Spooky Good Imaging in hotseat position with lowback chair

For a couple of years now I've been optimizing the living room stereo for listening at at the center back of the room.  I created a new EQ for this called the "Movie EQ."  By necessity it had to be very different from the "Hotseat EQ" for listening closer to the speakers.  The Hotseat EQ sounds very boomy in the back of the room.  The periphery of the room magnifies the room modes, whereas the center position tend to cancel them out too much.

During this time I haven't bothered to sit at the old hotseat position.  I've loathed the idea of dragging my full back recliner back into the center position, especially since I try to show at least one movie a week.  I also feared all my optimizing for the back of the room had messed up the sound at the old hotseat, though I retained the original Hotseat EQ for the subs as a separate memory (which can be quickly loaded at pushbutton through my Insteon/Universal Devices home control system).  But I have no separate EQ for the midrange which has gone through a lot of changes too, and changes I made to the speaker angling cannot be easily reversed.

I bought a new lightweight low back folding chair for the hotseat position, but hadn't gotten around to trying it yet until the weekend just past.  I feared the hotseat position would be all messed up now.  But I started trying it on Saturday and then on Sunday found a truly wonderful position, about further 6 inches back from the original hotseat position with the recliner.  The new position is probably required because I changed the speaker angling to solidify the center image when listening in the back.  The speakers are less canted inwards, which is necessary to get a stable center image at all in the back of the room.

Listening to Citizen Steely Dan Disc 3, in this new hotseat position with a lowback chair, with the hotseat EQ selected, the imaging was Spooky Good.  It seemed like Donald Fagan and all the other musicians were right there in front of me. Nothing I've heard elsewhere for a long time has given me that same feeling.

The back of the room "Movie" listening position simply does not and cannot compare with this.  A much wider angle between the speakers is required, and there may be other advantages of being close up to the speakers.  There is much more direct sound from the speakers.  Some have described this as being like "Giant Electrostatic Headphones," even when both speakers are still in front.  (There are problems with trying to put both speakers exactly on the sides for stereo, though it can be interesting to play with.)

I do really need to use the hotseat position and EQ for serious listening, which I should do more often.

At the new hotseat, the speakers are about 45 degrees to my left and right on either side.  I'm about 5 feet from the plane of the speakers, and they are 5 feet apart.  But where to measure this is tricky.  If I measured from the middle of the speakers they would be over 6 feet apart, making the angle greater than 45 degrees.  But I now believe what is most important is the angle to the closest part of the active membrane, which is only about 2 inches inwards from the outer edge of the speaker towers.  That angle is somewhat less than, but still close to 45 degrees.  I would guess it is about 40 degrees.

I think the relative speaker position and angling is the number one most important thing in getting Spooky Good Imaging.

It was my friend Tim who originally told me, over 15 years ago, that I should experiment with sitting close to the speakers at a very wide angle, greater than the 30 degrees often suggested.  He was right.

He also said the exact position of the speakers and angling might not be the same for every person, because our outer ears are not the same.

If you keep the speakers at a fixed position, you move forwards along the center image to find the place where the center image disappears.  That means the angle is too wide and you are too close to the speakers.  So then you move backwards until the center image reappears.  Around where the center image reappears is the best possible position (or angle between the speakers).  Further back, and you get proportionally less direct sound, and the image becomes more diffuse.

That was essentially what I did on Saturday to find the new listening position.

Alternatively, if the listening position is fixed, you could move the speakers in and out from the wall, but for me and my speakers that would be exceedingly challenging for both physical and technical reasons.  I have the sub to panel delays dialed in precisely, for one thing, and changing the panel positions would mess that up.

Even if you ultimately end up moving the speakers, it's far easier to experiment by moving yourself closer and farther from them.

In the new position, the center is so perfect and well focused it's clear that my optimizing the speaker position, angling, and delays for the back has not messed them up for the hotseat.  If anything, it's far better than it was before now.

I purposefully did not angle the speakers outwards as much for the back as might have been beneficial there, hoping it would not mess them up for the hotseat, and my compromise for the back has indeed worked out perfectly for the front while being "good enough" for the back.

