[William Sommerwerck]

I'm getting confident that the issue that warrants the use of highest
possible samplerate when making the sampling is reducing aliasing
and getting antialias filters well-out of the audio band.

This is also a theoretical advantage of SACD.

The problem is... what happens when you downsample for commercial release?
You still need to low-pass filter the signal.

[Luxey]

On Mar 7, 3:33*pm, "William Sommerwerck"
wrote:


The problem is... what happens when you downsample for commercial release?
You still need to low-pass filter the signal.

May be, at the time ther's nothing in the region, everything's already
upshifted.

[Peter Larsen[_3_]]

Luxey wrote:

On Mar 7, 3:33 pm, "William Sommerwerck"
wrote:

The problem is... what happens when you downsample for commercial
release? You still need to low-pass filter the signal.

May be, at the time ther's nothing in the region, everything's already
upshifted.

Not correct. The issue is another: you can do the low pass filtering in the
digital domain and it does not have to be done in real time.

Kind regards

Peter Larsen

[Scott Dorsey]

geoff wrote:
PStamler wrote:
William mentioned that few microphones have significant response over
20kHz. I've looked at some of my students' recordings using a spectrum
analyzer, and there was plenty of 20kHz+ signal present. I don't know
exactly which mics were used on which recordings -- most probably
various Neumanns on most of them -- but there was lots of signal up
above the audible range.

Naa, those were produced by the Aural Exiter (aka clipping) !

The exciter is definitely NOT clipping. The distortion spectrum is totally
different.
--scott

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

[Scott Dorsey]

Arny Krueger wrote:
"PStamler" wrote in message

What's that got to do with whether microphones produce significant
output over 20kHz?

It means that few if anyy ever actually hear it. Therefore, it is largely
sonically moot.

No, it's actually very important. Because, if we cannot hear it, we need
to filter it out of the chain as quickly as possible to prevent audible
intermodulation products from being generated. But if we _can_ hear it,
then we need to preserve it.
--scott

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

[[email protected]]

On Tuesday, March 6, 2012 5:30:35 PM UTC-5, Peter Larsen wrote:
I recorded a string quartet using very good violins yesterday ... the cleanest record-replay was at 192 kHz sample rate.

Unless you recorded the same performance at different sample rates using parallel identical equipment, you are comparing the performances much more than the sample rate.

If you did record using identical parallel gear, please post some clips. I've played in orchestras many times, and I'm familiar with how good quality violins sound. If your recording at 44.1 KHz really does sound like plastic toy violins, I think you did something wrong. :-)

--Ethan

[hank alrich]

wrote:

On Tuesday, March 6, 2012 5:30:35 PM UTC-5, Peter Larsen wrote:
I recorded a string quartet using very good violins yesterday ... the
cleanest record-replay was at 192 kHz sample rate.

Unless you recorded the same performance at different sample rates using
parallel identical equipment, you are comparing the performances much more
than the sample rate.

If you did record using identical parallel gear, please post some clips.
I've played in orchestras many times, and I'm familiar with how good
quality violins sound. If your recording at 44.1 KHz really does sound
like plastic toy violins, I think you did something wrong. :-)

--Ethan

I took Peter's plastic violin comment to mean that in this recording
situation something was lost in the tracks captured at the lower sample
rate as compared to the higher sample-rate captures, not that the
violins actually sounded like they came from Wal-Mart.

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

[Scott Dorsey]

Mike Rivers wrote:

When Sennheiser came out with the MKH-800 mic, they made a
big deal that it had reasonably mic-like response (my term,
not theirs, meaning that it was actually useful, not a lab
curiosity) up to 40 kHz or so. The idea was that there was
finally a mic that could take advantage of 2x sample rates.

Tannoy had a 50 kHz tweeter about the same time. Don't know
what ever became of that, but the MKH-800 is still around
and, as far as I know, still around $3,000.
http://www.sennheiserusa.com/media/p...oductSheet.pdf

There was a huge burst of that sort of thing when high sampling rate
systems came out. Pioneer had the supertweeter speakers before Tannoy
even, and Sanken and Schoeps both came out with mikes designed for
ultrasonic response at about the same time.

