Doc modding Marantz imperial 7

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karatestu
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Re: Doc modding Marantz imperial 7

Unread post by karatestu »

Semi omni speakers still suffer from room related peaks and nulls just the same as P&S :roll: Working with the room still has it's limitations I think.

With only the use of an audio equipment set up and test CD and my ears I ran through the various frequencies from 20 Hz to 20 KHz. There was output from 20Hz although not very loud. The loudness plateaued about 64Hz and was ok until I got to 125 Hz where volume dipped quite abruptly and then back up at 180 Hz where it was louder than 100 Hz. Above that all seems to be ok until I get to 1.2 KHz where there is a peak.

What can I gather from this ? :think: :think: :think: Not sure.

The null at around 125 Hz to 150 Hz is probably due to destructive cancellation from bass bouncing off the front wall. I haven't checked to see if the distance and quarter wave length at that frequency tie in . It probably does as both 12 " and 5" drivers both exhibited roughly the same null when tested on their own.

Not sure what the peak just after the null is about or the peak at 1.2 KHz but both the 12 " and 5" displayed this behaviour when tested on their own. When all tested together the results were the same. I can't imagine that all three drivers in free space have nulls and peaks at these frequencies so I will have to blame it on the room. :angry-steamingears:

Crossover between tweeters and the rest seems ok with no obvious to the ear peaks or nulls.

There is nothing that can be done for destructive cancellation I gather. Peaks can be removed with EQ but I aren't going down that route. #livewithit
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Re: Doc modding Marantz imperial 7

Unread post by karatestu »

Daniel Quinn wrote:
Wed Oct 07, 2020 4:19 pm
Listening to music plays a part in my rehabilitation. My right foot still doesn't move. But that fact belies a whole host of recent changes.

When I listen to music, my foot doesn't move, but my brain tells me and it feels like it is tapping.
So you are still making progress :dance: Sounds like a very long road to recovery though :cry:

Chiropractor told me nerve damage takes the longest to heal in the human body. I am still having problems with my back. Although I don't have much pain in my back itself, my groin and top of left leg still feels partly numb like when you have had an injection at the dentist and it is starting to wear off.
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Re: Doc modding Marantz imperial 7

Unread post by karatestu »

After 12 KHz my hearing nosedives :violin:

I suppose it's a bit daft to conduct a frequency response test with nothing but your ears which may not have a flat FR of their own :shock:

Plus as the frequency gets higher on the test tone CD the intervals between each frequency becomes larger. There will be other peaks and dips which fall in between the frequencies I tested.
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Re: Doc modding Marantz imperial 7

Unread post by Ithilstone »

karatestu wrote:
Wed Oct 07, 2020 5:24 pm


I suppose it's a bit daft to conduct a frequency response test with nothing but your ears which may not have a flat FR of their own :shock:
Depends - if you do it for yourself then there is no point in using any other equipment other than your ears. It might tell you something you do not hear anyway ;] and why worry then? ;]

Different thing would be to check what your speakers do to average Joe
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Re: Doc modding Marantz imperial 7

Unread post by karatestu »

Ah, the voice of reason (Tomasz). :grin:

I read on diyaudio.com lots of golden eared speaker builders can hear the effect of a fly landing on the mixing console in the recording studio when the music was captured :roll: I have come to the conclusion that I am not golden eared and have no hope in being so. Lots of the speakers I have heard I have liked. Maybe I am too easily pleased. Oh it's good to be easily pleased.

DSP seems to be the go to answer for all evils these days. I could try and design a notch filter for that 1.2 KHz peak but I am not going to as that would dishonour the design principles and The Doc.

I put my diy speakers up against my two pairs of B&W (CM1 & P4). My little two way CM1 have a simple xover - single coil on mid bass and single cap on tweeter. They are very polte and lacking dynamic compared to the P4 and especially my diy jobs. The emotion is just not there and instruments sound like they have been squashed and compressed.

The P4 fare much better. They are more lively and musical than the tiny CM1 but they are a small two way floorstander with 6.5" tapered line quarter wave transmission line bass loading. I was happy with these speakers for 20 years until my upstream electronics spotlghted their inadequacies - mainly the kevlar break up modes not masked by a single order electrical low pass filter and a metal dome tweeter crossed over at 3.5 KHz.

