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[Ivan Beaver]

Rational Acoustics and AFMG do a great job providing training to use Smaart and Systune (respectively).  What they can't do in 20 hours is give the operator the experience of:  making bad measurements, measuring for the wrong thing, or making measurements without a clear idea of why a particular measurement is being made.


Aligning to one spot is just that, one spot.  You can't align for 3 or 7 or 28 places, what you can try to do is minimize the differences.  In the end, there is still one physical place in the venue where things come together in time.  Over thinking the impossible distracts from applying our principles of alignment and optimization to the things that *are* possible.

One of the hardest things for people to get their head wrapped around is "IS this a good measurement?"

Either in position or parameter settings on the computer.  If not-WHY?

I ALWAYS question my self when I see something I don't expect-WHAT is causing it?  Is it the measurement position or the actual device being measured.

As the saying goes "If you don't know the answer BEFORE you start to measure-how do you know you are getting a good measurement?"

One of the things that REALLY REALLY started to help me understand what was going on in a sound system in multiple seats was using multiple mics.

I built my own rig (back when we could only input 1 mic at a time into a measurement system) that allowed me to calibrate my mics and to easily switch between them (without having to plug and unplug each one).

I would spread the mics ( around the room.  That was I could see what was happening at different seats when I adjusted different parts of the system, without having to get up and move the mic around.

It allowed me to get much more information much faster and get a better overall result.

One of the things that many people miss is how much one part of the sound system affects other parts that aren't anywhere near the coverage pattern.  Being able to see it is a REAL eye opener.

And it leads to looking for better tools to use that don't interfere as much.

[Mark Wilkinson]

Thank you Tim, 'Copy' to all you say....

I think a mistake I make in responding to measurement questions, is taking for granted that folks understand common limitations , and reply as if their only question is about best technique. (limitations such as being able to align to only one place unless subs are adjacent to mains)

I mean, I barely even try to align subs to mains indoors in the small rooms I see...I'm almost always far more concerned with how the subs interact with the room and stage.
Placement and cutting out modal boosts with eq get most my attention.  If any tuning time is left, I might try to align, but most often I just wing it with a ball park guess based off known 'stacked' timing.
And in fairness to the video that started this thread, he was talking about inside.......and doing more than I usually do !

Ivan, I really concur with the multiple mic learning experience.  I had the opportunity to build a fairy large audio room (11,000 cu ft) and spend several years measuring it with bunches of speaker/sub placements.  (I still remember the joy in learning what corner stacked labhorns could do hehe.) I set up multiple mics, and used both pink and sine waves.  Pink variations were of course large, but the sines blew my mind...could not believe the variations even foot by foot.  Ultimately, apart from the fore mentioned attention to speaker/sub placement and cutting room modes with eq, the real gains came from acoustic treatment.  Taming early reflections, getting an even RT60, and a few other acoustic techniques made for a great sounding room. 
Never could get smaart to work for aligning inside haha....had to go outdoors to learn how to do it!

On the reality of being not able to align to more than one place when subs and mains are separated...

I've found very good alignment with speakers right on the sub, but with subs on the ground and speakers on a stick 14ft in air, alignment of course loosens up moving away from 'aligned to' location.
OK, that's just the way it works..critical xover region will suffer some as you move around.

But what I'm thinking is why not at least minimize the critical  region of xover overlap...more than typical....
Why not run as steep a x-over as possible between the subs and mains? 
I'm toying with 96db right now @100Hz , and it's kinda cool how bass sounds like it stays with the main despite moving the main around.  This is inside, so no real telling till I get outside....and there's also the added delay issue from steeper xover to contend with....so a lot of dunnos still ..
   
Any experience here? Pitfalls beyond delay?
thx

[Ivan Beaver]

But what I'm thinking is why not at least minimize the critical  region of xover overlap...more than typical....
Why not run as steep a x-over as possible between the subs and mains? 
I'm toying with 96db right now @100Hz , and it's kinda cool how bass sounds like it stays with the main despite moving the main around.  This is inside, so no real telling till I get outside....and there's also the added delay issue from steeper xover to contend with....so a lot of dunnos still ..
   
