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How to find out my active speaker's Xover details? 7 years 10 months ago #12778

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just my two Yamaha HS80M (both in same distance to my head).

Are you measuring both speakers at once? :ohmy:
The resulting comb filtering would explain those phase discontinuities up top for sure...

Please post the file of a raw impulse of one speaker playing at a 1 or 2m distance, with the speaker at its normal position.

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How to find out my active speaker's Xover details? 7 years 10 months ago #12779

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No no, not at once. This is only the left speaker.

Here they are:

File Attachment:

File Name: measurementFeb24.txt
File Size:1,175 KB

Thank you in advance, pos!
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How to find out my active speaker's Xover details? 7 years 10 months ago #12780

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I would have preferred an impulse instead of an amplitude and phase file, as it would have let me apply some smoothing and/or windowing, and look at the impulse.
I would also have made the polarity and time offset setting easier.
Here is the correction (no need for phase EQ, and low frequency phase measurement is most certainly not accurate enough to be corrected anyway)

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How to find out my active speaker's Xover details? 7 years 10 months ago #12781

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By the way, the phase discontinuities with the -1440°/+1440° phase range appear to be a bug, I will have to fix that.
Thanks for the report ;)
In the meantime stay with the standard -180°/180° wrapped range... :whistle:

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How to find out my active speaker's Xover details? 7 years 10 months ago #12782

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Here's the impulse response txt:


I'm a bit confused now - how have you done the correction?
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How to find out my active speaker's Xover details? 7 years 10 months ago #12783

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Here's the impulse response txt:

I was not able to load your impulse: it looks overly smoothed.

I'm a bit confused now - how have you done the correction?

I used your measurement, which was not ideal but still usable. It was also truncated to 15kHz up high.
Just for the sake of it I loaded it in HOLM and smoothed it a bit (I should have done this from the start ;)), and then applied the correction that I posted above:

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How to find out my active speaker's Xover details? 7 years 10 months ago #12792

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Now I wonder, if something with my measurement or settings is wrong or if rePhase just isn't able to linearize phase shifts of this extent?

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How to find out my active speaker's Xover details? 7 years 10 months ago #12799

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and then applied the correction

Did you apply it in HOLM or in rePhase?

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How to find out my active speaker's Xover details? 7 years 10 months ago #12802

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Now I wonder, if something with my measurement or settings is wrong or if rePhase just isn't able to linearize phase shifts of this extent?

It is right there, post #12780 of this thread .
Just enter those parameters and you are set.
Don't forget the polarity and time offset setting when importing the measurement: as already said setting these correctly is key when tackling a correction.

and then applied the correction

Did you apply it in HOLM or in rePhase?

HOLM can do a convolution between two responses.
One was your original measurement (in blue), the second one the generated correction from rephase (not shown), and you see the result of the convolution in red.
This is the response you should obtain when measuring the output of your openDRC (with correct polarity and time offset)
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How to find out my active speaker's Xover details? 7 years 10 months ago #12811

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Oh sorry - now I finally got, that the extreme phase shifts were caused by my phase range setting. With -180/180 it looks like on your screenshot.
--> So all I have to do now, is to flatten the FR with the Paragraphic Gain EQ and do some finetuning with the Paragraphic Phase EQ, right? And then generate the impulse to import it to the FIR on my OpenDRC, yes?

But just for understanding, to do the correct settings also for the right channel:
- Which measurement value defines the -36 µsec timeoffset and if "invert polarity" is activated or not?
- How did you amount to the "vented hight Q" box setting at 40 Hz and the subsonic setting "butt 12dB/oct" at 35Hz?

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How to find out my active speaker's Xover details? 7 years 10 months ago #12819

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Oh sorry - now I finally got, that the extreme phase shifts were caused by my phase range setting. With -180/180 it looks like on your screenshot.

Yes, that is a bug I will have to fix.

--> So all I have to do now, is to flatten the FR with the Paragraphic Gain EQ and do some finetuning with the Paragraphic Phase EQ, right? And then generate the impulse to import it to the FIR on my OpenDRC, yes?

