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Old 14th September 2007, 02:01 PM   #2011
gedlee is offline gedlee  United States
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Quote:
Originally posted by john k...



And is this unedited 3-way system, on axis impulse response impossible too? Sorry that it is inverted. My mic inverts. (~60k/sec sampling rate, first 3 msec).

Click the image to open in full size.
John

An impulse response from an acoustic source must average to zero as there can be no DC response to any acoustic signal. What you have shown is indeed in error in that clearly the LF data is missing, thus somehow the missing data must be "assumed" and clearly here it was "assumed" to be the same as the passband - which is impossible. I am not saying that it is not possible to make a bad measurement that does not average to zero,but I am saying that in all of my work I test the DC to make sure that it is well below the passband or I don't trust the data. You should do likewise.

You will never see me show an acoustic impulse response that does not average to zero.
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Old 14th September 2007, 02:06 PM   #2012
soongsc is offline soongsc  Taiwan
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Quote:
Originally posted by gedlee


John

An impulse response from an acoustic source must average to zero as there can be no DC response to any acoustic signal. What you have shown is indeed in error in that clearly the LF data is missing, thus somehow the missing data must be "assumed" and clearly here it was "assumed" to be the same as the passband - which is impossible. I am not saying that it is not possible to make a bad measurement that does not average to zero,but I am saying that in all of my work I test the DC to make sure that it is well below the passband or I don't trust the data. You should do likewise.

You will never see me show an acoustic impulse response that does not average to zero.
DC offset in the mic preamp and input circuit? the last time I did a calculation, it seemed almost unavoidable to have some DC in the signal path, which should show up in measurement data.
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Old 14th September 2007, 04:41 PM   #2013
mige0 is offline mige0  Austria
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Hi

Quote:
Originally posted by soongsc

Well thinking about direct sound, at the first cone mode, I'm visualizing in the VC moving first, part of surface close to where VC and Cone connect move a bit later, then finally the outer edge of the cone follows. Throughout this first 1/4 cycle is the onset, if compensation prevents this part from being faithfully reproduced, then the sound will not be as good as a more well behaved cone. I won't be upset if I cannot let you understand this part.

Soongsc, don't make a common mistake.
EVRY TIME the air is moved by the speaker there is this same time delay between the current in the voice coil and the air getting pressurised. This is simply due to the fact that every material has its maximal speed of information transport ( sound speed here in the voice coil and the membrane not to mention that the membrane has to build up speed before pressure is created).
This is independent of resonance or not.
In case of equalisation you compensate phase to force the membrane to be exactly in time.

Its just like compensating a certain electrical filter ( high Q peak ) with its mirror function ( high Q notch ).
The main question that is left is how linear the driver really acts in its resonant frequency - as JohnK said.



---------------------------

Again any suggestions for EQ units that can compensate more accurate than a DCX ?
EQ the peak of the driver like shown makes a big difference. I would like to investigate that further.
As it seems even a minor notch of 3 dB makes tweaking necessary to keep a good balance.


Greetings
Michael

By the way is it only me that gets all the " the " in this thread displayed in beautiful red ?
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Old 14th September 2007, 05:00 PM   #2014
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Default Symetrix or Ashly?

What about these two options for DSP.

C
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Old 14th September 2007, 05:19 PM   #2015
mige0 is offline mige0  Austria
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Default DSP EQ with high Q and fine F stepping

Hi

Quote:
Originally posted by chrismercurio
What about these two options for DSP.

C

Chrismercurio,


The Ashly Protea 3.24 don't have finer steps than the DCX ( 80 Hz compared to 60 Hz at 2 kHz ).
Symetix is more in zone mixers no ?


Greetings
Michael

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Old 14th September 2007, 05:48 PM   #2016
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Default I'm checking on the model

we use for Theater installations on Symetrix. In the meantime...there is some interesting software being used over here.

http://www.diyaudio.com/forums/showt...30#post1301530

C
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Old 14th September 2007, 05:58 PM   #2017
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Quote:
Originally posted by gedlee


John

An impulse response from an acoustic source must average to zero as there can be no DC response to any acoustic signal. What you have shown is indeed in error in that clearly the LF data is missing, thus somehow the missing data must be "assumed" and clearly here it was "assumed" to be the same as the passband - which is impossible. I am not saying that it is not possible to make a bad measurement that does not average to zero,but I am saying that in all of my work I test the DC to make sure that it is well below the passband or I don't trust the data. You should do likewise.

