[ST]

Hi everyone,

Why is it that a room with slightly higher response level below 80hz
sounds more engaging than an average flat frequency response room.
When I say flat it doesn't mean ruler flat but overall 5 db
difference across the spectrum except for 5 or 6 peaks more than 12db
peak.

Your opinion, please.

Regards,
ST

[Arny Krueger]

"ST" wrote in message


Why is it that a room with slightly higher response level
below 80hz sounds more engaging than an average flat
frequency response room.

My first question would be: "How do you know that is generally true"?

[ST]

On Apr 6, 11:42*pm, "Arny Krueger" wrote:
"ST" wrote in message



Why is it that a room with slightly higher response level
below 80hz sounds more engaging than an average flat
frequency response room.

My first question would be: "How do you know that is generally true"?

Because after repositioning the loudspeakers and adjusting the room
treatment. It gave slight boost at frequencies below 100hz by 5db to
8db which sounded better to five different type of persons. It is
maybe possible that the highs are affected too because of the removal
of room treatment. But the bass with little bit of extra punch is
pleasing than before. I measured in two other listening rooms using
Denon test CD and SPL and generally found that being the case.

Regards,
ST

[Gary Eickmeier]

"Arny Krueger" wrote in message
...
"ST" wrote in message


Why is it that a room with slightly higher response level
below 80hz sounds more engaging than an average flat
frequency response room.

My first question would be: "How do you know that is generally true"?

The MacIntosh "room curve" may have been the earliest awareness of this
principle. The Mac technicians were trained to EQ their systems' rooms not
to flat, but to a room curve that emphasized the low end and tapered off the
highs. It is now a well-recognized and respected principle - as far as my
reading tells me.

The technical reason for this is that the field-type system of auditory
perspective depends partly on the playback room to assist the speakers and
recording in the illusion of placing a performance in front of you. In the
concert hall, there is this same natural fall-off of the high frequencies
and buildup or reinforcement of low frequencies by the surfaces in the hall
itself. Because the recordings are made with the mikes placed relatively
close to the orchestra, and the desired perspective is of them performing at
a distance in front of you, not on top of you, and because the room is
usually too small to do this all by itself, it is desirable to help the
process along by EQing to this room curve.

It is easier to visualize why this surprising aspect of the system works by
looking at the reproduction as a model of the real thing, with the sound
field "shaped" to mimic the larger space in both perspective, direct to
reflected ratios, and frequency response curves.

Gary Eickmeier

[[email protected]]

On Apr 7, 7:32*am, ST wrote:

My first question would be: "How do you know that is generally true"?

Because after repositioning the loudspeakers and *adjusting the room
treatment. It gave slight boost at frequencies below 100hz by 5db to
8db which sounded better to five different type of persons.

So how do you know it was flat in the first place? Unless you've
measured and know it's flat to start with, you don't. If so you have
no basis for your generalization.

Also, what are you after? If you want accuracy, accuracy doesn't
necessarily sound "good". Accuracy means telling the truth, and the
truth is often unpleasant.

Lots of extremely expensive speakers are engineered not to be flat,
because they will then sound good extremely on some subset of
recordings. Or they sound exciting in the typical show room and so
sell better. Of course they will then sound even worse than flat
speakers on some other subset of recordings. In any event many if not
most rooms are not "flat" with any speaker.

[Arny Krueger]

wrote in message
...

On Apr 7, 7:32 am, ST wrote:

My first question would be: "How do you know that is generally true"?

Because after repositioning the loudspeakers and adjusting the room
treatment. It gave slight boost at frequencies below 100hz by 5db to
8db which sounded better to five different type of persons.

So how do you know it was flat in the first place? Unless you've
measured and know it's flat to start with, you don't. If so you have
no basis for your generalization.

Exactly.

Also, what are you after? If you want accuracy, accuracy doesn't
necessarily sound "good". Accuracy means telling the truth, and the
truth is often unpleasant.

I'm under the impression that as long as boomy, thuddy bass is avoided,
there may be an effect along the lines of "More, deeper bass is better".

Lots of extremely expensive speakers are engineered not to be flat,
because they will then sound good extremely on some subset of
recordings. Or they sound exciting in the typical show room and so
sell better. Of course they will then sound even worse than flat
speakers on some other subset of recordings. In any event many if not
most rooms are not "flat" with any speaker.

Right. For example small rooms have a built-in bass boost that starts at a
frequency that goes lower as the room gets larger.

[Howard Davis]

What is really desirable is flat frequency response AT THE LISTENING
POSITION - the "sweet spot."

The speakers need not be perfectly flat but should be reasonably so over the
full audio spectrum. High end speakers plus a good subwoofer is usually
needed to achieve this. Room acoustics are usually very unflat due to
resonances and reflections, but this can be improved by proper acoustic
conditioning and speaker placement.

Then the serious audiophile uses, or hires a tech to install and set up, a
GRAPHIC EQUALIZER - at least 10 band, preferably more - to minimize the
nonlinearities in the frequency response at the primary listening position.
I use a DOD 231Q 31- band dual (2 channel) graphic EQ. The settings for
both channels are the same as the speakers are matched and symmetrically
located.

Truly flat response input to ear, even if only within +/- a few db over the
entire audio spectrum, will absolutely amaze you!

[[email protected]]

On Apr 7, 1:28*pm, wrote:
Lots of extremely expensive speakers are engineered not to be flat,
because they will then sound good extremely on some subset of
recordings. *Or they sound exciting in the typical show room and so
sell better.

Or they were "designed" in the absence of reliable
measurement and test methods and engineering
expertise, regrettably.

[Gary Eickmeier]

"Howard Davis" wrote in message
...
What is really desirable is flat frequency response AT THE LISTENING
POSITION - the "sweet spot."

This is exactly opposite of the truth. Speakers that are flat to a measuring
microphone in an anechoic chamber will have a natural, desirable fall-off of
the high frequencies in a good sized room, as will live instruments. This is
provided they have a very wide, even radiation pattern, preferably dipole or
omni, and are placed a few feet from all walls.


Then the serious audiophile uses, or hires a tech to install and set up, a
GRAPHIC EQUALIZER - at least 10 band, preferably more - to minimize the
nonlinearities in the frequency response at the primary listening
position.
I use a DOD 231Q 31- band dual (2 channel) graphic EQ. The settings for
both channels are the same as the speakers are matched and symmetrically
located.

You've got to be careful here. "Flat" at one point in the room could be
greatly mistaken. What is desirable is a power response that takes on a room
curve with slightly tapering high frequencies. You measure this by moving
the microphone over an area, not just in one spot, and averaging.

Truly flat response input to ear, even if only within +/- a few db over
the
entire audio spectrum, will absolutely amaze you!

And blister your ears off.

Gary Eickmeier

[Bob]

"ST" wrote in message
...
On Apr 6, 11:42 pm, "Arny Krueger" wrote:
"ST" wrote in message



Why is it that a room with slightly higher response level
below 80hz sounds more engaging than an average flat
frequency response room.