Correct, but if you want to do EQs I would suggest taking several measurements at different positions in order to avoid correcting things that only exists at one point in space (diffraction, specular reflexions, ...) and use smoothing/gating/caution (!). I would suggest staying away of high Q corrections unless you are really sure you know what you are doing. In general it is safer to only correct the speaker itself, not the effect of its environment.
In any case, you should do the amplitude corrections first, using phase-minimum EQs, and only then use separate phase EQ (with greate care and even more caution than amplitude EQ).

But just for understanding, to do the correct settings also for the right channel

I would suggest using the *exact* same setting (exact same impulse in fact) for both channels. If ever you need to use different corrections (they must be really close, or something is wrong somewhere), set the centering to "middle" to ensure both corrections will imply the same delay (you should keep the "float" centering option though, as its impact on delay is negligible), or be prepared to compensate any difference within the openDRC...

- Which measurement value defines the -36 µsec timeoffset and if "invert polarity" is activated or not?

This one is not easy to answer...
With your speakers polarity will always be reversed, but time offset might have to be reset from one measurement to another, depending on the way REW does its impulse centering.
Let's see how it does, and if necessary I will explain you how to proceed by looking at the impulse (easiest way) or at the phase frequency response.

- How did you amount to the "vented hight Q" box setting at 40 Hz and the subsonic setting "butt 12dB/oct" at 35Hz?

Partly guess, partly trial and error.
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How to find out my active speaker's Xover details? 7 years 10 months ago #12845

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Thank you so much for taking time to answer my questions, pos! You really help a lot...

In general it is safer to only correct the speaker itself, not the effect of its environment.

Why that? Actually all I want is to clear the room's effect.

In any case, you should do the amplitude corrections first, using phase-minimum EQs, and only then use separate phase EQ (with greate care and even more caution than amplitude EQ).

Okay, let me guess the reason: because of the "pre-ringing" of the FIRs? I just tested it with some extreme linear-phase EQ settings (just to hear the difference for the first time) and it sounds horrible. So minimum-phase EQs won't introduce the pre-ringing?

I would suggest using the *exact* same setting (exact same impulse in fact) for both channels.

Hmmm.. My speakers have pretty different responses, especially in the bass area. So why not correct them individually?

Just to explain, if it matters: I don't have a real dedicated studio, I'm about to achieve a passably linear sound in my home studio for music creation and mixing (not mastering). I have a pretty symmetric setting with wideband absorbers in front & back end and on the ceiling. But I'm not able to install basstraps or superchunks here. And on the left side there are shelves, while I have 2 windows on the right side. I know, that's not ideal.. just want to optimize it as far as possible with room EQing.

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How to find out my active speaker's Xover details? 7 years 10 months ago #12925

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Just to show you my differences in the bass and why I think, there're separate EQs on the left and right channel necessary:
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How to find out my active speaker's Xover details? 7 years 10 months ago #12969

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Hello Phranky, sorry for the delay.

In general it is safer to only correct the speaker itself, not the effect of its environment.

Why that? Actually all I want is to clear the room's effect.

Just to explain, if it matters: I don't have a real dedicated studio, I'm about to achieve a passably linear sound in my home studio for music creation and mixing (not mastering). I have a pretty symmetric setting with wideband absorbers in front & back end and on the ceiling. But I'm not able to install basstraps or superchunks here. And on the left side there are shelves, while I have 2 windows on the right side. I know, that's not ideal.. just want to optimize it as far as possible with room EQing.

This is a complicated matter. The danger with "room" correction is that if you can correct things that are only valid at one location (where the measurement was taken), such as diffractions, specular reflections, room modes buildups or suckouts (*never* compensate modal suckouts!), etc.
Also, you should try to only compensate the direct sound (and close reflexions if you cannot avoid them, say up to a few ms), not the power response. This is probably different in the bass where our "integration window" is longer.
Short answer is you should never use a single measurement for that kind of correction. You should either do multiple measurement at different positions inside your the listening window and go through them one by one in rephase to see the effect of your correction (using drag and drop), or better still do an averaging of said measurements (I think REW can do that). Also use gating and smoothing. All these techniques have their merits and should be used together, with multiple measurement and analysis....
In my opinion the only differences in corrections between channels should be there to compensate for driver or passive filters differences, or in the bass if you have symmetry issues with your room or placement. Above a few hundreds of Hz well placed passive solutions like the panels you are using should be able to take care of any symmetry issue.