*****

I think having a lowback chair is good too.  When you are sitting in a high back chair, you get near reflections from the back of the chair in your ears.  These reflections matter more for imaging than for frequency response, but the frequency response is affected too, as I noticed when doing measurements.  I generally have always done my measurements with the chair in place because that is that is what you do when listening.

There's a long story here.  When I bought the current chair around 2008 I intended to take it to a furniture repair place and see if I could have the back shortened.  I knew this would be expensive but I figured it would be worth it.  I never got around to doing that.

The cheap folding chair I bought, which has flexible and ventilated plastic seat and back, is adequate but not as comfortable as I would like.  I think I will explore other options, like a director's style chair I already have, or others.

*****

The Citizen Steely Dan series became available in 1993.  While many people think of it as merely a repackaging of Steely Dan albums, it's a whole new mastering that was never heard before.  You knew about this if, like me, you read Steely Dan blogs around that time.  I also knew something like this was needed for a long time.  I don't know how it compares with later Steely Dan CD's, I suspect any high resolution versions were made like this too.

It turns out that one of my best friends was a techie hanger with Steely Dan.  He once invited me to the back room for a Steely Dan recording session.  I passed on this opportunity around 1976.  I had never heard of Steely Dan before.  Oops!

Later, I missed a chance to dub a copy of the master tape Steely Dan was circulating among its members.  I could not get there on time.  I ended up getting a second generation cassette tape dub (using my Nakamichi 600, and copying a tape another friend of mine had made on his Nakamichi 600) and it was a revelation.  A whole world of missing detail was on that tape that I never heard from the vinyl records.

(I think I can hear my friends voice on the master tape, and possibly on at least one song on Katy Lied.  You can also hear his girlfriend, who had originally been Denny Dias's girlfriend, playing some auxiliary percussion.  I could have been there too, dammit.)

For quite awhile I felt this was because tape is superior to vinyl.  But that wasn't the reason,  The reason was that the circulating master tape lacked the final Dolby processing and reverb that was added to the vinyl pressing (and also to early CD's).

There was a big technical problem here.  Steely Dan did all their recordings from at least Katy Lied onwards with dBX.  They needed dBX because of the incredible amount of overdubbing they did to make the final albums.  But the record companies only used Dolby A when making the final masters to be pressed on vinyl records.  So the final albums were retouched with EQ, reverb, and other tricks to sound OK and cover up the error in decoding.

In the Citizen Steely Dan recordings, they went back to the original masters, made with dBX, and decoded them with dBX the way they should have been.  Then they added the touch of reverb that may be less heavy handed than in the original vinyl.  The result is Steely Dan the way it should have been.  And it is incredible.  Even better than my second generation tape dubs, which have a lot of tape hiss (my friend and I chose not to use Dolby B at all).

This may be more relevant to some of the discs in the Citizen Steely Dan series than others, but it appears to be especially true for Citizen Steely Dan Disc 3.  I'm not sure if all the albums had the processing error.  I think it was true for the albums from Katy Lied to Aja.  I have never heard the songs on Disk 3 sound as good as they did last Sunday night.  It was Spooky.

*****

Among the changes I made when tuning my system for the back of the room listening position was changing the relative channel levels and delays.  I didn't know how this would work for the hotseat position, but I now know it worked out great for that too.

*****

I worked for 3 months on the subwoofer EQ for the hotseat position in 2024.  I ended up making giving it very smooth and nearly flat bass response.  Probably for many reasons, it is not as smooth now and could use a bit of rework for the new hotseat position.  But still, it is incredibly good.  It gives a somewhat higher bass level than I achieved for the Movie EQ in the back.

I wondered how this could be, given that the Hotseat EQ has many more and deeper cuts to eliminate room modes.  Well, there are two reasons.  The gain compensation for the sub Movie EQ has a -9dB setting for gain compensation.  For the Hotseat EQ this is 0dB.  I also added a 6dB boost for one frequency range in the Hotseat EQ.  Neither of these changes could simply be applied to the Movie EQ, but I might try doing similar things.

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.

Magico, Sealed vs Ported

All of the Magico speakers I have heard (M and S) have been outstanding, with transparency so good as to make me question my lifelong obsession with electrostats.  Few dynamic speakers do that for me.

They're also sufficiently expensive I doubt I'll ever one any.

But stimulated by Magico I see a fad for sealed box speakers in general arising, and I think that needs to be qualified.