Any experience here? Pitfalls beyond delay?
thx

One of he reason for not using high order filters is the ringing that happens with really steep slopes.

I try to stay at 24dB/oct or less.

The best "sounding" filter is a simple 1 pole/6dB/octave.  But often it does not provide enough protection for louder systems that push drivers.

If the drivers are not going to be pushed, then shallower slope filters are better.

[Mark Wilkinson]

One of he reason for not using high order filters is the ringing that happens with really steep slopes.

I try to stay at 24dB/oct or less.

The best "sounding" filter is a simple 1 pole/6dB/octave.  But often it does not provide enough protection for louder systems that push drivers.

If the drivers are not going to be pushed, then shallower slope filters are better.

Aah ringing...yes.  Gosh, I've been doing tons of research on this. 
And I get the power handling issue with low slope, thx

I've found what you say about steep filters is very well documented for normal minimum phase filters.
And some of the better explanations  show that ringing is the term used for the phase distortion that occurs with steep recursive filters......that freq response can be fine, but time response gets real sucky. 

Linear phase filters seem to be another animal though. ( I should have said the 96dB xover attempt is linear phase).
I read debate as to whether linear phase pre-ringing is audible.....

So far, I can't hear any pre-ringing...but who knows how well I hear ? 
I can measure a very small amount of pre-ringing, but it really appears inconsequential.

Below is an impulse plot of a main high passed with 96dB at 100Hz.  You'll see a wee bit of pre and post ringing.

Right below it for comparison, is an impulse of an x-32's I/O that was used as the soundcard for the main's measurement, so the impulse of the soundcard is part of the impulse of the main.  The x-32's AD/DA conversion is nearly as ringy as the speaker !

I'm just not seeing speaker ringing worth paying attention to ??
Are there better ways to identify ringing?

[Charlie Hughes]

Aligning to one spot is just that, one spot.  You can't align for 3 or 7 or 28 places, what you can try to do is minimize the differences.  In the end, there is still one physical place in the venue where things come together in time.  Over thinking the impossible distracts from applying our principles of alignment and optimization to the things that *are* possible.

Several years ago I presented a method that I think can be very useful for aligning subwoofers & full-range loudspeakers while minimizing the variations within a large audience area.  A video of the presentation can be seen at the link below.
https://www.youtube.com/watch?v=yoASUFGPWwg

A PDF of the slide is available on my website.
http://excelsior-audio.com/Publications/AES129_RH_Charlie_Hughes_Subwoofer_Alignment_with_a_Full-Range_System.pdf

Hope this helps.

[Keith Broughton]

The "phase wrap" can be eliminated by selecting phase unwrap in the menu.

Now there is a tidbit I missed!
I have quite a time trying to phase align as the "wraps" are so dense, I can't get any good data.
Seemed sooooo easy in the class 
Have to set up a test scenario and figure this out.
Are you referring to the "phase smoothing" in the TF tools?

[Doug Fowler]

It's not Smaart, but....

When AFMG were working on the alignment tool built into Systune, I was looking at every angle to find an easier path for beginners with Systune. 

One thing I know for sure (probably the only thing :-) is beginners don't realize frequencies are arriving at different times, and that low frequencies are substantially behind, sometimes 25-30 msec at the bottom of a subwoofer's range.  Add in the effect of the low pass filter and it gets confusing, quickly.  Here is what I came up with, and actually presented this in the Renkus-Heinz booth at Infocomm a few years back: 

I don't think you can do this with Smaart but it works with Systune.

Display phase in one window and group delay in the other.  Assuming you're going to delay the subs, capture the LF from the mains with long averaging.   Group delay will be jumpy, and possibly show negative time at some frequencies due to the nature of the calculation, which is very sensitive to noise.   Let it average and when it stabilizes, capture it.  At this point you can use delay finder and use that value to get the phase display looking "normal".

Examine the frequencies at which you are going to align, let's say 80 Hz.   The arrival time(s) around 80 Hz will be quite clear.  Now mute the LF and unmute the subs, and let it average.  Leave the delay offset in place.  Look at 80 Hz area and you will see the arrival time(s) from the subs.  Start adding delay slowly until the subs match the captured LF trace. 

