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Time Alignment with Wavelets

boxerfan88

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Starting a new thread rather than side tracking the original DSP related thread.

With reference to this post, I am trying out this recommended oscilloscope method:


I got these results:

Footstool L-R-Sub.jpg


Looking at the waveform of L vs R vs Sub, the shape of Left waveform envelope seems badly mangled.
I re-measured about 1m away from MLP, the shape of the mangling is consistent.
Is that caused by the room and it is normal?

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Me thinks in the home setting, the impulse method seems easier ... just use REW to display the time difference between the peaks...

2025-05-10_1527 roomeqwizard Overlays.png
 
Good on you for trying. I wanted to do this all week but things kept getting in the way. But I have been thinking about it and I was going to start a thread to report my results. Looks like you beat me to it ;)

First things first - I realised that there is probably a lower frequency limit where this method will work. This limit is determined by the dimensions of your room and the position of the DUT and microphone. This determines whether you are able to capture the entire envelope of the wavelet before the reflection arrives and contaminates it. I am ignoring floor, ceiling, and side wall bounce because those reflections are in phase and in the same direction as the direct sound from the subwoofer (in my room anyway!), I am only concerned about the rear wall. Although the reflection itself is in phase, the distance it has to travel might make it out of phase by the time it arrives at the observation point.

1746863699976.png


So I did a little calculation. In my case, x = 3m, and y = 9m. Using the speed of sound (343m/s) and the equation (t = d/1000c), x = 8.74ms, and y = 26.2ms. This means I have (26.2ms - 8.74ms) = 17ms of reflection-free window.

However, the wavelet is 6.5 cycles long, and the time of the wavelet needs to be accounted for when we calculate the reflection-free time. Since my XO frequency is 50Hz, I need (number of wavelets * 1000/frequency) 130ms for the wavelet to fully emerge from the DUT. This is much longer than my reflection-free window, which means that I should see the rear reflection contaminate the wavelet after only 17ms.

If you paid attention to that video, they used a free-field measurement for an elevated speaker array, and a ground plane measurement for the sub. That is why the measured impulses looked so clean - no reflections! If we want to translate this method to a domestic listening room, we MUST account for reflections. It will change the shape of our waveform and stretch it, and make the results very difficult to interpret.

I concluded that if I want this to work, I need a shorter wavelet (fewer cycles) and a higher frequency. And also a larger listening room to delay the reflections, but that ain't happening. I was starting to have my doubts if this method would work at all in a listening room. It can't be used for polarity or phase if the waveform is distorted by the reflection. Neither can it be used for time alignment because of the slow energy build-up of the subwoofer which gives the appearance of pre-ringing. All I needed was an experiment to confirm/deny it.

Now let's come to your graphs. I think that it is heavily contaminated by reflections. Both your L and R graphs have an obvious reflection, you can see the coke bottle shape of the measurement - more obvious in L, more subtle in R. But what disturbs me is that the amplitude of your left speaker is much lower than the right. Either the reflection on the right is arriving so early that it increases the apparent amplitude, or you have some kind of channel imbalance.

Anyway, thank you for doing this. You have saved me the trouble! Even before I started, I had concluded that the method was unlikely to work. It should be great if you are a pro and you want to time align speakers and subs in a stadium, but maybe not so useful for normal speakers and normal listening rooms. Of course, I haven't done anything except think about it a bit. I don't have my own experiment to show.
 
I had a try out of curiosity, same settings, same distance, etc, different frequencies to see the impact.
1 m distance from speaker.

Same size pic so you can see them in a sequence for ease:

80.PNG
80Hz

240.PNG
240Hz (x-over here)

500.PNG
500Hz

Time is visible at the grid.

Edit: pic size fix
 
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If you paid attention to that video, they used a free-field measurement for an elevated speaker array, and a ground plane measurement for the sub. That is why the measured impulses looked so clean - no reflections! If we want to translate this method to a domestic listening room, we MUST account for reflections. It will change the shape of our waveform and stretch it, and make the results very difficult to interpret.

Yes, that was my learning from this experiment. This wavelet/oscilloscope method works better for large halls where first reflections are far far away in time, not so good for home environment.

Now let's come to your graphs. I think that it is heavily contaminated by reflections. Both your L and R graphs have an obvious reflection, you can see the coke bottle shape of the measurement - more obvious in L, more subtle in R. But what disturbs me is that the amplitude of your left speaker is much lower than the right. Either the reflection on the right is arriving so early that it increases the apparent amplitude, or you have some kind of channel imbalance.

Yeah, I was kinda worried too. I repeated the wavelet/oscilloscope measurement with super nearfield setup (i took a snap of the L-channel before running the experiment).

super nearfield pix.jpg



Resulting in:

Nearfield LR.jpg



Phew!!! Nothing wrong with my speakers. It's the room!!!