In the 1970's when I was becoming an audiophile, sealed box speakers were becoming unpopular.  The physics of ported speakers had become well understood, so they were no longer the shot-in-the-dark they had originally been.  And with a ported speaker you could get deeper bass and freedom from bass distortion that were unobtainable in comparatively priced sealed box speakers.

Affordability is still an issue for most people.  And many kinds of music require full bass response.

If you cover up the ports on a typical value priced but well designed ported speaker, it doesn't sound cleaner like Magico.  It sounds muffled as if you've trapped much of the sound in the box and it can't get out.

What makes Magico speakers work so well is the very rigid aluminum enclosures.  Sealed boxes don't sound anywhere near as good in wood, at low frequencies much of the radiation is from the box rather than the speaker drivers.  The sealed box design puts a lot more pressure on the box, and if the box isn't rigid enough, it doesn't sound good.

When you go for very solid cast or extruded aluminum enclosures, you are inevitably going to end up with speakers that are very expensive, too expensive for most people.

And if you want full frequency response which much music demands, either you are going to need a very large sealed box speaker, or a subwoofer.  And then you are throwing away the simplicity of the design, you will need to do a lot of system tuning and room matching to get it to work right, and likely your sub will be ported to some degree throwing away the 'phase accuracy' of sealed box speakers (which I don't actually think is all that important, but if that was what attracted you to sealed boxes then you should know you may have to ultimately sacrifice it anyway).

BTW, all speakers sound 'cleaner' if you electronically roll off the bass.

Decent high fidelity is only achieved if you have response down to 30 Hz, really high fidelity requires response to 20 Hz, and hyper high fidelity is enhanced by response to 15 Hz.

Whenever you extend the response below 150 Hz in ordinary home sized rooms you are going to need EQ to deal with room modes.  And as you extend down each octave further, the requirement gets greater.

But you can't use EQ much to extend the response of a sealed speaker, because the sealed speaker bass headroom rolls off as soon as the bass response rolls off.  Boosting the bass will largely increase distortion rather than the sonic fundamentals.

OTOH, ported speakers generally have lots more bass headroom, and you can do more with equalization.

People who have focused on Home Theater and similar things already know these details which many two channel enthusiasts (of which I am one) may be unaware.

Introducing the new better MakePlaylist

MakePlaylist is a computer program I started writing in 2022 to generate playlists for background music, which I like to play nearly all the time since then.  Every day, at the click of a mouse, I create a playlist chosen randomly from all the folders of music I consider good for background music.  I generally use Roon to do the actual playback, but many if not most music playing programs will read the standard .m3u playlist files MakePlaylist generates.

I never thought any random playing system before mine worked correctly, and now I see why.

Unlike most random play functions, MakePlaylist by default uses Random-Without-Replacement (RWOR) meaning I will hear everything once before anything is played twice, and this is true not for just one playlist but every playlist you make until you have actually heard everything in your collection (or selected folders).  The effect of this is profound.  More typical Random-With-Replacement (RWR) (or RWOR in just one session) can take 7 times longer to actually play everything at least once, and some things will by random chance be played dozens of times more than that.*  Those things being played a lot may well be the things you like least.  And if the random drawing system is biased, it could be worse.  

Managing RWOR across multiple sessions requiring keeping a history, and that is one thing MakePlaylist does.

MakePlaylist has many other features, including 4 different random modes in all, folders can be weighted relative to one another, and you can select subsets of files based on creation date and/or file type.  You can make playlists with either Albums or individual Songs.

Although MakePlaylist is built around the idea of random selection, you can easily prepend chosen titles, or shuffle in favorites and fairly new items.  MakePlaylist comes with another program ShufflePlaylist to shuffle multiple playlists together.  And TrimPlaylist lets you trim the early part of a playlist that you have actually played, or remove the following items you didn't play from the played history--which is crucial in keeping the RWOR system working as well as it can, but only if you only listen to a handful of titles in very long playlists with huge libraries will RWOR give slower coverage than typical RWR.  I've determined there is a set of parameters where that happens using another included program TestPlaylist which runs coverage tests using as many runs of MakePlaylist as necessary to get to full coverage of a set of folders.

