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Multi-Way Conventional loudspeakers with crossovers

rePhase, a loudspeaker phase linearization, EQ and FIR filtering tool
rePhase, a loudspeaker phase linearization, EQ and FIR filtering tool
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Old 15th November 2019, 09:30 PM   #2841
emailtim is offline emailtim  United States
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Hi BYRTT,

Quote:
Originally Posted by BYRTT View Post
For main quistion will say that is normal and expected because when we manipulate some typical excess phase lag in time domain it has to be a pre operation view into impulse or step response graphs that would cost some overall processing systen lag to repair.

In general looks you have a good feel on stuff

As some general tip for where you are now say everything is really perfect based, then remember than any wish for other house curve adjustment or left verse right channel calibrations has to be global adjustments. What i mean is stay away any house curve adjustments per selective band pass unlesh you really happen find some errors there, because even small EQ changes per pass band will often or probably need a new time allignment setting and that operation can often be a big workload.
Thank you for the feedback. Much appreciated. I am sure this isn't my last pass, I am trying to get an 8 channel DAC to replace my current gear.


======================== %< snip >% ========================


Hi Fluid,

Quote:
Originally Posted by fluid View Post
Those graphs look good Remember to listen to the correction and see if you like it, I have made many graphs that look great and sound terrible.
It sounds quite good. I previously tried DRC Designer and would get good and bad sounding results most assuredly due to the pilot's ignorance use of the tool.

Quote:
Originally Posted by fluid View Post
So you have an active crossover between dipole subs and main speakers with an unknown passive crossover?
I have stereo OB/Dipole subs and 3-way planar/ribbon speakers (8 channels) and a stereo 3-way active XO (6 channels) and 2 passive unknown internal XOs.
There is an active known 40Hz XO between the sub and bass.
There is an active known 250Hz XO between the sub and (mid/tweet) combo.
There is a passive internal XO between the (mid/tweet) combo that the factory suggests is @ 3,000Hz, but doesn't elaborate on slopes. That is the unknown that I approximated by braille.

I currently have no way of adding delays with the current XOs. I am waiting on an 8 channel DAC and then I can time align the drivers. I will then need to know what to use in REW to gauge delays. I see REW has an "estimate IR delay" function, a "time align" function and a "IR align" function and need to learn the differences. I can also switch to linear XO's at that point.

Quote:
Originally Posted by fluid View Post
As Byrrt said the change in the step response before the peak and the increased preringing in the impulse is the result of the phase manipulation. You have gone from 1700 degrees of phase turn to 360 degrees 20-20k so something has to give.
If I get the 8 channel DAC and switch to linear XO's, there should be less phase issues to correct if I understand them correctly.

Quote:
Originally Posted by fluid View Post
Are these in room averaged listening positions measurements like in SwissBears tutorial?
Not yet, I haven't moved the mic. I wanted to verify [before/after], [measured baseline/predicted results/measured results] without adding any additional variables into the equation while learning the software. It is hard to reliably repeat 9 different measurements at different X,Y,Z locations.

I have quite a bit of absorption and diffusion in the room and the current filters sound good at the sweet spot as well walking around the room.

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Originally Posted by fluid View Post
If so a flat room response will tend to be too bright. You could try adding in a room curve in REW to see how you like it.
I personally prefer a flat "house curve". If my ears were 30 years younger, that maybe another story.

Last edited by emailtim; 15th November 2019 at 09:45 PM.
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Old 15th November 2019, 10:28 PM   #2842
fluid is offline fluid  Australia
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Quote:
Originally Posted by emailtim View Post


Hi Fluid,



It sounds quite good. I previously tried DRC Designer and would get good and bad sounding results most assuredly due to the pilot's ignorance use of the tool.
Sounding good is the aim so that's good news

DRC Fir is very easy to get wrong and tweaking the parameters is not easy.

Give Gmad's method and filters a try, I find this to be the easiest way to use DRC

A convolution based alternative to electrical loudspeaker correction networks

Quote:
I currently have no way of adding delays with the current XOs. I am waiting on an 8 channel DAC and then I can time align the drivers. I will then need to know what to use in REW to gauge delays. I see REW has an "estimate IR delay" function, a "time align" function and a "IR align" function and need to learn the differences. I can also switch to linear XO's at that point.
You would be better to use the acoustic timing reference in REW

This is a page from minidsp, explaining the use of it, just ignore the minidsp stuff that doesn't apply

Measuring Time Delay

Quote:
If I get the 8 channel DAC and switch to linear XO's, there should be less phase issues to correct if I understand them correctly.
If you use a linear phase crossover properly time aligned with drivers EQ'd flat either side there should be nothing to correct. You will still get the same wobble in the step and IR.

