The theory of the punctiform acoustic source.

I think momentarily over the theory of the punctiform acoustic source.

Since each box has a periodical phase shift, also the virtual origin of the source changes with the frequency. One has then the impression that with deep frequencies the acoustic source is further in front in the area.

Thus the signal would have to be corrected. My considerations go by that that you have to split the signal into partial windows and with a fractionale fourier transformation the phase position could be corrected for each step separately. This would have to be a kind of Signalprocessor or a PC which is quick enough to split the signal in smallest possible parts. What do you think about? Are there already such devices?

Greetings
Ralf

Hi Ralf,

Yes, this has been done...certainly in gross steps,
such as "time aligning" drivers physically, using
delay lines to simulate a point source (ala Quad),
and digital delay lines with active crossovers.
Higher-end pro audio controllers (usually for
large venues) can have the capability for a more
contiguous phase correction vs. frequency. Not
a hard job for a DSP once so programmed.

Even the "gross" phase corrections (large or per
driver steps) can help significantly...but perhaps
this is not your question...

-grumpy

Quote:

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Originally posted by grumpy
Hi Ralf,

Even the "gross" phase corrections (large or per
driver steps) can help significantly...but perhaps
this is not your question...

-grumpy

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Hi grumpy,

yes, that is, what I meant.

One must know, as large the phase shift is with a fixed frequency, in order to adjust these. I imagine in such a way that over a reference measurement over the entire frequency range one seizes, which shift characteristic has the system. Thus the Algorhytmen can be determined for the individual windows. Thus an approximate phase shift of 0 degrees would have to be obtained to be able...
Since at the time of the phase correction in the same instant already is not foreseeable, as the signal looks, however no real time correction can separate via a Delay one time near correction to only take place. In relation to the original signal delay would have to be accepted as it were. Does the DSP`s, which you know work in this procedure?
Can anybody follow me?

:confused: :rolleyes: :)

Quote:

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Originally posted by Ralf

Can anybody follow me?

:confused: :rolleyes: :)

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:hmm: :hmm: Difficult, really difficult.....

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Originally posted by Guido
:hmm: :hmm: Difficult, really difficult.....

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Do you mean with "difficult" that it is to be understood with difficulty or difficulty to realize?

might have a look at:
http://www.diyaudio.com/forums/showthread/t-7552.html

or closer to home, the klein-hummel web page
(PRO C 28) for a pro example.

regards,

-grumpy

Quote:

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Originally posted by Ralf
Do you mean with "difficult" that it is to be understood with difficulty or difficulty to realize?

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Let's discuss this on thursday with a beer.

Quote:

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Originally posted by grumpy
might have a look at:
http://www.diyaudio.com/forums/showthread/t-7552.html

or closer to home, the klein-hummel web page
(PRO C 28) for a pro example.

regards,

-grumpy

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Hi grumpy,

thanks for the infos.
The PRO C 28 has the functions I imagine, but the price of 3950,-- Euros is hard...

I saw a software solution, which is mentioned on the diyaudio board: BruteFIR for Linux. Linux is my preferential operating system. I write server applications and software thereby. Therefore the solution assures to me much.

With a somewhat better sound card and a PC (1-2 GHz) with sufficient main memory (0.5-1Gb) the same result might to be produced be able. However the programming of the FIR filters needs a little bit time. But with the PRO C 28 it is the same case. The possibility of a reference measurement with automatic generation of filters is missing to both solutions. I will try that out at opportunity and will report then details.

Greetings
Ralf

Here I summarized again, which advantages among other things a DSP (digital signal processing) can bring:

- phase correction for bass reflex systems
- horn correction
- phaseindependent correction of the amplitute
- digital frequency switch
- equalizing
- room resonance correction
- and so on

These advantages are not realizable with similar devices. If one has its filters in the grasp, the result would have to be overwhelming. Theoretically thereby also a wrongly computed box (e.g. too small) would have to be so compensatable that the hearing result and the frequency characteristic suggest other dimensions...

Possibly such equipment is also a nice additive for the Lansing Heritage Project...

What you think about? Is that worth a few considerations? Or is digital signal processing for a fan of old technology rather a thorn in the eye?

:confused: :confused: :confused:

while you're on the linux/FIR kick, you may
want to check out DRC

http://freshmeat.net/projects/drc/?topic_id=114

which might give you a leg up on generating FIR parameters in an automated fashion.

cheers :cheers:

-grumpy

Quote:

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Originally posted by Ralf

What you think about? Is that worth a few considerations? Or is digital signal processing for a fan of old technology rather a thorn in the eye?

:confused: :confused: :confused:

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Personally I do not like this kind of wizarding with signals (as you may know).
Keep the software guys out of my studio monitor madness!!

As said, let's discuss this on thursday;)

Quote:

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Originally posted by grumpy
while you're on the linux/FIR kick, you may
want to check out DRC

http://freshmeat.net/projects/drc/?topic_id=114

which might give you a leg up on generating FIR parameters in an automated fashion.

cheers :cheers:

-grumpy

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Thanks a lot, grumpy

You know the necessary programs. Do you have already experiences thereby?

If yes, which sound card you recommend? The author of bruteFIR suggests the RME Audio Hammerfall (RME9652 and RME9636). This does not give it however any longer.