I have had no luck in measuring speaker response at low frequency's, I
can
measure from ~60Hz to 20Khz with near 100% repeatability (How accurate
is
questionable)


You measure it with microphones and ancillary equipment that is up
to the task. For myself, I have a number of Bruel & Kjaer, ACO,
GR and pther microphones that have verified flat response to well
below 20 Hz. Some of the B&K 1/2" capsules, for example, are within
+-1/2 dB from approximately 3 Hz to 30 kHz and above.

Can you give a recommend model of Bruel that you like?

Summerised I have tested in a number of different situation and with a
collection of mics and such. what I have found is that it is easy to get a
mike with a flat response from ~60Hz to ~20Kz, it seems that below this is
where a specialist product is required.

The remainder of the measurement and analysis chain has similar
properties: the primary measurement chain is DC coupled, for example.

Secondly, the size of the venue required for accurate measurements
is inversely proportional to the frequency you need to measure. Even
using techniques such as gated or windowed measurement, the distance
to the first reflection surface is a prime determinant of how low
you can measure. You want to measure 20 Hz accurately? Then you need
to find a room where the distance between the speaker/microphone and
the NEAREST surface is a minimum of 25 feet.

I have tested a few speakers and
basically I cannot get accurate (repeatable) data at much below 60Hz,
even
using a borrowed shure KSM141 stereo pair in omnidirectional gives vague
results ($3000 for the mic's!)

$3000 for microphones that were NEVER designed to be used as measurement
microphones, ESPECIALLY at low frequencies. These are recording
microphones, NOT measurement microphones.

I have come to the conclusion that you are
really measuring air displacement at anything under 30Hz and a flat
panel
with a transducer is the only repeatable method of measuring the output,

That may be the conclusion you came to, but that conclusion just happens
to be quite wrong. The physical stimulus that the ear responds to as
sound are periodic pressure variations of a sufficient amplitude and
within certain frequency limits. That's it. As long as a device can
detect these pressure variations, it can be used to measure sound.

The problem with you big flat panel method is that it assumes, quite
incorrectly, that the imnpinging waves are planar: unless you are VERY
far away from the speaker, such the wavefronts are no longer spherical,
it isn't going to work. The smaller the diaphgragm of the microphone,
the less it is affected by such a problem. That's one reason why
measurement microphones have very small diaphragms: they are essentially
point transducers over a wide range of frequencies.

how one calibrates said device is open for discussion.

One calibrates it by throwing it in the nearest landfill and going
out an learning the proper ways of measuring acoustic phenomenon.

You comments are most welcome but not vey helpful, I seem to be getting the
feeling you are giving me the 'I've be doing it for decade and got really
expensive gear, so my methods, results and opinions are gospel - do it my
way or be burned at the stake as a heratic'

One starts to wonder if perhaps you are one of the 'Experts' that 'corrects'
the measurments to what you 'know' is correct. Your attitude in your
responses seems to show this blinded mentality.

(Oh one firm told me that they use a laser to measure the low frequency
of
their speakers, check this out for novel! They place a small piece of
reflective foil on the base driver, and shine a laser beam on it, they
then
apply signal and measure via 'laser' the deflection,

Well, gee golly, since it can be shown on physical first principles
that the requirement for a constant sound pressure level (that would
mean flat frequency response) from a piston radiator is simply a
displacement which goes as the reciprocal of the square of frequency,
then if you know the displacement, which you can measure with a pretty
high degree of accuracy, then you can, over the piston range of the
driver, DIRECTLY and UNAMBIGUOUSLY determine the total acoustic power
as:

Pa = p/(2 pi c) * (Sd w^2 X)^2

where

p = density of air, typ. 1.18 kg/m^3
c = velocity of sound, typ 343 m/s
Sd = emissive area of the diaphragm in m^2
w = radian frequency
X = displacement of the diaphragm, in m.

the method is HARDLY novel at all, as it is well understood and utilized
in the field. If provides, for example, a means of measuring acoustical
power output without the confounding innaccuracies of microphones, rooms
and such, though the microphone innaccuracies are not a problem if you
use proper microphones to begin with.

I do undestand the theory, and I expect it would work well if the speaker
was moving in piston motion. (An some of the car sub woofers with ridged
aluminium cones would no doubt at low frequency)

However typical stereo speakers using either doped paper or pp cones won't,
as you know the LF breakup happens quite early on these type of drivers due
to the tradeoff of trying to give a wide response range with only 2 or 3
drives in most cases and stiff cone suspension.

More to your notion that it is "novel," you might want to modify that
opinion when you discover the technique is described in nearly every
text on acoustics.

-Noted and yes I have seen it mention and was taught such things, but it was
always theory and I had not heard of it done on audio speakers (however I do
know it is done on building structures to measure characteristic)

I do know that laser scanning of drive to measure the breakup is done, but
this was not what they were doing (So they said)

Compare that to using recording microphones whose measurement capabilities
are entirely unknown, in a room of unknown characteristics, using unknown
poorly calibrated and undoubtedly poorly controlled techniques by someone
who has little or know experience in measurement and acoustics...

I'd not bet good money on getting ANY reliable data out of the latter.

Well I have to say getting reliable data from 'experts' is not very
successful!

Ethan Winer wrote:

Not smoke and mirrors, acoustic interference. That's the real issue. And
that's why even 1/12th octave pink noise tests are useless to obtain a true
response within a room.

Bbuut ... "true response" as measured at any fixed location does not
apply for human beings because they do not ever stay at a fixed
location, they move around.

The quarter octave smoothened (based on eight octave measurement) that I
proposed on a previous version of my site has a much better correlation
to how things actually sound and overlooks stray and irrelevant
reflections from furniture and whatever.

Reflections that directly influence imaging are of course not "stray and
irrelevant", but if you want to measure the bass response of a
loudspeaker in a bookshelf, i. e. with lots of objects nearby then it
appears to me that there is no other way to do it than using a way of
measuring that is not clouded by statistical noise.

None of us live inside a anechoic chamber

Yes, but that's an entirely different issue. You said your stated goal is to
measure the *speaker's* response. Well, if you meausre it in a room you're
measuring far more of the *room's* response.

For measurements to be a useful aid in system alignement they should
correlate to the listernes experience of sonic balance. Narrow band
combfilter images tend to confuse rather than guide, it seems to me that
they focus on the room and omits the loudspeaker and the listeners
experienced impression thereof.

--Ethan


Kind regards

Peter Larsen

--
************************************************** ***********
* My site is at: http://www.muyiovatki.dk *
************************************************** ***********

Ethan Winer wrote:

Not smoke and mirrors, acoustic interference. That's the real issue. And
that's why even 1/12th octave pink noise tests are useless to obtain a true
response within a room.

Bbuut ... "true response" as measured at any fixed location does not
apply for human beings because they do not ever stay at a fixed
location, they move around.

The quarter octave smoothened (based on eight octave measurement) that I
proposed on a previous version of my site has a much better correlation
to how things actually sound and overlooks stray and irrelevant
reflections from furniture and whatever.

Reflections that directly influence imaging are of course not "stray and
irrelevant", but if you want to measure the bass response of a
loudspeaker in a bookshelf, i. e. with lots of objects nearby then it
appears to me that there is no other way to do it than using a way of
measuring that is not clouded by statistical noise.

None of us live inside a anechoic chamber

Yes, but that's an entirely different issue. You said your stated goal is to
measure the *speaker's* response. Well, if you meausre it in a room you're
measuring far more of the *room's* response.