I would suggest making a version of the correction without the phase linearization (just bypass those parts in Rephase and generate another filter) but keeping the same EQ from REW and see if you can tell the difference between them and then whether you actually have a preference either way.

This is one area where the graph will look significantly better with the correction but the sound may not reflect what you are seeing. Or it might


Quote:
Not yet, I haven't moved the mic. I wanted to verify [before/after], [measured baseline/predicted results/measured results] without adding any additional variables into the equation while learning the software. It is hard to reliably repeat 9 different measurements at different X,Y,Z locations.
You don't have to get the mic in exactly the same spot as the averaging works to remove a lot of the positional differences. Try a 'moving head' average by taking 4 to 8 measurements within the space your head would occupy and use that as a base for correction.

Quote:
I have quite a bit of absorption and diffusion in the room and the current filters sound good at the sweet spot as well walking around the room.


I personally prefer a flat "house curve". If my ears were 30 years younger, that maybe another story.
Maybe the combination of treatment and directivity make that better, ultimately it's your preference that counts
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Old 16th November 2019, 03:12 PM   #2843
mark100 is offline mark100  United States
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rePhase, a loudspeaker phase linearization, EQ and FIR filtering tool
Quote:
Originally Posted by BYRTT View Post
Maybe because we dont want to improve the group delay or timing isolated on its own is because the physical amplitude signal really isnt there and we then end break physics for timing in natural sounds, know its too simplistic but take below theoretical example from a recording chain, red curve is raw bandwidth of a acoustic bass guitar with 1st order stop bands at 41Hz and 7kHz, and if we feed or say cascade that bandwidth thru a very good microphone bandwidth with 1st order stop bands at 6Hz and 20kHz we end up a bit limited domain in the blue curve. Amplitude and timing (phase) for that new blue curve is what the other musicians plus mix and master process will base their cooperation or work on and therefor will guess if we change blue curves original blue phase to be the red phase we add a timing distortion because the wider amplitude performance of red curve we never get back because of the cascaded chain.
Hi BYRTT, been thinking about this.
But i keep coming back to .....i dunno....


Do low frequency natural sounds necessarily have any low freq rolloff, and hence group delay ?
Like a single note from a pipe organ or the lowest piano key?
I don't think so...could easily be wrong though.

If limited bandwidth sounds in nature don't necessarily have phase rolloff on the ends on their bandwidth, I think linear phase is the only true reproduction method...all the way down as low as possible.

If limited bandwidth sounds in nature do have phase rolloff, my mind is getting ready to be expanded, and i'd love the clarity that understanding would bring.

Now none of the above is to say that recordings will sound better fully lin phase...I can easily see how recording and mastering techniques/practices can be tuned to make playback sound better fully min phase.
But i think when we prefer (rightfully) min phase playback, we are just matching crooked sounding to a crooked made recording.
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Old 16th November 2019, 11:52 PM   #2844
fluid is offline fluid  Australia
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Quote:
Originally Posted by mark100 View Post
Hi BYRTT, been thinking about this.
But i keep coming back to .....i dunno....


Do low frequency natural sounds necessarily have any low freq rolloff, and hence group delay ?
Like a single note from a pipe organ or the lowest piano key?
I don't think so...could easily be wrong though.

If limited bandwidth sounds in nature don't necessarily have phase rolloff on the ends on their bandwidth, I think linear phase is the only true reproduction method...all the way down as low as possible.

If limited bandwidth sounds in nature do have phase rolloff, my mind is getting ready to be expanded, and i'd love the clarity that understanding would bring.

Now none of the above is to say that recordings will sound better fully lin phase...I can easily see how recording and mastering techniques/practices can be tuned to make playback sound better fully min phase.
But i think when we prefer (rightfully) min phase playback, we are just matching crooked sounding to a crooked made recording.
I think you could go round and round making arguments either way and never get to the "right" answer

I can't think of a low frequency musical instrument without a resonating chamber of some sort, pipe, sound box etc. All of those will behave as minimum phase devices and have rolloff's associated with that. Which will change phase and introduce group delay.

How can a natural sound have unlimited bandwidth and the phase not follow the amplitude?

If you make a speaker with a flat amplitude response with no rolloff of any kind until outside of the 20-20K region then you have a flat phase throughout the realistic range of hearing. If it rolls off minimum phase from there on it falls in the range of perception rather than hearing.