My first question would be: "How do you know that is generally true"?

Because after repositioning the loudspeakers and adjusting the room
treatment. It gave slight boost at frequencies below 100hz by 5db to
8db which sounded better to five different type of persons. It is
maybe possible that the highs are affected too because of the removal
of room treatment. But the bass with little bit of extra punch is
pleasing than before. I measured in two other listening rooms using
Denon test CD and SPL and generally found that being the case.

Regards,
ST

Seems to me that the premise of " sounds more engaging" will, in a large
part be determined by the type of sound that the people doing the listening
have become conditioned to. I think the subjective response of people who
are used to listening to flatter ( more accurate) reproduction would be
different than that of people conditioned by the type of sound reproduction
that they usually hear.
Back in the late 60's and early 70's when newer speaker designs extended the
high frequency response of speakers people that had listened to the likes of
Advents and AR speakers claimed that the newer designs were "too bright". In
reality they were hearing a larger frequency response than they had heard in
the past. After some time hearing all the detail missing with their older
equipment, when they relistened to material on their older designed speakers
they now could tell what they had been missing. Low bass response has
progressed significantly over the years in the same fashion.

[[email protected]]

On Apr 8, 7:35*am, "Gary Eickmeier" wrote:
"Howard Davis" wrote in message

...

What is really desirable is flat frequency response AT THE LISTENING
POSITION - the "sweet spot."

This is exactly opposite of the truth. Speakers that are flat to a measuring
microphone in an anechoic chamber will have a natural, desirable fall-off of
the high frequencies in a good sized room, as will live instruments. This is
provided they have a very wide, even radiation pattern, preferably dipole or
omni, and are placed a few feet from all walls.

Put dipoles or omnis close to the walls and you get a real mess. The
listening room and the concert hall simply do not work on the same
principles. In fact they are pretty close to opposite. Concert hall
reverb is an important component in live acoustic music. It is just a
distortion in the listening room.

[Sonnova]

On Mon, 6 Apr 2009 07:43:48 -0700, ST wrote
(in article ):

Hi everyone,

Why is it that a room with slightly higher response level below 80hz
sounds more engaging than an average flat frequency response room.
When I say flat it doesn't mean ruler flat but overall 5 db
difference across the spectrum except for 5 or 6 peaks more than 12db
peak.

Your opinion, please.

Regards,
ST

Simply put: You are, ostensibly, trying to reproduce a performance that
already exists, not create a new one. Anything that adds or subtracts
anything from the original performance is veering away from that goal. It's
OK, if that' what you want, but it's not "high-fidelity".

[Gary Eickmeier]

"Sonnova" wrote in message
...

Simply put: You are, ostensibly, trying to reproduce a performance that
already exists, not create a new one. Anything that adds or subtracts
anything from the original performance is veering away from that goal.
It's
OK, if that' what you want, but it's not "high-fidelity".

It sure seems like that would be the case, but it's not. Think about a
mult-miked pop or even most classical and jazz recordings that are very
close-miked and then mixed down to a master recording according to the
judgement and taste of the engineer. Some recordings are even made with all
instruments isolated from each other so that the engineer has max control
over the mix with minimal bleed-thru.

There is no "original event" as such, witnessed from the best seat in the
house by the perfect microphone - unless you are a gigantic man from Mars
with 37 ears or live under the lid of a piano or hovering over the drum kit.

Gary Eickmeier

[Sonnova]

On Fri, 10 Apr 2009 07:25:55 -0700, Gary Eickmeier wrote
(in article ):

"Sonnova" wrote in message
...

Simply put: You are, ostensibly, trying to reproduce a performance that
already exists, not create a new one. Anything that adds or subtracts
anything from the original performance is veering away from that goal.
It's
OK, if that' what you want, but it's not "high-fidelity".

It sure seems like that would be the case, but it's not. Think about a
mult-miked pop or even most classical and jazz recordings that are very
close-miked and then mixed down to a master recording according to the
judgement and taste of the engineer. Some recordings are even made with all
instruments isolated from each other so that the engineer has max control
over the mix with minimal bleed-thru.

There is no "original event" as such, witnessed from the best seat in the
house by the perfect microphone - unless you are a gigantic man from Mars
with 37 ears or live under the lid of a piano or hovering over the drum kit.

Gary Eickmeier


That's a separate issue. Whatever the source, however ineptly or wrong-headed
is the original recording, we, as listeners, are still, basically, trying to
reproduce it. The goal is to accurately reproduce the soundfield captured by
that recording and to present it, in our listening environment, as little
changed from the original as possible. That makes flat frequency response,
measured at your listening position very important. Luckily, the human ear
has a lot of latitude in this respect and will accept less than perfect
frequency response and still regard the result as high-fidelity.

[Howard Davis]

"Gary Eickmeier" wrote in message
...
"Howard Davis" wrote in message
...
What is really desirable is flat frequency response AT THE LISTENING
POSITION - the "sweet spot."

This is exactly opposite of the truth. Speakers that are flat to a
measuring
microphone in an anechoic chamber will have a natural, desirable fall-off
of
the high frequencies in a good sized room, as will live instruments. This
is
provided they have a very wide, even radiation pattern, preferably dipole
or
omni, and are placed a few feet from all walls.

It is not the opposite of the truth. The point at which your ears are
located are all that counts; it would be impossible to equalize for flat
response over a large part of, no less an entire room. If a rolloff of the
highs sounds more natural to you, turning down the treble control a bit will
give you that.

Then the serious audiophile uses, or hires a tech to install and set up,
a
GRAPHIC EQUALIZER - at least 10 band, preferably more - to minimize the
nonlinearities in the frequency response at the primary listening
position.
I use a DOD 231Q 31- band dual (2 channel) graphic EQ. The settings for
both channels are the same as the speakers are matched and symmetrically
located.

You've got to be careful here. "Flat" at one point in the room could be
greatly mistaken. What is desirable is a power response that takes on a
room
curve with slightly tapering high frequencies. You measure this by moving
the microphone over an area, not just in one spot, and averaging.

Setting equalization for flatter response over a larger area, based on the
averaging of measurements, certainly can be done, but this compromises the
flatness at a smaller "sweet spot."

Truly flat response input to ear, even if only within +/- a few db over
the entire audio spectrum, will absolutely amaze you!

And blister your ears off.

I fail to see how.

Gary Eickmeier

[Digiman]

"Sonnova" wrote in message
...
On Fri, 10 Apr 2009 07:25:55 -0700, Gary Eickmeier wrote
(in article ):

"Sonnova" wrote in message
...

Simply put: You are, ostensibly, trying to reproduce a performance that
already exists, not create a new one. Anything that adds or subtracts
anything from the original performance is veering away from that goal.
It's
OK, if that' what you want, but it's not "high-fidelity".