I suppose they had to find a balance for a simple two way and most "experts" would say the mid bass needs a higher order low pass to roll it off faster. My view is that it should have been a 3 way with the bass to mid xover much lower and a mid capable of going much higher than 3.5 kHz to cross over to the tweeter at say 5KHz. The P4 tweeter dos have a third order high pass on the tweeter but even so, I think too much is asked of it.

My diy speakers blow them both away even though they still have their little issues which need ironing out. The scale of them is off the scale :lol: The B&W's in comparison make you think that the instruments are miniature and played by dwarves with a music killing dose of compression slapped on in the recording process. A drum part on my diy jobs show all the small and large dynamic changes. One of the skills that make a gifted drummer is the dynamics even within a single fill or groove. The filterless semi omni's show all these dynamic variations with ease. I am howver guilty of comparing a speakers abilty to the sound i hear playing my own drum set. When I practice on my own my favourite grooves, licks or phrases are all ones that combine subtle low velocity hits with short fills etc of varying degrees of higher velocity. That's what I look for in speakers (rhythmically ).

At one point I worried that this musical dynamic ability was the product of a roller coaster frequency response but after comparing to other speakers and doing a crude frequency sweep I believe this is not the case. The B&W P4's hint at these dynamic swings just enough to convince me that they are on the recording. This is all with well recorded material by the way. Even my diy speakers can't help a modern noise wars compressed up to the bollox èxcuse for a recording. I still like some of this stuff but learn to just enjoy it whatever the quality.

Then there is the tone of instruments and the decay of the notes. You don't get decay of notes with Naim electronics but with NVA this information is rich and textured. The wrong speaker on the end of nva amps can throw all that harmonic information away (imo). Filterless bass and mid bass drivers just seem to me to hold on to that harmonic info and texture and project it in to the room for our enjoyment.

So I for one can forgive a speaker with nulls and peaks as long as they don't stick out like a sore thumb. During the doping process (adding a little then listening before adding any more required) you can hear the gradual attenuation of these offensive peaks to a point where they are not noticeable.

Some say doping a driver is wrong because it slows the bass or some such. Maybe it does but I value the benefits higher than this drawback. Mass of kevlar cones is more than paper so maybe the best approach is to use drivers with low mass paper cones which also have a better behaved break up region and dope those - they will likely need less dope.

One more thing. Mid bass drivers used up to say 5 kHz start to beam at a certain frequency. With doped drivers they are giving output almost right up to their natural roll off. This beaming is not the best thing for off axis reflections as the frequency response is different to that on axis of the driver. I expect firing drivers to the ceiling muddies the waters some what but to what degree I know not.
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Re: Doc modding Marantz imperial 7

Unread post by karatestu »

Another thought about doping paper cones. A lot of them I have looked at have weak motors (BL) compared to these Chinese kevlar PA drivers. That might not be a good idea as far as controlling the cone when it has had the added weight of dope applied.

Even comparing the size of the magnet of my B&W 6.5" kevlar drivers to the little 5" is an eye opener. The magnet is considerably larger on the 5" than that of the B&W. Compliance of the surround and spider will be all different as well.

I suppose all one can do is chose a driver with a strong motor (BL) compared to weight of the cone (mms). All this talk of T&S parameters would have brought a flaming from the Doc. Sorry Doc but I need to get my head around a few things :roll:
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Re: Doc modding Marantz imperial 7

Unread post by karatestu »

When playing the test tones CD through just the 12 inch woofer I found audible output right up to 10 KHz :shock: It was very quiet compared to the bass frequencies but it did surprise me a little (newbie know nothing) that output went that high.

The point source brigade (can you hear them coming) would likely comment that my speakers are a total failure. Several drivers of different sizes (12 " & two 5") all playing the same frequencies (over lapped) all the way up to 10KHz and possibly beyond. Add to that the fact there are four tweeters with only a first order high pass filter all firing in different directions .

Well of course the tweeters are no where near time aligned with the other drivers. The front firing tweeter is considerably closer than the acoustic centre of the 12" and 5" drivers. The up firing tweeter is somewhere in between and the side firing tweeters are time aligned with the bass and mid bass but not when you are sat in the centre between the speakers - toe in could sort that though I suppose.

But then the woofer and tweeter aren't time aligned on Cube's either with the front firing tweeter being much closer to your ears. I doubt the first order filter magically time aligns them but then I don't understand that aspect enough yet to know one way or the other. All I know is B&W have often moved tweeters forward slightly to compensate for crossover phase anomalies (Nautilus Diamond tweeter jobbies).

