Keith_W DSP system

[3ll3d00d]

I have managed to time align the subwoofers now

To my mind, it seems like you are conflating time alignment and phase alignment when mostly you seem to be concerned with the latter, the benefit of which is generally how the two sum (which is the bit of info not shown so far)

 

[dualazmak]

In any way, sub to woofer XO/EQ/Gain/Phase/Time tunings are in narrow low Fq zone, say within 30 Hz to 70 Hz Fq zone. Furthermore, the sound quality of this Fq zone is very much (or much more) dependent on our room acoustics, room mode, room environment, in our individual specific listening room. The actual air sound at listening position (especially when all the L&R SP drivers are singing together) does not always reflect the theoretical details of our/your intensive fine DSP tuning efforts. Our home listening environment can never to be that of huge anechoic laboratory chamber.

Consequently, at the end of story, all of our DSP and room treatment efforts towards better and optimal sound quality at listening position should be achieved only by various combination of "compromises" in DSP and room treatments in our own individual room environment, and they would differ our room to room, gears to gears, DSP to DSP. Yes, there would be no single best approach nor single best policy, indeed.

Keith @Pute Audio nicely wrote here that "You must hear equipment in your own room in your own system, compare unsighted (close your eyes) if there isn’t an immediately apparent difference/improvement. To go further, but if there isn’t a significant improvement then don’t change anything; the largest gains are speakers and room." for which I fully agree.

 

[Ze Frog]

Always been interested in the plasma tweeter technology, never heard one and likely never will unless I D.I.Y one one day as a fun science project. How is the sound?

 

[Keith_W]

What is done for live-sound sub to main alignments, is to choose a sub low-pass and a main high-pass, such that when time alignment is correct, the phase traces completely overlay each other for as wide a frequency range as possible.

That is very useful, thank you! I have a question though - if the slopes of the two phase traces are different, how do they get them to overlay over a wide frequency range? As the graph of my system shows, it is only possible to get the two traces to touch at a single point. Acourate has a series of steps where the phase of individual drivers can be linearized, and in fact I have performed that procedure. I will show you.

This is the phase linearization of the subwoofer. You can see the before trace in brown, where it wraps around a few times (vertical scale is in radians). Blue is after phase linearization. These are nearfield measurements, and the blue trace is a measured response, not a sim.

And this is the phase linearization of the woofer. Again, brown is before, and blue is after. Once again ... nearfield measurements, actual measured response.

So now we take the nearfield phase linearized sub, and phase linearized woofer, and place the mic at the MLP. The above is what we get. The slopes are widely divergent. Obviously the room has interfered with the response by quite a lot. Note that these curves are measured phase with no phase extraction, i.e. it shows both the min phase response and the excess phase.

Dr. Uli thinks that it is not worthwhile doing a nearfield linearization of the subwoofer, because he feels that the nearfield measurement would be contaminated by the room response. So although I can get the curve perfectly flat, it is not a representation of reality. But I wonder about correcting the MLP response.

I think I may be going about this the wrong way. Perhaps what I should be doing is not doing a nearfield phase linearization. Instead, maybe I should measure both at MLP, derive the difference in the slopes, and designing a reverse AP filter to correct that difference. Right now it is not clear in my head whether I should correct for the difference in min phase, or excess phase, or whatever. Just to be clear, I am using FIR filters - so correction for the amplitude response automatically corrects the minphase response (the graphs I am showing you also have corrected amplitude response). Any guidance you guys can provide will be gratefully received.

And FYI, I am not listening to the phase linearized filters that I showed you. I ended up removing all the nearfield phase linearizations and only do frequency amplitude linearization and time alignment at the crossover frequency. It sounds pretty amazing, and I am happy with it. But then you always wonder if you could do better ...

(EDIT) A dim lightbulb has gone off in my head. I will measure the subwoofer and the woofer at the MLP and look at the phase response. I will remove excess phase, and then design a reverse AP filter to correct for the minphase response. The Schroder frequency in my room is 106Hz (calculated), which means the end of the transition zone 4Fs is 424Hz. Because the woofer is low passed at 500Hz, I should be able to correct the phase all the way up to slightly above 4Fs.

So yes, is is possible to get the phase of a sub and main to align over a wide range of frequencies. Via the time honored phase trace overlay method.
We home audio folks who can stand latency have it easy....just use complementary linear pass xover between sub and main (and sufficient taps!)
Live sound folks who have to stick with IIR, are the ones who have to work,....much harder to get phase rotations to overlay than straight flat lines.
Hope this all made sense/ helps....