It sure seems like that would be the case, but it's not. Think about a
mult-miked pop or even most classical and jazz recordings that are very
close-miked and then mixed down to a master recording according to the
judgement and taste of the engineer. Some recordings are even made with
all
instruments isolated from each other so that the engineer has max control
over the mix with minimal bleed-thru.

There is no "original event" as such, witnessed from the best seat in the
house by the perfect microphone - unless you are a gigantic man from Mars
with 37 ears or live under the lid of a piano or hovering over the drum
kit.

Gary Eickmeier


That's a separate issue. Whatever the source, however ineptly or
wrong-headed
is the original recording, we, as listeners, are still, basically, trying
to
reproduce it. The goal is to accurately reproduce the soundfield captured
by
that recording and to present it, in our listening environment, as little
changed from the original as possible. That makes flat frequency response,
measured at your listening position very important. Luckily, the human ear
has a lot of latitude in this respect and will accept less than perfect
frequency response and still regard the result as high-fidelity.

Makers of $20000 speakers will tell you otherwise... (but I totally agree)
On a sidenote, some speaker manufacturers, and audio critics make a big
point of breaking in the speakers.
Lately I've been reading some articles on the matter, and especially one,
(was it The Audio Critic?)
claims that the "breaking-in" never ends. The point being that breaking-in
happens every time you start
using the speakers, in the first1-3 seconds of playing music. The dramatic
tales of cold, sterile, bassless, dead sound
from factory-fresh speakers are clearly dismissed.
In technical terms, the only element of a speaker that could theoretically
change characteristics during time and use is the suspension.
It follows that if it did so, then the manufacturers who has spent countless
hours calculating cabinet volume and frequency response
would have a serious problem. The logical conclusion is that the
"breaking-in" happens in the listeners ear, while the listener
gets used to the sound from his new speakers.
And then we're back to the main point; that the ear will accept less than
perfect frequency response...
*when it's used to that particular frequency response*
______________
Digiman

[Sonnova]

On Fri, 10 Apr 2009 14:28:23 -0700, Howard Davis wrote
(in article ):

"Gary Eickmeier" wrote in message
...
"Howard Davis" wrote in message
...
What is really desirable is flat frequency response AT THE LISTENING
POSITION - the "sweet spot."

This is exactly opposite of the truth. Speakers that are flat to a
measuring
microphone in an anechoic chamber will have a natural, desirable fall-off
of
the high frequencies in a good sized room, as will live instruments. This
is
provided they have a very wide, even radiation pattern, preferably dipole
or
omni, and are placed a few feet from all walls.

It is not the opposite of the truth. The point at which your ears are
located are all that counts; it would be impossible to equalize for flat
response over a large part of, no less an entire room. If a rolloff of the
highs sounds more natural to you, turning down the treble control a bit will
give you that.

Then the serious audiophile uses, or hires a tech to install and set up,
a
GRAPHIC EQUALIZER - at least 10 band, preferably more - to minimize the
nonlinearities in the frequency response at the primary listening
position.
I use a DOD 231Q 31- band dual (2 channel) graphic EQ. The settings for
both channels are the same as the speakers are matched and symmetrically
located.

You've got to be careful here. "Flat" at one point in the room could be
greatly mistaken. What is desirable is a power response that takes on a
room
curve with slightly tapering high frequencies. You measure this by moving
the microphone over an area, not just in one spot, and averaging.

Setting equalization for flatter response over a larger area, based on the
averaging of measurements, certainly can be done, but this compromises the
flatness at a smaller "sweet spot."

Truly flat response input to ear, even if only within +/- a few db over
the entire audio spectrum, will absolutely amaze you!

And blister your ears off.

I fail to see how.

The poster is alluding to the fact that real music does not have flat "power
response". IOW, real music rolls-off in intensity as the frequency rises.
Ostensibly, a signal that creates the same SPL at, say, 10KHz as it does at 1
Khz or 100 Hz will be ear-bleedingly bright. Since most people over 30 have
gradually falling high frequency hearing, and because most ensembles are
miked far enough away so that high-frequency content follows the
inverse-square law of sound propagation (intensity falls-off at a rate
inversely proportional to the square of the distance) or, has been EQed at
the mixing desk to avoid being to "too hot", then it's probably not all that
important.

[Sonnova]

On Fri, 10 Apr 2009 14:29:11 -0700, Digiman wrote
(in article ):

"Sonnova" wrote in message
...
On Fri, 10 Apr 2009 07:25:55 -0700, Gary Eickmeier wrote
(in article ):

"Sonnova" wrote in message
...

Simply put: You are, ostensibly, trying to reproduce a performance that
already exists, not create a new one. Anything that adds or subtracts
anything from the original performance is veering away from that goal.
It's
OK, if that' what you want, but it's not "high-fidelity".

It sure seems like that would be the case, but it's not. Think about a
mult-miked pop or even most classical and jazz recordings that are very
close-miked and then mixed down to a master recording according to the
judgement and taste of the engineer. Some recordings are even made with
all
instruments isolated from each other so that the engineer has max control
over the mix with minimal bleed-thru.

There is no "original event" as such, witnessed from the best seat in the
house by the perfect microphone - unless you are a gigantic man from Mars
with 37 ears or live under the lid of a piano or hovering over the drum
kit.

Gary Eickmeier


That's a separate issue. Whatever the source, however ineptly or
wrong-headed
is the original recording, we, as listeners, are still, basically, trying
to
reproduce it. The goal is to accurately reproduce the soundfield captured
by
that recording and to present it, in our listening environment, as little
changed from the original as possible. That makes flat frequency response,
measured at your listening position very important. Luckily, the human ear
has a lot of latitude in this respect and will accept less than perfect
frequency response and still regard the result as high-fidelity.

Makers of $20000 speakers will tell you otherwise... (but I totally agree)
On a sidenote, some speaker manufacturers, and audio critics make a big
point of breaking in the speakers.
Lately I've been reading some articles on the matter, and especially one,
(was it The Audio Critic?)
claims that the "breaking-in" never ends. The point being that breaking-in
happens every time you start
using the speakers, in the first1-3 seconds of playing music. The dramatic
tales of cold, sterile, bassless, dead sound
from factory-fresh speakers are clearly dismissed.
In technical terms, the only element of a speaker that could theoretically
change characteristics during time and use is the suspension.
It follows that if it did so, then the manufacturers who has spent countless
hours calculating cabinet volume and frequency response
would have a serious problem. The logical conclusion is that the
"breaking-in" happens in the listeners ear, while the listener
gets used to the sound from his new speakers.
And then we're back to the main point; that the ear will accept less than
perfect frequency response...
*when it's used to that particular frequency response*
______________
Digiman


There is probably more mythology associated with the hobby of stereo than any
other avocation (with the possible exception of devil worship ;-). Myrtle
wood blocks, speaker cables the size of municipal fire-hoses, $4000/meter
interconnects, special power cables, elevators for speaker cable,
free-standing wooden "sculptures" to enhance "imaging" and green pens all
make the audio snake-oil salesman and the gullible, neurotic audiophile
life-long partners in an endless bunko dance.