Then there is the fact that there are tweeters facing in four directions which will be smearing imaging and the like. It all adds up to an point source imaging freaks worst nightmare. Good job I am not one of them :guiness;

More speaker related bollox to come after I've had a cup of tea and called the grain merchant (money grabbing bar stewards they are)
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Re: Doc modding Marantz imperial 7

Unread post by karatestu »

A quote from Roy Johnson of Green Mountain Audio. I have come across his writings before and find them to be in line with much of what I believe. Sadly he passed away summer of 2019 :(


gma952b65 posts
09-20-2002 2:22pm
Time coherence is as important as the amplitude response measurements typically taken.

A time-domain snapshot would show the pressure spreading away from that cabinet- a disturbance that contains both high and low 'frequency' components, better thought of as quickly rising/falling pressures overlaid with slower rising/falling ones. When we hear them in their original sequence, we remark "what a gifted musician!"

There are serious challenges and outright limitations to achieving 'perfect' time coherence:

-We are limited by the drivers having finite bandwidths before any crossover is applied. We need perfect pistons in the treble, and response to DC in the bass.
-A driver's electrical characteristics change with the power applied (temperatures rise), which means the crossover points, thus the phasing, change dynamically.
-We have the issue of cabinet reflections. A tweeter in a large cabinet face is like putting a woofer in a corner, speaking wavelength-for-wavelength. Even if that face is bevelled, or felted (felt does not absorb 100%)

How can you tell a cabinet face is a problem? Just pick any point on that face and compute the `round trip distance for sound to get over to there from the dome tweeter and from there on out to your ear. Compare that to the direct path distance from the dome to your ear. You'll see that the path-length DIFFERENCE is greater than 1/4 wavelength of any lower-range sound from the tweeter (or mid), which means the reflection smears over the direct wave- it is not coherent.

This tweeter (and mid) 'splash' off of the front panel is 'corrected' in those large cabinet face designs by crossing over the tweeter higher than the mid's crossover point (and the mid higher than woofer)- which de-focuses the image and makes the dynamics sluggish, as now the drivers are 'a little out of phase' over their ENTIRE ranges.

If it wasn't 'corrected' by staggering those crossover points, the tweeter (or mid) would measure too loud in its lower range, as the reflections boost its 'bottom end response' when measured with test tones or pink noise, or even MLSSA- which is why this phenomena is not discussed- it's not visible with std. tests. But it is audible.

To hear what really good time-alignment and lack of reflections do for the clarity of the performance and the musicality, please listen to a single element headphone- Grado's or some Stax electrostatics, etc. And listen to a single-mic recording on them, such as a Harry James Sheffield disc- you can clearly hear what each musician is doing, in any part of the spectrum, which is the benefit of a time-coherence transducer.

Then play a crummy recording and see if it is less irritating- it will be. That's because the transducer has less phase distortion, which only distorts the original distortion. You normally hear distorted distortion, which is a multiplicative process, never additive. And why it's better to improve the source components first, before say, changing the amplifier or speaker wires.

Speakers which are very sensitive to your choice of amplifiers have a phase-response that is all screwed up- which magnifies any problems the amplifier has. Also they use a wierd crossover circuit that's causing the phase problems to begin with. These circuits also put a difficult load on the amplifier, as their extra parts store energy instead of passing it on as soon as possible.

Finally, time-coherent speakers can sound great on cheap stereos, for the reasons above, but only if they employ very simple first-order crossover circuits whose few parts can actually respond to every nuance the amplifier can muster. The evidence is heard even in cheap Sony headphones (which have little phase shift) plugged into any stereo.

Also, most crossover-circuit parts cannot pass the most delicate signals, which makes the music bland. Most crossovers also use far too many of those lower-fi parts.

We don't see too much written in the press about time-coherence, as the math is confusing at first- not suitable for a casual article. I wrote one for Audio Ideas Guide magazine, and it's still hard for me to wade through without re-reading.

The best dealer has worked hard to hear `most every brand set up well, whether he carries it or not. This industry wouldn't be where it is without those retailers (which are few).

Hope this helps. Basically, trust your ears and use them to verify that a dealer knows what good sound really is.

Roy Johnson
Green Mountain Audio
DIY inspired by Richard "The Doc" Dunn RIP

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Re: Doc modding Marantz imperial 7

Unread post by karatestu »

Another quote from Roy

gma952b65 posts
09-23-2002 12:44pm
They are not tuned to your room, but "focused" to your listening position via moving the mid and tweeter in the C-2 and new C-3. You are setting what we call the "Sound field Convergence" for time coherency at your seating spot. This is done with a tape measure and takes a few minutes max.

