TIme Alignment Answer from Linkwitz

Linkwitz explains why I had to delay the signal to the subwoofers to get time alignment with the panels.  At the end of this incredibly informative page on Crossovers.

The woofer's natural highpass behavior causes a phase lead which is probably far from zero at the crossover point and therefore affects the addition of the woofer and midrange outputs.  This can be corrected by placing a first order allpass in the midrange channel which simulates the highpass phase shift of the woofer.

I don't think I like his solution better than mine (I delay the woofer channel instead of phase shifting the midrange channel) but his analysis is crystal clear despite being totally counter-intuitive.

One normally doesn't think of woofers as being a high-pass device, especially subwoofers.  They are supposed to go lower than we can hear after all.  But lower than we can hear, is still infinitely higher than DC.  At some point above DC the woofer HAS TO cut off (except for fan-type subwoofers, and they require massive amounts of power and cannot be made cheaply).

AND, here's the rub.  The lower the low cutoff, the greater the phase lead!  So, woofers will naturally lead higher frequency drivers, since the former have lower low cutoff.

My solution of delaying the subwoofer channel can work perfectly at a single frequency, but not other frequencies because the problem may not be that the woofer is simply delayed...it has a highpass phase lead, which means the delay varies with frequency.  In that sense, Linkwitz' solution is better, it brings the woofer and midrange into alignment at all frequencies.  Mine might be  better because I'm not introducing more allpass behavior in the midrange.

With a reasonably steep crossover, I don't believe the mismatch in allpass behavior is such a problem generally (at least in digitally crossed over systems like mine that can easily apply delay instead), and especially in my system where the "midrange" panels can go nearly as low as the subwoofers.  However, it's worth keeping in mind.  It's an incredibly important insight.  It seems to suggest that my friend who was often criticizing speakers with recessed woofers especially as not being "time aligned" may have been all wrong.  Also explains why one of the least costly "transient correct" speakers of all time, the Spica's, have deeply recessed woofer.  I vaguely recall, back in the day, that a defender of Spica or some such mentioned the woofer lead issue, but I never understood it before at all.  I thought it was the crossover that was causing the woofer to "lead," and in that case the distances still needed to be equal so the crossover would work correctly.  But the problem is, it's not the crossover that causes the woofer to lead, it's the difference between the woofer and the midrange drivers in themselves, their differing low frequency cutoffs.  Ideally both should have identical low frequency delays, but they don't.

However it still troubles me that this "lead" might not be applicable entirely to transients.  That's where my intuitions regarding how filters work seriously breaks down.

It seems to me that information cannot travel through a system in negative time, and that information does not travel through woofers hugely faster than midranges.  Any kind of "phase lead" must represent some kind of information loss, like the front part getting lopped off.  But in saying that, I still can't see how this would apply to the woofer/midrange situation.

Perhaps it's because I'm not thinking of the information loss to the midrange--it is losing the lowest frequency information (because of it's own highpass behavior at least) relative to the woofer.  The lopping off of this low frequency information is possibly causing it to be delayed MORE.  The woofer is being delayed LESS because it is not losing as much low frequency information.

I don't really know how to understand this yet, but it does seem that to work properly in a system with a crossover, the woofer must be delayed even if this means it doesn't start contributing information to the listening position until a later time than the midrange.  And that is what one sees in a decently engineered but nevertheless allpass speaker impulse...in which the high response starts and rolls into the lower responses.  But how would it work with transient perfection?  Perhaps the greater phase correction is needed for the woofer itself than the crossovers.

It seems that electrical engineering has many different abstractions, and issues arise with mixing the abstractions the wrong ways.

Although we are talking about the low pass function of a woofer (Kellogg-Rice dynamic woofer), but actually, as far as the abstract high pass function nature of it, we could just as well be talking about a capacitor in a series-capacitor followed by shunt resistor circuit (following some AC generator), which is another high pass function if we idealize it slightly.

In that circuit, current must lead voltage in the capacitor.  For abstraction purposes, we assume the resistor has no parasitic capacitances and inductances, and as such it samples the voltage at its terminal instantaneously.  (Somehow when people talk about the capacitor in the voltage leading thing, they rarely talk about the load.)

But what does this mean for information traveling from the generator to the load?  I'm thinking out loud here.

At the moment the generator applies a voltage, that voltage appears across the capacitor and the load. But the capacitor is having none of it.  Instead, it all appears across the load.  As current flows through the load, it flows into the capacitor and therefore a voltage appears across the capacitor AFTER it has appeared across the load.  The capacitor is soaking up the signal AFTER it has started. So in the long run, DC or a Step would be quenched all into the capacitor and none into the load.

In what sense, then, is the signal "leading" into the resistor?  In the sense that the maximum rate of rise for the load is at 0 degrees, whereas the maximum rise in the AC signal itself is at 45 degrees.  As soon as the input signal has started, the rate of change in the load begins falling, as would happen at 45 degrees.

The voltage at the load...which is the output of the low pass filter, is therefore 45 degrees ahead of the input signal.

However, none of this means that start time of signal is changed.  It simply starts at 45 degrees rather than 0.  One may be tempted to draw sine waves like arches but they aren't.  The first 45 degrees is a gradually inflecting upwards, starting like nothing.  OTOH, the output of a high pass filter starts at maximum upwards movement.

Now, for a larger capacitor this leading time is longer.  So the high pass filter retains the maximum upwards characteristic longer.  I still find it baffling that this means the larger capacitor "leads" more. It seems to me more that it holds on to the lead longer.

CIVIL: Capacitor: I leads V, and by 90 degrees, Inductor: Lags.

The Capacitor is maximally charging at 0 degrees into the input signal, which means the current is at maximum at that point, which means the voltage across the load leads the AC input signal by 90 degrees, because the voltage across a resistive load has the same phase as the current flowing through it, and because the current flowing through the capacitor must be the same as the current flowing through the load.