There may well be something to initial speaker break-in. When things like
speaker cone surrounds are first manufactured, they are going to be stiffer
than they will be after flexing for a few thousand cycles. This loosening of
the suspension can be easily demonstrated. Take a piece of cardboard like a
shirt card, and bend it. Then with your hands, bend it back and forward until
it looses all of its stiffness at the bend. Soon, it won't even stay where
you bend it, but will just drop with gravity. Ostensibly, this is what
happens to the flexible areas on loudspeakers and it seems to me that this
cannot help but change the free-air resonance of the driver by making the
cone more compliant and less resistive to movement. Whether one can actually
HEAR the difference between two identical drivers, one which has a 100 hours
of break-in on it and another, straight out of the box, is another matter.
Maybe, but I tend to doubt it.

[[email protected]]

On Apr 10, 7:25*am, "Gary Eickmeier" wrote:
"Sonnova" wrote in message

...

Simply put: You are, ostensibly, *trying to reproduce a performance that
already exists, not create a new one. Anything that adds or subtracts
anything from the original performance is veering away from that goal.
It's
OK, if that' what you want, but it's not "high-fidelity".

It sure seems like that would be the case, but it's not. Think about a
mult-miked pop or even most classical and jazz recordings that are very
close-miked and then mixed down to a master recording according to the
judgement and taste of the engineer. Some recordings are even made with all
instruments isolated from each other so that the engineer has max control
over the mix with minimal bleed-thru.

There is no "original event" as such, witnessed from the best seat in the
house by the perfect microphone - unless you are a gigantic man from Mars
with 37 ears or live under the lid of a piano or hovering over the drum kit.

Gary Eickmeier

Sounds like you are proposing a bad fix for bad recordings at the
expense of every decent to excellent recording. Poorly engineered
mulitmiked recordings splashed all over the walls isn't my idea of
high end audio.

[ST]

On Apr 11, 3:43 am, Sonnova wrote:
On Fri, 10 Apr 2009 07:25:55 -0700, Gary Eickmeier wrote
.........


That's a separate issue. Whatever the source, however ineptly or wrong-headed
is the original recording, we, as listeners, are still, basically, trying to
reproduce it. The goal is to accurately reproduce the soundfield captured by
that recording and to present it, in our listening environment, as little
changed from the original as possible. That makes flat frequency response,
measured at your listening position very important. Luckily, the human ear
has a lot of latitude in this respect and will accept less than perfect
frequency response and still regard the result as high-fidelity.

Please correct me if I am wrong, my understanding is recording
engineers mix various musical piece based on what sounds good or right
as to how the composition should sound based on their own ears. But
human ears perceived loudness drops sharply below 250hz. Theoretically
if my room's frequency response is flat then I should able to repeat
the same. But let's say at x level the recorded loudness of 31.5hz is
100db and at 500hz is 44db then will my room produce the same loudness
at x level?

Or all this is only possible if the recording mix environment/room
frequency response is identical to my room. How do we know that ? And
my feeling is most music is meant for people who listen in normal
environment such as living hall without boom trap and other audiophile
gadgets. which generally gives out louder bass.

ST

[Gary Eickmeier]

"Sonnova" wrote in message
...

That's a separate issue. Whatever the source, however ineptly or
wrong-headed
is the original recording, we, as listeners, are still, basically, trying
to
reproduce it. The goal is to accurately reproduce the soundfield captured
by
that recording and to present it, in our listening environment, as little
changed from the original as possible. That makes flat frequency response,
measured at your listening position very important. Luckily, the human ear
has a lot of latitude in this respect and will accept less than perfect
frequency response and still regard the result as high-fidelity.

But the reproduction problem is a lot more complex than just placing two
speakers at an angle in front of you and keeping frequency response flat and
distortion low. If it were that simple, there would be no need for an entire
society devoted to audio reproduction, project after project of research
trying to figure out how to do it better.

The first step is to realize that the reproduction is a completely new piece
of art, or event. The goal of that work of art may indeed be to try to
transport you to the location of the original recording, or it may be a work
in and of itself, like some synthesizer stuff and pop recordings. In either
case, frequency response and distortion are but a small part of the problem,
or aspects of a sound event. If you consider the total event as the "thing"
that we are trying to reproduce, then you must attempt to come as close to
all aspects of the sound as you can. To do that, you've got to know more
than frequency response and distortion.

Consider photography as an example from the visual arts. The engineer in you
thinks that the photograph will be "accurate" if you can reproduce perfect
color and sharpness, and keep noise and distortions low. But if you consider
all aspects of the original scene that you are trying to reproduce with the
photo, you can easily see that the photo, no matter how "accurate," will not
fool anyone into thinking they are at the original scene. It is a new piece
of art in and of itself. But it has no three-dimensionality, is not as big
as the original, and not as bright in the highlights - to name just a few
aspects that have not been reproduced.

There's a lot more to it than that, but for now all I want to establish is
that the complex original sound field has been compressed into two or more
channels of electrical signal. How those signals are treated at the playback
end is the problem, and it is more complicated than just frequency response
and distortion, and a far more productive path to follow would be to create
a new sound event in your playback room - one that has more of the aspects
of the original and of good sound than your expressed engineering insticts
have suggested.

Gary Eickmeier

[Sonnova]

On Fri, 10 Apr 2009 19:26:12 -0700, ST wrote
(in article ):

On Apr 11, 3:43 am, Sonnova wrote:
On Fri, 10 Apr 2009 07:25:55 -0700, Gary Eickmeier wrote
........


That's a separate issue. Whatever the source, however ineptly or
wrong-headed
is the original recording, we, as listeners, are still, basically, trying to
reproduce it. The goal is to accurately reproduce the soundfield captured by
that recording and to present it, in our listening environment, as little
changed from the original as possible. That makes flat frequency response,
measured at your listening position very important. Luckily, the human ear
has a lot of latitude in this respect and will accept less than perfect
frequency response and still regard the result as high-fidelity.

Please correct me if I am wrong, my understanding is recording
engineers mix various musical piece based on what sounds good or right
as to how the composition should sound based on their own ears. But
human ears perceived loudness drops sharply below 250hz. Theoretically
if my room's frequency response is flat then I should able to repeat
the same. But let's say at x level the recorded loudness of 31.5hz is
100db and at 500hz is 44db then will my room produce the same loudness
at x level?

Or all this is only possible if the recording mix environment/room
frequency response is identical to my room. How do we know that ? And
my feeling is most music is meant for people who listen in normal
environment such as living hall without boom trap and other audiophile
gadgets. which generally gives out louder bass.

