Laser Measure

 I read about audiophiles using laser measuring devices on Stereophile and even say they're essential for serious audiophiles (to get your speakers precisely positioned) I've never bought one.

This looks like one I'd like to get, a Mileseey S7.  

https://mileseeytools.com/products/outdoor-laser-measure-tool-mileseey-s7?currency=USD&sku=18063677008810900264450910&utm_medium=cpc&utm_source=google&utm_campaign=Google%20Shopping&utm_source=google&utm_medium=PMAX&utm_campaign=S50&gad_source=1&gad_campaignid=23050170196&gbraid=0AAAAAoVRmp6WhX8rcXFFIy1Ypabp95nu0&gclid=CjwKCAjwqonVBhA4EiwA9wYJ3f5cErJTQ0TCJu2b96xTEdEz3BYFNRhfXWsYMBbCU4E1EmUYO8xjYxoC2y4QAvD_BwE

But since my income is limited, I plan to wait until I've sold surplus audio equipment of equal or greater value, which I really need to do right now for other reasons too but have been too busy with urgent household projects recently, mainly that I want to be prepared for the rain storms likely with the Super El Nino and have been working on critical hardscaping and drainage features all around the house, stuff I should have thought of 30 years ago before my house needed foundation repair, now I want to prevent it from needing even more foundation repair, fence repair, driveway repair, etc.

Polishing up the Bass

The two most recent posts revealed notable deficiencies in the bass response below 200 Hz.  The left channel has a minor depression around 90 Hz and the right channel has a minor depression around 50 Hz.  The former is more noticeable in the graphs.  However neither of these depressions coincides with a PEQ (programmable EQ) cut, so I can't fix either one simply by dialing back EQ.  And furthermore, nearby PEQ cuts were deliberately made quite narrow so as not to exacerbate them.  They are going to be tough to fix.

Probably the more audible issue anyway are the depressions in the range 120-180 Hz in both channels.  Those are entirely due to the panels, because the subs are nearly entirely cut off in those ranges (except just around 120) with very steep crossovers (phase linear 8th order pseudo Linkwitz Riley) that are already 6dB down at 120 Hz.  In the past I've sometimes boosted the sub response to fix such things, but I've decided it's best to keep the sub down below.  I'm already running the "subs" higher than most people at 120 Hz, but I can do that because my subs have output up to 300 Hz, very rigid cones, and I'm using 8th order filters on them (don't try that unless you also use phase linear Finite Element Response (FIR) filters like I do).  I cross over at 120 Hz because that way I avoid serious issues in both the subs (if crossed higher) and the panels (if crossed lower).  Notably the panels have fundamental resonances around 85 Hz which add distortion and can even cause the panels to rattle if driven too hard below 120 Hz.

The depressions around 160 Hz likely result from a combination of dipole bass cancellation and wall bounce.*  I already have the speakers 42 inches from the wall, and to put the wall bounce below the crossover point I'd need to move them at least another couple more feet from the wall, which is impossible in my multi-purpose living room where I also talk to guests and show movies on a projection screen.  Moving the speakers farther out the wall would also make it difficult to walk into the living room itself from the entry way.  Even a dedicated listening room would have the same problem unless I entered it only from the back, and I must also pass through the living room to get from the bedrooms and bathrooms to the kitchen.  The speakers are already as far out from the wall as I can manage with this house plan.

But it doesn't look like my old EQ settings are optimal any more.  I used a boost around 182 Hz in the left channel, and 133 Hz in the right channel.  And yet, in the measured response, the right channel has the depression which reaches from 140-180 Hz and the left channel has a very narrow depression right around 160 Hz.  It looks like I previously got these backwards, but also the work was done from the center room which may be different from the back, which the Movie EQ is intended to optimize, or there may have been other salient changes since I came up with the earlier boosts.

Even with these existing bass dips, the current Movie EQ settings are wonderful I think.  I watched the movie Contact (1997) on Sunday, which is full of deep bass, and the sound was full and beautiful, better than I recall hearing in theaters, and with no added "boom" as often happens in theaters.