You will never see me show an acoustic impulse response that does not average to zero.
Earl,

What leads you to the conclusion that the impulse responses which John and I posted do not average to zero?

As you know a normalized impulse response for a perfect system - flat from DC to daylight - is a single spike of magnitude 1. If the system is low-pass, then the spike is lower magnitude, dull and has a small finite width. If the system is high-pass it will have a sharp spike with a magnitude of nearly 1, and then a small undershoot with a slow decay back to zero. If the low-frequency cut off is at 20 Hz, then the undershoot and tail will be very low magnitude and very long - on the order of 10's of milliseconds. Given the very small magnitude and extended length of the tail due to the low-frequency cut off, one does not 'notice' it in the presence of the hash of the noise floor.

Of couse, this whole description is predicated on the assumption that the pass-band phase response is minimum phase. If you have a conventional 3-way speaker system with two crossover transitions between 20 Hz and 20 KHz, then the acoustic sum is not minimum phase.

The response will be like this:

Click the image to open in full size.

But if the same system has a minimum phase acoustic sum, then one gets this:

Click the image to open in full size.

Perhaps, there is a misunderstanding due to the common lack of familiarity with minimum phase acoustic sum systems.

Edward
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Old 14th September 2007, 07:29 PM   #2018
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Default Symetrix Symnet 8x8 DSP

http://www.symnetaudio.com/index.php...ow1=&Show2=188
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Old 14th September 2007, 07:33 PM   #2019
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Quote:
Originally posted by gedlee


John

An impulse response from an acoustic source must average to zero as there can be no DC response to any acoustic signal. What you have shown is indeed in error in that clearly the LF data is missing, thus somehow the missing data must be "assumed" and clearly here it was "assumed" to be the same as the passband - which is impossible. I am not saying that it is not possible to make a bad measurement that does not average to zero,but I am saying that in all of my work I test the DC to make sure that it is well below the passband or I don't trust the data. You should do likewise.

You will never see me show an acoustic impulse response that does not average to zero.
Earl,

We when through this before. This system has a 25 Hz Q=0.5 high pass cut off (woofer). The statement is correct that the signal must average to zero, but how? Relatively speaking the impulse has a very short, high amplitude positive spike. The area is approximately H x dt where H is the amplitude and dt is the width of the pulse (one sample here, ~0.0167msec) The negative part of the pulse, by comparison, has a very low amplitude, very long duration decay to zero from below for this system. While there is a little ringing in the impulse following the initial spike due to the nature of the system's low pass characteristic (tweeter cut off at about 25k Hz), the low frequency, negative swing is so low in amplitude that it's not visible on the vertical scale and if the time axis were extended room reflections would contaminate the result further. In fact, at the level the system was tested, the negative part of the impulse might even be lost in the noise floor of the measurement environment. That there is no visible negative swing or oscillation around zero is because there is not time/transient distortion introduced by the woofer/midrange or midrange/tweeter crossovers. These, since they occur at much higher frequencies relative the high pass corner of the present system, are typical much more visible in the impulse. May I suggest you look at the impulse response for a system with a band width defined by a 2nd order, 25Hz Q=0.5 high pass filter cascaded with a 25k Hz Q= 0.7, 2nd order low pass filter. You will see the initial part of the impulse looks very similar to what I have posted with the magnitude of the negative, after the initial tweeter ringing, on the order of 0.5% of the magnitude of the initial positive pulse. Average to zero? Yes. Necessarilly visible in the impulse response,? No.
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Old 14th September 2007, 07:46 PM   #2020
mige0 is offline mige0  Austria
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Default DSP EQ with high Q and fine F stepping

Hi

Quote:
Originally posted by EdwardWest



Perhaps, there is a misunderstanding due to the common lack of familiarity with minimum phase acoustic sum systems.

Edward

Edward, At least for me its hard to decipher that " minimum phase " thing not necessarily a problem of translation only -. would be great if you give a short survey .



-----------------------

chrismercurio, thanks for the link which lead me to

http://www.acourate.com/

something I will check out in more detail. If it were not a computer solution with all its drawbacks ( acoustic and electric noise and handling ) it seemed to be what I was looking for.

Sure worth a try anyway !

Greetings
Michael
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