For measurements to be a useful aid in system alignement they should
correlate to the listernes experience of sonic balance. Narrow band
combfilter images tend to confuse rather than guide, it seems to me that
they focus on the room and omits the loudspeaker and the listeners
experienced impression thereof.

--Ethan


Kind regards

Peter Larsen

--
************************************************** ***********
* My site is at: http://www.muyiovatki.dk *
************************************************** ***********

Our students in electroacoustics use this method in a lab on
loudspeakers. We use a microphone inside the box (Sennheiser MKE-2)
and software that can tilt the response by 12 dB/oct. It works great
(within the decibel compared to theoretical predictions of the
frequency response) down to the lower frequency limit of the
microphone and up to about half (or something) of the frequency of the
first standing wave inside the box. In theory, if you only could find
a microphone good enough, it would work down to DC. Works for closed
boxes and bass-reflex boxes.

The lab instructions are in swedish (sorry) but can be found at
http://www.speech.kth.se/~svante/elak/htlab.pdf
I think you can get an idea about it by looking at the pictures :-)

The software can be downloaded at
http://www.speech.kth.se/music/downloads/smptool/

I thought up my own method for eliminating the effect of the room on
bass measurement (although, no doubt it has been done before). The
main feature is to place the (pressure) microphone within the speaker
and to compensate by 12 dB/oct.

Normally I use a MLS and filter out higher frequencies, say above
100Hz, with a brick wall FIR filter using Coooledit. The 12 dB/oct
compensation is also done in a similar way, either to the input MLS
or to the output recording. This sort of pre-filtering can give a
large increase to the signal to noise of the meaasurement.

There must be some restrictions on the accuracy of this method but
the only one I can think off at the moment is that the wavelength must
be large compared to the internal dimensions of the speaker. The
meaasurement includes the effect of any port and of the enclosure
flexing.

There is probably an assumption that the air is compressed
adiabatically. The pressures involved are, of course, small if the mic
is not to be overloaded.

No, the pressures involved are HUGE, typically there is no problems
reaching 140 dB inside the box at low frequencies. So keep the levels
low in order to avoid overloading the microphone. (Or did I perhaps
misunderstand your statement?)

Our students in electroacoustics use this method in a lab on
loudspeakers. We use a microphone inside the box (Sennheiser MKE-2)
and software that can tilt the response by 12 dB/oct. It works great
(within the decibel compared to theoretical predictions of the
frequency response) down to the lower frequency limit of the
microphone and up to about half (or something) of the frequency of the
first standing wave inside the box. In theory, if you only could find
a microphone good enough, it would work down to DC. Works for closed
boxes and bass-reflex boxes.

The lab instructions are in swedish (sorry) but can be found at
http://www.speech.kth.se/~svante/elak/htlab.pdf
I think you can get an idea about it by looking at the pictures :-)

The software can be downloaded at
http://www.speech.kth.se/music/downloads/smptool/

I thought up my own method for eliminating the effect of the room on
bass measurement (although, no doubt it has been done before). The
main feature is to place the (pressure) microphone within the speaker
and to compensate by 12 dB/oct.

Normally I use a MLS and filter out higher frequencies, say above
100Hz, with a brick wall FIR filter using Coooledit. The 12 dB/oct
compensation is also done in a similar way, either to the input MLS
or to the output recording. This sort of pre-filtering can give a
large increase to the signal to noise of the meaasurement.

There must be some restrictions on the accuracy of this method but
the only one I can think off at the moment is that the wavelength must
be large compared to the internal dimensions of the speaker. The
meaasurement includes the effect of any port and of the enclosure
flexing.

There is probably an assumption that the air is compressed
adiabatically. The pressures involved are, of course, small if the mic
is not to be overloaded.

No, the pressures involved are HUGE, typically there is no problems
reaching 140 dB inside the box at low frequencies. So keep the levels
low in order to avoid overloading the microphone. (Or did I perhaps
misunderstand your statement?)

"Goofball_star_dot_etal" wrote in message

On Sun, 23 Nov 2003 12:29:42 -0800, "The Flash" wrote:

What type of / brand of mic are you using?



I was not trying to make a very accurate measurement so I just used a
cheap Panasonic WM-60AT price £2

http://rocky.digikey.com/WebLib/Pana...ata/WM-60A.pdf

http://www.digikey.com/digihome.html

Actually, this mic is known to be very accurate within the audio range. It's
major disadvantages are that it is a raw capsule with no supporting hardware
or circuitry, it tends to clip at moderately-high sound levels, and it has
questionable noise characeristics when used to pick up low-sound levels.
Your application seems to finesse these issues so it remains a good,
accurate mic for the application.

The Behringer ECM-8000 is arguably the next step up for under $40 in the US.
It is a finished microphone and has better performance at high sound levels
and w/r/t noise. When used for woofer measurements, it does have a
documented roll-off below 40 Hz that can be easily compensated for.

"Goofball_star_dot_etal" wrote in message

On Sun, 23 Nov 2003 12:29:42 -0800, "The Flash" wrote:

What type of / brand of mic are you using?



I was not trying to make a very accurate measurement so I just used a
cheap Panasonic WM-60AT price £2

http://rocky.digikey.com/WebLib/Pana...ata/WM-60A.pdf

http://www.digikey.com/digihome.html

Actually, this mic is known to be very accurate within the audio range. It's
major disadvantages are that it is a raw capsule with no supporting hardware
or circuitry, it tends to clip at moderately-high sound levels, and it has
questionable noise characeristics when used to pick up low-sound levels.
Your application seems to finesse these issues so it remains a good,
accurate mic for the application.

The Behringer ECM-8000 is arguably the next step up for under $40 in the US.
It is a finished microphone and has better performance at high sound levels
and w/r/t noise. When used for woofer measurements, it does have a
documented roll-off below 40 Hz that can be easily compensated for.

On 24 Nov 2003 00:24:57 -0800, (Svante)
wrote:

Our students in electroacoustics use this method in a lab on
loudspeakers.

I hope you teach them "inverted thinking" at the same time. :-)


We use a microphone inside the box (Sennheiser MKE-2)
and software that can tilt the response by 12 dB/oct. It works great
(within the decibel compared to theoretical predictions of the
frequency response) down to the lower frequency limit of the
microphone and up to about half (or something) of the frequency of the
first standing wave inside the box. In theory, if you only could find
a microphone good enough, it would work down to DC. Works for closed
boxes and bass-reflex boxes.


Thanks for the confirmation that the method works, although I had my
heart set on somebody "cleaning my clock" (objecting)

The lab instructions are in swedish (sorry) but can be found at
http://www.speech.kth.se/~svante/elak/htlab.pdf
I think you can get an idea about it by looking at the pictures :-)

I like pictures but you got me on the swedish

The software can be downloaded at
http://www.speech.kth.se/music/downloads/smptool/

I thought up my own method for eliminating the effect of the room on
bass measurement (although, no doubt it has been done before). The
main feature is to place the (pressure) microphone within the speaker
and to compensate by 12 dB/oct.

Normally I use a MLS and filter out higher frequencies, say above
100Hz, with a brick wall FIR filter using Coooledit. The 12 dB/oct
compensation is also done in a similar way, either to the input MLS
or to the output recording. This sort of pre-filtering can give a
large increase to the signal to noise of the meaasurement.

There must be some restrictions on the accuracy of this method but
the only one I can think off at the moment is that the wavelength must
be large compared to the internal dimensions of the speaker. The
meaasurement includes the effect of any port and of the enclosure
flexing.

There is probably an assumption that the air is compressed
adiabatically. The pressures involved are, of course, small if the mic
is not to be overloaded.

