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How to measure phase properly at home using REW?

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Krunok

Krunok

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How do you know it is calibrated in phase?
The point of calibrating is to have a flat curve when recording a flat system.

Well, for a start your mic is not calibrated at all. You have received a calibration file that contains amplitude and phase data from some guy on the Net and now you think your microphone is calirated? I've got bad news for you - it is not.

There are 2 ways you can get calibrated microphone:
- you can buy it with calibration file included as measured/calibrated by the manufaturer
- you can send your microphone for calibration to a specialised company

I advise you to read carefully info on this link. As John Mulcahy explains there you would do better without calibration file for Behringer ECM8000 than with the ones that are circulating the Net.
 
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Krunok

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Come on, you wrote that correcting the phase in the bass department sounded "awful". Then you wrote that a lot of ABX showed that you couldn't hear a difference. What's the truth?

How about reading carefully my posts? I said sound was "awful" because of ringing. I got ringing with filters where I lowered the phase in the LF region to make it flat instead of rising.

And yes, lot of ABX is actually showing no difference when you correct the time domain of a speaker.
 
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Krunok

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I also read that under 300 Hz, people should correct to have minimal phase and not linear phase in order to avoid pre-ringing. I for one have better results with linear phase set with a calibrated mic.

How abut posting your measurements with minimal phase vs linear phase filters?
 

daftcombo

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I am just trying to help.
You ask how to measure phase properly, I reply: "get a calibrated mic first".
If you think that phase doesn't matter because you read ABX reports telling so, why do you care?!
If a phase "corrected" with a non-calibrated mic sounds awful, either there's a problem with the mic, either it's your ABX reports that are strange.

As for me, I tried my calibrated mic on a pair of speakers that I bought to some other guy who corrected them to be flat. He used a different mic with a calibration file sent by the company. With my calibration file, my mic shows a quite flat curve as well. Without it, there's a rising. I consider it sufficient to say that I have better measurement with the calibration file than without.
 
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I am just trying to help.
You ask how to measure phase properly, I reply: "get a calibrated mic first".
If you think that phase doesn't matter because you read ABX reports telling so, why do you care?!
If a phase "corrected" with a non-calibrated mic sounds awful, either there's a problem with the mic, either it's your ABX reports that are strange.

As for me, I tried my calibrated mic on a pair of speakers that I bought to some other guy who corrected them to be flat. He used a different mic with a calibration file sent by the company. With my calibration file, my mic shows a quite flat curve as well. Without it, there's a rising. I consider it sufficient to say that I have better measurement with the calibration file than without.

Again - microphone is considered calibrated only by those 2 criterias I mentioniend in my previous post. According to that your mcirophone is not calibrated, and I certainly don't want to adjust my microphone calibratiton file so that my measurements look better.

I think we had enough talk about your microphone. Please, either contribute to this therad by posting your measruements with explanation how you made them or stop bombarding this thred about your "calibrated" microphone.
 

daftcombo

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This thread is not about my setup.
I'll just stop contributing to it. Good luck with your "proper" measurements.
 
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This thread is not about my setup.
I'll just stop contributing to it. Good luck with your "proper" measurements.

This thread is about sharing knowledge and experience of measuring phase of our home setups.

As you don't seem to have anything to contribute you won't be missed.
 
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Btw, as mentioned here it seems John Atkinson was placing microphone at 50" distance from the speakers when measuring MLs at owner's home.
 
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Krunok

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I will add this post to share the method that worked for me.

So, if doing manual roomEQ with rePhase and you want to do correction of phaser eposnse at LP you need to measure the uncorrected phase response reliably. Best way that I found was to make 9 sweeps per channel over the 1m2 area (which is slightly more than size of my 1-seat LP) - 8 over the edges and 1 at the center. Time aligne all 9 sweeps and make Vector average from them. Procedure is described in detail in "Bear_REW_rePhase_tuto.pdf" which can be easilly found on Net. DO NOT be confused with the fact that in that document it is suggeseted to use vector average for magnitude correction. IMHO this is wrong as magnitude of vector average differs substantially from standard average of same sweeps. On the other hand, standard average of sweeps correlates veri nicely with MMM RTA method, which is my preferred method when doing magnitude correction for the simple reason that it has the most samples.