ST

We don't have anyway of knowing how a recording sounded in the studio. We
don't know the equipment used, the monitoring speakers used, the room
dimensions, the room acoustics, none of that. That's why I am of the opinion
that studio recorded music cannot EVER be high-fidelity. You get
high-fidelity with some kind of stereo-pair microphone arrangement placed in
front of an ensemble in a real space and recorded directly. No mixing, no
monitoring other than headphones. The resultant recording (two channel, four
channel, whatever) goes directly to CD/SACD/Hi-Res and gets played back
through the requisite number of channels. If the playback chain is wide-range
and generally flat in frequency response and low in distortion, the listener
will get a reasonable facsimile of the original performance. Anything else is
a re-interpretation of the musicians intent. That's my opinion, anyway.

[Gary Eickmeier]

wrote in message
...
On Apr 10, 7:25 am, "Gary Eickmeier" wrote:

It sure seems like that would be the case, but it's not. Think about a
mult-miked pop or even most classical and jazz recordings that are very
close-miked and then mixed down to a master recording according to the
judgement and taste of the engineer. Some recordings are even made with
all
instruments isolated from each other so that the engineer has max control
over the mix with minimal bleed-thru.

There is no "original event" as such, witnessed from the best seat in the
house by the perfect microphone - unless you are a gigantic man from Mars
with 37 ears or live under the lid of a piano or hovering over the drum
kit.

Gary Eickmeier

Sounds like you are proposing a bad fix for bad recordings at the
expense of every decent to excellent recording. Poorly engineered
mulitmiked recordings splashed all over the walls isn't my idea of
high end audio.

So the only good recordings are purist two microphone, perhaps Blumlein pair
recordings?

"Splashed all over the walls"? Where did that come from?

Gary Eickmeier

[[email protected]]

On Apr 10, 9:18*pm, Sonnova wrote:
There may well be something to initial speaker break-in.
When things like speaker cone surrounds are first
manufactured, they are going to be stiffer than they
will be after flexing for a few thousand cycles.

Which, of course, is accomplished in a few seconds of
QC testing for rub and buzz, for example.

This loosening of the suspension can be easily
demonstrated. Take a piece of cardboard like a
shirt card, and bend it. Then with your hands,
bend it back and forward until it looses all of its
stiffness at the bend. Soon, it won't even stay where
you bend it, but will just drop with gravity.
Ostensibly, this is what happens to the flexible areas
on loudspeakers and it seems to me that this cannot
help but change the free-air resonance of the driver
by making the cone more compliant and less
resistive to movement.

Except that shirt cardboard and the material used to
make the suspension parts of loudspeakers are
vastly different materials. Try your same experiment
with a sheet of a polybutadene-styrene alloy, and
you will find a very different result: that flexing the
material enough will slightly raise its temperature,
and its mechanical compliance does change, not
because of "break-in", but because of the change
in temperature.

Put your piece of shirt cardboard and the piece of
polybutadene-styrene down for a minute and repeat
the experiment. The shirt cardboard will still be limo
and, well, as borken as it was when you broke it,
the polytbutadene-styrene allot will look and behave
just like it it when you first picked it up.

There's another flaw to your experiment: your shirt
cardboard gets flexed FAR beyond its linear elastic
limits and is not "broken in," it's just plain BROKEN.

[[email protected]]

On Apr 11, 3:20*am, "Gary Eickmeier" wrote:
wrote in message

...





On Apr 10, 7:25 am, "Gary Eickmeier" wrote:
It sure seems like that would be the case, but it's not. Think about a
mult-miked pop or even most classical and jazz recordings that are very
close-miked and then mixed down to a master recording according to the
judgement and taste of the engineer. Some recordings are even made with
all
instruments isolated from each other so that the engineer has max control
over the mix with minimal bleed-thru.

There is no "original event" as such, witnessed from the best seat in the
house by the perfect microphone - unless you are a gigantic man from Mars
with 37 ears or live under the lid of a piano or hovering over the drum
kit.

Gary Eickmeier

Sounds like you are proposing a bad fix for bad recordings at the
expense of every decent to excellent recording. Poorly engineered
mulitmiked recordings splashed all over the walls isn't my idea of
high end audio.

So the only good recordings are purist two microphone, perhaps Blumlein pair
recordings?

"Splashed all over the walls"? Where did that come from?

Gary Eickmeier- Hide quoted text -

- Show quoted text -

It comes from placing dipoles or omnies close to the back wall.

[ST]

On Apr 11, 12:49 pm, Sonnova wrote:

.........


We don't have anyway of knowing how a recording sounded in the studio. We
don't know the equipment used, the monitoring speakers used, the room
dimensions, the room acoustics, none of that. That's why I am of the opinion
that studio recorded music cannot EVER be high-fidelity. You get
high-fidelity with some kind of stereo-pair microphone arrangement placed in
front of an ensemble in a real space and recorded directly. No mixing, no
monitoring other than headphones. The resultant recording (two channel, four
channel, whatever) goes directly to CD/SACD/Hi-Res and gets played back
through the requisite number of channels. If the playback chain is wide-range
and generally flat in frequency response and low in distortion, the listener
will get a reasonable facsimile of the original performance. Anything else is
a re-interpretation of the musicians intent. That's my opinion, anyway.


Waterlily Acoustic claims so . Remember Tabala Rasa? But it sound so
dead to me.

[[email protected]]

On Apr 6, 10:43*am, ST wrote:
Hi everyone,

Why is it that a room with slightly higher response level below 80hz
sounds more engaging than an average flat frequency response room.
When I say flat it doesn't mean ruler flat but overall 5 db
difference across the spectrum except for 5 or 6 peaks more than 12db
peak.

Your opinion, please.

Regards,
ST

Well, if a flat frequency response is not desirable, then the program
material must need some help. If all aspects of the recording are
respected to their true source dimensions, there would be no need to
alter frequency response.

Frequency response of a speaker system within a given listening room
is important to accuracy, but not always the most important critical
factor.

Bass, and mid bass boost are known to improve the sound impressions
when playback is at low to moderate sound pressure levels. This is
reinforced by the inclusion of the "loudness" switch on many preamps
and receivers. However, when sound pressure levels approach the
actual performance levels, this feature is less pronounced in
effectiveness and desirability.

Playback of good quality signal at actual performance SPL should not
require any major equalization corrections to sound accurate. This
depends upon the room and speakers being matched and complementary to
enable a good stereo image and void of objectionable resonance or
standing waves.

Where is this going? I've had this deja vu before with this same
debate on the horizon. This has been beat to death without an
objective winner. Won't happen this time either.

Aaaarrrgh,
Don.

[Sonnova]

On Sat, 11 Apr 2009 07:23:38 -0700, wrote
(in article ):

On Apr 10, 9:18*pm, Sonnova wrote:
There may well be something to initial speaker break-in.
When things like speaker cone surrounds are first
manufactured, they are going to be stiffer than they
will be after flexing for a few thousand cycles.

Which, of course, is accomplished in a few seconds of
QC testing for rub and buzz, for example.

This loosening of the suspension can be easily
demonstrated. Take a piece of cardboard like a
shirt card, and bend it. Then with your hands,
bend it back and forward until it looses all of its
stiffness at the bend. Soon, it won't even stay where
you bend it, but will just drop with gravity.
Ostensibly, this is what happens to the flexible areas
on loudspeakers and it seems to me that this cannot
help but change the free-air resonance of the driver
by making the cone more compliant and less
resistive to movement.