No, the pressures involved are HUGE, typically there is no problems
reaching 140 dB inside the box at low frequencies. So keep the levels
low in order to avoid overloading the microphone. Or did I perhaps
misunderstand your statement?)

Yes I should have been more clear. I am economical with words to the
point of recklessness. I should have warned that it is possible to
damage the microphone, which is a good reason to use a cheap one to
perfect the setup. What I meant was that when the levels are adjusted
so that the mic. is operating linearly then it follows that the
physics will be right - the pressure changes inside the box will be
linear with volume displacement. It is a small signal measurement with
large mic signals. . . You may have to fit an attenuator (with low
output impedance between amp and speaker.

The actual compression/expansion will be somewhere between adiabatic
and isothermal and since isothermal involves heat exchange with the
mass of any stuffing I might expect the degree to which it tends
towards isothemal to change somewhat with frequency.

Making complementary filters by say, filtering an "impulse" with a
high-pass filter and then subtracting this from the original "impulse"
to form the high-pass, convolving these impulse responses with the
MLS to generate a low and high frequency test waveform, makes it
possible to "stitch together" a complete IR for the speaker using the
in-box method for the low end and a gated measurment outside the box
for the high end of the range. (you have to jiggle with time delay and
gain) Or, better still, just forget what I just wrote. . .

On 24 Nov 2003 00:24:57 -0800, (Svante)
wrote:

Our students in electroacoustics use this method in a lab on
loudspeakers.

I hope you teach them "inverted thinking" at the same time. :-)


We use a microphone inside the box (Sennheiser MKE-2)
and software that can tilt the response by 12 dB/oct. It works great
(within the decibel compared to theoretical predictions of the
frequency response) down to the lower frequency limit of the
microphone and up to about half (or something) of the frequency of the
first standing wave inside the box. In theory, if you only could find
a microphone good enough, it would work down to DC. Works for closed
boxes and bass-reflex boxes.


Thanks for the confirmation that the method works, although I had my
heart set on somebody "cleaning my clock" (objecting)

The lab instructions are in swedish (sorry) but can be found at
http://www.speech.kth.se/~svante/elak/htlab.pdf
I think you can get an idea about it by looking at the pictures :-)

I like pictures but you got me on the swedish

The software can be downloaded at
http://www.speech.kth.se/music/downloads/smptool/

I thought up my own method for eliminating the effect of the room on
bass measurement (although, no doubt it has been done before). The
main feature is to place the (pressure) microphone within the speaker
and to compensate by 12 dB/oct.

Normally I use a MLS and filter out higher frequencies, say above
100Hz, with a brick wall FIR filter using Coooledit. The 12 dB/oct
compensation is also done in a similar way, either to the input MLS
or to the output recording. This sort of pre-filtering can give a
large increase to the signal to noise of the meaasurement.

There must be some restrictions on the accuracy of this method but
the only one I can think off at the moment is that the wavelength must
be large compared to the internal dimensions of the speaker. The
meaasurement includes the effect of any port and of the enclosure
flexing.

There is probably an assumption that the air is compressed
adiabatically. The pressures involved are, of course, small if the mic
is not to be overloaded.

No, the pressures involved are HUGE, typically there is no problems
reaching 140 dB inside the box at low frequencies. So keep the levels
low in order to avoid overloading the microphone. Or did I perhaps
misunderstand your statement?)

Yes I should have been more clear. I am economical with words to the
point of recklessness. I should have warned that it is possible to
damage the microphone, which is a good reason to use a cheap one to
perfect the setup. What I meant was that when the levels are adjusted
so that the mic. is operating linearly then it follows that the
physics will be right - the pressure changes inside the box will be
linear with volume displacement. It is a small signal measurement with
large mic signals. . . You may have to fit an attenuator (with low
output impedance between amp and speaker.

The actual compression/expansion will be somewhere between adiabatic
and isothermal and since isothermal involves heat exchange with the
mass of any stuffing I might expect the degree to which it tends
towards isothemal to change somewhat with frequency.

Making complementary filters by say, filtering an "impulse" with a
high-pass filter and then subtracting this from the original "impulse"
to form the high-pass, convolving these impulse responses with the
MLS to generate a low and high frequency test waveform, makes it
possible to "stitch together" a complete IR for the speaker using the
in-box method for the low end and a gated measurment outside the box
for the high end of the range. (you have to jiggle with time delay and
gain) Or, better still, just forget what I just wrote. . .

Peter,

overlooks stray and irrelevant reflections from furniture and whatever ...
Reflections that directly influence imaging

I was mostly talking about the severe response problems that occur at low
frequencies. Mid and high frequency reflections can be tamed easily and
cheaply. Low frequencies bounce off the walls and collide in mid-air, and
that is much more difficult to correct.

--Ethan

Peter,

overlooks stray and irrelevant reflections from furniture and whatever ...
Reflections that directly influence imaging

I was mostly talking about the severe response problems that occur at low
frequencies. Mid and high frequency reflections can be tamed easily and
cheaply. Low frequencies bounce off the walls and collide in mid-air, and
that is much more difficult to correct.

--Ethan

"Ian" wrote in message
...
Your assertion that it is not possible to get low frequencies using an
8 inch driver turns out not to be correct. I have a pair where the LF
is done using 8 inch drivers in a bandpass enclosure. Those have a
measured response which is 1dB down at 30Hz, and an overall
response at 3dB down of 27Hz to 90Hz.

No mention of what the maximum SPL Vs distortion level is at 30 Hz though,
I see.
Simply disregard these and you can easily get 30 Hz from a single 4 inch
driver.
Of course you can't actually HEAR it :-)

TonyP.

"Ian" wrote in message
...
Your assertion that it is not possible to get low frequencies using an
8 inch driver turns out not to be correct. I have a pair where the LF
is done using 8 inch drivers in a bandpass enclosure. Those have a
measured response which is 1dB down at 30Hz, and an overall
response at 3dB down of 27Hz to 90Hz.

No mention of what the maximum SPL Vs distortion level is at 30 Hz though,
I see.
Simply disregard these and you can easily get 30 Hz from a single 4 inch
driver.
Of course you can't actually HEAR it :-)

TonyP.

Ethan Winer wrote:

overlooks stray and irrelevant reflections from furniture and
whatever ...

Reflections that directly influence imaging

Depends really, a 1" by 1" piece of wood carrying a bookshelf does not
gravely influence imaging, but it does void the conditions for a gated
measurement.

I was mostly talking about the severe response problems that occur at low
frequencies. Mid and high frequency reflections can be tamed easily and
cheaply.

Indeed, and it is easier to get a room too boring than to keep it
suitably live and yet good.

Low frequencies bounce off the walls and collide in mid-air,

This does depend on the wavelength.

and that is much more difficult to correct.

I can see how actually measuring the narrow interference patterns makes
sense in the special case of adding absorbers to the room, yes. Other
than that they are just statistical noise because of the non static
nature of most listeners.

I don't really think that one should "get lost" in the steady state
performance of a room by measuring with non-noise or non-warbled tones,
i. e. sinewaves except in the special case of finding the optimum
location for a bass trap.

Perhaps I should put that stuff about measuring without much equipment
up on my site again .... of course, now software rta is more common, so
it may be is less relevant, but I liked the correlation between the
resulting graph and actual experience of the tonal balance.

--Ethan


Kind regards

Peter Larsen

--
************************************************** ***********
* My site is at: http://www.muyiovatki.dk *
************************************************** ***********

Ethan Winer wrote:

overlooks stray and irrelevant reflections from furniture and
whatever ...