"Have it easy" is relative! If you don't have any training in sound and you are self-taught, it can be quite a struggle. Believe me, I am very aware of my own deficiencies in education, something which is driven home every time I look at those traces and ponder what I am going to do! I really hope this thread does not come across the wrong way, I am not presenting myself as some kind of authority. It's more like "this is what I have done, if you think my understanding is wrong, please correct me". So your post was very helpful - thanks again.

 

[Keith_W]

To my mind, it seems like you are conflating time alignment and phase alignment when mostly you seem to be concerned with the latter, the benefit of which is generally how the two sum (which is the bit of info not shown so far)

I thought that the two were the same, after all phase = time? If the two are different, I would love to read more about it. I will do my own search, but if you have links to good resources please feel free to post.

 

[Keith_W]

Always been interested in the plasma tweeter technology, never heard one and likely never will unless I D.I.Y one one day as a fun science project. How is the sound?

You know, I am rather hesitant to describe the sound because I don't know what measurement it correlates to. But it definitely sounds different to a normal tweeter. It has an ethereal quality and it sound appears to emerge from thin air.

I can show you an impulse response of the tweeter:

 

[ernestcarl]

I thought that the two were the same, after all phase = time? If the two are different, I would love to read more about it. I will do my own search, but if you have links to good resources please feel free to post.

Imagine that your left and right speakers are perfectly time aligned, but the polarity in one channel just so happens to be inverted -- both impulses still arrive in time, of course, but the phases appear shifted 180 degrees apart.

This might help: https://www.prosoundweb.com/time-phase-alignment/

 

[3ll3d00d]

I thought that the two were the same, after all phase = time? If the two are different, I would love to read more about it. I will do my own search, but if you have links to good resources please feel free to post.

The article posted above is a straightforward summary, as it says they are certainly related given a time delay is a frequency dependent phase shift but not the same. For a crossover, you typically look at phase alignment so as to check they sum constructively after having time aligned them.

 

[Keith_W]

Imagine that your left and right speakers are perfectly time aligned, but the polarity in one channel just so happens to be inverted -- both impulses still arrive in time, of course, but the phases appear shifted 180 degrees apart.

This might help: https://www.prosoundweb.com/time-phase-alignment/

That was a FANTASTIC article, thank you! When I was looking at my own curves, I realized that I was looking at similar concepts to what that article discusses, except that I was using the wrong terminology in my discussion. So, if I read my older post again and substitute "phase alignment" for "steady state", I would be effectively describing the difference between time alignment and phase alignment.

What the article does not say is how much the phase needs to be aligned. Suppose the phase discrepancy between two drivers is 3690 degrees (i.e. a 90 degree phase difference has been rotated 10 times). What do you do then? Rotate one driver 3690 degrees? Or rotate it 90 degrees so that the phase aligns?

The article also briefly discusses what to do when confronted with different phase slopes. Reverse all pass filter into one driver to get alignment.

While I grasp the concepts, I need to translate it into an Acourate workflow. I will think about it.

Again, thank you so much for that article. It really helped.

 

[3ll3d00d]

Rotate one driver 3690 degrees?

I guess you must be thinking of unwrapped phase here but that, in the sub region, can be quite corrupted by the room. Rotation in acourate is just a time delay so really you are talking about time alignment at this point and then trying to use a phase adjustment to align two drivers.

 

[Keith_W]

I guess you must be thinking of unwrapped phase here but that, in the sub region, can be quite corrupted by the room. Rotation in acourate is just a time delay so really you are talking about time alignment at this point and then trying to use a phase adjustment to align two drivers.

I guess I should rephrase my question with this illustration.

Here we can see the unwrapped phase response of two drivers (subwoofer and woofer). We note that they are neither time aligned, or phase aligned.

I know how to time align them, that is no problem. My question is about phase alignment. Do I need to:

1. Adjust the slope of both unwrapped phase curves so that they match in slope only (thus keeping time alignment)? or
2. Match the slope of both phase curves and overlap them (thus messing up time alignment)?

Furthermore, the measured response has both minimum phase and excess phase. What phase should I correct? Measured? Minphase? Excess phase?