If what you are looking for is room correction you should probably check a dedicated software like Align2 (see the dedicated section on this forum), that will take all the (hopefully good) decisions for you.
Jean-Luc is an expert in these things B)

In any case, you should do the amplitude corrections first, using phase-minimum EQs, and only then use separate phase EQ (with greate care and even more caution than amplitude EQ).

Okay, let me guess the reason: because of the "pre-ringing" of the FIRs? I just tested it with some extreme linear-phase EQ settings (just to hear the difference for the first time) and it sounds horrible. So minimum-phase EQs won't introduce the pre-ringing?

If you use minimum-phase EQs on a minimum-phase system (which a loudspeaker driver is), flattening the amplitude does also flatten the phase. Minimum-phase EQ is what you should use 99% of the time.
Phase-only EQ should only be used with for low Q correction, to help with crossover phase linearization.

I would suggest using the *exact* same setting (exact same impulse in fact) for both channels.

Hmmm.. My speakers have pretty different responses, especially in the bass area. So why not correct them individually?

You mean your room? If your speakers themselves have differences that large, by all mean change them :P . By the way, you should try to measure them at the same location, to see what is position-related, and what is speaker-related.
Anyway, if you do want to apply different corrections for the two channels, please note that in that case letting rePhase do an automated centering based on energy (default) will most probably lead to different implied delays for the two channels (see the indication given by rePhase in samples and ms after generation), which *will* sound bad. So in that case you should either use the "middle" centering, enforcing the same (suboptimal) centering on both channels, or compensate for the delay difference in the openDRC.
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How to find out my active speaker's Xover details? 7 years 10 months ago #13021

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No problem pos, I'm very thankful for so much detailed information!

The danger with "room" correction is that if you can correct things that are only valid at one location (where the measurement was taken), such as diffractions, specular reflections, room modes buildups or suckouts

My head / listening position really just moves in a circle of max. 20cm, but to play safe I'll do three measurements for each speaker/channel with different head locations. Then I'll average them in REW.
--> Is there a way to average the phase of these measurements?

*never* compensate modal suckouts!

Okay, so I'll only reduce gains, but won't boost any :)

Also use gating and smoothing.

Where is gating applied to the measurement in REW and what does it mean?

If you use minimum-phase EQs on a minimum-phase system (which a loudspeaker driver is), flattening the amplitude does also flatten the phase. Minimum-phase EQ is what you should use 99% of the time. Phase-only EQ should only be used with for low Q correction, to help with crossover phase linearization.

So I'll only use minimum-phase filters and then do rough adjustments with the Phase EQ. But won't use linear-phase filters. Thus I won't hear any pre-ringing, right?

if you do want to apply different corrections for the two channels, please note that in that case letting rePhase do an automated centering based on energy (default) will most probably lead to different implied delays for the two channels (see the indication given by rePhase in samples and ms after generation), which *will* sound bad. So in that case you should either use the "middle" centering, enforcing the same (suboptimal) centering on both channels, or compensate for the delay difference in the openDRC.

So is this the right conclusion: I would use the "energy" centering and then add the both channels' delay difference to the less delayed channel in the Open DRC?

Which measurement value defines the -36 µsec timeoffset and if "invert polarity" is activated or not?

if necessary I will explain you how to proceed by looking at the impulse (easiest way) or at the phase frequency response.

When I'm doing new measurements now, I worry that I need to know now, how to find out the correct time offset value :S

I hope I'm not digging to deep, also in your free time!! Hopefully these will be the last points to finally find the right approach for creating a linear reference sound for my music productions. Thank you so much!

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