If you look at this post from JohnPM it actually says that vector average feature was "added at the request of users who wanted a vector average to allow phase correction to be applied to the result".

After calculating vector average from all 9 measurements I applied FDW of 1/6 cycles and 1/6 smoothing.

These are phase response, step response and GD taken from LP after the correction with rePhase:

Phase.JPG


GD.JPG


Step.JPG
 

Pio2001

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Won't phase correction introduce pre-echo ?

How does the CEA 2010 Burst test signal sound after phase correction (REW / Generator / CEA-2010 burst, play, try various frequencies) ? The right result is the sharpest one.
 
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Won't phase correction introduce pre-echo ?

No, it will actually reduce it.
You will get minimum pre-echo once the excess phase is flat and close to zero.

How does the CEA 2010 Burst test signal sound after phase correction (REW / Generator / CEA-2010 burst, play, try various frequencies) ? The right result is the sharpest one.

I didn't test with those signals. However, what I have noticed is that after phase correction HF tones like cymbals etc sound noticeably clearer and more similar to how they sound when heard alive.

I tested with bulb drop recording from Focal Test CD No2 and that one also plays noticeably sharper and very "lively".
 
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Krunok

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One more thing: I noticed that some room EQ tools (for example DRC-FIR) have algorithms to reduce/eliminate pre-echo in step response. Unfortunately I don't know how they do it. I would appreciate very much if somebody who knows would chime in and explain how to do that when doing correction mannualy with rePhase.
 
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Won't phase correction introduce pre-echo ?

How does the CEA 2010 Burst test signal sound after phase correction (REW / Generator / CEA-2010 burst, play, try various frequencies) ? The right result is the sharpest one.

To reduce pre-ringing I tried to keep my system to be minimum phase, so with excess phase close to zero. Here is how it looks:

Excess phase_Left.JPG


Excess phase_Right.JPG
 

Pio2001

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To reduce pre-ringing I tried to keep my system to be minimum phase, so with excess phase close to zero. Here is how it looks:

It looks good... but it is very difficult to understand how the phase goes through the process of windowing, then averaging, and then smoothing.

It seems to me that smoothing a phase curve, at least, makes no sense. Smoothing an amplitude curve gives the average level of the frequency range, which is something that we can hear. But the average phase is not something that we can hear. I'm even not sure that it has any meaning at all.

On the contrary, the steepness of the raw phase curve is something that we can hear : it is the group delay.
Wikipedia :
the group delay is a measure of the slope of the phase response at any given frequency

Therefore smoothing the phase curve (here with 1/6th octave smoothing) must be avoided. And since smoothing is cumulative in REW, if smoothing was activated before you made the vector average of your measurements, it has become permanently embedded in your data.
I'm afraid that this turns the whole measurement useless.

And you have talked about windowing, too. Does this allow you to measure the phase of the direct sound only ? Until what frequency ?
If the direct sound has no pre-echo, obviously, the sound at the listening position won't. So that would be a good thing. But how can we be sure that the window allowed to reject completely any reflections ?

And what does averaging 9 measurements do to the phase ? If we are in a frequency range where only the direct sound is present, and if you time-align your impulses, that would mean averaging the phase of the speakers across a given angular listening window.
 
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Krunok

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It looks good... but it is very difficult to understand how the phase goes through the process of windowing, then averaging, and then smoothing.

It seems to me that smoothing a phase curve, at least, makes no sense. Smoothing an amplitude curve gives the average level of the frequency range, which is something that we can hear. But the average phase is not something that we can hear. I'm even not sure that it has any meaning at all.

On the contrary, the steepness of the raw phase curve is something that we can hear : it is the group delay.
Wikipedia :


Therefore smoothing the phase curve (here with 1/6th octave smoothing) must be avoided. And since smoothing is cumulative in REW, if smoothing was activated before you made the vector average of your measurements, it has become permanently embedded in your data.
I'm afraid that this turns the whole measurement useless.