Except that shirt cardboard and the material used to
make the suspension parts of loudspeakers are
vastly different materials. Try your same experiment
with a sheet of a polybutadene-styrene alloy, and
you will find a very different result: that flexing the
material enough will slightly raise its temperature,
and its mechanical compliance does change, not
because of "break-in", but because of the change
in temperature.

Put your piece of shirt cardboard and the piece of
polybutadene-styrene down for a minute and repeat
the experiment. The shirt cardboard will still be limo
and, well, as borken as it was when you broke it,
the polytbutadene-styrene allot will look and behave
just like it it when you first picked it up.

There's another flaw to your experiment: your shirt
cardboard gets flexed FAR beyond its linear elastic
limits and is not "broken in," it's just plain BROKEN.

Since it was just an example of what some say is happening with speaker
break-in, of course its hyperbole.

[[email protected]]

On Apr 11, 2:13*pm, wrote:
Well, if a flat frequency response is not desirable,
then the program material must need some help. *

Hardly the only possibility. Basically, it's because
the listener has an expectation of what it SHOULD
sound like, and the presentation needs to be adjusted
to meet that expectation. That includes all sorts of
possibilities, including program material needing
help, the listeners expectations at variance with
some reality and so forth.

If all aspects of the recording are respected to
their true source dimensions, there would be
no need to alter frequency response.

Simple not true: you assume that a neutral,
uncolored presentation is the ONLY correct one
for all listeners.

Frequency response of a speaker system within
a given listening room is important to accuracy,

Again, your premise assumes that a clear definition
of accuracy is agreed upon and the result is accepted
by all. It is not.

Playback of good quality signal at actual
performance SPL should not require any major
equalization corrections to sound accurate.

Again, you assume that a. everyone agrees upon the
same definition of "accuracy" and b. everyone agrees
that accuracy is THE goal.

This depends upon the room and speakers being
matched and complementary to enable a good
stereo image and void of objectionable resonance or
standing waves.

In a word, no. Among other things, you're assuming
that response magnitude is he only variable. It most
assuredly is not. The fact is that a standing wave in
a room can well be a non-mininum phase error, and
attempting to correct it with a non-minimum phase
speaker response or a different non-minimum phase
response will not work. Simply put: a complimentary
speaker response to a room error is complimentary
ONLY in one domain, (response magnitude vs
frequency) and is NOT other, equally critical domans
(e.g. phase of time vs frequency).

Where is this going? *I've had this deja vu before with this same
debate on the horizon. *This has been beat to death without an
objective winner.

Often time, by being clubbed to death by quite
non-objective or just plain "technical" wrong
arguments.

[Sonnova]

On Fri, 10 Apr 2009 21:48:37 -0700, Gary Eickmeier wrote
(in article ):

"Sonnova" wrote in message
...

That's a separate issue. Whatever the source, however ineptly or
wrong-headed
is the original recording, we, as listeners, are still, basically, trying
to
reproduce it. The goal is to accurately reproduce the soundfield captured
by
that recording and to present it, in our listening environment, as little
changed from the original as possible. That makes flat frequency response,
measured at your listening position very important. Luckily, the human ear
has a lot of latitude in this respect and will accept less than perfect
frequency response and still regard the result as high-fidelity.

But the reproduction problem is a lot more complex than just placing two
speakers at an angle in front of you and keeping frequency response flat and
distortion low. If it were that simple, there would be no need for an entire
society devoted to audio reproduction, project after project of research
trying to figure out how to do it better.

But there really isn't much more to it than that. The problem arises because
even today, with our advanced materials technology and good electronics, we
can't accomplish this simple task very well. That's why interpretive talent
has to be between the musicians and the final recording.

The first step is to realize that the reproduction is a completely new piece
of art, or event. The goal of that work of art may indeed be to try to
transport you to the location of the original recording, or it may be a work
in and of itself, like some synthesizer stuff and pop recordings. In either
case, frequency response and distortion are but a small part of the problem,
or aspects of a sound event. If you consider the total event as the "thing"
that we are trying to reproduce, then you must attempt to come as close to
all aspects of the sound as you can. To do that, you've got to know more
than frequency response and distortion.

While I agree, It's also largely irrelevant to the point.

Consider photography as an example from the visual arts. The engineer in you
thinks that the photograph will be "accurate" if you can reproduce perfect
color and sharpness, and keep noise and distortions low. But if you consider
all aspects of the original scene that you are trying to reproduce with the
photo, you can easily see that the photo, no matter how "accurate," will not
fool anyone into thinking they are at the original scene. It is a new piece
of art in and of itself. But it has no three-dimensionality, is not as big
as the original, and not as bright in the highlights - to name just a few
aspects that have not been reproduced.

That's my point. Since the process is imperfect, the process becomes the
limiting factor and therefore must be "manipulated" to produce a desired
result.

There's a lot more to it than that, but for now all I want to establish is
that the complex original sound field has been compressed into two or more
channels of electrical signal. How those signals are treated at the playback
end is the problem, and it is more complicated than just frequency response
and distortion, and a far more productive path to follow would be to create
a new sound event in your playback room - one that has more of the aspects
of the original and of good sound than your expressed engineering insticts
have suggested.

But it doesn't need to be if all we are trying to do is to capture a
performance for playback. Unfortunately, many different types of music don't
respond to those techniques.

[Honest Abe]

"Sonnova" wrote in message
...
On Fri, 10 Apr 2009 14:28:23 -0700, Howard Davis wrote
(in article ):

"Gary Eickmeier" wrote in message
...
"Howard Davis" wrote in message
...
What is really desirable is flat frequency response AT THE LISTENING
POSITION - the "sweet spot."

This is exactly opposite of the truth. Speakers that are flat to a
measuring
microphone in an anechoic chamber will have a natural, desirable
fall-off
of
the high frequencies in a good sized room, as will live instruments.
This
is
provided they have a very wide, even radiation pattern, preferably
dipole
or
omni, and are placed a few feet from all walls.

It is not the opposite of the truth. The point at which your ears are
located are all that counts; it would be impossible to equalize for flat
response over a large part of, no less an entire room. If a rolloff of
the
highs sounds more natural to you, turning down the treble control a bit
will
give you that.

Then the serious audiophile uses, or hires a tech to install and set
up,
a
GRAPHIC EQUALIZER - at least 10 band, preferably more - to minimize the
nonlinearities in the frequency response at the primary listening
position.
I use a DOD 231Q 31- band dual (2 channel) graphic EQ. The settings
for
both channels are the same as the speakers are matched and
symmetrically
located.

You've got to be careful here. "Flat" at one point in the room could be
greatly mistaken. What is desirable is a power response that takes on a
room
curve with slightly tapering high frequencies. You measure this by
moving
the microphone over an area, not just in one spot, and averaging.