This is not to say it's starting any earlier in response to any signal, but that it's starting at the 90 degree point instead of zero for AC waveforms.

I'm still not clear why the lower high pass filter would lead more.  However, if the lower high pass filter leads by 90 degrees of it's cutoff frequency, that would be "more leading" time.  I just don't see how it's cutoff frequency comes into play here.  It should be just 90 degrees.  The capacitor is maximally charging at 0 degrees, i.e. when the signal starts.

And even then, it's still even more baffling how the lower high pass filter leads more then the higher high pass.

The "more leading time" seems to be the correct reason and answer.  The leading DOES have to do with the cutoff of the RC circuit.  It leads by 90 degrees of that cutoff frequency, even if that frequency is no where in the input, that same effective leading is applied to everything along with the associated cutoff.  It's weird but I think I'm beginning to get it.

One again this has little or nothing to do with how fast the input starts, but the phase angles are going to contaminate the step response unless the leading is accounted for...this tends to mean delaying the "more leading" woofer...and it does seem desirable that it should be delayed, and also so that the impulse response "rolls into " the bass, rather than the bass in any way contaminating the leading edge of the impulse.  In fact that contamination is what happens is the woofer is not delayed, as the speaker with the lower highpass cutoff is going to have pre-mature phase angle from a dirac impulse--actually beginning to tend downward as the midrange is still going up, causing cancellation of the leading edge.

The exception would be designing each range of speaker for identical highpass response.  Some very quirky designs do that, not that I'm endorsing them in total, but it's easy to understand the desirability of it now.

The June 2019 TIme Alignment

Finally, as promised (but similar to a long line of unmet promises), here is the report on the living room system time alignment and adjustment near the beginning of June 2019 when I finally turned REW onto the living room system in order to do a measured time alignment which hadn't systematically been done since about 2014...something like forever for this ever changing system.

In May I obtained a Focusrite 2i4 interface so I could do full loopback with REW.  I had been unable to get my historic Emu 0404 to work on Windows 10, so I just decided to get the cheap interface everyone uses now (and which apparently still works on Windows 10, with the status of other interfaces such as the Lynx Hilo uncertain).  I already had a suitable calibrated mike I bought specifically for REW a few years ago.  You cannot use a USB mike during loopback because you must use the same audio interface for both input and output, and a USB mike only has an input interface.

When you do full loopback, in theory, you can simply have REW compute the delay time from the loopback electrical output to the measured acoustical stimulus.  Then, in theory, you can simply dial in the required compensation into your digital processing units.

Using a loopback is also preferred for getting a more accurate computed step response.  REW uses a stimulus which is a sine wave sweep from a low frequency to a high frequency.  The actual response to that is hard to interpret for phase response by eye.  So then REW analyzes the response to this sweep stimulus using FFT, and then it can compute what the response would be either to a dirac type impuse (the "Impulse" response), and a step type stimulus (the "Step" response).  Because it is the simplest possible stimulus, and arguably most informative, speaker designers and reviewers mostly evaluate the step response.  It should always be remembered, however, that REW did not actually measure this stimulus as such, it merely computed that, given the way the unit under test responded to the sweep, this is the way it would have responsed to an actual step.  The computed step response will have artifacts.  But if you were measuring an actual step, it would be highly random because of noise.  The swept sine stimulus, having more broadspectrum (across desired band) energy increases the S/N over what you could measure with an actual Step stimulus.

I had long done such compensation simply based on the physical distances of the speakers as seen from the listening position.  Roughly speaking, the supertweeters are about 1 foot back from the panels, and the subs are about 2 feet back.  Assuming a foot of distance to have about 1 ms delay (I'll use that approximation a lot in this post), I could then compensate by having NO DELAY (0 ms) dialed in to the subwoofer crossover unit, 1 ms delay dialed in to the supertweeter crossover unit, and 2ms dialed in for the panels.   In practice, I intentionally add 10ms to these numbers to facilitate more on-the-fly experiments and adjustments, since you can't further reduce a delay already at 0 ms.  So, a typical adjustment based on distances might be: 10ms delay on panels, 9ms delay on supertweeters, and 8ms delay on the subs.  (In practice, I tried to make the numbers accurate to 0.1ms by measuring to the nearest inch.)

But this is all approximations until you actually do acoustical measurements, especially when all speakers are nothing like point sources.  Real acoustical measurement, not distance approximation, is what has to be done to do the time alignment correctly.  But it's so difficult, and fraught with problems, I haven't actually done it very much.  In principle, I should do a new time alignment after any adjustment of any kind.

It was failure to do actual acoustical measurements (which many subjectophiles and speaker amateurs HAVE NEVER DONE) that led to a huge mistake which I continued for several years!  I started using different DACs on each way, without realizing that the DACs themselves have delay differences, and these differences change at each different sampling rate.  Once I discovered this issue, I quickly gave up on dialing in new compensation when the sampling rate changed.  Too much work.  But that was when I decided to add 10ms extra delay to each path.  However, it turned out that some DACs could add far more than 10ms delay for some sampling rates.  I ultimately decided the best (?) way to handle that variance is to eliminate it by using identical DACs on each way.  I went with inexpensive but decent measuring and pure sounding Emotive Stealth DC-1's (now discontinued) because they had all the needed features: AES digital input, balanced and unbalanced outputs, exact level adjustment.  Audio Science Review puts this DAC down (somewhat unfairly IMO) however if I had to make the choice now, I'd go with the similarly inexpensive DAC they recommend, which has even lower distortion and noise.