Reflections that directly influence imaging

Depends really, a 1" by 1" piece of wood carrying a bookshelf does not
gravely influence imaging, but it does void the conditions for a gated
measurement.

I was mostly talking about the severe response problems that occur at low
frequencies. Mid and high frequency reflections can be tamed easily and
cheaply.

Indeed, and it is easier to get a room too boring than to keep it
suitably live and yet good.

Low frequencies bounce off the walls and collide in mid-air,

This does depend on the wavelength.

and that is much more difficult to correct.

I can see how actually measuring the narrow interference patterns makes
sense in the special case of adding absorbers to the room, yes. Other
than that they are just statistical noise because of the non static
nature of most listeners.

I don't really think that one should "get lost" in the steady state
performance of a room by measuring with non-noise or non-warbled tones,
i. e. sinewaves except in the special case of finding the optimum
location for a bass trap.

Perhaps I should put that stuff about measuring without much equipment
up on my site again .... of course, now software rta is more common, so
it may be is less relevant, but I liked the correlation between the
resulting graph and actual experience of the tonal balance.

--Ethan


Kind regards

Peter Larsen

--
************************************************** ***********
* My site is at: http://www.muyiovatki.dk *
************************************************** ***********

"Arny Krueger" wrote in message
...
The Behringer ECM-8000 is arguably the next step up for under
$40 in the US.
It is a finished microphone and has better performance at high
sound levels
and w/r/t noise. When used for woofer measurements, it does
have a
documented roll-off below 40 Hz that can be easily compensated
for.

Does anyone have a correction chart for the ECM-8000? The
datasheet shows flat to 40Hz or so but no data below.

"Arny Krueger" wrote in message
...
The Behringer ECM-8000 is arguably the next step up for under
$40 in the US.
It is a finished microphone and has better performance at high
sound levels
and w/r/t noise. When used for woofer measurements, it does
have a
documented roll-off below 40 Hz that can be easily compensated
for.

Does anyone have a correction chart for the ECM-8000? The
datasheet shows flat to 40Hz or so but no data below.

"Rusty Boudreaux" wrote in message

"Arny Krueger" wrote in message
...
The Behringer ECM-8000 is arguably the next step up for under $40 in
the US. It is a finished microphone and has better performance at
high sound levels and w/r/t noise. When used for woofer
measurements, it does have a documented roll-off below 40 Hz that
can be easily compensated for.

Does anyone have a correction chart for the ECM-8000? The
datasheet shows flat to 40Hz or so but no data below.

Interesting, if my recollections are correct, they've changed it since the
last time I looked.

"Rusty Boudreaux" wrote in message

"Arny Krueger" wrote in message
...
The Behringer ECM-8000 is arguably the next step up for under $40 in
the US. It is a finished microphone and has better performance at
high sound levels and w/r/t noise. When used for woofer
measurements, it does have a documented roll-off below 40 Hz that
can be easily compensated for.

Does anyone have a correction chart for the ECM-8000? The
datasheet shows flat to 40Hz or so but no data below.

Interesting, if my recollections are correct, they've changed it since the
last time I looked.

Sorry to offend you Dick this was not my aim, my termanology is not the
best. My training in the field of audio engineering is what I got in the
first year of my engineering course about a decade ago. The first year
covered all engineering topics, years 2 and 3 were specilised. In the first
year the field covered were Heat and Vent, Audio, Power Electrical,
Televison, Computer and Math. Lots and Lots of theory and lots of Lab work.
Quite why the course were run this way was beyond me, There were students
from all around the country doing the First year who the had to go somewhere
else to do years 2 and three!

The big thing was that they only taught the topics if people were enrolled
in that stream. That year we had 1 person doing Heat and Vent and 1 doing
Televison and 1 doing Power Electrical, I seem to remember 2 or 3 doing the
Audio enginnering side and the rest were doing Computer engineering. We
ended up doing all manner of often bizzar stuff. Of the stuff we did I
actually found the Audio side very interesting (as did almost every one it
seemed) and enjoyed playing with amplifiers and speaker and mics in the
anechoic chamber using the spectrum analyzers and such. All the digital
stuff was good but the Heating and Vent stuff was mindless.

I almost quit the course when we spent day after day designing H&V for
buildings, the lecturor / tutor was some old fossil they dragged out of the
grave and he had us doing all drawings on paper and refused to allow
calculators to be used all longhand had to be shown, oh we were allowed to
use sliderules to check our answers! (No one in the class had a slide rule)
after 8 weeks of this he strangly disappared and they roped in an Electrical
tutor, lo and behold we now started using Autocad and doing the Drawings on
PC's! Thru the year a lot of people left by the end we were down to only
about 20 out of 30, the guy doing TV left mid year and we asked if the TV
modules could be dropped but typical we had to do them and you had to pass
them! At the end of the year 20 odd people had passed Stage 3 Televison and
none were going on to stages 4 and 5.
(Stage 3 was a certificate). I have always had an interest in Audio and have
it as a hobby for a long time now. (The above just made me remember the
Army had a class doing the above courses at the same time but they had army
instructors teaching them, although that had the same 50% for a pass they
had a 66% or better 'prefered' it was easy to see who failed to get a
'prefered' mark as the were the ones required to run round the fields and
stand to eat in the cafeteria! It seemed to motivate them!)

I am interest in why a bad speaker sound bad and why a good speaker sound
good (Yet both can be of similar design). My comments on LF response were
not clarified very well. I do know that you can get good low frequency
response with 8 inch speakers, its just it doesn't seem the norm from what I
have found and getting less common all the time (I don't like this trend to
small speakers and smaller enclosers with ports)

I really like dome midranges, I cannot understand why these are not used
more often.

I feel that the conventional 3 drive infinite baffle(and the acoustic
suspension) speaker systems using a soft dome tweeter and either a 4 inch
midrange or a dome mid and a 10 or 12 inch base unit is inherently a good
design sound wise. (Exept it takes a lot of room and tend to be little
ineffcient)

I have a dislike of bandpass and bass reflex designs. (I own a few bass
reflex speakers and bandpass subs)

I have only ever tried one transmisson line and have formed no opinion other
than it was a brute of a thing weighing in at 220 pounds per cabinet used a
B139 kef LF unit, unsure of who made or designed it but it was a quality bit
of wood work!

I find cone tweeter seem to have good frequency response but very poor axial
characteristics
I also find that metal dome tweeters have great response excellent axial
characteristics but I don't like the sound they give in real use.

I have very serious doubts about the wave of 5.1 surround sound speaker
systems being sold and the most unnatural audio characteristics that any I
have tried generate (It seems that have active subs rather than passive subs
may be a change for the worse but I don't know for sure)

Ok, I have probably really rilled up a few people with my opinions, (Best to
get my asbestos undies on).

So sorry again Dick for any offence. I thought you were doing a King in the
castle tossing the bones down to the starving peasents and saying 'Go make
soup'!

Sorry to offend you Dick this was not my aim, my termanology is not the
best. My training in the field of audio engineering is what I got in the
first year of my engineering course about a decade ago. The first year
covered all engineering topics, years 2 and 3 were specilised. In the first
year the field covered were Heat and Vent, Audio, Power Electrical,
Televison, Computer and Math. Lots and Lots of theory and lots of Lab work.
Quite why the course were run this way was beyond me, There were students
from all around the country doing the First year who the had to go somewhere
else to do years 2 and three!