And lastly ... given that I am high passing the mains with a steep slope ... should I even bother? My understanding is that I should only care about phase alignment if the subs and mains are being used full range (and thus substantial potential for strange peaks and cancellations to occur).

(EDIT) after thinking about it a bit more, I think I have answered my own questions.

1. The slopes of the curves need to match, and then they need to be rotated so that they are in phase. They do not need to overlap. This is because getting the two curves to overlap would mean substantial time adjustment which will send the initial impulse far off. And we are only concerned that the two drivers are in phase with each other.

2. Given that the sub and the woofer in my system are crossed over, there will be minimal overlap between the frequency bands, so the difference in phase will not produce too many problems. And if there are, I can measure them at the MLP. I already have these measurements, and the ONLY issue i see is a narrow but deep notch at 40Hz. I don't think it is worth fixing. You can see it here:

3. If, after all this, I decided that I am a sucker for punishment / love self-abuse / love creating unnecessary work for myself and I STILL want to align the phase slopes, then I would convolve a reverse AP using excess phase as my guide slope. This is because it is the room which is messing up the phase response, and thus the excess phase. I think using the measured phase is probably the wrong approach because it will over-correct the phase, and there is no point correcting the min phase response since that has already been corrected by the FIR filter. Would you agree with this reasoning?

 

[bothu]

I have managed to time align the subwoofers now. I will say a few things about subwoofer time alignment which will hopefully help someone out there. I took a leaf out of Dr. Uli's book, and decided that I will use multiple methods to cross-check subwoofer time alignment. I have gone over the procedure so many times and spent so much time staring at the curves that I could write a treatise about it, but I suspect that even the eyes of DSP nerds on ASR will glaze over in utter boredom. So here is the abbreviated version.

Keith's rules of subwoofer time alignment:

1. You can align for the initial impulse OR the steady state, but not for both. You have to choose. This law is immutable, it is due to physical behaviour of your drivers AND the behaviour of bass in the room (it may appear to change phase due to reflections before it arrives at your mic). There is no way to DSP around it.

2. If you align for the steady state, only one frequency can ever be time aligned, and all other frequencies will diverge from perfect time alignment. You may need to employ additional strategies to achieve better time alignment.

3. Every time you do a sweep, the result will be different. I have measured variances between 0.02 - 0.1ms. This is normal, don't get too hung up about it. You are unlikely to hear a subwoofer misalignment of 0.1ms.

4. Different filters will produce different delays, so it is important to repeat the time alignment procedure every time you design a new filter for the sub, or for the woofer.

5. The importance of a good cup of coffee can not be overstated.

So why should you time align for the initial impulse? Answer: if you are high passing the main speakers. In this case, the alignment of lower frequencies of the main speaker does not matter because the sub will be handling those frequencies. Why should you time align for the steady state? Answer: if you are using both your mains and subs to produce bass.

As proof of my first statement, that it is impossible to achieve time alignment for both the initial impulse and the steady state, I offer this graph as proof:

View attachment 354567

This is a sweep of the subwoofer and the woofer, convolved with a 50Hz sinewave to better show the time behaviour of the drivers. You can see that I have rotated the subwoofer to align with the woofer at the initial impulse, but doing this messes up the steady state. Almost as if the subwoofer has inverted polarity! But not so, look closely at the initial impulse and you will see the shape of the deflection of both drivers is the same.

If we align for the steady state at 50Hz, the initial impulse is no longer aligned, as we can see here:

View attachment 354568

However, there is not much good in doing this. This is because the phase rotates across the frequency range of both the subwoofer and the woofer. What this means is that only one frequency can ever be time aligned, and all other frequencies will diverge from perfect time alignment. I will show this graph as proof:

View attachment 354566

This is the unwrapped measured phase of both the sub and the woofer, zoomed in the frequency band of interest (20Hz - 100Hz) and the vertical scale adjusted. Using the above method (sinewave convolution), I rotated the sub and woofer by the values I derived. As you can see, perfect alignment at 50Hz. But nowhere else. Some of those delays are almost 10 radians apart. I posted the calculation to convert radians to time here - we are talking delays of up to 30ms! Now 30ms may not matter if it was in isolation, but if you have two drivers playing the same frequency 30ms apart it is bound to cause problems.

It MAY be possible to extract excess phase and use the min phase version to design some kind of all pass filter that will straighten the phase of both the sub and the woofer to get them to align a bit better. But since I don't need to do it because I am high passing the main speakers, I did not try this experiment.