This stuff is all still available although it's not quite as aggressively
pushed as it was when it first came out.
--scott
--
"C'est un Nagra. C'est suisse, et tres, tres precis."

[Les Cargill[_4_]]

Peter Larsen wrote:
Arny Krueger wrote:

"William wrote in message
...

"Peter wrote in message
k...

I recorded a string quartet using very good violins yesterday. The
shortest possible signal chain pre A/D conversion, so it was a good
chance to compare the actual sound quality of an R44 using
different sample rates, on that actual recorder the cleanest
record-replay was at 192 kHz sample rate. Equipment used: quality
condenser mic, minimalist preamp, Edirol R44, Sennheiser HD25
headphones.

"Cleanest" in what way?

Probably clean from the experimental controls that normally used for
this kind of evaluation when it matters. E.g. ITU BS 1116-1.

It is quite rare for you not to get the point made. Said point is that for a
valid test of a recording system you need audio that is not previously
recorded because otherwise it is undefined what it is you are testing.

As for your comment in another post to the effect that you have found it
possible to sample rate convert a high sample rate recording down and then
up again with no obvious audible artifacts that is just what I would expect
and correlates well with my findings when doing high sample rate recordings
and then downsampling them, ie. they remain cleaner that recordings made at
44.1 kHz sample rate.

I'm getting confident that the issue that warrants the use of highest
possible samplerate when making the sampling is reducing aliasing and
getting antialias filters well out of the audio band. There does not at an
initial glance seem to be any consideration of that issue in the literature
referred to in this thread, it has instead become bandwidth religion.



My understanding is that oversampling converters handle this.

if you think that's true, I'd think it could be measured fairly easily,
without bothering with any subjective stuff at all. Indeed, a
single pulse should prove it true or not.

Kind regards

Peter Larsen




--
Les Cargill

[[email protected]]

On Wednesday, March 7, 2012 12:12:11 PM UTC-5, hank alrich wrote:
I took Peter's plastic violin comment to mean that in this recording
situation something was lost in the tracks captured at the lower sample
rate as compared to the higher sample-rate captures, not that the
violins actually sounded like they came from Wal-Mart.

That's what's wrong with so much audiophile wording - it's unclear. One man's "musical" is another man's "harsh." But my main point is that tests of subtle differences such as sample rate must be done using the same performance. This is a very common testing error, recording something one way then making a *second* recording another way.

--Ethan

[PStamler]

On Mar 7, 6:49*am, "Arny Krueger" wrote:
"PStamler" wrote in message

...
On Mar 6, 4:36 pm, "Arny Krueger" wrote:



"PStamler" wrote in message

....

William mentioned that few microphones have significant response over
20kHz. I've looked at some of my students' recordings using a spectrum
analyzer, and there was plenty of 20kHz+ signal present. I don't know
exactly which mics were used on which recordings -- most probably
various Neumanns on most of them -- but there was lots of signal up
above the audible range.

What exactly does "plenty of 20 KHz+" mean in dB @ various frequencies?
I don't have quantitative figures, merely noted that as levels on the
spectrum analyzer rose and fell with the music, there was significant
content 20kHz. In some instances, greater than between 10-20kHz
(possibly due to microphone resonances in the ultrasonic regions).
Then there is the problem of the human ear's construction.
(1) Note that the Fletcher Munson curves fall off at approx 12 dB/octave
above 5 KHz. That's for a normal young adult.
(2) Hearing is based on hair cells within the Organ of Corti on the
Basilar
membrane, with the high frequency related hairs first. The first cells
generally respond to frequencies just below 20 KHz, and there are no hair
cells that respond to higher frequencies.
(3) Masking. If the spectral content of the sound being heard is not
rising
at a rate of 12 dB/octave then lower frequencies will control the
critical
band and the higher frequencies, even if audible by themselves, will be
masked and not heard.
What's that got to do with whether microphones produce significant
output over 20kHz?