It satisfies both minimum phase target and linear in the important regions.

I don't think you can ever get to a consensus on the idea of listening to the same type of system as the engineers did at mixing and mastering unless there is an international standard on it. So you are left deciding for yourself what you think sounds right or best.

It doesn't make sense to me to create a speaker that could not exist without the aid of linear phase processing, to make the phase not follow the amplitude. I prefer the idea of using the correction to return the speaker to the best natural response it could have. Not that this helps answer whether it is right or wrong
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Old 17th November 2019, 12:23 AM   #2845
wesayso is online now wesayso  Netherlands
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I agree with Fluid. If we were able to create the ideal speaker, where one point in space would play our desirable bandwidth, it would still act as a minimum phase source.
As do natural sounds in our universe, so why would we want something different from our (not so perfect) speakers?

I tried to get that minimum phase timing at the listening position... By using enough room treatment and a pair of full range arrays plus EQ/DSP. I've listened to an entirely linear correction as well as phase following the band pass behaviour at that spot. I definitely prefer the latter. Acting as a minimum phase bandpass device.

Some songs just didn't sound right with a complete linear phase correction, a bit pushed, unnatural, at least in my perception. Its both a feel and listen experience and if both those perceptual tools match it just clicks into place.

Do as you wish though, you're free to make that choice for yourself.
I can't think of a single reason why any instrument wouldn't follow minimum phase, except when it has been manipulated.
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Old 17th November 2019, 04:46 PM   #2846
BYRTT is offline BYRTT  Denmark
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rePhase, a loudspeaker phase linearization, EQ and FIR filtering tool
Think you have good points there to support physics fluid and wesayso including everybody should feel free about what settings to use or prefer in phase domain, thanks joining/post about subject : )
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Old 17th November 2019, 04:54 PM   #2847
mark100 is offline mark100  United States
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Hi fluid and wesayso, thanks for comments...good food for thought...

fluid, i think you are spot on with the observation that a speaker with flat amplitude response 20-20K, or rather flat through audibility, satisfies both minimum and linear phase. I think it's more than satisfies...I think in such a case minimum phase equals linear phase.

(This is of course assuming there are no response anomalies throughout the audible spectrum. And I guess it's appropriate to define rolloff questions in terms of low frequency only)

The thing I keep coming to, is linear phase in a speaker does not mean that the recording gets altered from minimum to linear phase. It just means the recording will be reproduced exactly as recorded, doesn't it?

Whatever minimum phase rolloff is in the recording will be reproduced exactly as is by a linear phase speaker system i think.

If we add an additional degree of minimum phase rolloff via our speaker tuning, aren't we doubling up rolloff.
I guess if the recording was mastered listening to monitors that had rolloff, it might sound best if we could match the rolloff of the monitors used. Or less to no rolloff, if mastered on headphones or the new breed of linear phase monitors that are showing up in pro circles.

As far as natural sounds having low freq rolloff...wow, I wish I could wrap my head around what's going on. I've been thinking....how do you even ascertain what the phase response of a natural sound is? What do you compare it too?
I can visualize a natural sound has some combination of frequencies that originate at particular times, but is that timing of origination due to limited bandwidth? Doesn't seem so...seems like it's due to the nature of the origination..and not a lack of bandwidth...does a lion's roar lack bandwidth and hence have group delay rolloff ....I dunno....

Anyway, like you guys said...round and round and never be right
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Old 17th November 2019, 05:45 PM   #2848
Oabeieo is offline Oabeieo  United States
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So if I’m understanding this right

In nature any “sound” is a vibration of something. Something is vibrating to make the sound
Could be anything, at whatever frequencies these things vibrate for them to become louder the amplitude has to increase which means longer duration between cycles which suggests minimum phase

Even in the complex signal like hitting a stick on a object where two things vibrate as a result of an energy transfer, the vibrations are bigger as in more force and more sound is created as the duration between peaks in longer , again minimum phase.

So it makes sense to reproduce such things the system would want to be free from any artifacts of its own in the entire spectrum to faithfully reproduce whatever types of signals are given to it. In a simplistic way of saying it.

In nature it’s not possible to say or claim if something has phase rolloff or not as that’s contrary to what we use as a standard. As you have to look at the ends in which the thing was vibrating.

If a stick was beat across a rock in which is its final end, the stick will vibrate as a result of you hitting it with no scientific analysis done on how the stick was it. It’s end is it stops vibrating and resumes whatever shape is left after and continues it’s life as a stick.