The big thing was that they only taught the topics if people were enrolled
in that stream. That year we had 1 person doing Heat and Vent and 1 doing
Televison and 1 doing Power Electrical, I seem to remember 2 or 3 doing the
Audio enginnering side and the rest were doing Computer engineering. We
ended up doing all manner of often bizzar stuff. Of the stuff we did I
actually found the Audio side very interesting (as did almost every one it
seemed) and enjoyed playing with amplifiers and speaker and mics in the
anechoic chamber using the spectrum analyzers and such. All the digital
stuff was good but the Heating and Vent stuff was mindless.

I almost quit the course when we spent day after day designing H&V for
buildings, the lecturor / tutor was some old fossil they dragged out of the
grave and he had us doing all drawings on paper and refused to allow
calculators to be used all longhand had to be shown, oh we were allowed to
use sliderules to check our answers! (No one in the class had a slide rule)
after 8 weeks of this he strangly disappared and they roped in an Electrical
tutor, lo and behold we now started using Autocad and doing the Drawings on
PC's! Thru the year a lot of people left by the end we were down to only
about 20 out of 30, the guy doing TV left mid year and we asked if the TV
modules could be dropped but typical we had to do them and you had to pass
them! At the end of the year 20 odd people had passed Stage 3 Televison and
none were going on to stages 4 and 5.
(Stage 3 was a certificate). I have always had an interest in Audio and have
it as a hobby for a long time now. (The above just made me remember the
Army had a class doing the above courses at the same time but they had army
instructors teaching them, although that had the same 50% for a pass they
had a 66% or better 'prefered' it was easy to see who failed to get a
'prefered' mark as the were the ones required to run round the fields and
stand to eat in the cafeteria! It seemed to motivate them!)

I am interest in why a bad speaker sound bad and why a good speaker sound
good (Yet both can be of similar design). My comments on LF response were
not clarified very well. I do know that you can get good low frequency
response with 8 inch speakers, its just it doesn't seem the norm from what I
have found and getting less common all the time (I don't like this trend to
small speakers and smaller enclosers with ports)

I really like dome midranges, I cannot understand why these are not used
more often.

I feel that the conventional 3 drive infinite baffle(and the acoustic
suspension) speaker systems using a soft dome tweeter and either a 4 inch
midrange or a dome mid and a 10 or 12 inch base unit is inherently a good
design sound wise. (Exept it takes a lot of room and tend to be little
ineffcient)

I have a dislike of bandpass and bass reflex designs. (I own a few bass
reflex speakers and bandpass subs)

I have only ever tried one transmisson line and have formed no opinion other
than it was a brute of a thing weighing in at 220 pounds per cabinet used a
B139 kef LF unit, unsure of who made or designed it but it was a quality bit
of wood work!

I find cone tweeter seem to have good frequency response but very poor axial
characteristics
I also find that metal dome tweeters have great response excellent axial
characteristics but I don't like the sound they give in real use.

I have very serious doubts about the wave of 5.1 surround sound speaker
systems being sold and the most unnatural audio characteristics that any I
have tried generate (It seems that have active subs rather than passive subs
may be a change for the worse but I don't know for sure)

Ok, I have probably really rilled up a few people with my opinions, (Best to
get my asbestos undies on).

So sorry again Dick for any offence. I thought you were doing a King in the
castle tossing the bones down to the starving peasents and saying 'Go make
soup'!

"The Flash" wrote in message ...
Your assertion that it is not possible to get low frequencies using an
8 inch driver turns out not to be correct. I have a pair where the LF
is done using 8 inch drivers in a bandpass enclosure. Those have a
measured response which is 1dB down at 30Hz, and an overall
response at 3dB down of 27Hz to 90Hz.

Ok, your using a pair of 8 inch drivers not a single driver, a band pass
enclosure to give base reinforcment. I have a Isobaric bandbass using two 8
inch drivers and yes it will produce low frequencies well, however its
response range is quite limited and its low effientcy is masked by its class
H amplifier inbuilt. I still doubt that you can get a true 'Hifi' response
from a system using a 8 inch driver per cabinet for the LF.

A couple of misconceptions here.

All else being equal, more drivers do not result in lower cutoff
frequencies, they result in more total acoustic power capability,
since you've increased the total displacement volume of tyhe system.

So-called (actually, mis-named) 'isobarik' systems are a way of
obtaining a given response in half the volume, since the arrangement
results in half the total equivalent compliance volume and thus
requires half the enclosure volume for the same response function.

As to "true hifi response," such a term needs definition to be
anything but meaningless jargon, in precisely the same way that
a "specification" like "20Hz-20kHz" is meaningless without
qualification.

THe fundamental efficiency/enclosure volume/cutoff frequency equation
will ALWAYS rule. But, interestingly enough, driver diameter simply
does not enter into that relation. Specifically, the relation:

n0 = kn Vb F3^3

where n0 is reference efficiency, Vb is enclosure volume, F3 is is
low frequency cutoff and kn is the efficiency constant has NO term
in it in any way depedent upon driver diameter. This directly refutes
your assertion that it's not possible to get your "unqualified" 'hifi
response from an 8 inch driver. One needs to simply balance the three
terms of efficiency, enclosure volume and cutoff frequency and you're
there. You seem to intimate that low efficiency is incompatible with
'hifi response," for example, an unjustifiable viewpoint in light
of the lack of qualification of 'hifi response.'

"The Flash" wrote in message ...
Your assertion that it is not possible to get low frequencies using an
8 inch driver turns out not to be correct. I have a pair where the LF
is done using 8 inch drivers in a bandpass enclosure. Those have a
measured response which is 1dB down at 30Hz, and an overall
response at 3dB down of 27Hz to 90Hz.

Ok, your using a pair of 8 inch drivers not a single driver, a band pass
enclosure to give base reinforcment. I have a Isobaric bandbass using two 8
inch drivers and yes it will produce low frequencies well, however its
response range is quite limited and its low effientcy is masked by its class
H amplifier inbuilt. I still doubt that you can get a true 'Hifi' response
from a system using a 8 inch driver per cabinet for the LF.

A couple of misconceptions here.

All else being equal, more drivers do not result in lower cutoff
frequencies, they result in more total acoustic power capability,
since you've increased the total displacement volume of tyhe system.

So-called (actually, mis-named) 'isobarik' systems are a way of
obtaining a given response in half the volume, since the arrangement
results in half the total equivalent compliance volume and thus
requires half the enclosure volume for the same response function.

As to "true hifi response," such a term needs definition to be
anything but meaningless jargon, in precisely the same way that
a "specification" like "20Hz-20kHz" is meaningless without
qualification.

THe fundamental efficiency/enclosure volume/cutoff frequency equation
will ALWAYS rule. But, interestingly enough, driver diameter simply
does not enter into that relation. Specifically, the relation:

n0 = kn Vb F3^3

where n0 is reference efficiency, Vb is enclosure volume, F3 is is
low frequency cutoff and kn is the efficiency constant has NO term
in it in any way depedent upon driver diameter. This directly refutes
your assertion that it's not possible to get your "unqualified" 'hifi
response from an 8 inch driver. One needs to simply balance the three
terms of efficiency, enclosure volume and cutoff frequency and you're
there. You seem to intimate that low efficiency is incompatible with
'hifi response," for example, an unjustifiable viewpoint in light
of the lack of qualification of 'hifi response.'

Peter,

This does depend on the wavelength.

Sure. The big problems are between maybe 40 and 300 Hz, and that's a pretty
wide range.

non static nature of most listeners.