And you have talked about windowing, too. Does this allow you to measure the phase of the direct sound only ? Until what frequency ?
If the direct sound has no pre-echo, obviously, the sound at the listening position won't. So that would be a good thing. But how can we be sure that the window allowed to reject completely any reflections ?

And what does averaging 9 measurements do to the phase ? If we are in a frequency range where only the direct sound is present, and if you time-align your impulses, that would mean averaging the phase of the speakers across a given angular listening window.


I smoothed the phase curve after the averaging. But ok, let's throw vector averaging out of the picture for the moment as results with the single sweep taken from center of the LP are practically identical. Also, I increased FDW to 12 cycles for this example.

Here they are:

Phase, no FDW, no smoothing
Phase_noFDW_noSMOOTHING.JPG


GD, no FDW, no smoothing

GD_noFDW_noSMOOTHING.JPG


Phase, FDW 12 cycles, no smoothing
Phase_12cyclesFDW_noSMOOTHING.JPG


GD, FDW 12 cycles, no smoothing
GD_12cyclesFDW_noSMOOTHING.JPG


Phase, FDW 12 cycles, 1/6 smoothing
Phase_12cyclesFDW_1-6SMOOTHING.JPG



GD, FDW 12 cycles, 1/6 smoothing
GD_12cyclesFDW_1-6SMOOTHING.JPG


As you can see I applied 1/6 graph smoothing just to see "the woods from the trees", otherwise the graphs is simply too choppy to see anything.
It probably also doesn't reflect how we hearr things as our hearing doesn't have resolution to catch all those narrow high peeks.
 
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Krunok

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Anyway, I'm pretty sure in these 2 things:

- the ammount of pre-ringing depends on the structure of used filters. Filters with higher Q and more gain/attenuation will generate more pre-ringing

- minimum phase system has less pre-ringing, and for that reason pre-ringing reduced when I corrected the phase to be minimum phase

I have set of filters generated by DRC-FIR tool and in them pre-ringing has been avoided, however I'm not using them as they sound inferior to the filters I made mannually with rePhase:

DRC-FIR, IR
IR_DRC-FIR.JPG



My filters, IR. As you can see it does have slightly more pre-ringing than IR of DRC-FIR filter but it seems pretty insignificant to me.

IR_My-filters.JPG




I have noticed that at least DCR-FIR (but I think it goes for other room EQ tools as well) is in a way "obsessed" with getting the excess phase flat and close to zero. That means that phase curve of both channel has to closely follow minimum phase curve and that very often leads to phase mismatch which than leads to bass cancellation when measuring both speakers response. That also means GD will suffer as phase curve will be less flat when it strictly follows minimum phase curve so GD, being derivation of phase curve, will suffer.

IMHO those tools also put too much emphasis on pre-ringing although I haven't really seen any research on pre-ringing audibility and certainly not that it is worth sacrificing GD response to reduce ringing. Or ..?
 
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Krunok

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And you have talked about windowing, too. Does this allow you to measure the phase of the direct sound only ? Until what frequency ?

In these examples I used Frequency Dependent Windowing of 12 cycles. If the width is in cycles a 12 cycle window has a width of 120 ms at 100 Hz (12*10ms), 12 ms at 1 kHz (12*1ms) and 1.2 ms at 10 kHz (12*0.1ms).

FDW of 10,12 or 15 cycles is very common and used by majority of room EQ tools, although smaller windows can also be used (like 6 or 3 cycles) to get smoother curves.
 

JohnPM

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And since smoothing is cumulative in REW, if smoothing was activated before you made the vector average of your measurements, it has become permanently embedded in your data.
I'm afraid that this turns the whole measurement useless.
That is not correct. REW's Vector average, like other trace arithmetic operations on impulse response data, uses unsmoothed data generated from the impulse responses. The result is smoothed using whatever smoothing was previously applied, but that smoothing can be removed or changed as desired.
 

Pio2001

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GD, no FDW, no smoothing

View attachment 46738

This picture is interesting.
The large peaks at 130, 170 and maybe 240 Hz are significant.

Narrow peaks can be ignored, as they are probably measurement artifacts : when the test signal becomes very weak, at the bottom of dips in the frequency response, the phase response suddenly jumps 180° away. Since the signal is barely detectable, the software can't always read if the jump is positive or negative (+180 or -180° both lead to the same value), so it may draw a peak instead of a dip in the group delay graph, and conversely.