It means that few if anyy ever actually hear it. Therefore, it is largely
sonically moot.

Not if the concern is audible IM products generated by ultrasonic
signals, which I thought was one of the original article's chief
complains about high-sample-rate tracks.

I don't know if this is really an issue, but I at least wanted to add
the data point that there *is* signal up in the inaudible regions with
modern condenser mics. Presumably high-sample-rate recordings would
retain that signal, and it *might* cause IM distortion. I still want
to see some data showing that it *does* cause IM, in real recordings
played on real systems.

Peace,
Paul

[Scott Dorsey]

wrote:

That's what's wrong with so much audiophile wording - it's unclear. One man=
's "musical" is another man's "harsh." But my main point is that tests of s=
ubtle differences such as sample rate must be done using the same performan=
ce. This is a very common testing error, recording something one way then m=
aking a *second* recording another way.

My personal favorite was the Kangawa Institute paper where they played one
piece of music recorded at 44.1 ksamp/sec, and another different piece of
music recorded at 96 ksamp/sec. They measured alpha brain wave activity
among the listeners and found that people listening to the higher sample rate
recording were more relaxed than the people listening to the lower sample
rate recording.

No word about what the actual music was, my guess is it was the Sex Pistols
and Mozart....
--scott

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

[Trevor]

"Scott Dorsey" wrote in message
...
Arny Krueger wrote:
It means that few if anyy ever actually hear it. Therefore, it is largely
sonically moot.

No, it's actually very important. Because, if we cannot hear it, we need
to filter it out of the chain as quickly as possible to prevent audible
intermodulation products from being generated. But if we _can_ hear it,
then we need to preserve it.

IF you can preserve it without "audible intermodulation products from being
generated", then there is no need to filter it out "as quickly as possible"
regardless of whether you can hear it.
IF you can't preserve it without "audible intermodulation products from
being generated", you need a better recording system.
IF you are talking about aliasing however, you simply need to learn how to
properly deal with that.

Trevor.

[Trevor]

"Arny Krueger" wrote in message
...
Back in 2002 I did a very similar experiment by recording some music that
was very rich in 20 KHz content and/or had very wide dynamic range
courtest of close micing and a very quiet room.

I bet it sounds the same as the DVD-A.

I downsampled my hi-rez recordings down to 16/44 and then upsampled them
back up to 24/96 so that they technically were the same. Of course the
upsamping was totally benign, neither restoring that which was lost nor
adding any but microscopic artifacts.

In blind tests using a variety of listeners of various ages, a
statistically significant result was not found, either for all listeners
as a group, or for each listener by him or herself.

And I bet you would have been surprised if the result was otherwise, I know
I'd be looking for *your* mistake if it was :-)

Trevor.

[Scott Dorsey]

In article , Trevor wrote:

No, it's actually very important. Because, if we cannot hear it, we need
to filter it out of the chain as quickly as possible to prevent audible
intermodulation products from being generated. But if we _can_ hear it,
then we need to preserve it.

IF you can preserve it without "audible intermodulation products from being
generated", then there is no need to filter it out "as quickly as possible"
regardless of whether you can hear it.

But you can't. All systems are nonlinear. You don't know to what degree
the nonlinearity is a problem. You don't know how nonlinear the user's
playback system is. So filter it out.

IF you can't preserve it without "audible intermodulation products from
being generated", you need a better recording system.
IF you are talking about aliasing however, you simply need to learn how to
properly deal with that.

I'm not talking about aliasing, I'm talking about nonlinearity in ordinary
everyday transducers and analogue electronics.
--scott

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

[William Sommerwerck]

Presumably high-sample-rate recordings would retain that
[ultrasonic] signal, and it *might* cause IM distortion.

With respect to recording and then playing back at high data rates, this is
unlikely, simply because I doubt speakers and amps are /that/ badly behaved.