I dislike adding boosts in my digital system because they can be headroom reducing.  Imagine you have a sine wave that sweeps from low to high very quickly (so as not to damage anything) at maximum level (0dB).  If you have a boosted range, it will digitally clip in the boosted range.  Now you would probably never want to run such a test, but if you happen to be listening at 0dB or nearby, as I often do for movies which have very low average level usually (to allow for big peaks) 0dB transients can sometimes occur in these ranges.

However, IIRC, I deliberately cut all panel inputs to the panel crossover device (a miniDSP-DRC with all digital I/O) by 6dB to prevent clipping due to asynchronous digital sampling.  I truly hate ASRC, it "resamples" a digital signal rather than just passing the data along unchanged, and therefore creates the possibility of digital clipping where there was none before.  I wish all digital were done fully synchronous as the Tact does it using 1999 digital chips, but hardly anyone does it that way anymore because they've all been scared by the "jitter" parade, meanwhile the importance of jitter is vastly overstated, and of no consequence at all for a device which is digital in and digital out.  I think there are still a few chips that support fully synchronous operation over SPDIF without a separate timing line--I use a separate timing line with my Lavry AD converter for truly perfect transfer--but it's simply not used much anymore otherwise--the miniDSP's don't let me select it because they don't have a separate timing line.  The Behringer 2496 DEQ's are also fully synchronous just like the Tact.  But sadly miniDSP decided to do things the "new" way, which introduces the possibility of digital clipping unless you are using their analog outputs.  To be absolutely safe from digital clipping, therefore, you have to reduce the level to the internal digital resampler by 6dB as I am doing.  This is no serious loss in the 24 bit or higher domain that the resample is working in.

Then I re-boost the signal 3dB in the DEQ's, which are fully synchronous.  That leaves me about 3dB headroom in the worst case.  I did things this way to allow for the 3dB boosts I was using before I started developing the Movie EQ, when I zeroed them all out for a fresh start.

So 3dB boost is all I can permit with my current setup.  In previous experience, that can be surprisingly useful, and you'd probably not want to use more than 3dB boost anyway even if you had the digital headroom for it.

I started with the Left channel and started boosting +3dB around 160 Hz with 1/6 octave.  But this created the condition where 160 Hz was higher than the 180 Hz band.  I pushed the PEQ frequency higher and higher, until at 170 Hz the two bands were equal, which also allowed me to broaden the PEQ to 1/3 octave without raising the already elevated band at 140 Hz, a pretty good compromise, and the "suggested EQ boost" fell exactly as much as with 1/6 octave centered at 160 Hz.   There's still a big visible dip, but the lowest point on the dip is not much more than other dips in the response above 200 Hz.  Also notice that the suggested EQ boost at 160 Hz has dropped from +7.6 dB (very bothersome) down to +5.2dB (meh). Don't aim for perfection, that usually ends up horrible, do what you can with limited EQ boosts like 3dB.   I've previously found such small improvements uniformly beneficial when larger or more complicated ones ones can get tiresome.  "Ship it."  

Also note that +/- response numbers vary enormously with filter bandwidth.  I could make this look nearly flat with octave bands, but's that nothing like music.   I use the minimum bandwidth and maximum fft size (which has huge effect on resolution) available to me to see the finest detail I can.  A sine wave generator or MLS generator would be better but requires much more complicated setup.

Left Channel before adjustment

Left Channel with boost at 170 Hz 1/3 octave

Duplicating this filter in the right channel wasn't any good.  It raised the respose at 180 Hz compared to 160 and 140.  140 happens to be already depressed, rather than peaking as it does in the other channel.  So I lowered the PEQ frequency to 160 Hz and broadened it 3/4 octave, which gives a fairly smooth U shaped response in the region centered at 160 Hz, probably the best that can be done with only one 3.0 dB boost, and the suggested EQ boost has dropped over 3dB (probably lucky sampling and less than that, it never falls as much as you have boosted and you're lucky if it's 0.5dB less...on a second measurement I got 4.8dB suggested which would be a 2.7dB drop from 7.5 suggested before adjustment but similar looking spectrum).