So in regards to natural rolloff of a loudspeaker, it seems to me at least you would want. To try to correct anything that would be heard to have a flat phase and magnitude, if the rolloff is part of what you hear than it too should be made flat.

If the loudspeaker falls short of the spectrum and rolls off early, I think it makes sense to look at in which end are you measuring this thing. Just taking into account in how we look at graphs and what dictates to us what is or not linear phase and how we get there (an impulse perhaps) and the characteristics of in which we translate those signals could give us a better look into what we’re trying to do.

Being that we view these things in the electro/mechanical acoustical logmerithic fashion (ex. As high frequency band gets louder the phase moves forward electrically) than it makes somewhat sense to me to not linearize the low side but do linearize the high side to keep continuity on which we read these graphs

Or am I way off base?
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Old 17th November 2019, 05:46 PM   #2849
nc535 is offline nc535
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re' natural sounds
Keep in mind that they may be represented by a Fourier series. Thinking of musical tones this way is especially helpful. A tone is the sum of its fundamentals and harmonics weighted by complex coefficients that carry both magnitude and phase information. Pick something simple like a square wave, play with rephase paragraphic phase equalization and you can both see and hear the effects of phase distortion.

Re' minimum phase rolloff at the high and low ends of the range
I think that has more to do with reducing the number of taps required to reproduce the filter characteristic and or reduce pre-ringing than it does the accuracy of reproduiction
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Old 17th November 2019, 10:19 PM   #2850
fluid is offline fluid  Australia
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Quote:
Originally Posted by mark100 View Post

fluid, i think you are spot on with the observation that a speaker with flat amplitude response 20-20K, or rather flat through audibility, satisfies both minimum and linear phase. I think it's more than satisfies...I think in such a case minimum phase equals linear phase.
Exactly, if the bandwidth is large enough the two end up being the same.


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Originally Posted by mark100 View Post
The thing I keep coming to, is linear phase in a speaker does not mean that the recording gets altered from minimum to linear phase. It just means the recording will be reproduced exactly as recorded, doesn't it?
I agree with the phase part and I see the point you are making, but I don't think you are looking at the whole picture. The problem comes when you try to keep the phase flat but let the amplitude fall. You fix one problem and create another. Flat phase without flat amplitude is just the flipside of your idea of adding group delay to a recording from a speaker, two wrongs don't make a right

The recording won't be reproduced exactly if the phase is flat and the amplitude rolls off. It will if the phase and amplitude are flat.

If you can't make the phase and amplitude flat you are left with a trade off and you have to pick your poison.


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Originally Posted by mark100 View Post
I guess if the recording was mastered listening to monitors that had rolloff, it might sound best if we could match the rolloff of the monitors used. Or less to no rolloff, if mastered on headphones or the new breed of linear phase monitors that are showing up in pro circles.
Maybe but how many different speakers was that recording listened to and tweaked on before it got to you. Which one do you pick, the last one in the chain at mastering who had no involvement in the recording process or the mix engineer, or the tracking engineer.......

There is also a difference between something that has had it's phase corrected in an anechoic environment to something that has the phase corrected at the listening position as wesayso promotes.

Quote:
Originally Posted by mark100 View Post
As far as natural sounds having low freq rolloff...wow, I wish I could wrap my head around what's going on. I've been thinking....how do you even ascertain what the phase response of a natural sound is?
A simple way is to record it and have REW generate an minimum phase version. Valid at the point the microphone was placed.


Quote:
Originally Posted by mark100 View Post
I can visualize a natural sound has some combination of frequencies that originate at particular times, but is that timing of origination due to limited bandwidth? Doesn't seem so...seems like it's due to the nature of the origination..and not a lack of bandwidth...does a lion's roar lack bandwidth and hence have group delay rolloff ....I dunno....

Anyway, like you guys said...round and round and never be right
The two mechanisms aren't separate, there are timing variations due to the times at which different parts of the instrument are excited, i.e. finger presses the key, hammer strikes the string. It takes time for the sound to move through the parts or chambers of an instrument. Same with a lions roar, the vocal chords resonate, the mouth and lungs act like resonant chambers changing the sound and amplifying it. The chambers will have roll off. No natural sound has unlimited bandwidth.

The phase will follow the amplitude response and group delay will be there. Sampling theorum shows that you can recreate the input if the sampling frequency is high enough. If something only produces sound in a limited range increasing the bandwidth won't change the sound, the change will come if you limit the bandwidth below where the sound is being produced. Like in the graphs Byrtt showed earlier.
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