On the other hand, if high-sample-rate recordings are downsampled, there has
to be correctly applied LP filtering, or you're going to get aliasing (which
is a form of IM) within the audible passband. It is my opinion that this is
one of the reasons -- probably /the/ reason -- that digital recordings
sometimes sound "hashy", and transfers from analog recordings hardly ever
do.

[Jason[_14_]]

In article 9e063818-9dbc-4f6b-aba7-bb2680d8b588
@x17g2000yqj.googlegroups.com, says...

William mentioned that few microphones have significant response over
20kHz. I've looked at some of my students' recordings using a spectrum
analyzer, and there was plenty of 20kHz+ signal present. I don't know
exactly which mics were used on which recordings -- most probably
various Neumanns on most of them -- but there was lots of signal up
above the audible range.

Peace,
Paul

Last Sunday, I recorded a local ensemble. One of the pieces called for
the violinist to play a sustained harmonic that was so high that I was
having trouble hearing it. (So was she.) I recorded it at 24/96 with an
NT-4. Audition's spectral dislay shows a fair amount of signal as high as
34kHz.

Jason

[Anahata]

On Wed, 07 Mar 2012 17:47:40 -0500, Jason wrote:

Last Sunday, I recorded a local ensemble. One of the pieces called for
the violinist to play a sustained harmonic that was so high that I was
having trouble hearing it. (So was she.) I recorded it at 24/96 with an
NT-4. Audition's spectral dislay shows a fair amount of signal as high
as 34kHz.

I take it the specified note itself wasn't at 34kHz?

There's no doubt that sounds exist above 20kHz, and there's no reason why
a mic shouldn't respond to them, but I think the significant part of that
story is that the violinist, who was surely closer than anyone to the
sound source, could hardly hear it herself. Recording sounds we can't
hear and thus running the risk of intermodulation products that we *can*
hear seems pointless.

Microphones, amplifiers and speakers are often not well specified for
frequencies above 20kHz, and there are good reasons to expect some of
those components to be non-linear outside their expected working range.

--
Anahata
--/-- http://www.treewind.co.uk
+44 (0)1638 720444

[William Sommerwerck]

"anahata" wrote in message
o.uk...

Microphones, amplifiers and speakers are often not well specified for
frequencies above 20kHz, and there are good reasons to expect some of
those components to be non-linear outside their expected working range.

There are no such good reasons. The equipment doesn't "know" that human
hearing ends at 20kHz.

[Luxey]

On Mar 7, 4:06*pm, "Peter Larsen" wrote:
Luxey wrote:
On Mar 7, 3:33 pm, "William Sommerwerck"
wrote:
The problem is... what happens when you downsample for commercial
release? You still need to low-pass filter the signal.
May be, at the time ther's nothing in the region, everything's already
upshifted.

Not correct. The issue is another: you can do the low pass filtering in the
digital domain and it does not have to be done in real time.

* Kind regards

* Peter Larsen

I don't know why, some of my messages do not appear, necvermind, I'll
repeat:

yesterday, I wanted to add "not to mention everything's already
digital", but could not be bothered, now I can only play smart ass.

Also, I had one about sentences containing both "young girls" and "old
men with no hope". That one did not show either.

[Luxey]

On Mar 8, 10:33*am, anahata wrote:
On Wed, 07 Mar 2012 17:47:40 -0500, Jason wrote:


There's no doubt that sounds exist above 20kHz,

Yes, it's called ultrasound, it's for dogs, or whatever, not for
humans.

[John Williamson]

William Sommerwerck wrote:
"anahata" wrote in message
o.uk...

Microphones, amplifiers and speakers are often not well specified for
frequencies above 20kHz, and there are good reasons to expect some of
those components to be non-linear outside their expected working range.

There are no such good reasons. The equipment doesn't "know" that human
hearing ends at 20kHz.


Not a good reason, maybe, but the designers "know" that nobody can hear
anything above 20kHz, so they're not always as careful as they might be
about handling frequencies above that, especially if they can save a few
pennies.

--
Tciao for Now!

John.