This is why I like Acourate - it is not a "DSP black box" where measurement goes in, and result comes out. You have no idea what the "black box" did and whether the result is correct. I know that some software packages don't even let you change what the software has decided, nor do you even get the tools to check yourself. Acourate forces you to think and make decisions, and you learn so much in the process.

(EDIT) after all that I forgot to post a step response to show proper driver time alignment. So here it is:

View attachment 354570

What does a time aligned system sound like? Well, it is easier to describe what a non-time aligned system sounds like. If you have the ability, you should deliberately mess up the time alignment of your system. You will hear smearing and a loss of clarity. The image is unstable and seems to shift depending on pitch (the worst offenders are instruments with a wide frequency range like a grand piano). Bass sounds "slow" and laggy. Dynamics are affected, and the system sounds lazy and "constricted". With time alignment, there is incredible transparency, slam, and punch.

Thank You for this excellent work !
I bow my bold head to my knees pointing to the south hemisphere

Bo Thunér in Sweden.

 

[3ll3d00d]

We note that they are neither time aligned, or phase aligned.

How can you tell they are time aligned from this view?

I think an unwrapped phase view is much harder to read btw because it's so much harder to see the actual slope of each curve.

My understanding is that I should only care about phase alignment if the subs and mains are being used full range

Phase alignment is important for any crossover. You can look at how sine waves sum to understand how far off they have to be before it becomes destructive (more than 120 degrees basically). Looking at the frequency response in that region is the simplest way to see this.

 

[Keith_W]

How can you tell they are time aligned from this view?

You are right, I am thinking in a sloppy fashion again. If the two slopes are very close to each other, then time alignment will cause the two slopes to intersect. But if they are far apart, then I need to switch to a different view to demonstrate time alignment. Sorry about that.

And BTW I was probably editing my post above while you were typing your response. Would appreciate your comments on the parts I edited (at the end, indicated by EDIT).

I think an unwrapped phase view is much harder to read btw because it's so much harder to see the actual slope of each curve.

Would you prefer to look at the Group Delay (negative derivative of unwrapped phase slope)?

Phase alignment is important for any crossover. You can look at how sine waves sum to understand how far off they have to be before it becomes destructive (more than 120 degrees basically). Looking at the frequency response in that region is the simplest way to see this.

This is the FR of the drivers individually and together.

Red = sub, Green = woofer, Brown = together. All measurements were taken from the MLP.

There is a notch at 47Hz. This is very close to the sub and woofer XO point at 50Hz. What I just noticed, which may be interesting, is that it has a central peak at 50Hz with notches either side of it, a "W" shape. I wonder if that is due to phase cancellation. Fortunately, I have the measurement with the sinewave convolved, so I can look at the phase behaviour at 50Hz:

You can see that when the initial impulses for the sub and the woofer are aligned, the phases are 180 deg out of phase. So indeed it does look like there is phase cancellation at 50Hz.

So that's it. Decision made: all pass filter and then remeasure.

 

[gnarly]

"Have it easy" is relative! If you don't have any training in sound and you are self-taught, it can be quite a struggle. Believe me, I am very aware of my own deficiencies in education, something which is driven home every time I look at those traces and ponder what I am going to do! I really hope this thread does not come across the wrong way, I am not presenting myself as some kind of authority. It's more like "this is what I have done, if you think my understanding is wrong, please correct me". So your post was very helpful - thanks again.

Easy is Relative indeed
Honestly, what you are doing is far harder than anything I've tried to do yet. I still feel room correction at listening position is over my head.
I have only become pretty good at quasi-anechoic multi-way speaker tuning. And even with that I take the easy way and only use linear-phase xovers via FIR.
The IIR sub-to main alignments that live sound folks have to do are much more difficult than what I do, ime, having done both.

And I totally hear you re being self-taught...for me, it leaves holes and doubts in what I think i know.




That said, let me try to give my idea as to your question and info below.

That is very useful, thank you! I have a question though - if the slopes of the two phase traces are different, how do they get them to overlay over a wide frequency range? As the graph of my system shows, it is only possible to get the two traces to touch at a single point. Acourate has a series of steps where the phase of individual drivers can be linearized, and in fact I have performed that procedure. I will show you.

This is the phase linearization of the subwoofer. You can see the before trace in brown, where it wraps around a few times (vertical scale is in radians). Blue is after phase linearization. These are nearfield measurements, and the blue trace is a measured response, not a sim.

