Yes.
Hi,
Maybe with this kind of correction:
using phase EQ (range +-180°)
something like this,or several narrow band,according to the measurement (and the frequency of the null).
I'm trying to do like a shelf filter for phase, something that'll increase phase shift by a constant amount based on the frequency. Using a parametric EQ to simulate a shelf filter shape isn't exactly ideal. Is there a way to do a shelf filter shape phase adjustment in rePhase?
Suggest try start on "Paragraphic Gain EQ" tab and make those shelf filters data you think about which will create IIR filter phase shift, now go to "Paragraphic Phase EQ" tab and use whatever it takes to flatten phase shift again and maybe it helps using example three banks or more where one is set Q=1 next is Q=2 on then Q=4, in the end when phase is corrected flat then disable those IIR shelf filters and you end up have the phase shift only. Into these two tabs there is also convenient help commands in the "Tools" and "Preset" buttons.
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Hello
What you need in such a situation is simply delay and polarity switch
That's a really smart idea. I'm gonna try that.
Wouldn't that turn my speaker into a cardioid in the woofer range?
Yes, woofers are in the same box.
Oh you're right, my bad. I forgot how speakers work lol.
I can't do a delay + polarity because the 2 woofers share the same airspace.
Hi bcodemz, i tend to think very simple and very basic, pls don't mistake my questions for anything other than trying to understand at that level
Don't all drivers share the same airspace? I'm just having a hard time seeing exactly what you're trying to do ...sorry..
You are right, forget what I said, I overlooked the "woofer" part
If those woofers are limited to LF then I suppose the path difference between the two will not matter and you can just keep them in phase (assuming 360° radiation) with no delay and no specific correction.
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Is it a version of the general problem with a tube and a woffer in the end?
If it is a inside wall in the end of the tube, first cancellation, -180 degrees, is where the wave have traveled half the wavelength.
That is tubelength•2 = wavelength/2
With a woffer in each end the same cancellation apply as the woffers act as tha inside wall.
In adittion we have a cancellation where
Tubelength = wavelength/2
As the woffers cancelling each other out, -180 degrees.
Then the question: Can FIR filters on the drivers be such that these (first) cansellations are negated. Then the tube sub can be used at higher frequencies.
But I don't know if the filter exits? Is is just a 1 order allpass phase shelving filter?
Guess only the largerst wavelength cancellation can negated?
If it is a inside wall in the end of the tube, first cancellation, -180 degrees, is where the wave have traveled half the wavelength.
That is tubelength•2 = wavelength/2
With a woffer in each end the same cancellation apply as the woffers act as tha inside wall.
In adittion we have a cancellation where
Tubelength = wavelength/2
As the woffers cancelling each other out, -180 degrees.
Then the question: Can FIR filters on the drivers be such that these (first) cansellations are negated. Then the tube sub can be used at higher frequencies.
But I don't know if the filter exits? Is is just a 1 order allpass phase shelving filter?
Guess only the largerst wavelength cancellation can negated?
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Does rePhase correct major acoustic offset between drivers on a downstream crossover network?
For example, I have in mind a project design which includes the following:
miniDSP 2x4HD ->
CH1: feeds left channel HF passive crossover (tweet + midrange)
CH2: feeds left channel LF active crossover (midwoofer + LFE)
CH3: feeds right channel HF passive crossover (tweet + midrange)
CH4: feeds right channel LF active crossover (midwoofer + LFE)
This is a nearfield 2.1 setup, with the mono sub located centrally underneath a desk. Each satellite would have 3 drivers, consisting of a tweeter, midrange, and midwoofer. The LF active crossovers would both split their respective inputs into midwoofer + LFE, and the LFE would be summed into a mono signal. Altogether this is a total of 7 unique passbands, with 5 amp channels required.
So going back to the original question - would the acoustic offset between the midwoofers and the mono sub be eliminated so that everything is linear phase from top to bottom?
For example, I have in mind a project design which includes the following:
miniDSP 2x4HD ->
CH1: feeds left channel HF passive crossover (tweet + midrange)
CH2: feeds left channel LF active crossover (midwoofer + LFE)
CH3: feeds right channel HF passive crossover (tweet + midrange)
CH4: feeds right channel LF active crossover (midwoofer + LFE)
This is a nearfield 2.1 setup, with the mono sub located centrally underneath a desk. Each satellite would have 3 drivers, consisting of a tweeter, midrange, and midwoofer. The LF active crossovers would both split their respective inputs into midwoofer + LFE, and the LFE would be summed into a mono signal. Altogether this is a total of 7 unique passbands, with 5 amp channels required.
So going back to the original question - would the acoustic offset between the midwoofers and the mono sub be eliminated so that everything is linear phase from top to bottom?
OK what about this kind of FIR filter? I want a variable delay that is directly proportional to the frequency.
For example, I want a 1ms delay at 100Hz, 0.5ms delay at 200Hz, 0.25ms delay at 400Hz, etc and everything in between, so the delay increases proportionally for every frequency within a specified band, say 100-400Hz.
I'm trying to do a perfect cardioid response for every frequency as specified in Patrick Bateman's beer budget beam steering thread.
For example, I want a 1ms delay at 100Hz, 0.5ms delay at 200Hz, 0.25ms delay at 400Hz, etc and everything in between, so the delay increases proportionally for every frequency within a specified band, say 100-400Hz.
I'm trying to do a perfect cardioid response for every frequency as specified in Patrick Bateman's beer budget beam steering thread.
Have a read of this link and see if it makes any sense to you https://dsp.stackexchange.com/questions/32155/how-to-design-a-digital-filter-with-a-frequency-dependent-phase-delay
The example is for very high frequencies but I suppose the theory is the same. You may need to use another program to create your filter coefficients.
The example is for very high frequencies but I suppose the theory is the same. You may need to use another program to create your filter coefficients.
Time delay:
*Different FIR length create different time delay, when filters are not skewed.
*Skew the filter. (That may affect response)
*Time delay outside the FIR filter/rePhase
*An all pass filter can give you a time delay less than the sampling interval.
Guess the two first options are the best for mid bass.
(Of course the room will dictate the phase response other than one plase in space.)