Right Channel before adjustment

Right Channel with boost at 160 Hz 3/4 octave

Right Channel with same boost later

Final PEQ settings for today 9-7-26 are

Left EQ 9-7-26

Right EQ 9-7-26

I played the entire album Bass Ecstasy by Bass Erotica at -1.4dB total (-1.3 being applied by Roon's album leveling) in the center back at it was captivating.  3D image, wall to wall, with impressively deep powerful tuneful and never unpleasant bass notes.  I remember when I could hardly listen to this at -10dB because of all the room boom.  I measured 93.1 dBC "fast" peaks on my SPL Pro app at the listening position.  dBC de-weights the deepest bass which will have peaks much above that measured flat.

The Movie EQ sounds great all around the house because of suppressing the primary house modes, and the way the Acoustats couple to entire space.  The Hotseat EQ exaggerates some house modes to get flat response at a close seating position.

Technical Notes

*The way dipole cancellation works is similar to a 6dB/octave lowpass filter that starts where the minimum width of the dipole is 1/4 wavelength and has cutoff (-3dB) where the minimum width is 1/2 wavelength and simply get worse all the way down from there. 

https://www.diyaudio.com/community/threads/dipole-phase-cancellation-in-esl-speakers.204246/

 The Acoustats are 20 inches wide so the rolloff begins around 170 Hz...just what we are seeing here, and goes down from there with the cutoff at 85 Hz and below.  However, the Acoustat diaphram is tuned so that it has a resonance around 85 Hz which cancels out the dipole cancellation from about 160 Hz down to around 40 Hz, giving the speaker a 40-50 Hz minimum response, at which point it is further assisted by room modes typically down to 20-30 Hz.  The diaphram gets looser with age so that the resonance goes lower and may not correct the response nearly all the way up to 160 Hz as designed, but instead extend further down, thus explaining the dip I see in that around 160 Hz, or perhaps it never does all that well around 160 Hz anyway.  Looser diaphrams from aging also explain why I get rattling if I drive the Acoustats too loudly below 120 Hz.  Brand new Acoustats would play louder in that region without rattling, but would ultimately reach their limits too.

Rear wall cancelation would be worst right up to the back wall (since the back of the dipole is out-of-polarity) and reverses to maximal addition where the distance to the rear wall and back is 180 degrees, then back to maximum subtraction where the distance to the rear wall and back is 360 degrees, with a neutral point in between.

maximal addition: 2D = wavelength / 2, then 2D = 3 * wavelength/2, ...

wavelength = 4D, 4/3 D

maximal subtraction 2D = wavelength, then 2D = 2 * wavelength, ...

wavelength = 2D, 1D

The average distance of the acoustats from the wall is about 3.5 feet or 42 inches.

maximal addition wavelength = 168 inches, 56 inches

maximal addition frequencies = 80 Hz, 241 Hz

maximal subtraction wavelength =  84 inches, 42 inches

maximal subtraction frequencies = 160 Hz, 321 Hz

More Imaging Improvements, need to get both level and delay correct

Previously I determined I needed about a 0.1 ms delay added to the right channel to bring the center image into the exact center position.  It's funny because the speakers look to be exactly equidistant, theoretically I could achieve the same result by moving the right speaker about 1.5 inches back, but it's already as far back as it can go because of the subwoofer and other equipment being in the way.  And it's strangely hard to dial in the correct position acoustically by moving the speakers which is very difficult to do especially considering how very important the angling is too.

It was also very hard to come up with the 0.1ms delay number itself, if I just play with the delay setting with the remote control I almost invariably came up with larger delays, which then were later found to introduce other issues.  The 0.1 ms delay is the minimum which causes the image center to move at all, and that's what the correct setting seems to be as it doesn't introduce other issues, and then somehow it seems right afterwards if not during the adjustment process itself where the ear keeps saying "more, more" until everything goes to hell.

The testing involved listening to mono'd sources and trying to get everything in the phantom center.