This time around, my plan on simply dialing in the REW computed delay for each way was swiftly defeated by an apparent fault with the way REW computes delay for my subwoofers.  I have written a long post about this previously.  I first noticed that the delay times being computed by REW for the subwoofers were about 3ms longer than I had expected.  Then, looking at the computed step response, it appeared that REW was computing the zero point as the beginning of the second hump in the impulse response.  This appears to be an error in REW--while the second hump is slightly larger than the first, it is nearly the same height, and does not appear to be a digital or noise artifact (though there are lots of those, so it's not certain either).  It may have something to do with peculiarities in my system--the rear wall reflection is my current best theory.  This is an example of the computed step response for the Right subwoofer.  The red line shows where REW believes the actual step response is beginning:

It shocked me to see that I got this same result even with No Crossover, which I finally realized I could safely do simply by restricting the range of the sweep to below 300 Hz.  In tests a few years back I decided it might not be safe to send a full range signal--especially a dirac--to the subwoofer with no internal or external crossover being applied.  So I feared sending audio to the sub without crossover.  But this is safe when the input stimulus itself is restricted.  Coincidentally REW defaults to a range below 250 Hz, making it safe to run on most subs without crossover, until further adjustments by the user.  Here is the computed step response with No Crossover:

So, the smaller leading hump which REW's delay calculation ignores has nothing to do with the use of a crossover, which had been the leading theory until I took that measurement (and it's still hard to get out of my mind, because I remembered either lowpass or highpass was supposed to LEAD...but actually it's the highpass that's supposed to LEAD).

Along with the double hump is the extremely curious fact that the beginning of the first hump actually has less delay than the output of the panels, even though the subs are further back.  The beginning of the second hump, where REW seems to think the step response begins, has about 3ms more delay than would be guess from the physical location of the speakers.

Now, REW seemed to give me two choices: go with the officially determined "delay time" which put the sub about 6 feet back from the panels instead of the approximate 3 foot distance, or go with the time of the beginning of the first hump, which put the subs ahead of the panels by about a foot (???)

I was so frustrated by this, I decided to look at something else: the notch in the panel response around 225 Hz which I had long assumed to result from rear reflection/cancellation.  I discovered that moving the speakers further out from the wall could change this a lot, seeming to almost eliminate the problem, and I decided to optimize THIS before doing the actual time alignment.  As I changing the Acoustat position, and looking only at the frequency response btw,  I simply left the delay compensation as I had previously set it by distance approximation.  Starting from the "original" position from a few years ago, which had the 225 Hz suckout, I was getting a step response like this:

Notice the little peak at just over the top of a large hump.  That little peak is the beginning of the Acoustat step response, wheras the hump is the first hump of the subwoofer step response.

When I finally decided to move the Acoustats exactly 6 inches further out (for a total of about 39 inches from the wall) because it removed the suckout at 225Hz in the frequency response I was looking at, the step response looked like this:

You can see the little peak of the Acoustat step response moving forwards in time, as the Acoustats are being moved closer to the listening position, just about to the highest point of the first subwoofer hump.  But even moved 6 inches forwards (which is about 5 inches closer to the listening position) the subwoofer response still seems to start first.  Notice you can also see the little wiggle where the supertweeter is starting about 1 ms later than the Acoustats.  At this point I had not yet moved the supertweeters forward to match the Acoustats, which I couldn't completely do anyway, and it appears their initial adjustment was slightly wrong also, but less than 0.5 ms as you can see in the earlier graph where the supertweeter start is a bit more subtle.

And now, for something completely different, here is what the Acoustat step response looks like just by itself (with the low and high pass crossovers I am using):

It pretty much starts with a spike upwards, as it should.  Somehow the amplitude appears much greater than it does in combination with the woofer response, partly that's due to normalization and the curious nature of step response computation.

I measured and stared at the subwoofer step response for a long time, and finally made a wild guess about where the beginning of the woofer step should have been computed by REW.  I then used the computed panel delay, and MY computed subwoofer delay, to figure out a correct delay compensation.  Sadly I was not taking notes about any of this, it was just a guess, and I tried it, and this combined Acoustat and Sub alignment is about good as anything I've done since, which have only been minor excursions away from this point anyway:

The Acoustat step now merges with the woofer step so that the intial rise is a tall spike, as it should be.  It still possibly looks like there is some preceding bass response, but that's down by at least 40dB, and is probably just background noise and artifact.  I could not make the step look any better by delaying the bass much more than the above.   I tried adding a bit more delay, and the step got worse:

OK, so whatever that was, it looks like too much subwoofer delay has pushed the initial Acousat spike down into the hump in some bass noise valley that precedes the actual bass response, which you can see kicking in a few ms after the initial spike with a blast.

Comparing these pictures begins to get subjective again, I admit, and I wish I had something more solid to latch on to.  It did seem to me that I could try to maximize the length of the leading edge of the spike.  That would indicate that all drive units are contributing maximally to that leading edge.  Now, since REW always places the top of the spike at 0dB (btw, I have not yet calibrated the Focusrite interface itself) maximizing the length of the leading edge curiously means that the leading edge should "start" as low as possible.  You can see that operating in the above two graphs.  In the first graph, what I am calling the noise floor joins the leading spike at -46dB, with the top of the spike at -7dB, so I would count this as showing a leading edge of 39dB.  In the second graph, the noise floor joins the leading spike at -30dB, so the leading edge of only 23dB, clearly inferior.

What isn't so easy to interpret, however, is the the negative spike excursion below the joining point means.  I think it is mostly artifact, but also should be as low as possible indicating complete cancellation or some such.  Anyway, as the delay is adjusted up and down the distance downward changes and also a "gap" seems to open up between the preceding bass-hump noisefloor and the leading edge of the step, making even more a subjective call each time.  As I can't remember the actual adjustment for each measurement (my poor note taking on display again) I'm just going to show a bunch of them, in the order I took them, and where I stopped and why.

Trial 3 has leading edge 24dB, very poor.

Trial 4 has leading edge 30dB (but look at how low it goes also--actually to the bottom of the full scale not shown).

Trial 5 has leading edge 28dB, but deep spike again.

Trial 6 has leading edge 47dB and deep valley beforehand, best so far.

Trial 7 has leading edge 32dB, very poor.

Trial 8 has leading edge leading edge 31dB, very poor.

Stepping back in opposite direction now, Trial 9 has leading edge about 35dB.

