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Multi-Way Conventional loudspeakers with crossovers

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Old 14th September 2007, 08:32 PM   #2021
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Default mige0

The residential integration firm I work for only uses it in home theater installs... and find it to be of high quality and extremely flexible.

I do not know how it would measure up in a nice 2 channel scenario.

C
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Old 14th September 2007, 09:39 PM   #2022
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Quick question for John K. On the NaO with the rear mounted tweeter, was it wired in phase(bipole) or out of phase(dipole) to match the rear output of your mids? And did did you pad down the tweeters output compared to the front? Any other thoughts on the matter would be appreciated, I'm working on a similar design idea with my first open back speaker. Thanks.
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Old 14th September 2007, 10:07 PM   #2023
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Quote:
Originally posted by mige0
---------------------------

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



The EQ I used for the drivers in the response shaping was done using SoundEasy's digital equalizer. It works by dividing the target response curve by the measured driver response to find the transfer function for the filter. Then the PC processes the whole thing for play back. With the correct sound card up to a stereo 5-way system digitally.
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Old 14th September 2007, 10:11 PM   #2024
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Quote:
Originally posted by Caferacer
Quick question for John K. On the NaO with the rear mounted tweeter, was it wired in phase(bipole) or out of phase(dipole) to match the rear output of your mids? And did did you pad down the tweeters output compared to the front? Any other thoughts on the matter would be appreciated, I'm working on a similar design idea with my first open back speaker. Thanks.

I do connect it out of phase with the front and I have the option to switch it on and off. It's not really padded down. Also, it doesn't make a big difference if it's in or out of phase except right around the crossover point. The separation between the front and rear tweeter is such that the front are rear tweeters are uncorrelated which means the operate more like two independent sources.
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Old 14th September 2007, 10:18 PM   #2025
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Default Minimum phase???

Quote:
Originally posted by mige0
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 .
Michael,

I have created a (VERY) short introduction to the difference between minimum and non-minimum phase systems. Please have a look at my five-minute introduction to the minimum-phase concept for a two-way speaker.

Best Regards,
Edward
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Old 14th September 2007, 10:59 PM   #2026
mige0 is offline mige0  Austria
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Default Re: Minimum phase???

Hi


Quote:
Originally posted by EdwardWest


Michael,

I have created a (VERY) short introduction to the difference between minimum and non-minimum phase systems. Please have a look at my five-minute introduction to the minimum-phase concept for a two-way speaker.

Best Regards,
Edward

Edward, thanks a lot for your extra work. I'll try to digest it the right way.

I alway tried to visualise " minimum phase " with somthing like " time coherent ". Though I was close its obviously not exactly what describes it completely.

Greetings
Michael
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Old 14th September 2007, 11:13 PM   #2027
mige0 is offline mige0  Austria
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Hi

Quote:
Originally posted by john k...



The EQ I used for the drivers in the response shaping was done using SoundEasy's digital equalizer. It works by dividing the target response curve by the measured driver response to find the transfer function for the filter. Then the PC processes the whole thing for play back. With the correct sound card up to a stereo 5-way system digitally.

JohnK, would you say that a correction of a Q=20 peak is possible that way ( I am aware but don't care of the limitations sound wise in this case ) ?

Did you have to process the wav file by " SoundEasy " and than play it or is it a convolution file that is computed and then can be used with a plugin like SIR_1011 ?
IIRC you mentioned somewhere at the start of this giant thread that you also use a computer based crossover. Is it somthing like this

http://www.thuneau.com/allocator.htm

?


Greetings
Michael
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Old 14th September 2007, 11:14 PM   #2028
mige0 is offline mige0  Austria
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Hi

congratulations Lynn for jumping over the 2k mark stone
( and not loosing track though we all do our best )

Greetings
Michael
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Old 15th September 2007, 02:05 AM   #2029
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Quote:
Originally posted by mige0
Hi




JohnK, would you say that a correction of a Q=20 peak is possible that way ( I am aware but don't care of the limitations sound wise in this case ) ?

Did you have to process the wav file by " SoundEasy " and than play it or is it a convolution file that is computed and then can be used with a plugin like SIR_1011 ?
IIRC you mentioned somewhere at the start of this giant thread that you also use a computer based crossover. Is it somthing like this

http://www.thuneau.com/allocator.htm

?


Greetings
Michael
SoundEasy does it all. I doesn't create wave files. It generates the required transfer function and emulates then digitally using FIR convolution. Please don't confuse FIR convolution with linear phase crossovers as is commonly done by many. FIR can do any type of causal filter, linear phase, minimum phase, arbitrary phase.
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Old 15th September 2007, 05:29 AM   #2030
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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.

I bet I'm not the only one trolling this thread that does not fully appreciate the logic of what you are saying. In an electrical system, I can create any arbitrary average DC value I like with a half sine, saw tooth, or whatever. Why would it be impossible to to reflect this acoustically, assuming the room was sealed? Surely the air in the room is compressed for duration t of the impulse, and to amplitude A You can't really do this perfectly, but if the impulse causes the cone to move from rest to position +A, and back to rest, rather than to -A before returning to rest, why would the average area under the curve be zero? I appreciate that bandwidth limitations on the top end will result in undershoot, and on the low end prevent compresion of the air for any usable time, but at the end of the day, didn't we really compress the air in the room briefly? For a continuous function, what you say makes sense, but an impulse is not a continuous function. I am missing something.

Dick
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