I routinely build my playlists like this: a header section containing the newest files less than 3 days old, and a middle section which combines randomly selected items from my entire library shuffled with favorite items and further shuffled with items less than 60 days old.  An alternative to mixing in favorite items would be to use folder weighting, which I do in some cases.

Using such a method (illustrated in one of the scripts) one doesn't need to manually create header playlists for new items, a header is automatically prepended with the newest files.

In addition to making standard .m3u playlists, MakePlaylist can make folders of "links" pointing to the original files.  These folders take very little space and can be deleted later.  Some playing programs can't read playlists, but they can either load a folder of files, or you can cut and paste files from that folder into the playing program. 

So then there is another program to shuffle links-folders ShuffleLinks, and one to pre-pend one links folder to another PreLinks.  (With ordinary playlists, you can append and prepend them with the terminal cat command, but a special program was needed to "prepend" ordered links folders that MakePlaylist creates.)

MakePlaylist requires Tcl, which is generally available on MacOS and linux systems.  On MacOS you can simply download the MakePlaylist distribution and run the programs right away.

On Microsoft Windows you must load something like Windows Subsystem for linux, and then install Tcl from Hombrew or other online source.

You can download MakePlaylist at SourceForge here.  MakePlaylist is Free Software.

Since the Tcl source code is the program, it's entirely transparent, even with a very basic understanding of programming you can see what it is doing, and the code is very well documented.

You will download a .zip file, and if it was not unzipped automatically during download, you can just click on it and it will unzip into a folder now named:

MakePlaylist_all_10131

(or whatever the latest version is)

Open that folder in Finder, and it looks like this:

If you want to read the installation instructions, open 1st.README.txt.

At this point, you can simply open the bin folder and click on (or use Open With Terminal) to run any of the included programs.  The first time, MacOS will refuse to run them, but then you go to the Privacy and Security panel of your System Settings, and it will say MakePlaylist was blocked, and you can give it permission to run.

Formal installation is described step by step in the file 1st.README.txt, which you can click on to read.  It is just moving the files in the bin folder to your HOME bin folder, if you have one,  or use the included bin folder as the beginning of a new one.  And then adding that bin folder to your shell startup, which is now .zshrc on macs, if it isn't already there.  And then you can customize as needed and move the User Scripts ... files to your Desktop or other location.  And then to run those scripts from the Desktop, you can just click on them, to construct basic playlists you might want daily (these are non-interactive).  That is how I use MakePlaylist, through scripts that often combine playlists in different ways.

MakePlaylist programs run in 3 different ways.  You can just click on programs to run them.  You can open a Terminal and give MakePlaylist commands.  Or you can write scripts that use MakePlaylist and other programs.  Once created, those scripts can be just clicked on to run.

Although I myself have generally been using MakePlaylist based scripts, I realized recently I could make the program accessible to more people by making all the programs interactive so beginners can get used to them before writing scripts.

One cool thing that MakePlaylist can do is list either the files which have been played, or the files that have not yet been played.  Or it can show how many files have been played each number of times, either from the beginning of MakePlaylist history or in the current epoch (since that last time that all files were played and a history reset was done).  In my background music folder (~/itunes) this looks like this:

itunes% MakePlaylist -epoch -statc *

Played   Titles

------   ------

     1   126

     0   610

This shows that I am using the default RWOR mode, since all files have been played either once or twice since the last reset.

With TestPlaylist, one can run tests on MakePlaylist to see when and if it will actually play all the files.  With the default RWOR mode, this is pretty boring.  The "1" plays number gets larger until the "0" number goes away, and you get a result that looks like this:

Iteration 25

Played   Titles

------   ------

     2   18

     1   714

Options are -albums

Length is 30

Thus, it took 25 playlists of 30 albums each to cover the entire collection of albums at least once, and only the last 30 album playlist started playing some playlists a second time.

Using RWR (MakePlaylist option -wrd) the result is very different:

 Iteration 164

Played   Titles

------   ------

    14   4

    13   7

    12   11

    11   33

    10   43

     9   65

     8   104

     7   119

     6   115

     5   92

     4   66

     3   32

     2   32

     1   8

Options are -wrd -albums

Length is 30

With RWR, it took 164 playlists until full coverage was achieved.  When it was finally achieved with the 164th playlist, some albums had been played as many as 14 times.