Setting equalization for flatter response over a larger area, based on
the
averaging of measurements, certainly can be done, but this compromises
the
flatness at a smaller "sweet spot."

Truly flat response input to ear, even if only within +/- a few db over
the entire audio spectrum, will absolutely amaze you!

And blister your ears off.

I fail to see how.

The poster is alluding to the fact that real music does not have flat
"power
response". IOW, real music rolls-off in intensity as the frequency rises.
Ostensibly, a signal that creates the same SPL at, say, 10KHz as it does
at 1
Khz or 100 Hz will be ear-bleedingly bright. Since most people over 30
have
gradually falling high frequency hearing, and because most ensembles are
miked far enough away so that high-frequency content follows the
inverse-square law of sound propagation (intensity falls-off at a rate
inversely proportional to the square of the distance) or, has been EQed at
the mixing desk to avoid being to "too hot", then it's probably not all
that
important.

A truly flat reproduction system will recreate the relative sound levels
over the audio spectrum exactly as they were recorded. I expect that when
the recording techniques are correct, and the recording microphone to final
reproduction at the ears is truly flat, you will hear exactly the sounds
that were recorded.

[Sonnova]

On Sat, 11 Apr 2009 03:20:33 -0700, Gary Eickmeier wrote
(in article ):

wrote in message
...
On Apr 10, 7:25 am, "Gary Eickmeier" wrote:

It sure seems like that would be the case, but it's not. Think about a
mult-miked pop or even most classical and jazz recordings that are very
close-miked and then mixed down to a master recording according to the
judgement and taste of the engineer. Some recordings are even made with
all
instruments isolated from each other so that the engineer has max control
over the mix with minimal bleed-thru.

There is no "original event" as such, witnessed from the best seat in the
house by the perfect microphone - unless you are a gigantic man from Mars
with 37 ears or live under the lid of a piano or hovering over the drum
kit.

Gary Eickmeier

Sounds like you are proposing a bad fix for bad recordings at the
expense of every decent to excellent recording. Poorly engineered
mulitmiked recordings splashed all over the walls isn't my idea of
high end audio.

So the only good recordings are purist two microphone, perhaps Blumlein pair
recordings?

As far as I'm concerned, that's true.

[Arny Krueger]

"Sonnova" wrote in message


There may well be something to initial speaker break-in.
When things like speaker cone surrounds are first
manufactured, they are going to be stiffer than they will
be after flexing for a few thousand cycles.

That's a supposition, for which I see zero evidence.

It is highly improbable that something will change rapidly over flexing over
a few thousand cycles, and then suddenly stop changing at the point where it
performs ideally, and stay that way from then on.

For openers, a speaker cone flexes at a rate equal to the highest frequency
it reproduces. This can easily be 20 KHz. The alleged thousand flexures can
thus take place in the first 20 milliseconds of operation. This varies
sharply from audiophile lore, which has break-in taking place OVER a period
of days or weeks.

My friend who actually test speakers for this sort of thing finds that
speakers may change their properties during the first fraction of a second
to several seconds of operation, every time they are restarted after a LONG
period of rest. The speaker then stabilizes. In general nobody is likely to
notice a rapid change like this.

[[email protected]]

On Apr 11, 3:20*am, "Gary Eickmeier" wrote:
wrote in message

...





On Apr 10, 7:25 am, "Gary Eickmeier" wrote:
It sure seems like that would be the case, but it's not. Think about a
mult-miked pop or even most classical and jazz recordings that are very
close-miked and then mixed down to a master recording according to the
judgement and taste of the engineer. Some recordings are even made with
all
instruments isolated from each other so that the engineer has max control
over the mix with minimal bleed-thru.

There is no "original event" as such, witnessed from the best seat in the
house by the perfect microphone - unless you are a gigantic man from Mars
with 37 ears or live under the lid of a piano or hovering over the drum
kit.

Gary Eickmeier

Sounds like you are proposing a bad fix for bad recordings at the
expense of every decent to excellent recording. Poorly engineered
mulitmiked recordings splashed all over the walls isn't my idea of
high end audio.

So the only good recordings are purist two microphone, perhaps Blumlein pair
recordings?

A "good" recording is any recording that is deemed so by the listener.
If one is looking for the best illusion of an original acoustic
performance from a single perspective then I'd say the only recordings
that really excel are indeed purist recordings such as those recorded
with the Blumlein paired microphones. If you are listening to Pink
Floyd Dark Side of the Moon or some other studio creation naturally
the standards of excellence become much more subjective. But IME even
those recordings tend to sound their best on the same hardware that
best supports the purist recordings.

[[email protected]]

On Apr 10, 9:48*pm, "Gary Eickmeier" wrote:
"Sonnova" wrote in message

...

That's a separate issue. Whatever the source, however ineptly or
wrong-headed
is the original recording, we, as listeners, are still, basically, trying
to
reproduce it. The goal is to accurately reproduce the soundfield captured
by
that recording and to present it, in our listening environment, as little
changed from the original as possible. That makes flat frequency response,
measured at your listening position very important. Luckily, the human ear
has a lot of latitude in this respect and will accept less than perfect
frequency response and still regard the result as high-fidelity.

But the reproduction problem is a lot more complex than just placing two
speakers at an angle in front of you and keeping frequency response flat and
distortion low.


Of course it is a lot more complex than that for the simple reason
that there is more in the entire chain of events than a speaker system/
listening room interface involved in stereo recording and playback.

If it were that simple, there would be no need for an entire
society devoted to audio reproduction, project after project of research
trying to figure out how to do it better.


Actually if it were that simple the problems still would not be solved
so I don't think your argument holds water. You will not find a ruler
flat,distortion free speaker system on this planet. One person's
simple is another's life labor.


The first step is to realize that the reproduction is a completely new piece
of art, or event. The goal of that work of art may indeed be to try to
transport you to the location of the original recording, or it may be a work
in and of itself, like some synthesizer stuff and pop recordings. In either
case, frequency response and distortion are but a small part of the problem,
or aspects of a sound event. If you consider the total event as the "thing"
that we are trying to reproduce, then you must attempt to come as close to
all aspects of the sound as you can. To do that, you've got to know more
than frequency response and distortion.



You had me until you said frequency response and distortion are a
small part of the problem. In a stereo recording and playback system
where the goal is to create an illusion of an original acoustic event
from a designated perspective you can't dismiss frequency resonse and
distortion as trivial. They are pretty much everything that determines
the success of the illusion. What else is there?



Consider photography as an example from the visual arts. The engineer in you
thinks that the photograph will be "accurate" if you can reproduce perfect
color and sharpness, and keep noise and distortions low. But if you consider
all aspects of the original scene that you are trying to reproduce with the
photo, you can easily see that the photo, no matter how "accurate," will not
fool anyone into thinking they are at the original scene. It is a new piece
of art in and of itself. But it has no three-dimensionality, is not as big
as the original, and not as bright in the highlights - to name just a few
aspects that have not been reproduced.