51dB !!! This appears to be the winner, especially if you take the beginning of the initial spike at -58dB, then the initial spike is 51dB.  The supertweeter is already being added in and that might account for some of the wiggle at the start--it appears pretty well aligned too.  Alternatively, you could interpret the beginning of the spike as high as -48dB, in which case this is not the winner, but close.

With hand wave you could almost say this is almost looking like a pretty good step response, with a tall initial spike and pretty well filled in right after that.

This is clearly not as good, a step backwards, with the leading edge at no more than 40dB.

After the 11th measurement, I believe I went back to the preceding one, Alignment 10, as the best.  I can't be sure of this, because I didn't take notes, so it might have been one of the other pretty good ones, such as The First Guess.  But I'm pretty sure it was the 10th, because I could still remember the values for the 10th when I did the 11th, and the 11th was clearly a step backwards.

The curious delays in the DSP boxes to achieve the best alignment are:

Subwoofer: 6.3 ms delay
Supertweeter: 4.6 ms delay
Panels: 5.3 ms delay

So the subwoofers are delayed 1 ms MORE THAN than panels, though they are also about 2 feet further back.  That is the second issue I don't understand.

I have already ruled out the lowpass crossover as causing this.  It also probably not because of delays in the complex interface box of the Acoustats, because the delays required to align the supertweeters and the panels is about what one would expect based on distance.

This should not be confused with the first issue I don't understand, which is why REW is apparently computing the subwoofer delay time incorrectly.  This is an entirely separate issue, as I did my alignment by the graphical means shown above, and NOT by using REW's delay estimates, especially for the subwoofer.  However, there might be some connection I cannot yet fathom between the two.  Or maybe not.

Now some might find it outlandish to write so much and not say a word about how it sounds.  Of course it sounds wonderful!  My "electrostatic" bass is now even more well integrated into the sound, without being the least bit less impactful.  Bass lines are easier to follow.  And in many other ways it sounds better overall.

But you should expect that I would feel this way, and it's probably not best to fully trust the audio judgements of the audio investigator, and probably not the measurements either.

Next month it could be different.

Weird Woofer Step Response in REW

I noticed and mentioned this strange problem over a week ago, after I had decided how to do the 3 way system time alignment on the living room system.  I do not trust the Delay times computed by the Room EQ Wizard (REW) program for the subwoofer output.  I found a different way of doing the time alignment using the summed response, but more about that in a future post.  In this post I'm just going to show the problem with the REW computed delay time, which is a very serious issue in my opinion.  I can imagine many people using the delay times to time align their system, as I almost did, and it would be wrong, at least if it's like mine.  (I don't really know what is causing these weird results, perhaps it's something that mainly applies only to me.  As I am writing this post I've completed about six different experiments to try to get REW to calculate the subwoofer delay correctly, but none have worked so far.)

Examining the (computed) step response REW is showing for the subwoofer shows the problem with REW's delay computation.

Notice how the step response begins (that's where it clearly rises above the noise) with a series of humps.  The first hump is not as high as the second, and the second one is the tallest.  FWIW, the black line, which represents the Schroeder Integral, seems to start pretty much where I would consider the bass step to have begun.  But that is not where the computed delay time would lead you.

The computed delay time, which doesn't show up in this graph for no good reason, but rather in an information box for "the measurement" in the REW program, is 12.05 ms +/- 0.042 relative to loopback.  (I am taking loopback at the line loop output of the subwoofer, so it is seeing the same signal as the subwoofer amplifier.  The crossover inside the subwoofer is turned off, so this signal has already been low-passed with 24dB/octave Linkwitz Riley.  I have tried taking full range loopback and it makes no difference.)

If you find 12.05ms on the graph, it is within the rise of the second larger hump, not the first hump.

Now the impulse shown above had a long line of predecessor measurements, which looked basically identical.  I first did loopback using the full range output of my preamp, where it's the easiest to tap off the loopback for my 3-way system.  If I take the loopback at the speaker inputs of the Acoustats, it doesn't include the bass or super tweeter signals--that didn't work at all when measuring the subwoofer by itself.

So then I got more sophisticated, and took the loopback at the subwoofer itself, which has a plate amplifier.  This plate amplifier conveniently already has a balanced "loop output" which I routed bact to my Focusrite interface (it required the attenuator being straight up to not clip) which is also handling the microphone input and stimulus output.

Well then I suspected that the double hump might represent the EQ, of which I am applying 8 additional typically very narrow band and up to -12dB notch corrections to the subwoofer room response.

Not wanting to erase my evolved EQ settings by mistake, I waited until I had time to photograph all the settings in my 3 Behringer units, THEN I saved them to memory (not wanted to do that--which might cause overwriting the current settings by mistakenly selecting "load" rather than "save" until I took the pictures first), then I turned off all the EQ's except for the two HC's which represent the linkwitz riley 24dB crossover.  Then I power cycled the Behringer DEQ (because sometimes it doesn't seem to immediately respond to changes in HC or LC filters, until power cycled) with the subwoofer turned off, then I turned the subs back on.  That's what the picture above represents: No EQ except the crossover itself.

Disabling HC eq's in the Behringer is tricky.  It doesn't want to turn HC or LC filters off when you simply turn them off with the controls.  The "cut" EQ's are "sticky" and remain turned on.  I tried power cycling and even that didn't always work either.  This led to several days of confusing measurements.

Finally I figured out how to shut the EQ's off, AND how to verify the number of HC filters you have enabled (both are important!).

The verification is done by using the REW "generator" function to play a fixed 200 Hz sinewave signal at a convenient level, not too loud.  I set it to -16dB.  The actual subwoofer should be turned off, but even if it isn't, -16dB isn't bad.

The most foolproof way to ensure there are no HC filters enabled is to use the Memory function to reload the "Initial Data" of the EQ.

After that is done, I go to the Level display, choosing in particular the bargraph which has the greatest resolution.  With no EQ active, the 200 Hz tone output level (analog or digital) should be the same dB level, within a few 0.1dB's, of the input level.

