Q: When is it done?
A: When I'm fed up trying to make it any better.
I think I've finally figured out why my 3 time alignment methods don't match. It's very simple. My speakers do not have perfect phase response. Added to that, my room is not an anechoic chamber, so anything past the first reflections is potentially changing everything past that reflection.
So assuming there there are errors in phase response in either speaker driver at the crossover frequency, those will mean that if you dial in the actual correct time alignment, the magnitude of the frequency response at the crossover frequency will in fact not be maximized. It would only be maximized if there were no phase errors and no reflections, or if in some chance the phase errors happened to cancel out and no applicable reflections.
That problem is going to affect the cancellation method also. Phase errors mean that you will not get perfect cancellation at the correct alignment, but instead you may get better cancellation at some other relative delay which is not the correct alignment.
Now, the full on approach to fixing this is applied by many tireless audiophiles like Linkwitz, who carefully measure the phase response for each driver, and meticulously perfect the phase response of each driver as much as possible with electronic EQ, either immediately before, after, or within the electronic crossover electronics.
I am not going to be using that full on approach in the immediate future. And especially I am not going to be measuring the phase response and having some computer program compute the ideal inverse (as is standard practice today in home theater and the fancy room equalization). My approach is to avoid adjusting to exact measurements, because measurements are never exact or fully revealing. Companies that cook up automated measurement systems put in many person years of research and development time, and it still isn't perfect or perfectly reliable.
Instead, I believe in compensating to the model, not the measurements. Actually if you read Linkwitz you see this is exactly what he does. He figures out what is happening by combining lots of measurements and his understanding of relevant theories, then uses that to model of the drivers or whatever, and then compensates for that model. That's a lot of work and at this point I do not understand anything well enough to model it very well. My attempts to fix things are currently based on a best first approximation.
So it is, then, that having studied several alternatives, it now appears to me the best way for me to do a time alignment is pretty much the simple and intuitive way. I align so that the wavefronts from each speaker arrive at the listening position at the same time. This is where the current time alignment project started, and then I took detours exploring two alternatives I though might give more precision, but instead they just provided more confusion.
This is only complicated a little bit by the fact that the wavefronts themselves have artifacts. The output of every speaker seems to have phase lead and/or digital pre-ringing. The phase lead is possibly caused by the high-pass characteristic of each driver. None goes to DC but some go deeper than others. The subwoofer has by far the longest phase-lead because it has the lowest high pass frequency.
The first best approximation is to align each wavefront at the beginning of it's large positive polarity response to a positive transient impulse signal. (Assuming the drivers are all in-polarity, as mine are, otherwise use the first large negative polarity response.)
That's it. My examination of both the maximum crossover frequency cancellation method, and the maximum crossover frequency summation method, has revealed to me that neither can be used unless your driver phase responses are something like perfect. Good luck with that. Otherwise, they just lead you astray.
This is actually good and useful, because it means time alignment can be done fairly quickly and efficiently as compared with the nearly exhausting summation and cancellation methods, where you must do a large number of tests to zero in on the best summation or cancellation.
The kind of alignment I now recommend could be done with any simple device that generates a transient pulse and records the following sound output--for which a microphone and storage oscilloscope would work. It so happens I have a fancy FFT based analyzer that can do this as well as endless other tricks. But most of the tricks are useless for this simple alignment, except as a cross verification that you haven't messed other things, like the frequency response, up too badly.
No comments:
Post a Comment