Analogies prove nothing but if they did you would have just shot down
your own argument. In fact we do use photographic images all the time
in the film industry to do just that, fooling the audience into
thinking what they are seeing is the real thing rather than a
photographic image. The trick is simple. the image has to be in the
right scale and has to match the lense used to take the image with the
audience's perspective. It's trick that can be done quite effectively
in real life too. The final image simply has to be printed or
projected in the correct scale and displayed so that the lense used to
capture the image will corispond with the veiwers perspective. But
anyway.......



There's a lot more to it than that, but for now all I want to establish is
that the complex original sound field has been compressed into two or more
channels of electrical signal.


Actually it hasn't. You pointed it out yourself. " The first step is
to realize that the reproduction is a completely new piece of art, or
event. The goal of that work of art may indeed be to try to transport
you to the location of the original recording," That is essentially
right. You can't record the original three dimensional soundfield of
the original event. all a recording engineer can do with consideration
of how stereo recording and playback works, is choose a single
listener perspective and engineer a recording that best creates the
illusion of the original event from that single perspective.



How those signals are treated at the playback
end is the problem, and it is more complicated than just frequency response
and distortion, and a far more productive path to follow would be to create
a new sound event in your playback room - one that has more of the aspects
of the original and of good sound than your expressed engineering insticts
have suggested.

IME if the recording engineer has talent and taste and a basic
understanding of this simple idea that stereo recording at it's best
will capture a performance from a single listener perspective with the
idea of playback creating an illusion of that experience, the best way
to create that illusion is with relatively low distortion speakers and
minimal listening room coloration. I'm not saying that all distortion
is always bad. I think there are inherent limitations in the system as
a whole and I *suspect* that like spice in fine foods, a little bit
can enhance the final product. But I think dipoles or omnis placed
near the back wall is more than just a little distortion added as
spice. It's like a proverbial stick of pepper.

[Sonnova]

On Sat, 11 Apr 2009 17:57:44 -0700, Arny Krueger wrote
(in article ):

"Sonnova" wrote in message


There may well be something to initial speaker break-in.
When things like speaker cone surrounds are first
manufactured, they are going to be stiffer than they will
be after flexing for a few thousand cycles.

That's a supposition, for which I see zero evidence.

I just love to be quoted out of context. Did you not see the rest of my
comment where I said that While it was possible, I DOUBT IT?

It is highly improbable that something will change rapidly over flexing over
a few thousand cycles, and then suddenly stop changing at the point where it
performs ideally, and stay that way from then on.

That is correct.

For openers, a speaker cone flexes at a rate equal to the highest frequency
it reproduces. This can easily be 20 KHz. The alleged thousand flexures can
thus take place in the first 20 milliseconds of operation. This varies
sharply from audiophile lore, which has break-in taking place OVER a period
of days or weeks.

My friend who actually test speakers for this sort of thing finds that
speakers may change their properties during the first fraction of a second
to several seconds of operation, every time they are restarted after a LONG
period of rest. The speaker then stabilizes. In general nobody is likely to
notice a rapid change like this.

Precisely what I would expect. Yet I have seen speakers where the
manufacturer's instructions tell the new owner that the speaker won't sound
its best until AFTER a 100 hrs or so of break-in. Why propagate mythology?

[[email protected]]

On Apr 12, 11:48*am, Sonnova wrote:
On Sat, 11 Apr 2009 17:57:44 -0700, Arny Krueger wrote
(in article ):

"Sonnova" wrote in message


There may well be something to initial speaker break-in.
When things like speaker cone surrounds are first
manufactured, they are going to be stiffer than they will
be after flexing for a few thousand cycles.

That's a supposition, for which I see zero evidence.

I just love to be quoted out of context. Did you not see the rest of my
comment where I said that *While it was possible, I DOUBT IT?

It is highly improbable that something will change rapidly over flexing over
a few thousand cycles, and then suddenly stop changing at the point where it
performs ideally, and stay that way from then on.

That is correct.

For openers, a speaker cone flexes at a rate equal to the highest frequency
it reproduces. This can easily be 20 KHz. The alleged thousand flexures can
thus take place in the first 20 milliseconds of operation. This varies
sharply from audiophile lore, which has break-in taking place OVER a period
of days or weeks.

My friend who actually test speakers for this sort of thing finds that
speakers may change their properties during the first fraction of a second
to several seconds of operation, every time they are restarted after a LONG
period of rest. *The speaker then stabilizes. In general nobody is likely to
notice a rapid change like this.

Precisely what I would expect. Yet I have seen speakers where the
manufacturer's instructions tell the new owner that the speaker won't sound
its best until AFTER a 100 hrs or so of break-in. Why propagate mythology?

Well here is a story that you all might find hard to believe but I
have witnessed the phenomenon first hand. Knowing that I was the owner
of the original Martin Logan CLSs and that i was less than impressed
by the CLS IIs and the CLS IIas, my Martin Logan dealer had me over to
audition the CLS MkIIzs. We did side by side comparisons and sure
enough the IIz upgrade was indeed a significant upgrade. So I upgraded
my Martin Logan CLSs. They sounded like crap. I brought them back and
sure enough in side by side comparisons my new CLS IIzs were
defective. The dealer assured me that it was a "break in" thing and I
had to give the speakers 100 hours. I took the speakers home and ran
them continuously for four days and nights at fairly high SPLs. Then I
immedietly took them back to the dealer without so much as a second
listen. I simply did not believe that any sort of break in could fix
the problems and that the ungraded parts had to be defective in some
way. Well, guess what? They were pretty much identical to the display
models. I'm pretty sure I didn't imagine differences that weren't
there the first time around. They literally sounded as though there
was some defect before the "break in."

[Gary Eickmeier]

wrote in message ...

It comes from placing dipoles or omnies close to the back wall.

OK now this is getting really frustrating. Where did "placing dipoles or
omnis close to the back wall" come from?

Gary Eickmeier

[John Stone]

On 4/11/09 1:06 PM, in article ,
" wrote:

On Apr 11, 3:20*am, "Gary Eickmeier" wrote:
wrote in message

"Splashed all over the walls"? Where did that come from?

Gary Eickmeier- Hide quoted text -

- Show quoted text -

It comes from placing dipoles or omnies close to the back wall.

Ok, so misplacing dipoles and omni's causes problems. That's a given. But
poorly placed monopole speakers will also cause problems, and they too
"splash sound all over the walls", just in a different manner. So is the
solution to not splash sound all over the walls, or for the walls to
completely absorb that sound? You wouldn't think so if you've ever heard a
pair of speakers played in an anechoic chamber.

[Gary Eickmeier]

wrote in message
...

But I think dipoles or omnis placed
near the back wall is more than just a little distortion added as
spice. It's like a proverbial stick of pepper.

You raise some potentially interesting points, but I am not going to go any
further unless you explain where "dipoles or omnis placed near the back
wall" came from. Need to know you are actually listening.

Gary Eickmeier