When one HC at 100 Hz is enabled, the level at 200 Hz drops about 12dB.  When 2 HC's are enabled, the level at 200 Hz drops about 24dB.

Then, the second HC can usually be disabled by turning it off, then pressing the "reset EQ" button in the PEQ second page.  If that doesn't work, power cycling might help.

When there is only one HC enabled, turning it off and pressing "reset EQ" doesn't seem to work.  In that situation, the working solution is to reload the initial data again.

Until I developed these strategies, I was beginning to fear that that two HC's couldn't be run at the same time because the result did not seem to be a steeper slope than just one.  (It turned out, I was not correctly turning off the second one, so I was comparing 2xHC with 2xHC.)  Then I feared the subwoofer itself (Ultra PB13) had serious rolloff above 100 Hz even with no HC filters (and the crossover in the subwoofer itself has always been turned off).  That resulted from not correctly getting both HC's turned off, which may require that loading the inital data step I just mentioned.

Anyway, when I finally figured out how to set the crossover correctly, and verify it, with all these variations, the subwoofer step response stayed fairly similar, and in every case the "delay time" seems incorrect in about the same way.

Here's the step response of the woofer measured with 24dB/octave crossover again, this time slightly differently but the same as in the next two graphs.  Because of how I captured this image from the "Impulse" and not the "Filtered IR" tab of REW, the Schroder Integral is not available.  The leftmost red dotted line is showing where the delay ends and the step proper begins (this info is shown in this unfiltered version, but not the "Filtered IR").  It is showing the beginning of the step at the beginning of the second hump, or 12.5ms in this measurement using loopback from the sub loop output.

With the 12dB/octave crossover, the humps look thinner somehow, but the beginning of the step is still being calculated to be at the beginning of the second hump at 12.5 ms.

With NO HC filters (no crossover!) the humps are even narrower, with the first hump turning into two humps, and now the computed delay falls at the beginning of the third hump, at 15.3ms.

In no case does REW compute the delay to begin at the beginning of the first hump, but that's where my other method seems to land for the time alignment with the beginning of the panels and the super tweeters.  The step responses for the panels and supertweeters does not have this issue at all, the computed delay begins right where the first hump rises from the noise.

Attend Live Performance, Or Not?

Somehow, I have always strongly believed that it is beneficial, to mind, heart, and soul, to attend live musical performances.  Especially, what I feel is the grandest of them all, symphony orchestras, but YMMV.

It has always seemed to me that if you listen to a live type of music, its best in many ways in an actual live performance.  (Not all kinds of music are like this.  Some music is a form of art that takes months to years to fully realize in a recorded medium--it never existed "live" anywhere.)  The live performance is the real thing that a reproduction is merely a fairly limited copy of.

In a number of ways, the true quality of live music cannot in principle or practice be reproduced, even by the best systems, and certainly such systems as you or I know, don't come close.

1) Dynamic range...the awesome dynamics of a live performance are hard to believe beforehand.  Even a bit of unnecessary audience noise does not usually obscure this...the awesome low and ethereral background noise and reverberation, contrasted with the mind blowing yet effortless peaks.

2) Lack of interference from in-room reflections and modes...the reflections you hear in a live concert ARE the awesomely low ones of that concert hall itself, which could only be superimposed on the annoying short-delay ones in your listening environment that tend to obscure the various lines of music.  The best of the best is the famous Tanglewood.  It's entirely open except the front where the orchestra is, and the top, with a higher roof than can be imagined beforehand.  The result: complete freedom from audible reverberation.  Never have I heard orchestral sections with such clarity.  But generally Symphony Halls are very much better than your living room anyway, Tanglewood is just even better than that.

3) Lack of all kinds of audio distortion (any deviation from the original): including accurate pitch (now assumed with digital media but it was not always so), lack of wow, flutter, and jitter (though I think digital jitter is vastly overrated with post 1997 equipment, some earlier equipment was obviously not good, now it's superstition), noise, harmonic distortion, intermodulation distortion, (I'm not including the various TIM, SID, etc, because I believe those are properly already part of and included by harmonic distortion and intermodulation distortion) and dynamic modulations of the above (dynamic modulation of the noise level, for example, being more than just noise itself).  People think these have all been conquered.  The best electronic equipment is probably good enough already, but the inevitable speakers are not, FAR from it, regardless of hype or salesmanship.  Nobody denies that speakers can be distinguished blind by trained listeners.  And there are the microphones too, they are far from audibly indistiguishable as well, in any honest account.

Now, some may argue that audio reproduction is not perfect, but it's fully "good enough" never to benefit from live performance, which has it's hassles and irritations as well.  But it's not clear to me, how you would know that, without having spent much time at both, and when you find people who have done both, they will always say if you love recorded live music, you cannot merely experience it reproduced, you MUST experience live music live also, as much as you can.  Such people, including me, will agree with all I've said so far, and may introduce other factors as well:

4) Learning from others.  When you attend a live concert, you somehow selected the concert.  But you didn't necessarily select all of the music (or any!).  And within he musical slections, there are also infinite options: such as to play this phrase louder or softer or faster or slower or with more vibrato...  Ultimately, when you attend a live performance, you are giving up some degree of control to someone else, who has selected the music and chosen how to play it.  So--you are experiencing someone else's artistic choices.  This is good.  This is how to experience new things, to learn and grow.  Every performance is different, but that does not mean any are wrong, just different.  (Sometimes there are wrongs too, but they are so small usually I hardly notice if at all in professional performances.)  So each new concert is full of learning, as I have said, for the mind, heart, and soul.  Even if it's a work of music you've heard before in it's most definitive performance (of which, there are usually many too).  This is true with recorded music also, but you are less open to new things with a piece of recorded music you've heard before.  Even within the range of all the recorded versions of something, each new live performance is a new thing, from which new aspects and angles can be learned.  And when this is being driven by someone elses choices--it brings greater connection to the world (and we're just talking about the learning aspects of this so far) at the present moment.  (Understandings of and intepretations even of music written long ago are constantly evolving.  Your favorite iconic recording from 1958 can only reflect the understanding o that time.)