And this is the phase linearization of the woofer. Again, brown is before, and blue is after. Once again ... nearfield measurements, actual measured response.

So now we take the nearfield phase linearized sub, and phase linearized woofer, and place the mic at the MLP. The above is what we get. The slopes are widely divergent. Obviously the room has interfered with the response by quite a lot. Note that these curves are measured phase with no phase extraction, i.e. it shows both the min phase response and the excess phase.

I think that you achieved excellent flat phase for both the sub and the woofer, at each of their respective nearfield measurement positions.

To the extent the phase trace measurements diverge at a new common measurement position (such as MLP (that means main listening pos, correct?),
I think the divergence would be due to physical distance/geometry differences. Time-alignments, fixed constant time.
(This is ignoring the influence/contamination of room reflections, and what I experience when aligning subs-to main outdoors.)

I have to admit I don't fool with the ideas of minimum phase, excess phase or group delay anymore....for the purpose of speaker tuning alone.
After I do a driver by driver magnitude and phase flattening, like you did above, all it takes to get both phase and time alignment correct, is to get constant time delays set.
The delays inevitably end up very close to the physical distances between driver sections' acoustic centers...and provide a bit of a smell test for the overall correctness.

Let's me end up with speaker response like for this 5-way, showing the mag and phase traces for each section at measurement position.
(The top red trace in the magnitude pane is the Coherence of the 750Hz to 4kHz band..ignore it...)

So time alignment is my guess as to the divergence at MLP.

Thanks for your postings on Acourate...someday I may graduate basic speaker tuning and take on room corrections too.
But so far, I'm happy to get speakers and subs right (which I always stack indoors same as outdoors, to know tuning holds up indoors prior to room response).
Then i just knock down room modes with some simple PEWs after determining optimal speaker positions, and tune tonality to taste globally.
It's why i use the q-sys setup...to be able to separate tasks. If i do end up using Acourate, or some other room correction, it will most likely be to do global correction prior to the speaker processing I've already done.

Don't mean to be belaboring what i do in your thread....just wanted to give you perspective on how/whether to value my comments.

 

[3ll3d00d]

Fortunately, I have the measurement with the sinewave convolved, so I can look at the phase behaviour at 50Hz:

No need to do that as that helps with time alignment, can you show the (unwrapped) phase response?

 

[Keith_W]

No need to do that as that helps with time alignment, can you show the (unwrapped) phase response?

Umm, I did? See the post that you were responding to, the one where you replied with: "I think an unwrapped phase view is much harder to read btw because it's so much harder to see the actual slope of each curve."

 

[3ll3d00d]

Umm, I did? See the post that you were responding to, the one where you replied with: "I think an unwrapped phase view is much harder to read btw because it's so much harder to see the actual slope of each curve."

Sorry, brain fail, I meant wrapped.

 

[Keith_W]

Sorry, brain fail, I meant wrapped.

Not sure what you are going to see here (I can't make any sense of it myself), but here you go.

 

[Keith_W]

Anyway, I have put in a reverse AP filter into the woofers and the subs to get the phases to align better.

This shows the unwrapped phase of the left woofer compared to the left sub, before and after. Note that in the "before" curve, the phase slopes of the woofers and subs appear to be converging above the 50Hz XO point. With the reverse all pass filter in place, the two curves run more parallel. The same is true to a lesser extent below 30Hz, but that is getting quite far away from the crossover point so IMO it matters less.

Before: if I align the initial impulse of the left sub and left woofer, the phase alignment goes off. Almost 180 degrees out of phase.

After: If the initial impulses are aligned, the phase alignment looks much better. Not perfect, but much better. So now we ask ourselves whether this has any improvement on the previously observed cancellation at the 50Hz crossover point?

This is a full range sweep of the left channel, before and after (i.e. with the subs and the woofers playing together). The two curves have not been normalized and are slightly different volume. Nevertheless, I see about a 14db improvement just above the crossover point. The dip is still there, but it is no longer so objectionable.

This exercise has proven to me that phase alignment via convolution of a reverse all pass filter can get rid of cancellation at the XO point. I did not manage to completely ameliorate the dip, I could potentially do a better job with the reverse AP filter. Or I could leave it ... after all it's only 10Hz wide Sheer laziness is telling me to leave it. I have spent way too much time worrying about one dip in the curve already!