Good point. If one is in the habit of walking around constantly while
listening to music, then the response problems will move around too. But if
you sit in one place (some of us do that), the peaks and nulls - especially
the nulls - can be extremely deep. I recently measured an untreated 16x10x8
room and the response at 122 Hz was 31 dB lower than the response at 108 Hz.
This is a HUGE drop in level across a very narrow frequency range! And it's
absolutely typical of small, untreated rooms.

I don't really think that one should "get lost" in ... sinewaves except in
the special case of finding the optimum location for a bass trap.

I disagree. If you think about it, other than snare drums and cymbals, all
music consists mainly of steady sine waves. A kick drum is basically a click
with a sine wave that decays. An electric bass is mainly a fundamental sine
wave with a second harmonic sine wave. When you play pop music using these
as the main low frequency instruments, you ARE playing sine waves. So if you
want to know how accurate the response is for these typical sources, then
static sine waves are the ONLY sensible test.

--Ethan

Peter,

This does depend on the wavelength.

Sure. The big problems are between maybe 40 and 300 Hz, and that's a pretty
wide range.

non static nature of most listeners.

Good point. If one is in the habit of walking around constantly while
listening to music, then the response problems will move around too. But if
you sit in one place (some of us do that), the peaks and nulls - especially
the nulls - can be extremely deep. I recently measured an untreated 16x10x8
room and the response at 122 Hz was 31 dB lower than the response at 108 Hz.
This is a HUGE drop in level across a very narrow frequency range! And it's
absolutely typical of small, untreated rooms.

I don't really think that one should "get lost" in ... sinewaves except in
the special case of finding the optimum location for a bass trap.

I disagree. If you think about it, other than snare drums and cymbals, all
music consists mainly of steady sine waves. A kick drum is basically a click
with a sine wave that decays. An electric bass is mainly a fundamental sine
wave with a second harmonic sine wave. When you play pop music using these
as the main low frequency instruments, you ARE playing sine waves. So if you
want to know how accurate the response is for these typical sources, then
static sine waves are the ONLY sensible test.

--Ethan

Hi Ethan,

Ethan Winer wrote:

I don't really think that one should "get lost" in ... sinewaves except in
the special case of finding the optimum location for a bass trap.

I disagree. If you think about it, other than snare drums and cymbals, all
music consists mainly of steady sine waves. A kick drum is basically a click
with a sine wave that decays.

Let us take the deceptively simple example of a piano. Way long time ago
Scientific American published a paper about the piano and its tuning. It
is a very non-simple sound source, including that resulting
emanated/perceived frequencies are non-constant as I remember that
paper.

It is also in my experience one of the three sound sources that combined
tell all about the properties of loudspeakers and room combined. The
other two sources are a or a few violins, i.e. a chamber music ensemble
and a full symphony orchestra, well recorded, something romantic is fine
because of the probable fullness of the sound. The latter will reveal
all room colorations that need to be address because it will
"de-compose" and "segment" if there are - ahem - issues.

An electric bass is mainly a fundamental sine
wave with a second harmonic sine wave. When you play pop music using these
as the main low frequency instruments, you ARE playing sine waves. So if you
want to know how accurate the response is for these typical sources, then
static sine waves are the ONLY sensible test.

No contest. Static sinewaves will by their behavior tell you the exact
frequency of a problem. Frankly I plain do not like to use them in my
apartment, if not for any other reason then because it is an apartment
and not a house and currently I have peace with my neighbors, I like to
keep it that way. I think we see this differently because - as I
understand you - it is your occupation to solve room problems fast.

The listening test above will tell you whether there is a problem that
matters. Rock and pop music are great for rapid cross-over tuning but
they are not the fastest possible way to determine minor problems. Major
problems, such as a resonance that is the same between multiple walls
due to impractical room dimensions, yes - that will be very obvious. I
recall one listening room where the only resort was to use a narrow
notch filter at such a fundamental room node, 52-55 Hz could not be
reproduced in that room.

The TACT stuff is of course an interesting solution WHEN the major
problems are solved, but I can not - on the basis of the demonstration I
was at - make my mind up as to whether it really is an advantage. But it
could be great fun to be into tuning its parameters ...

--Ethan


Kind regards

Peter Larsen



--
************************************************** ***********
* My site is at: http://www.muyiovatki.dk *
************************************************** ***********

Hi Ethan,

Ethan Winer wrote:

I don't really think that one should "get lost" in ... sinewaves except in
the special case of finding the optimum location for a bass trap.

I disagree. If you think about it, other than snare drums and cymbals, all
music consists mainly of steady sine waves. A kick drum is basically a click
with a sine wave that decays.

Let us take the deceptively simple example of a piano. Way long time ago
Scientific American published a paper about the piano and its tuning. It
is a very non-simple sound source, including that resulting
emanated/perceived frequencies are non-constant as I remember that
paper.

It is also in my experience one of the three sound sources that combined
tell all about the properties of loudspeakers and room combined. The
other two sources are a or a few violins, i.e. a chamber music ensemble
and a full symphony orchestra, well recorded, something romantic is fine
because of the probable fullness of the sound. The latter will reveal
all room colorations that need to be address because it will
"de-compose" and "segment" if there are - ahem - issues.

An electric bass is mainly a fundamental sine
wave with a second harmonic sine wave. When you play pop music using these
as the main low frequency instruments, you ARE playing sine waves. So if you
want to know how accurate the response is for these typical sources, then
static sine waves are the ONLY sensible test.

No contest. Static sinewaves will by their behavior tell you the exact
frequency of a problem. Frankly I plain do not like to use them in my
apartment, if not for any other reason then because it is an apartment
and not a house and currently I have peace with my neighbors, I like to
keep it that way. I think we see this differently because - as I
understand you - it is your occupation to solve room problems fast.

The listening test above will tell you whether there is a problem that
matters. Rock and pop music are great for rapid cross-over tuning but
they are not the fastest possible way to determine minor problems. Major
problems, such as a resonance that is the same between multiple walls
due to impractical room dimensions, yes - that will be very obvious. I
recall one listening room where the only resort was to use a narrow
notch filter at such a fundamental room node, 52-55 Hz could not be
reproduced in that room.

The TACT stuff is of course an interesting solution WHEN the major
problems are solved, but I can not - on the basis of the demonstration I
was at - make my mind up as to whether it really is an advantage. But it
could be great fun to be into tuning its parameters ...

--Ethan


Kind regards

Peter Larsen



--
************************************************** ***********
* My site is at: http://www.muyiovatki.dk *
************************************************** ***********

"Arny Krueger" wrote in message
...
"Rusty Boudreaux" wrote in message

"Arny Krueger" wrote in message
...
The Behringer ECM-8000 is arguably the next step up for under $40 in
the US. It is a finished microphone and has better performance at
high sound levels and w/r/t noise. When used for woofer
measurements, it does have a documented roll-off below 40 Hz that
can be easily compensated for.

Does anyone have a correction chart for the ECM-8000? The
datasheet shows flat to 40Hz or so but no data below.

Interesting, if my recollections are correct, they've changed it since the
last time I looked.


That's interesting to me as well ;-)

I'm sure I have a mic response chart that goes below that, and doing
nearfield
measurements on a bandpass bass enclosure gave virtually perfect match
to expected down to 30Hz (HP35670, frequency response error from
expected well under 1dB, BUT can't remember if this was before or
after getting the mic replaced after failing).

Arny, do you have a source (and date) for the documented rolloff?

Regards
Ian

"Arny Krueger" wrote in message
...
"Rusty Boudreaux" wrote in message

"Arny Krueger" wrote in message
...
The Behringer ECM-8000 is arguably the next step up for under $40 in
the US. It is a finished microphone and has better performance at
high sound levels and w/r/t noise. When used for woofer
measurements, it does have a documented roll-off below 40 Hz that
can be easily compensated for.