5) Gathering with others, others who share your love of music and live performance.  This is not promoted in the USA as well as it was in UK, in my very limited experience there.  When any performance ends at The Albert Hall, the bar there opens up, and people hang around for and hour or two afterwards, possibly meeting new similarly minded people.  That is how it SHOULD be, IMO.  In the USA, when the performance ends, the venue closes, and there are no additional chances to gab with others previously unknown.  However, there is the general issue in the USA that the lack of public transportation means that people should be fit to drive.  There may be generally smaller groups in the USA as well: more singles.  Anyway, the gathering thing even such as it exists is still important, I think any concert fan will tell you.

6)  Supporting the art.  Great recorded performances of live music aren't created in a vacuum, but in an ecosystem of development which is more supported financially and otherwise by live performance fees than recorded royalties, and the dispersion of this ecosystem of development, keeping it aligned with the heart and soul of every community--absolutely demands local live performances.  In effect, those who listen to recorded music alone are "free riders" taking advantage of the musicians and the development of music itself that live performance actually makes possible.  Furthermore,  even more "concertgoers" are essential in that development, by responding differentially to what resonates with them.

Now, some people aren't impressed by such things as 4-6.  They express  "I already know/have what I want" and "What's in it for me" attitudes.  These are short sighted and selfish,  and some people may be shortsighted and selfish in one or more ways.  There is nothing positive to be said about short sightedness or selfishness, however, and it would be better to help dissuade people from such mindsets, with better music education for example, for starters.  I myself am writing this post to help understand these issues and persuade as well.

A critical thing for me is (4) and things related to it, and a broader theme I've been hoping to develop:

There is isn't really only one Absolute Sound.

I mean this in a variety of ways.  Firstly, and with greatest import, we never experience anything the same way twice!  Each experience is colored by the memories and thinking that goes with it.  If we listen to something a second time, our memories and thinking have already been changed by hearing it the previous time, as well as all other thoughts and experiences in the interval.

Orchestra Maestro's often exploit this fact with a trick, they play every totally new work of music (which the audience will necessarily never have heard before, because it has never been played before) twice.  Sometimes without warning.  It is always said, and I have always experienced it, as not just a different piece of music, but totally different in a large number of ways.  Perhaps a different genre, for example.  It is then a shocking surprise that it was actually the same work of music, played the same, as much as humanly possible.

Now each new performance, or even audition, will highlight different things.  This is all the more true if there are deliberate or unavoidable changes.  This change in the highlighting of different things is not a disadvantage, it is a path to greater learning, personal resonance, and attachment.

It is claimed by some that music is art, but reproduction is just engineering.  This is wrong in many ways, but principally in that it reflect this obsession with the Absolute Sound, as it was laid down, in one way, at one time.

But the very essence of the true absolute sound, is that it is NEVER exactly the same.  So in freezing a slice of the past, and not permitting any deviation, we are also destroying its very essence.

Thus I have already argued that at least in this one way, analog reproduction, through turntable or tape, is more like live sound precisely in the fact that it is never twice the same.  Everytime it is slightly different, highlighting things slightly differently, and helping to enhance a broader understanding of the music itself, even if recorded at one long lost slice in time long ago.

Now all this being said, I do appreciate the effortless noise and distortion free character of digital recorded sound anyway.  I'm not saying we need turntables (though some people do) to stir things up.  But what I'm trying to open people's mind to, is that there is an advantage of listening to the same thing (not that other things might often be better) in different ways.  Not always on closest-to-perfection system A, but pleasant and surrounding system B, for example.  Neither one is an absolute, but they complement each other by shining light through different facets.  And certainly neither is perfect either.

Ultimately, the absolute, is only in one's mind, and it is the sum of all the experiences of something, different or similar as they may be.

I also feel strongly that as the ultimate executive designer of all my audio systems, I am an artist also, extracting from cans and making virtual performances, each one different.

I was going to say a lot about how perception itself is not a passive thing, music does not strain through us like as a feeder fish, but a constructive thing, each time we construct the sound and the meaning and from only a small part of the available information, and never exactly the same parts.  It is this ultimate construction, in our minds and hearts, that is the ultimate art of music, and it includes every being in that chain, including ourselves.

Now, certainly, in a live performance, you have a whole variety of tools you can use.  Every slight change in angle of your head, for example, brings a different set of information of which a stereo recording can only have one version.  You are choosing how to sample from a much larger sphere of information.  And your eyes and body are part of the whole sensation as well.

So the difference between recorded and live is as that between swimming in a fishbowl and swimming in an ocean, as far as the variety of direct information that is available.  A reproducing system can, at best, only produce the perfect fishbowl, and the truth is, nothing even comes close to that.  Audio reproduction is fundamentally an illusion, not true recreation of an infinitely complex soundfield.

Dolby Pro Logic IIx vs DTS Neo 6

I've been playing the various surround formats in my kitchen system and comparing them, as applied to two channel sources, primarily FM radio, classical and jazz stations.  I have an outdoor antenna, an ultimate state of the art Kenwood L-1000T FM tuner, and a Yamaha HTR-5790 receiver from 2005.

Contrary to other self-appointed experts like Stacy Spears, so far I find DTS Neo 6 to be the most pleasant sounding of the 30 or so available options.  All the various Dolby options like PL IIx Music, add some kind of hashy grain to the front stereo channels if not the rear channels as well.  I do not use a center channel speaker, I believe that is unsuited for music and refer to people like Siegfried Linkwitz to back me up.  I set in the proper listening spot for stereo, and a center channel speaker would block my video monitor too.

