but I don't get much feedback, especially on
> the audio (but the audio is usually just ambient noise).

Well, I'm not at all surprised that you're not getting any
comments. With something like that, only you know what it
really sounded like, and unless it's really interesting,
probably nobody else cares.

[Mike Rivers]

I give up, since you won't seem to. But I'm right about this
and you're just guessing and arguing.

[hank alrich]

Mike Rivers <mri...@d-and-d.com> wrote:

> On 2/10/2012 6:36 PM, Mxsmanic wrote:
> > Mike Rivers writes:
> >
> >> The frequency may be the same, but not all transformers are
> >> the same. It's not the sweep that makes the noise, it's all
> >> the things rattle around that aren't supposed to be transducer.
> >
> > But they should all rattle at the same frequency, since the sync
> > frequency is locked.
>
> I give up, since you won't seem to. But I'm right about this
> and you're just guessing and arguing.

He could use a grasp of the concept that what is generated at the source
will differ from what is received and that several variables will affect
what is received.

[Mxsmanic]

Mike Rivers writes:

> Well, I'm not at all surprised that you're not getting any
> comments. With something like that, only you know what it
> really sounded like, and unless it's really interesting,
> probably nobody else cares.

True. A few people have mentioned that they prefer the videos without music
(all but two have no music), which rather surprised me, but that's fine with
me, since it costs money to license background music.