The phase response of the C-2/C-3 will thus be- by definition- compromised everywhere else. But by many HUNDREDS of degrees less than speakers with higher-order crossovers.

We do note that with a first-order crossover, off-axis comb filtering does not seem to be an issue except on test tones and pink noise.

Moving completely up and out of the main listening plane (where off-centre listening was still fine), we hear the depth of field decreasing, as the time domain becomes progressively "warped" from bass to treble. But at least there are no sudden "jumps" in the relative acoustic phase from driver to driver as there are with high-order crossovers, no "twitchiness".

Standing up, the time delay warp across the spectrum is still far less than with high-order-crossover speakers. We do not hear the sound field degenerate into "a wall of sound"- plenty of ambience remains.

High-order designers are getting better at smoothing out the abrupt jumps in phase- for less "separate tweeter-separate woofer" effect.

They are also getting better at damping the ringing present in the highest-order crossovers, but the result is a hard load on the amplifier.

What you hear from the smoothing and damping is less image depth, less dynamic impact, and less rhythmic definition (finesse) anywhere around those crossover points. Which is why, quite often, the approved "audiophile recordings" used to demonstrate them are so bland performance-wise. Not much there to challenge the speakers. Something aggressive won't be pleasant- no Zappa allowed...

I noted in some postings above, references to the possible lack of phase shifts in minimal-driver speakers- single panel electrostats, Jordan module, Lowther, etc.

In a single driver, phase shift won't be caused by an electrical crossover- there isn't one!. The signal remains free of crossover parts distortions/haze too.

However, any driver has mass, suspension, and damping (by the suspension's resistive losses and the amplifier). Thus it is a "damped harmonic oscillator"- in a Physics 101 book.

A harmonic oscillator has a 1/4 wave's worth of time-delay down at its low-frequency resonance, compared to the midrange tones. For a sealed woofer with -3dB at 40Hz (close-mic'd measured), that means 1/4 of 1/40th of a second, or 1/160th of a second=6.25 milliseconds. That doesn't sound like much time delay, but it is ~7 feet of distance, at the speed of sound.

Put two microphones on a piano- one for the left hand, one for the right; both equally close to the strings/soundboard. Now, impose 6.25 milliseconds delay between those two mics- that is, between the lowest notes and the mid-scale notes.

Imagine what the piano would sound like if the right hand tones got to the microphone seven feet sooner than the left hand's lowest notes, because that's what's happening as you slide down the scale for ANY loudspeaker- and it's a gradual change in phase, which is why we don't complain too much. Any driver does this- headphones, Walsh, electrostats, Lowthers...

A damped, harmonic oscillator also has a high-frequency limit, imposed by its moving mass- which equals phase shift in the highs, or time delay.
It has phase shift in the low frequencies, because it has mass bouncing on a suspension (it's a mass/spring system), as described above.
And since it cannot be an infinitely-rigid cone, it has cone breakup too, which imposes a ragged phase error across the roll-off region, a raggedness that changes with loudness too.

If it has a whizzer cone for the highs, like a Lowther, then there is a time-delay (phase shift) between cone and whizzer, seen as a wiggle in the driver's impedance curve. At that mechanical crossover frequency, the idea is that the cone stops moving as the whizzer starts moving.

Yet the amount of time delay between those two parts is far more than some electrical crossovers would've imposed. Even as the whizzer moves, the cone is also breaking up- parts of it "rattle on" in non-pistonic motion, so the phase change is not smooth with frequency.

Finally, since the forward edge of the whizzer is un-terminated (not damped or otherwise constrained), it has its own breakup modes. Which makes complex, loud, high tones sound hazy, fizzy, fuzzy or dirty (depending on the whizzer's breakup modes).

Any mechanical transition also changes its characteristics with loudness, humidity (possibly) and aging of the materials. You are asking a piece of paper, plastic or glue joint to flex, predictably, for billions of cycles (per week), and flex in a completely linear, proportional manner on the very softest sound and the very largest- often simultaneously.

A mechanical transition is also happening at the leading and trailing edges of the ripple moving down a Walsh driver, or spreading out across the face of a Manger diaphragm. It is hard to find driver materials that do not change very much in flexibility with age, or humidity, or loudness- which means the designers of those two drivers were truly ingenious to get as far along as they did. What those drivers offer is minimal phase shift in their mid-bands- which is good. But neither one can handle low bass, nor is very sensitive.