Does anyone have a correction chart for the ECM-8000? The
datasheet shows flat to 40Hz or so but no data below.

Interesting, if my recollections are correct, they've changed it since the
last time I looked.


That's interesting to me as well ;-)

I'm sure I have a mic response chart that goes below that, and doing
nearfield
measurements on a bandpass bass enclosure gave virtually perfect match
to expected down to 30Hz (HP35670, frequency response error from
expected well under 1dB, BUT can't remember if this was before or
after getting the mic replaced after failing).

Arny, do you have a source (and date) for the documented rolloff?

Regards
Ian

"Tony Pearce" wrote in message
u...

"Ian" wrote in message
...
Your assertion that it is not possible to get low frequencies using an
8 inch driver turns out not to be correct. I have a pair where the LF
is done using 8 inch drivers in a bandpass enclosure. Those have a
measured response which is 1dB down at 30Hz, and an overall
response at 3dB down of 27Hz to 90Hz.

No mention of what the maximum SPL Vs distortion level is at 30 Hz
though,
I see.
Simply disregard these and you can easily get 30 Hz from a single 4 inch
driver.
Of course you can't actually HEAR it :-)

TonyP.



Tony, please note I said "bandpass enclosure". Some of the excursion/output
relations change.

You (perhaps) cannot "hear" 30Hz, although that depends ;-), however it does
make a significant difference to the listening experience (and I DO NOT MEAN
listening to harmonics - I am a member of the AES, and I pretty fully
understand
the issues here). (Waits for the positioning/room size/modes/listener
placement
"squeaks".)

Maximum SPL vs distortion level is just fine for reasonable levels in a
reasonable
(UK) size room.
Regards
Ian

"Tony Pearce" wrote in message
u...

"Ian" wrote in message
...
Your assertion that it is not possible to get low frequencies using an
8 inch driver turns out not to be correct. I have a pair where the LF
is done using 8 inch drivers in a bandpass enclosure. Those have a
measured response which is 1dB down at 30Hz, and an overall
response at 3dB down of 27Hz to 90Hz.

No mention of what the maximum SPL Vs distortion level is at 30 Hz
though,
I see.
Simply disregard these and you can easily get 30 Hz from a single 4 inch
driver.
Of course you can't actually HEAR it :-)

TonyP.



Tony, please note I said "bandpass enclosure". Some of the excursion/output
relations change.

You (perhaps) cannot "hear" 30Hz, although that depends ;-), however it does
make a significant difference to the listening experience (and I DO NOT MEAN
listening to harmonics - I am a member of the AES, and I pretty fully
understand
the issues here). (Waits for the positioning/room size/modes/listener
placement
"squeaks".)

Maximum SPL vs distortion level is just fine for reasonable levels in a
reasonable
(UK) size room.
Regards
Ian

Your assertion that it is not possible to get low frequencies using an
8 inch driver turns out not to be correct. I have a pair where the LF
is done using 8 inch drivers in a bandpass enclosure. Those have a
measured response which is 1dB down at 30Hz, and an overall
response at 3dB down of 27Hz to 90Hz.

Regards
Ian


Ok, your using a pair of 8 inch drivers not a single driver, a band pass
enclosure to give base reinforcment. I have a Isobaric bandbass using two 8
inch drivers and yes it will produce low frequencies well, however its
response range is quite limited and its low effientcy is masked by its class
H amplifier inbuilt. I still doubt that you can get a true 'Hifi' response
from a system using a 8 inch driver per cabinet for the LF.

Your assertion that it is not possible to get low frequencies using an
8 inch driver turns out not to be correct. I have a pair where the LF
is done using 8 inch drivers in a bandpass enclosure. Those have a
measured response which is 1dB down at 30Hz, and an overall
response at 3dB down of 27Hz to 90Hz.

Regards
Ian


Ok, your using a pair of 8 inch drivers not a single driver, a band pass
enclosure to give base reinforcment. I have a Isobaric bandbass using two 8
inch drivers and yes it will produce low frequencies well, however its
response range is quite limited and its low effientcy is masked by its class
H amplifier inbuilt. I still doubt that you can get a true 'Hifi' response
from a system using a 8 inch driver per cabinet for the LF.

"Arny Krueger" wrote in message
...
"Rusty Boudreaux" wrote in message
Does anyone have a correction chart for the ECM-8000? The
datasheet shows flat to 40Hz or so but no data below.

Interesting, if my recollections are correct, they've changed
it since the
last time I looked.

I just downloaded the pdf. The stated response is 15Hz-20kHz but
no dB qualifier. The magnitude plot shows essentially flat down
to 60Hz but no data below. However, they do show response to
30kHz...down 10dB.

http://www.behringer-download.de/ECM...00_C_Specs.pdf

"Arny Krueger" wrote in message
...
"Rusty Boudreaux" wrote in message
Does anyone have a correction chart for the ECM-8000? The
datasheet shows flat to 40Hz or so but no data below.

Interesting, if my recollections are correct, they've changed
it since the
last time I looked.

I just downloaded the pdf. The stated response is 15Hz-20kHz but
no dB qualifier. The magnitude plot shows essentially flat down
to 60Hz but no data below. However, they do show response to
30kHz...down 10dB.

http://www.behringer-download.de/ECM...00_C_Specs.pdf

I have yet to find ANY speaker system using an 8 inch driver that has the
ability to produce 40hz or lower frequencys as they all have fallen so far
down in output level as to be useless.

Some 8" studio monitors can produce sub 40hz meaningfully, though not very
far below, and they split the drivers at a crossover. HR824's fall off
rapidly below 39hz but 40hz is definitely present, at least my chest thinks
so. The test tones are definitely there. I can feel and sort of hear
*something* (or is that my headphones on my desk vibrating...).

I can't see a single speaker being very flat or even close at the full range
of 20-20k. There seems to be some fundamental limitations to most
loudspeaker designs that would preclude this.

If you find this holy grail, please post the manufacturer he )

(Oh one firm told me that they use a laser to measure the low frequency of
their speakers, check this out for novel! They place a small piece of
reflective foil on the base driver, and shine a laser beam on it, they
then
apply signal and measure via 'laser' the deflection, Also they place a
passive radiator 1 meter infront of the driver unit and use the same
method
to measure its deflection.

This seems reasonable. If it is vibrating at 20 hz, and measurable, it
produces 20 hz. No one has ever said they have a single speaker with a flat
frequency response from 20-20k, just that the speaker will respond audibly
to this frequency range. This is why they all post 20-20k response, most
without a frequency response curve. It wouldn't be pretty or sell speakers.

"The Flash" wrote in message
...
I have had no luck in measuring speaker response at low frequency's, I can
measure from ~60Hz to 20Khz with near 100% repeatability (How accurate is
questionable)

I have questioned a number of speaker builders and a couple of companies
that 'tune' speakers the answers given on what they measure with and how
the
do the tests give serious rise to the claims at low frequency.

Almost any speaker system sold today claims 20Hz to 20Kz response yet this
is so far from the truth I cannot understand how they dare claim such
figures.




How do you measure the responese at 20Hz?, I have tested a few speakers
and
basically I cannot get accurate (repeatable) data at much below 60Hz, even
using a borrowed shure KSM141 stereo pair in omnidirectional gives vague
results ($3000 for the mic's!) I have come to the conclusion that you are
really measuring air displacement at anything under 30Hz and a flat panel
with a transducer is the only repeatable method of measuring the output,
how
one calibrates said device is open for discussion.