[William Sommerwerck]

"John Williamson" wrote in message
...
William Sommerwerck wrote:
"anahata" wrote in message
o.uk...

Microphones, amplifiers and speakers are often not well specified
for frequencies above 20kHz, and there are good reasons to expect
some of those components to be non-linear outside their expected
working range.

There are no such good reasons. The equipment doesn't "know" that
human hearing ends at 20kHz.

Not a good reason, maybe, but the designers "know" that nobody can
hear anything above 20kHz, so they're not always as careful as they
might be about handling frequencies above that, especially if they can
save a few pennies.

One argument in favor of reproducing ultrasound is that IM products produced
by the ear's non-linearities might be audible. There have been systems that
use these non-linearities to produce "full-range" sound from dinky piezo
speakers.

[Arny Krueger[_4_]]

"geoff" wrote in message
...
PStamler wrote:
William mentioned that few microphones have significant response over
20kHz. I've looked at some of my students' recordings using a spectrum
analyzer, and there was plenty of 20kHz+ signal present. I don't know
exactly which mics were used on which recordings -- most probably
various Neumanns on most of them -- but there was lots of signal up
above the audible range.


Naa, those were produced by the Aural Exiter (aka clipping) !


The classic design for the Aphex Aural Exciter produces even order nonlinear
distortion. This is not the same as the symmetrical clipping we see on
certain modern CDs. Symmetrical clipping produces odd-order nonlinear
distortion. Two very different animals!

[Arny Krueger[_4_]]

"William Sommerwerck" wrote in message
...
I'm getting confident that the issue that warrants the use of highest
possible samplerate when making the sampling is reducing aliasing
and getting antialias filters well-out of the audio band.

This is also a theoretical advantage of SACD.

The problem is... what happens when you downsample for commercial release?
You still need to low-pass filter the signal.


In most system design, you do the bandpass limiting as early in the process
as possible to avoid problems due to nonlinear distortion, etc.

Therefore, if we could practically put the bandpass limiting into the
microphones, that would the place to do it.

[Peter Larsen[_3_]]

Arny Krueger wrote:

Therefore, if we could practically put the bandpass limiting into the
microphones, that would the place to do it.

My opinion is that this is one of the reasons why dynamic microphones can be
a good choice on a drumkit, the other being the lower output signal and thus
less risk of input clipping.

Another can of worms tho' ... but worrying about stressing systems with
"useless" treble energy comes across as very sensible.

It also makes sense to consider that in the context of high sample rate
recordings. With this example 192 kHz sample rate sounded best on site, but
the energy plot of the recording shows noise increasing with 12 dB pr.
octave from 50 kHz and up, and it may quite possibly be unwise to bother a
poweramp and treble units with.

Anybody know whether an Oade modified R44 has the same noise spectrum?

Kind regards

Peter Larsen

[Peter Larsen[_3_]]

wrote:

On Wednesday, March 7, 2012 12:12:11 PM UTC-5, hank alrich wrote:
I took Peter's plastic violin comment to mean that in this recording
situation something was lost in the tracks captured at the lower
sample rate as compared to the higher sample-rate captures, not that
the violins actually sounded like they came from Wal-Mart.

That's what's wrong with so much audiophile wording - it's unclear.
One man's "musical" is another man's "harsh." But my main point is
that tests of subtle differences such as sample rate must be done
using the same performance. This is a very common testing error,
recording something one way then making a *second* recording another
way.

You're quite right, it was the same performance, it was the rehearsal.

--Ethan

Kind regards

Peter Larsen

[Peter Larsen[_3_]]

wrote:

On Tuesday, March 6, 2012 5:30:35 PM UTC-5, Peter Larsen wrote:
I recorded a string quartet using very good violins yesterday ...
the cleanest record-replay was at 192 kHz sample rate.

Unless you recorded the same performance at different sample rates
using parallel identical equipment, you are comparing the
performances much more than the sample rate.

If you did record using identical parallel gear, please post some
clips.