Now inspect the phase values in the phase window, and fine tune from there.  It works, trust me.  What we're doing here is using a somewhat unstable calculation (group delay) to get us close, then fine tuning with phase until the slopes match.  The value of using a group delay window is no phase interpretation is required.  It's dirt simple: frequency is X axis, time is y axis. 

If I ever get a chance I'll post an example.  This is off the top of my head, but I'm pretty sure this is how I did it.

[Mark Wilkinson]

Several years ago I presented a method that I think can be very useful for aligning subwoofers & full-range loudspeakers while minimizing the variations within a large audience area.  A video of the presentation can be seen at the link below.
https://www.youtube.com/watch?v=yoASUFGPWwg

A PDF of the slide is available on my website.
http://excelsior-audio.com/Publications/AES129_RH_Charlie_Hughes_Subwoofer_Alignment_with_a_Full-Range_System.pdf

Hope this helps.

Thanks Charlie,

I stumbled onto your youtube video sometime last year when I was going nuts trying to line up impulse peaks on a 3-way main.   
Your video, which i only half understood  , led me to examine a perfect impulse response using 1/3 oct filters.  And then I saw the need to align first arrivals..........

It still melts my mellon that a single narrow spike (perfect impulse) can contain all the exploded impulse responses shown by filtering.
I'm left thinking that what the perfect impulse really contains, is a single sample of each frequencies "first arrival" or rather single sample of initial propagation. And that the exploded responses are just mathematical extrapolations out into time....
Is this thinking even on the right path?  I find this stuff hard to conceptualize....

I also noticed that the 1/3 octave impulse peaks showed a delay from time zero, that appears to be about a constant 2.8X the period of the filter center.  For example, 1000Hz showed 2.8ms delay, 100Hz 28ms delay, etc.  Does this sound right? What's going on?

Sorry, too many questions...! 
Mark

[Mark Wilkinson]

It's not Smaart, but....

When AFMG were working on the alignment tool built into Systune, I was looking at every angle to find an easier path for beginners with Systune. 

One thing I know for sure (probably the only thing :-) is beginners don't realize frequencies are arriving at different times, and that low frequencies are substantially behind, sometimes 25-30 msec at the bottom of a subwoofer's range.  Add in the effect of the low pass filter and it gets confusing, quickly.  Here is what I came up with, and actually presented this in the Renkus-Heinz booth at Infocomm a few years back: 

I don't think you can do this with Smaart but it works with Systune.

Display phase in one window and group delay in the other.  Assuming you're going to delay the subs, capture the LF from the mains with long averaging.   Group delay will be jumpy, and possibly show negative time at some frequencies due to the nature of the calculation, which is very sensitive to noise.   Let it average and when it stabilizes, capture it.  At this point you can use delay finder and use that value to get the phase display looking "normal".

Examine the frequencies at which you are going to align, let's say 80 Hz.   The arrival time(s) around 80 Hz will be quite clear.  Now mute the LF and unmute the subs, and let it average.  Leave the delay offset in place.  Look at 80 Hz area and you will see the arrival time(s) from the subs.  Start adding delay slowly until the subs match the captured LF trace. 

Now inspect the phase values in the phase window, and fine tune from there.  It works, trust me.  What we're doing here is using a somewhat unstable calculation (group delay) to get us close, then fine tuning with phase until the slopes match.  The value of using a group delay window is no phase interpretation is required.  It's dirt simple: frequency is X axis, time is y axis. 

If I ever get a chance I'll post an example.  This is off the top of my head, but I'm pretty sure this is how I did it.

Thanks Doug,

If I get what you're saying...the basic idea is to match group delays at crossover frequency for the coarse adjustment...
And that this should get rid of phase wraps...and then allow fine tuning via phase overlay.

Just trying to make sure it sinks in....

Mark

[Merlijn van Veen]

Regarding subwoofer placement in small venues. Instead of doing trial and error, reverse the roles.

Put the sub at the actual listening position where you want it to sound good, turn it on and start listening at candidate positions where the sub could end up.

The spot where it sound best or the the way you like, is where you'll put it. It's a matter of simple reciprocity.


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