(Oh one firm told me that they use a laser to measure the low frequency of
their speakers, check this out for novel! They place a small piece of
reflective foil on the base driver, and shine a laser beam on it, they
then
apply signal and measure via 'laser' the deflection, Also they place a
passive radiator 1 meter infront of the driver unit and use the same
method
to measure its deflection.

The apply a 'correction factor' and the produce the frequency response
data
(company builds very expensive car and home audio subs!)

I have yet to find ANY speaker system using an 8 inch driver that has the
ability to produce 40hz or lower frequencys as they all have fallen so far
down in output level as to be useless.

Some 8" studio monitors can produce sub 40hz meaningfully, though not very
far below, and they split the drivers at a crossover. HR824's fall off
rapidly below 39hz but 40hz is definitely present, at least my chest thinks
so. The test tones are definitely there. I can feel and sort of hear
*something* (or is that my headphones on my desk vibrating...).

I can't see a single speaker being very flat or even close at the full range
of 20-20k. There seems to be some fundamental limitations to most
loudspeaker designs that would preclude this.

If you find this holy grail, please post the manufacturer he )

(Oh one firm told me that they use a laser to measure the low frequency of
their speakers, check this out for novel! They place a small piece of
reflective foil on the base driver, and shine a laser beam on it, they
then
apply signal and measure via 'laser' the deflection, Also they place a
passive radiator 1 meter infront of the driver unit and use the same
method
to measure its deflection.

This seems reasonable. If it is vibrating at 20 hz, and measurable, it
produces 20 hz. No one has ever said they have a single speaker with a flat
frequency response from 20-20k, just that the speaker will respond audibly
to this frequency range. This is why they all post 20-20k response, most
without a frequency response curve. It wouldn't be pretty or sell speakers.

"The Flash" wrote in message
...
I have had no luck in measuring speaker response at low frequency's, I can
measure from ~60Hz to 20Khz with near 100% repeatability (How accurate is
questionable)

I have questioned a number of speaker builders and a couple of companies
that 'tune' speakers the answers given on what they measure with and how
the
do the tests give serious rise to the claims at low frequency.

Almost any speaker system sold today claims 20Hz to 20Kz response yet this
is so far from the truth I cannot understand how they dare claim such
figures.




How do you measure the responese at 20Hz?, I have tested a few speakers
and
basically I cannot get accurate (repeatable) data at much below 60Hz, even
using a borrowed shure KSM141 stereo pair in omnidirectional gives vague
results ($3000 for the mic's!) I have come to the conclusion that you are
really measuring air displacement at anything under 30Hz and a flat panel
with a transducer is the only repeatable method of measuring the output,
how
one calibrates said device is open for discussion.

(Oh one firm told me that they use a laser to measure the low frequency of
their speakers, check this out for novel! They place a small piece of
reflective foil on the base driver, and shine a laser beam on it, they
then
apply signal and measure via 'laser' the deflection, Also they place a
passive radiator 1 meter infront of the driver unit and use the same
method
to measure its deflection.

The apply a 'correction factor' and the produce the frequency response
data
(company builds very expensive car and home audio subs!)

Rusty Boudreaux wrote:

I just downloaded the pdf. The stated response is 15Hz-20kHz but
no dB qualifier. The magnitude plot shows essentially flat down
to 60Hz but no data below. However, they do show response to
30kHz...down 10dB.

My incomplete understanding is that there is no reason why that type of
microphone should roll off early other than series capacitance.

http://www.behringer-download.de/ECM...00_C_Specs.pdf

Not checked.


Kind regards

Peter Larsen

--
************************************************** ***********
* My site is at: http://www.muyiovatki.dk *
************************************************** ***********

Rusty Boudreaux wrote:

I just downloaded the pdf. The stated response is 15Hz-20kHz but
no dB qualifier. The magnitude plot shows essentially flat down
to 60Hz but no data below. However, they do show response to
30kHz...down 10dB.

My incomplete understanding is that there is no reason why that type of
microphone should roll off early other than series capacitance.

http://www.behringer-download.de/ECM...00_C_Specs.pdf

Not checked.


Kind regards

Peter Larsen

--
************************************************** ***********
* My site is at: http://www.muyiovatki.dk *
************************************************** ***********

So-called (actually, mis-named) 'isobarik' systems are a way of
obtaining a given response in half the volume, since the arrangement
results in half the total equivalent compliance volume and thus
requires half the enclosure volume for the same response function.

My Isobric has been constructed to improve LF ouput (it seems) not reduces
size as i can tell as it has a ~12 cubic foot cubed primary with a ~ 6 cubic
secondary, I guess that they have decided that the cone mass of an 8 inch
drive in isobaric has advantages over that of an 12 inch driver (at the size
of the brute it could easily use a 12 inch drive, one thing did occur to me
is that perhaps due to its age that duel VC speakers were not avalible and
the only method of getting a phase and anti phase driver easily was to use
two drivers.

As to "true hifi response," such a term needs definition to be
anything but meaningless jargon, in precisely the same way that
a "specification" like "20Hz-20kHz" is meaningless without
qualification.

I guess what I desire is a reasonably flat response fro 32 Hz to 15Khz with
gradual roll off above and below. If that is 'hi fi' or not is questionable.
I spent some time years ago analysing CD's to find usual and lowest
frequency's (plus HF, lots of CD's have significant output above 15Khz
(components up to 25Khz+ can be present) but at my age thats starting to get
above what I can easily here!) LF it seems stops round the ~30 Hz, few
instraments produce much less than this (yes I know that alot can) a lot of
the very LF on disk is artificial (1/2 frequency echo and its that it is
possibly added to give 'space' to the recordings) unless you are a pipe
organ freak under 30Hz seems to unused (but not in Dolby Prologic, ES or
such as the LF in these type video recordings is huge it seems (and
unnatural in my books) )

THe fundamental efficiency/enclosure volume/cutoff frequency equation
will ALWAYS rule. But, interestingly enough, driver diameter simply
does not enter into that relation. Specifically, the relation:

n0 = kn Vb F3^3

where n0 is reference efficiency, Vb is enclosure volume, F3 is is
low frequency cutoff and kn is the efficiency constant has NO term
in it in any way depedent upon driver diameter. This directly refutes
your assertion that it's not possible to get your "unqualified" 'hifi
response from an 8 inch driver. One needs to simply balance the three
terms of efficiency, enclosure volume and cutoff frequency and you're
there. You seem to intimate that low efficiency is incompatible with
'hifi response," for example, an unjustifiable viewpoint in light
of the lack of qualification of 'hifi response.'

kn = efficiency constant derived from driver (this figure will improve with
driver diameter, thus a large driver will allow a a small enclosue to
produce the same output at the same frequency with less power - correct me
if wrong)

However in order to get usable (at say 32Hz) levels the enclosure volume
(8 inch) will become so large as to be impactical in a lot of cases. This
equates to the 'You can put an airplane engine in a submarine' but who wants
too?

An enclose that is over 32 cubic feet will meet significant resistance when
placed in the home! (Test this by asking you wife if you can shift 2 small
fridges into the lounge!)

I tend to deal with and try and work in what can be achieved reasonably
easily in real world situations, absolutes and what is theoretically
possible don't fit! (But are well worth knowing and considering (Hey I own a
leak stereo 30 once discribed as all the power a home stereo would ever need
and enough th have your neighbours complaining! This we all know not to be
true!))

As a side issue I see that companies are producing very large (24 inch
diametre) woofers now. how well do these work and are they worth the
premiums being asked for them?