Ethan, I used the same recorder, stopped it, changed sample rate, started
again and listened and took he earphones off and listened to the ensemble. I
started out at 48 kHz since that usually is my "house standard", tried 44.1
to hear if it was OK enough to skip the sample rate conversion, it
definitely wasn't - sounded just like my Fostex MR8HD (which I hold in
higher and higher regard) and then tried 96 - cleaner than 48, but 192 just
was more open with better spatial rendering.

I can not post this recording, it would be simpler to so do, but I'm not
gonna ask and won't post without asking.

--Ethan

Kind regards

Peter Larsen

[[email protected]]

On Thursday, March 8, 2012 11:00:28 AM UTC-5, Peter Larsen wrote:
I can not post this recording, it would be simpler to so do, but I'm not
gonna ask and won't post without asking.

Surely nobody would object to a pair of ten-second snippets.

--Ethan

[hank alrich]

wrote:

On Thursday, March 8, 2012 11:00:28 AM UTC-5, Peter Larsen wrote:
I can not post this recording, it would be simpler to so do, but I'm not
gonna ask and won't post without asking.

Surely nobody would object to a pair of ten-second snippets.

--Ethan

Sometimes one just does not ask a client for that permission. Seriously.

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

[Scott Dorsey]

Peter Larsen wrote:

Ethan, I used the same recorder, stopped it, changed sample rate, started
again and listened and took he earphones off and listened to the ensemble. I
started out at 48 kHz since that usually is my "house standard", tried 44.1
to hear if it was OK enough to skip the sample rate conversion, it
definitely wasn't - sounded just like my Fostex MR8HD (which I hold in
higher and higher regard) and then tried 96 - cleaner than 48, but 192 just
was more open with better spatial rendering.

I can believe that, and I hear stuff like that all the time. But I am not
willing to attribute it to the standard so much as to artifacts in the
conversion electronics.

Take your 192 file, downsample to 48, then upsample back to 192. If you
can hear a difference between the original and the resampled data, THEN
it's time to start investigating artifacts in the resampling procedure.
If THAT turns out clean, ONLY THEN can you reasonably attribute it to
the sampling rate rather than the implementation.
--scott
--
"C'est un Nagra. C'est suisse, et tres, tres precis."

[geoff]

Arny Krueger wrote:
"geoff" wrote in message
...
PStamler wrote:
William mentioned that few microphones have significant response
over 20kHz. I've looked at some of my students' recordings using a
spectrum analyzer, and there was plenty of 20kHz+ signal present. I
don't know exactly which mics were used on which recordings -- most
probably various Neumanns on most of them -- but there was lots of
signal up above the audible range.


Naa, those were produced by the Aural Exiter (aka clipping) !


The classic design for the Aphex Aural Exciter produces even order
nonlinear distortion. This is not the same as the symmetrical
clipping we see on certain modern CDs. Symmetrical clipping produces
odd-order nonlinear distortion. Two very different animals!

Who said anything about 'symetrical' or CDs (or digital) wrt clipping ?

Also, I missed the smiley off the end my sentence ;-)

geoff

[Arny Krueger[_4_]]

"geoff" wrote in message
...
Arny Krueger wrote:
"geoff" wrote in message
...
PStamler wrote:
William mentioned that few microphones have significant response
over 20kHz. I've looked at some of my students' recordings using a
spectrum analyzer, and there was plenty of 20kHz+ signal present. I
don't know exactly which mics were used on which recordings -- most
probably various Neumanns on most of them -- but there was lots of
signal up above the audible range.


Naa, those were produced by the Aural Exiter (aka clipping) !


The classic design for the Aphex Aural Exciter produces even order
nonlinear distortion. This is not the same as the symmetrical
clipping we see on certain modern CDs. Symmetrical clipping produces
odd-order nonlinear distortion. Two very different animals!

Who said anything about 'symetrical' or CDs (or digital) wrt clipping ?

In general, digital clipping and the clipping we see on most CDs with
clipping is highly symmetrical.