Getting the image exactly centered is surprisingly important in creating a holographic image.  Just a little bit off and and you have ping pong stereo with no depth.  Of course you cannot with multiple listeners siting side by side...that is one flaw in acoustical stereo.  You can get correct centering for multiple listeners only if they sit in front and back of each other, but that leads to other issues.

I don't like center channels either.  While that is the obvious solution if you MUST have multiple listeners side-by-side with at least some semblance of image centering for all, center channels always sound unnatural to me when I am at the sweet spot myself.  When you have a correctly centered phantom center, you have depth, width, and everything, with a center channel you just have left to middle to right, end of story.

But when I did this testing some weeks ago, and posted about it here, I did not at all try to equalize the loudness of the two channels, which has to be done with a different kinds of test.  Getting the loudness equal (and not influenced by delay) requires playing the same signal in one channel, then the other channel, and adjusting until they have the same loudness and general sound.

I did this playing pink noise only in one channel, then the other channel.  I'm shocked that there is no test CD with this kind of test.  I had to make one by taking a pink noise recording (from Stereophile Test Disc 2 IIRC) and then using Audacity to mute one channel or the other and making a new audio file from that.  Sadly I cannot distribute this test because it uses Stereophile's original "content."  I will someday have to regenerate it using the pink noise feature of Audacity itself, and that will also involve determining if that is as good as the pink noise from Stereophile, so I keep putting that off.

I suppose most audiophiles use an analog preamp with a balance control you can swing from far left to far right so they don't need such test signals.  But not only do most digital preamps lack such a feature, many purist preamps don't have balance controls either, and some that do don't allow you to swing from far left to far right.

The pink noise in each channel can be measured objectively with an SPL meter with "C" weighting.  You do not want to use the more well known "A" weighting, because that is for noise which is the stuff you don't want to hear.  The "C" weighting is for Content, the stuff you are trying to hear.  A flat contour gets it wrong too.  I use an SPL app on my iPhone to do the measurements nowadays, which gives me high accuracy and digital readout.  I use the slowest available response on my app, which is Slow and then do a mental averaging of the number.

I was shocked to find the right speaker was about 1 dB louder.  I wondered if it was because I'm using two vintages of Acoustat 2+2's on either channel.  But actually, I had previously attenuated the left speaker by 2.3 dB in the Levels menu of my Tact 2 Digital Preamp.  I had long forgotten about that.  I think I was trying to compensate for the delay difference by only using level adjustment, which is impossible.

I got the best result dialing back the attenuation to 1.0 dB.  Then the speakers measured the same loudness.  And when that was done, I also noticed the speakers sounded much more similar.

I then tried re-adjusting the delay in the left channel.  But the best image centering still seemed to occur with 0.1ms delay.  It seems like delay is the more powerful localization mechanism, so even if the levels don't quite match (within one dB anyway) the center image position is almost entirely controlled by the relative delay.

But getting the levels matched better than one dB, and with the correct delays too, it greatly improves the depth and realism of the image, I determined in subsequent listening.

Current settings:

Delay: 0.00 ms Left, 0.10 ms Right

Level: -1.0dB Left, 0dB Right

This was in center back seating position of the room, I think it should work in the closer up position too.  Sometime in the past few months I had eliminated ALL EQ's from the Acoustat feed because I figured they were not well optimized for the back of the room (and maybe not for the more frontal listening position either).

The overall response seems fairly flat until you get to midbass frequencies 200 kHz and below where there are some dips, those are problematic to adjust because I'd need to add "boost" which I try to avoid as it is headroom lowering.  The subwoofer feed has several EQ's to deal with room modes, but the subs are cut off with a steep FIR linear filter above 120 Hz  (FIR linear filters are digital magic which introduce no phase shift) so the subs can't help with dips above 120 Hz caused by back of room reflection and dipole bass loss.

Though I've done this in the past, I generally do not like leaking some sub base above 120 Hz to "help" the panels.

The left and right channels measure about the same from 300 Hz on up.  They sounded very different before the level adjustment, after the level adjustment they still sound a bit different.

I need to dial in EQ's better AND measure and adjust bass and panels separately, and then go back and check levels again.