I suspect this is because Dolby surround processes use various complex "steering" and "feedback" algorithms to force sounds to particular speakers.  This "works" as reported by Stacy and others in moving movie soundtrack sounds to the correct speaker better than other systems.  (It might be helping that movie sound is basically made for if not by Dolby processes in the first place.)  Dolby also applies non-linear compression to the surround channels and perhaps elsewhere, so they don't "stick out."

Well, that may be fine for movies, but it's crap for music.

In contrast, it seems, and I haven't looked at the Wikipedia article yet (one of the few places where you can often find information instead of just hype and opinions) but it seems like DTS Neo 6 may be pretty much leaving the front channels alone.  Well, it would make sense that would be an appropriate thing to do for stereo music sources, especially if you are not using a center channel speaker, and you care about the sound of the music.

Actually, DTS may not be leaving the front channels alone, in the Wikipedia explanation it claims to have 12-19 channels of steering.  Well I don't hear much more than ambience, I think somehow it's less "forced" than Dolby PLIIx, and may use less audio-quality-corrupting steering processes, or simply use them less.  DTS the company was established as a "higher quality" competitor to Dolby and is occasionally recognized as such though not by all.

This is a preliminary finding, but I felt moved to get it out there because it has always seemed to me that Dolby is taking over the world, not necessarily because they are the best, and they have endless sycophants to back them up.  So I want it to be known that there are some who don't see it this way, even if their critique is not yet fully validated.

I wish someone would make a surround system that simply maps, using only phase information as primitive matrixes do, any number of input channels to any number and placement of output channels.  Without any nonlinear processes such as steering, feedback, bucket brigade delays, or Dolby Noise Reduction.

I see now that in competition with Dolby Atmos, which gives hemispheric presentation but "requires" certain layouts, DTS has DTS X, which is more flexible, lets you set up speakers anywhere within the hemisphere, and encoded sound objects in the datastream are mapped to them using an open and license-free framework.  This is great, and would be what I want for discrete...AND for simulated surround from 2 channels.  But does it also process 2 channel inputs to add simulated surround?  It appears not.  I may still have to do that myself.  But there is apparently Neuro DTS X which can translate "legacy" sources into DTS X, so maybe they have all the pieces for this now.

REW on the living room system

To start the month of June out with another big but long, very long, delayed project, I decided to try to tackle the time alignment of the living room system using my newly learned tool REW.  This is, of course, my number one serious system, with subwoofers, "full range" electrostatic speakers (which are crossed over at both ends by me btw, I am not a minimalism true believer obviously, though I tried the opposite approach for many years), and super tweeters, with each crossover and numerous parametric EQ corrections, each "way" handled by a separate Behringer DEQ box in pure digital, then to identical DACs and suitable amplifiers.  The speakers for each channel are clustered but they are not coincident, so at the listening position each contribution would be offset by the different distances, if not other factors.  (Previously, I didn't much consider these "other factors," but if there's one lesson to be learned this time, those other factors may be the important ones.)  I can easily compensate for the different delays by adding additional delay in the DEQ boxes to compensate, making the total delay for each way exactly the same as experienced at the listening position.

Ever since the living room system got "serious", when I got the Acoustats and the TacT 2.0 RCS digital preamp, I started using the Tact to measure, adjust, and align my system.

I didn't like the actual room correction part of the Tact system (however, I'm not claiming I gave it a totally fair trial either, but I did try it a couple times, though I am predisposed against such "automatic" things).  But the frequency response measurement done by the Tact prior to generating corrections generally seemed quite good, with the strange quirk that you needed to look at two different spectras to get the whole picture--which could only exist in your mind.  The two different spectras actually both covered the entire range, but with different emphasis.  Why it did this, or what the two spectra really meant, was never clear to me, but you did clearly get more bass resolution in the LF response curve than the HF response curve.

But for time alignment, the Impulse Response the Tact showed was not very clear in showing the correct time alignment.  It was also quite hard to interpret, once again with two different versions.  But twiddling the delay knobs on the DEQ's for each frequency range, I felt, sometimes, like I had, probably, made it better.

It was still very troublingly equivocal.  I went through this exercise several times, and always meant to make a complete post showing all the complexities, but in the end, I kept away at it, like the proverbial gambler, hoping for the big win instead, the method that would clearly and intuitively reveal the correct time alignment, but never catching anything that was completely intuitive and self evident.

So each experience was like the the battles described by Stephen Crane in Red Badge of Courage in the sense I got so lost in it--the fog of war.  Which was good in some ways, but not so good in developing an ultimate understanding and methodology of how to do it, or even an honest feeling of certainty that I had finally done it right this time.  Just that at some point, each time, I found some little fit somewhere in the impluse response, where I at least imagined it showed the bass, midrange, and highs all starting at the same time, and I called it done, without coming close to being able to prove it was right, I just didn't want to look at it anymore.

Each time I thought, maybe, my final little trick needed a writeup on this blog.  But I could never get myself to do it.

I recall that in the end, the most trustworthy way of aligning the subs and the panels was to make separate impulse measurements of each, and get the signal starting time (shown on a fixed scale by the Tact, relative the the signal it is sending) of both to be the same.  It was easy to see the starting point of the panel transient.  It was not so easy to see the beginning of the bass impulse, because it starts very small and blends into the LF noise of the room and microphone.

The time scale wasn't fine enough to do this well for the super tweeters, and the crossover settings changed the appearance of the initial 0.1ms of the impulse quite a lot.  (I had many different crossover settings to choose from earlier on because I started using the DCX crossover units instead of the DEQ digital eq units.  Now I can only cascade 12dB/octave low or high cuts to achieve the crossover filters I want.  I switched to the DEQ's mainly because only the DEQ's have full digital I/O and I'm no longer stuck using their lousy built-in DAC's.)  At one point I felt I had "nailed" the timing and the crossover of the panels and supers for a near perfect impulse start.  But, I could never get myself to get the pictures online, also fighting with different kinds of smart phones and the demise of the picture sharing service I had originally used.  And now I no longer use the DCX which was far superior wrt time alignment and the crossover selections, but I felt I couldn't live without digital output.