But the fact that the group delay is around 80 ms at 130 and 170 Hz is a bit worrying. Except if these parts are at the bottom of dips in the amplitude response curve, as it is often the case. Then it doesn't matter, since these frequencies are nearly silent.

The hills at 240, 390 and 480 Hz are strange. I have nothing like this in the raw group delay graph from my listening position :

50_GroupDelayBefore.png


What's the sweep size used as a test signal ? For phase and GD measurement from the listening position, I think that the longest one is best (1M). In any case, the length of the test signal has a clear effect on this graph.
I don't know if the fact that the microphone is perfectly still during the measurement matters.

GD, FDW 12 cycles, no smoothing
View attachment 46740

So here, we have the group delay of only a part of the sound : the direct sound and its first reflections (it's completely anechoic only below, say, 1ms, that is only above 1000 Hz in this graph).

It's a bit difficult for me to understand what it means. I can understand what means the amplitude response with Frequency Dependant Window. It's the tonal balance of the attacks in the sound, with reverberations in the room left aside. I can imagine the spectrum changing with time, and being different in the first milliseconds, while we are hearing the direct sound of the speaker, then changing as the sound decays, while we are hearing the sound of the room.

But phase changing as sound decays ?
If I understand correctly, it gives an idea of the time coherence of the attacks that reach our ears before the sound begins to decay in the room.

The two previous peaks, at 130 and 170 Hz, are still visible here, reduced in amplitude. But we can't tell if they come from the direct sound (then it matters, as this represents the time coherence of the speaker) or from the first reflections in the room, that the 12 cycle windowing includes.
In the later case, they should be ignored, as they are just remnants of the peaks in the initial graph, not completely canceled because of the too wide windowing.

GD, FDW 12 cycles, 1/6 smoothing
View attachment 46742

[...]
It probably also doesn't reflect how we hear things as our hearing doesn't have resolution to catch all those narrow high peeks.

I don't think so. I think that what we hear is the original peak at 80 ms in the raw graph.

I have made some videos with random phase distortion. Unfortunately, all examples include pre-echo, which prevents to hear group delay alone, as pre-echo is much more audible. But anyway, here they are.
Listen to them with headphones, to avoid the group delay of your room masking everything.


In the first part of the video, the effect of amplitude alone is heard. Since the transformation is not minimal phase, an awful lot of pre-echo is audible in low frequencies.
In the second part of the video, the effect of phase alone is heard. Again, a lot of pre-echo can be heard in the bass.

This second video is a bit more intersting :


Here, the same random amplitude distortion as in the first video is added to the sound, but in two different versions, minimal, and linear phase alternately.
Although the pre-echo of the linear phase version is still obvious, we can also hear that the bass rhythm is slightly offsetted in time. This is the audible effect of group delay alone (+74 ms here).

The raw group delay curve of the linear phase example is this :

SimulNoGroupDelay.png


It peaks around 25 ms (it should have been flat, but the actual corrections are not perfectly accurate).

The group delay of the minimal phase version is that :

SimulGroupDelay.png


And the effect of this narrow peak of group delay is audible in itself, I think (although I can't prove it, since the delay alone can't be separated from time smearing).
If the musical beat had been perfectly centered around the frequency of 54 Hz, it would have been delayed by 100 ms. But since it is not exactly at that frequency, we can measure, on the musical waveform, that the actual delay is 74 ms only.

What I'm meaning is that what we can see, and hear in this example is a 74 ms delay in time, and this is only visible in the group delay graph when it is not smoothed.

Later, I started from the minimal phase version of the distorted file, and applied two reverse filters : one linear phase, and one minimal phase. This time I was able to compare the temporal effects without pre-echo in any of the two files. I must have these files somewhere on my hard drive. I ABXed them with a score of 16/16.
But the main audible difference was not the delay, it was the overall time smearing. Although there was no pre-echo, and all the bass was delayed after the initial attack, the time smearing was still audible. It was easier to focus on it than on the delay to pass the ABX test.
 
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