[Anahata]

On Thu, 08 Mar 2012 12:31:12 +0000, John Williamson wrote:

William Sommerwerck wrote:
There are no such good reasons. The equipment doesn't "know" that human
hearing ends at 20kHz.


Not a good reason, maybe, but the designers "know" that nobody can hear
anything above 20kHz, so they're not always as careful as they might be
about handling frequencies above that, especially if they can save a few
pennies.

Exactly. I was thinking particularly about amplifier circuits, which
typically run out of open loop gain at high frequencies and start
distorting as they approach slew rate limiting. If you're designing to a
spec and a tight budget, it's quite likely that your distortion figures
will go up substantially above 20kHz.

Similarly microphones may have *some* response over 20k, but it's
unllikely to be very flat up there unless extra effort and expense were
put into making it so, and getting a mic flat in the intended range is
tricky enough as it it.

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

[Jason[_14_]]

In article ,
says...

I take it the specified note itself wasn't at 34kHz?

It wasn't. It was audible and I sure can't hear that high!

... but I think the significant part of that
story is that the violinist, who was surely closer than anyone to the
sound source, could hardly hear it herself.

I was only about six feet from her - quite close. She is older than I am
and admits that her high-frequency hearing "isn't what it used to be."

[MarkK]

"Jason" wrote in message
...
In article ,
says...

I take it the specified note itself wasn't at 34kHz?

It wasn't. It was audible and I sure can't hear that high!

... but I think the significant part of that
story is that the violinist, who was surely closer than anyone to the
sound source, could hardly hear it herself.

I was only about six feet from her - quite close. She is older than I am
and admits that her high-frequency hearing "isn't what it used to be."

This is interesting.... we are discussing the audibility of an 18kHz vs 20
kHz vs 22 kHz upper cutoff.

I was very surprised the other day, I had occasion to set up a graphic
equalizer for a 2 dB flat shelf reduction of everything above 2 kHz. I was
surprised as to how VERY obvious that change was. With a flat 2 dB shelf
reduction the recording sounded mellow and warm and without the EQ set
flat, it was harsh.

My point is that the flatness of the mid range or the slope across the full
spectrum is much more significant to the sound then the exact upper limit of
the response. A 2 dB change over the mid range is VERY audible.

Mark

[Wrettyirraple]

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[hank alrich]

Jason wrote:

In article ,
says...

I take it the specified note itself wasn't at 34kHz?

It wasn't. It was audible and I sure can't hear that high!

... but I think the significant part of that
story is that the violinist, who was surely closer than anyone to the
sound source, could hardly hear it herself.

I was only about six feet from her - quite close. She is older than I am
and admits that her high-frequency hearing "isn't what it used to be."

A lifetime of playing violin can do that to a person.

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

[William Sommerwerck]

"MarkK" wrote in message
...

My point is that the flatness of the mid range or the slope across
the full spectrum is much more significant to the sound then the
exact upper limit of the response. A 2 dB change over the midrange
is VERY audible.

A 1/2 dB change is audible.

Phono pickups (most brands, most models) traditionally suffer(ed) from a
slight upper-midrange depression. Thirty (or more) years ago, Shure brought
out an improved version of the M91, the M93, which got rid of this dip.
Julian Hirsch dutifully reported that there was no particular difference in
the pickups' sound.

When I read that, I thought that Julian was lucky Shure had not sent Guido
and Vito to break his legs.

[Scott Dorsey]

In article , MarkK wrote:

I was very surprised the other day, I had occasion to set up a graphic
equalizer for a 2 dB flat shelf reduction of everything above 2 kHz. I was
surprised as to how VERY obvious that change was. With a flat 2 dB shelf
reduction the recording sounded mellow and warm and without the EQ set
flat, it was harsh.

My point is that the flatness of the mid range or the slope across the full
spectrum is much more significant to the sound then the exact upper limit of
the response. A 2 dB change over the mid range is VERY audible.

Yes, and if you have a typical 12 dB/octave filter and you make a boost or
cut at 30 Khz, there will be substantial change at 15 Khz and possibly audible
change even in the midrange.
--scott

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