(Speculation on my part-- maybe the 4-panel windows on the left side causes the bass to leak out, not providing sufficient side reflection, compared to the right side of the room is an entire wall with a door which would probably reflect more bass. So one side has less energy to overcome the reflected wave from the back of the room. Anyways, pure speculation on my part.)


Anyway, thank you for doing this. You have saved me the trouble! Even before I started, I had concluded that the method was unlikely to work. It should be great if you are a pro and you want to time align speakers and subs in a stadium, but maybe not so useful for normal speakers and normal listening rooms. Of course, I haven't done anything except think about it a bit. I don't have my own experiment to show.

Yup, I agree that this wavelet/oscilloscope method may not be so useful for home environment, and more suitable for large venues/Pro environment.

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I might digress here a little bit, but just to share something I have been experimenting with lately:
After applying delay values that I got from performing acoustic timing reference measurements I have applied an All Pass filter at the crossover frequency and playing around with the Q value very quickly resulted in some wonders actually
See here the before/after:
Post in thread 'Dayton Audio PS180-8 full range project' https://audiosciencereview.com/foru...ps180-8-full-range-project.62063/post-2279789
 
Yes, thanks for doing this @boxerfan88! I too have been giving it a lot of thought and watched the video series a couple more times since it popped up in the other thread. I knew from experience it would be difficult to get clean impulses in my room so thanks for crunching all the numbers @Keith_W.
 
(Speculation on my part-- maybe the 4-panel windows on the left side causes the bass to leak out, not providing sufficient side reflection, compared to the right side of the room is an entire wall with a door which would probably reflect more bass. So one side has less energy to overcome the reflected wave from the back of the room. Anyways, pure speculation on my part.)

FYI, windows do not cause bass to leak. An open window will cause bass to leak out - the sound goes out the window, never to return. But a closed window reflects some sound (usually higher frequencies). For lower frequencies, it is absorbed and re-radiated with some of it lost as frictional heat. Some of the energy is lost to the exterior so a closed window acts like a high-pass filter.

I don't know if that double impulse you see on your left wavelet is due to the window absorbing and re-radiating energy. It looks a bit too beautifully clean for that. It would be intriguing to repeat the experiment with the window open.
 
....
.... I am only concerned about the rear wall. Although the reflection itself is in phase, the distance it has to travel might make it out of phase by the time it arrives at the observation point.

View attachment 450029

So I did a little calculation. In my case, x = 3m, and y = 9m. Using the speed of sound (343m/s) and the equation (t = d/1000c), x = 8.74ms, and y = 26.2ms. This means I have (26.2ms - 8.74ms) = 17ms of reflection-free window.

However, the wavelet is 6.5 cycles long, and the time of the wavelet needs to be accounted for when we calculate the reflection-free time. Since my XO frequency is 50Hz, I need (number of wavelets * 1000/frequency) 130ms for the wavelet to fully emerge from the DUT. This is much longer than my reflection-free window, which means that I should see the rear reflection contaminate the wavelet after only 17ms.
.....
Yes, I essentially agree with you.
I have actually observed/performed very similar "identification of sound reflecting wall/plane" using rather extraordinary/unusually strong excitation of room air using 8-wave tone-burst signal of 500 Hz, as shared here on my project thread.
- Identification of sound reflecting plane/wall by strong excitation of SP unit and room acoustics: #498
WS003158.JPG


WS003160.JPG
 
I don't know if that double impulse you see on your left wavelet is due to the window absorbing and re-radiating energy. It looks a bit too beautifully clean for that. It would be intriguing to repeat the experiment with the window open.

Great idea. Repeated test with mic @ MLP. Windows closed. Windows open (all 4 panels).

Result:
  • with windows open, the shape of the "double impulse" becomes even clearer
  • with windows open, the tail (decay) becomes shorter.


WinClosedOpen_markup.jpg



My interpretation/speculation -- the 100Hz bass wavelength is so long, the windows are very thin, the "bass leakage" would be similar be it windows closed/open. A better test to confirm my original speculation is to build a "fake wall" on the left side that covers the window area. Too bad I don't have the means to build a fake wall on the left side to fully test my speculation ... :)

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Can you place "tentative" sound insulators (sound deadening materials) near to the rear wall? Just like I did in my post #498 on my project thread...
WS003161.JPG
 
Can I piggyback on this? I’m also currently trying the different methods listed in the dsp thread.
In the past I always used tape measurement as starting point and then reverse polarity method.
Never really thought about the ground delay before, so I was probably off by few cycles even with good summing.