Other methods turned out worse.  I could never get one of my old scopes to proper lock onto pulses, and the pulses themselves led to problems with my amplifier (then, the Krell), my Acoustat fuses and transformer, and differing with different DACs.

So now, I am leaving the 2001 Tact system behind (not really, in fact I think I'll try it again soon) and using the thoroughly modern, up-to-date, and free REW, everything is easy and fine, right???

Well, no, I'm back in a trench, but now it's even more unfamiliar.

Sorry to disappoint, but this is merely the INTRO!  I'll get the the details, and pictures (finally) I hope, in future installments...

It was clear from the unclear computed Impulse "Step Response", this was going to be another hard slog.  But it was also frustrating to again see what I'd long presumed to be a rear/reflection/cancellation at about 230Hz.  After a few measurments for the time alignment proved frustrating, and there was a curious and alarming notch in the computed step response at a couple ms, I though maybe I could fix that by something that had long been a no-no-not-again: moving the speakers.  (Note: it didn't fix the notch in the impulse, which remains unexplained.)

Yes, moving the speakers away from the wall.

Now some time long ago I had established the current position as the maximum distance away from the wall that was consistent with actually being able to walk through the hallway that adjoins the living room right at the front where the speakers are.

Any more, and I'd have to move the speakers back every night, and I am not about to do that sort of thing, I think, ever.  I wouldn't want "oh, I can't play the stereo now because I have to move the speakers" ever have to get in the way of actually using the speakers, though I find endless other excuses.

Well things may have changed slightly since that last major repositioning, which was 2015 or so.   One bookcase has been removed and the remaining bookcases moved back, so, viola, I can move the speakers a little further out into the room, as recommended by dipolar speaker experts, fans, and even Acoustat themselves, which suggested a "minimum" of 3 feet which I was just skirting at 33 inches, thinking myself to be an expert who could cheat a little.

So, I tried moving the right speaker out from the wall about as far as I could (currently limited by the speaker cables) about 8 inches.  It made a huge difference, and the notch at 230Hz notch completely disappeared (or at least appeared to) in the spectrum.

Backing up, I can now consistently measure each change (and this consistency means almost as much as getting true accuracy, which I don't necessary claim but try my best, using the correction curve with my calibrated microphone).

I found that as little as 2 inches out pretty much fixes the 230Hz notch, leaving only a small dip, and 8 inches obliterates it and makes a peak near 200Hz instead.  I settled on 6 inches out, which seemed to be acceptable for room passage also.

Now this 230Hz notch has been on my mind for a couple years now, and I had been planning to attack it using an entirely new set of EQ units, feed a separate 230 Hz bandbassed signal into the subs, etc.  I was just about to start ordering a new DEQ, since the previous spare has now gone into adjusting the back surrounds in the kitchen system--level mostly.

But if this approach works, of course, it is better (and simpler too).  There are some issues remaining, it's possible I didn't display the spectrum with enough resolution to see any remaining notch.

I then discovered the left speakers didn't have the 230 Hz notch problem at all.  But I dutifully moved that Acoustat out by exactly the same distance to match it's partner, which actually didn't seem to change the response much on that side.

Listening has also confirmed, this is a huge change for the better.  And it really didn't require REW at all, but REW gave the final kick in the pants, and it also showed how easy the problem was actually to fix, something that might not have been as obvious by other methods (such as using the 1/6 octave RTA in my smartphone, which is what I've mainly done the past few years, but it often isn't consistent enough to produce firm conclusions).

So, yes, I'm chalking up this enormous yet super simple improvement primarily to REW, and just part of using REW on the living room system for the first day, after attempting to do the time alignment again appeared so frustrating.

The second day, I did get back to the time alignment, which continued on the third day as well.

I am now using a separate audio interface with my laptop, a Focusrite 2i4, so that I can get a "loopback" which is required for REW to compute delay times.  I thought it would be as simple as reading off the delay times after measuring each driver.

Well indeed the delay times computed for the Acoustats and the supertweeters seemed perfectly accurate, about what I expected, and it seemed possible to do it this way.

However, the subs were again a sticking point.  The delay time computed for the subs appeared to be about 3 ms higher than anything I had ever dialed in before.

I took a look at the actual sub impulse response, and noticed something very peculiar.  The impulse response of the subs begins with two humps, and for some reason REW decides that the beginning of the subwoofer response is the beginning of the second hump.  It seems to ignore the first hump completely.  If I examine the entire dynamic range, the first hump seems to arise out of the background noise, so it does not appear to be an artifact.

Now, curiously, the beginning of the first hump is about 3 ms EARLIER than the panel response.  I do not understand why this would be true, and all the explanations I've come up with seem a bit shakey.

After a vast number of additional tests, however, I've decided that where the alignment of the bass needs to be done is in fact at the beginning of the first hump, not the second hump that REW identifies.

Listening seems to confirm this as well, never has the "you are there" feeling been so solid, and percussive instruments and basses sound much more real now.

One test that clinched it for me was noticing that if I started the panels and subs as I did before, based on the distance between them, the step response looked like a tower on top of a small mountain.  I presumed the small mountain to be the bass response, which was apparently already under way.

Actually, the trick method for doing the final adjustment was simply this: I adjusted until the straight line of the beginning of the step response went as far down as possible.  This actually makes the initial step as long as possible, but you can't get that result by looking at the height of the step, since the height is normalized it seems always the same.  But you can see how low it goes beforehand, and where it goes the deepest, you have dialed in the correct time alignment.

The point where the mountain preceding the vertical rise of the step response went away, was exactly the point where the first hump in the woofer-only step response starts.  And if I turn on the supertweeter also, you can see it begin right at that same point also.  So I have confirmation using REW in two other ways.

I plan to publish the pictures of all this soon, and also I'm going to continue with other tests, programs, and techniques until a clearer understanding of this arrives.

But meanwhile, once again, I think I've found the magic, and far better than ever.