[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[_4_]]

Mxsmanic 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.

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[_4_]]

"Mxsmanic" wrote in message
...

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 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 ,
Mxsmanic wrote:
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.

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.
--scott
--
"C'est un Nagra. C'est suisse, et tres, tres precis."

[Mark]

On Feb 7, 8:06*am, "Arny Krueger" wrote:
"Mxsmanic" wrote in message

...

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!

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 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

--
"C'est un Nagra. C'est suisse, et tres, tres precis."

[Scott Dorsey]

Mxsmanic 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.
--scott



--
"C'est un Nagra. C'est suisse, et tres, tres precis."

[Les Cargill[_4_]]

Mxsmanic wrote:
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.


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" wrote in message
...
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[_4_]]

"Mxsmanic" wrote in message
...
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 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.
--scott
--
"C'est un Nagra. C'est suisse, et tres, tres precis."

[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 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.
--scott
--
"C'est un Nagra. C'est suisse, et tres, tres precis."

[Arny Krueger[_4_]]

"Mxsmanic" wrote in message
...
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[_4_]]

"Mark" 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.

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/

[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.

[[email protected]]

On 2012-02-09 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[_4_]]

"Mxsmanic" wrote in message
...
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.

That's what I would expect.

[Arny Krueger[_4_]]

"Mike Rivers" wrote in message
...
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[_4_]]

"Mxsmanic" wrote in message
...

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]

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 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.


--
"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

[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 **** 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



--
"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

[Arny Krueger[_4_]]

"Mxsmanic" wrote in message
...
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.

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.

[Richard Webb[_3_]]

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 ****?

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.

[[email protected]]

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,

replace anything before at with elspider