With my new system it just didn't work, so I played a bit with the spectrogram.
It makes it very visible on one view, could see the ground delay mess, the summing, and with I guess enough precision (not by 1 ms).
It appeared the only pace where I could align with both sides was either below 30Hz or above 70Hz, in between it's a mess with opposite 20ms swings.
Now I'm studying all pass filters to fix this first.

So the spectrogram looks like a very viable option for sub alignment, have you had success with it?
 
Can I piggyback on this? I’m also currently trying the different methods listed in the dsp thread.
In the past I always used tape measurement as starting point and then reverse polarity method.
Never really thought about the ground delay before, so I was probably off by few cycles even with good summing.

With my new system it just didn't work, so I played a bit with the spectrogram.
It makes it very visible on one view, could see the ground delay mess, the summing, and with I guess enough precision (not by 1 ms).
It appeared the only pace where I could align with both sides was either below 30Hz or above 70Hz, in between it's a mess with opposite 20ms swings.
Now I'm studying all pass filters to fix this first.

So the spectrogram looks like a very viable option for sub alignment, have you had success with it?

In case if you would be seriously interested in my primitive but fully validated methods, I will be more than happy to share all the test tone signals (as well as the independent "analysis" software thereof) which I prepared and used in my measurements and tunings; if this would be the case, please simply contact me through PM person-to-person communication.
- Precision measurement and adjustment of time alignment for speaker (SP) units: Part-1_ Precision pulse wave matching method: #493
- Precision measurement and adjustment of time alignment for speaker (SP) units: Part-2_ Energy peak matching method: #494
- Precision measurement and adjustment of time alignment for speaker (SP) units: Part-3_ Precision single sine wave matching method in 0.1 msec accuracy: #504, #507

- Measurement of transient characteristics of Yamaha 30 cm woofer JA-3058 in sealed cabinet and Yamaha active sub-woofer YST-SW1000: #495, #497, #503, #507

- Identification of sound reflecting plane/wall by strong excitation of SP unit and room acoustics:
#498
 
Sure thank you!
Or maybe is it something I can replicate with rew, the burst? Will PM

I also have in mind to study the room later for some treatment, seems a nice way to find the right placement!
So far room simulations didn't work well, the room is just too weirdly shaped.
 
Great idea. Repeated test with mic @ MLP. Windows closed. Windows open (all 4 panels).

Result:
  • with windows open, the shape of the "double impulse" becomes even clearer
  • with windows open, the tail (decay) becomes shorter.

Well, it looks as if the window isn't the cause of the double impulse then. Studying your wavelets again, I can see that the left has a double impulse, and the right has double the amplitude and double the ringing. So I am guessing that the right also has a double impulse, but with the reflection buried in the main impulse. Meaning you have an asymmetric room, with the reflective path on the right shorter than the left.
 
Phew!!! Nothing wrong with my speakers. It's the room!!!

Exactly !!!

When we realize wavelets are the most precise alignment tool for low frequency work we have...and that even they are bogus once out away from speaker and into a small room....
well, how much faith can we have in small-room low frequency measurements.????? Little to none, imo.

Over in the state of dsp thread, i opined that...
in the direct field, close enough to speaker that direct energy dominates strongly, wavelets will show repeatability. They do, and predominantly the direct radiator (speaker).
Further back into the room, reflections enter into the wavelets' scope trace...hence the longer tails of the wavelets beyond the initial direct waveform.
It is what it is, truncate the tails and work with what appears to be the direct arrival....or just say like me, in-room low freq meas are bogus :p
 
Well, it looks as if the window isn't the cause of the double impulse then. Studying your wavelets again, I can see that the left has a double impulse, and the right has double the amplitude and double the ringing. So I am guessing that the right also has a double impulse, but with the reflection buried in the main impulse. Meaning you have an asymmetric room, with the reflective path on the right shorter than the left.

Interesting that you mention that my room could be asymmetric… it isn’t. My room is a square room, 4m x 4m.

Front wall is full brick wall.
Back wall is full brick wall.
Right side is brick wall with a wooden door.
Left side is brick wall with 4 large panel windows.
 
Serious question: is any of this audible? If so, what's the deviation tolerance?
 
Serious question: is any of this audible? If so, what's the deviation tolerance?
Good question and as far as I know there is not a lot of research on audibility threasholds for LF group delay, distortion, filter ringing, and driver misalignment. Common wisdom is our hearing is less sensitive at LF but based on Fletcher Munson curve that may not be the case. I recently switched from 2 small subs with ~30%- 50% distortion and 100 ms group delay @ 20 Hz to a very large 18" sub with ~ 20 ms group delay and less than 1% distortion @ 20 Hz and that change is very audible especially at higher listening levels.
 
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