After any kind of EQ adjustment where channels are equalized differently, you need to re-measure channel loudness as I just did, using single channel pink noise and a C weighted SPL meter, and not rely on image centering which is more controlled by delay when the channel levels are already fairly close.

Re-thinking Peterson Dip

For the last decade or so I've experimented with high frequency dips modeled after the Linkwitz Gundry dip (generally characterized as 3dB down at 3kHz with Q of 2, which is between a half octave and a full octave bandwidth) but on the basis that the issue is excess high frequency reverberation in playback compared to the direct sound vs direct because of small relatively live rooms generally, and not just in the region of cochlear resonance centered at 3kHz.  This reverberant field causes more high frequencies to enter the ear canal which begins roughly perpendicular to the direct sound axis than would happen in a live performance with the music sources in front.

I've generally started a roll off around 2kHz and going up to 12kHz or so, above which some added highs are beneficial.  But I also came to feel it imparted a certain dullness.  So I decided to rethink the "Peterson Dip" which is intended for dipolar full range speakers like Acoustats.

People with "box" speakers aren't bothered by this as much as people with full range dipoles like me.  The reverberant field is naturally rolled off in the highs and often the upper midrange as well with box speakers because of limited high(er) frequency dispersion in woofers and tweeters.

Now I'm trying a dip centered at 4kHz with an bandwidth of 2 octaves, and limited to 3dB maximum.  The "cutoff" points are then 2kHz and 8kHz, which includes what I consider most important--the "metallic" range around 6kHz, without necessarily rolling back the higher frequencies which add "air" or suppressing 1kHz which would remove clarity.  The Q is a comfortably low 0.667 with hardly any overshoot whereas a Q of 2 has significant ringing.  Previously I used 3 filters with a octave bandwidths or less to achieve a 2-12kHz dip, but unless you are truly cancelling resonances, which this is not, I think fewer filters are better.

Noting that my Acoustats measure fairly flat in the range 1kHz-16kHz, I decided to simply apply the new dip on top of their pre-existing response without getting deep into achieving the desired response acoustically for the time being.

It seems to provide about the desired effect on taming excess brightness and making it easy to listen at a higher levels too.

Looking at the resulting spectrum, the difference is hard to see on this scale, but the -3 dB cut actually looks a bit smoother too:

The sound is significantly more "laid back" with the cut on pink noise.  The speakers sound very "forward" without it.  You don't have to have golden ears to hear it if playing while adjusting the control, each -0.5dB difference at 4kHz is very audible.  The actual center frequency is 3990 Hz, the closest to 4kHz that the Behringer DEQ 2496 will let me select and I'm going to call it "4kHz" for simplicity.

All measurements are made from back of the room center position with the Movie EQ on the subs, no other PEQ's on the panels.  iPhone is being held with stand.  I'm standing about 18 inches to one side mostly.  Left channel level set in Tact preamp is -1.0 dB (see next post) which gives best measured balance with SPL meter C weighting.

Left Channel No Cut

 

Left Channel, -3dB at 4Khz

Right Channel, No Cut

Right Channel, -3dB at 4kHz

These 4kHz dips are currently the only PEQ's applied to the panels, since I set all the old  PEQ's to 0.0dB with the new Movie EQ in order to "start from scratch."  However I left all the old frequencies and bandwidths in place because they may be useful in setting new PEQ's all over again.  They are historical information.

The new panel EQ settings are now saved unambiguously as a new memory named "Sep26."  The meanings of the older settings are impossible to discern anymore.

I played the entire album Bass Ecstasy by Bass Erotica at -1.4dB total (-1.3 being applied by Roon's album leveling) in the center back at it was captivating.  3D image, wall to wall, with impressively deep powerful tuneful and never unpleasant bass notes.  I remember when I could hardly listen to this at -10dB because of all the room boom.  I measured 93.1 dBC "fast" peaks on my SPL Pro app at the listening position.  dBC de-weights the deepest bass which will have peaks much above that measured flat.

The Movie EQ sounds great all around the house because of suppressing the primary house modes, and the way the Acoustats couple to entire space.  The Hotseat EQ exaggerates some house modes to get flat response at a close seating position.