To check the transition to whizzer (happens in the high-voice range), see if something abrasive such as Janis Joplin, can be tolerated in that tone range. Then listen to her on a good headphone (has no phase shift in that tone range). The whizzer transition could be apparent on massed, loud strings- as wiry, steely, or strident- it all depends on where that mechanical crossover point is in the tonal scale, and how much you aggravate it.

No single driver can cover the whole audible range including low bass, unless it is a large single-panel driver, for which you have to sit exactly in the middle- exactly.

The Lowther drivers and Jordan 5" drivers do have some bass, but not enough to balance out the voice-range when listening > 10 feet away, and they have loudness restrictions: Their high efficiency comes from low moving mass, due to a short voice coil = minimal stroke available for mid bass and lower tones. Look up the x-max specs on the drivers- you'll be surprised.

If a design has a mid cone with no electric crossover, yet the tweeter does, then at the acoustic crossover point, you have:
the electrical phase shift of that tweeter's circuit,
plus the phase shift caused by the tweeter's having its own low-end resonance,
plus the phase shift and ringing at the mid cone's breakup modes(indicated by wrinkles in the impedance curve of that mid driver),
plus the emf sent back into the amplifier from those extra cone oscillations, which gets into the negative feedback loop.

If you have ANY kind of frequency-response roll-off, then you have phase shift (time delay) that gradually comes on as you approach those roll-off points, no matter what object in your stereo, or in the recording studio you examine- amp, mic, mixer, A/D & D/A converters, analogue tape recorders, disc-cutting lathes.

But all those devices have very little phase shift in the main part of the audible range. What they do impose comes on gradually, octave by octave, as you approach the devices' -3dB points, with the exception of certain microphones, like a Shure SM-57/58, that have a ragged phase shift in the sibilance range- lending a hard edge to the voice, often intentionally employed by the recording engineer.

It is the speaker's phase error vs. frequency that is much higher than anything else in the recording/reproduction chain. It is caused by high-order electrical or mechanical crossovers that "twist" or warp the phase in the mid-bass or low treble- wherever those crossover points are. It is caused by cones breaking up (or going soft, ala KEF and B&W) in the middle of their range, before they even get close to crossing over to the next driver. It is also caused by the drivers not being the same acoustic distance from the ear.

To find out what effect any crossover (mechanical or electrical, or combination thereof), has on music, listen to simple sounds that move through those frequency ranges- there you lose depth, clarity and dynamic expression. Also listen to a lot of musical instruments in person, up close, perhaps at a music store on a slow afternoon. Talk with the store's percussionist- let him show you why musicians pick certain cymbals, bell-trees, drum kits, sticks, mallets. Have the guitar person play you some differences in his gear.

Any speaker designer worth his salt needs to know, quite intimately, what goes on in the studio, in the musician's hands. After all, that's what needs to be heard on the other end.

Roy Johnson
Green Mtn. Audio
DIY inspired by Richard "The Doc" Dunn RIP

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Re: Doc modding Marantz imperial 7

Unread post by karatestu »

Bored yet ? Here's another


gma952b65 posts
01-16-2003 1:17pm
Phase shift in the bass is a given.

When you suspend a mass, so it can move, but with proper damping (so it does not vibrate on forever), you've created a "damped harmonic oscillator"- the term in a physics or engineering book.

When one tries to drive that mass with a "tone burst" signal (which sounds like "OOOO"), the mass takes time to reach full stroke. How much time depends on how high or low on the musical scale the "OOOO" lies. And it takes the same amount of time to then stop. After all, the energy didn't go anywhere- it just got delayed. Late start = late finish.

Whatever the amount of time delay, we also call it "phase shift" (# of degrees, 360/cycle).

A "perfect" moving system always has 90 degrees of phase shift at its resonance- a frequency calculated by using the mass, compliance, and damping values. This is in physics 101 texts.

A smaller woofer in a sealed box, properly damped, has a 50Hz resonance (50Hz = 1/50th second per cycle). It also has 90 degrees of phase shift at 50Hz, which means it has a time delay at 50Hz = 1/4th of 1/50th second = 1/200th second = 5 milliseconds delay.

Compared to what?

Relative to the time delay in the midband of that woofer (assuming it has a decent cone and no crossover). Approaching the midband range, the amount of delay declines to only a few percent of the test-tone's period. When we get ~3 times higher than the resonant frequency, at ~150Hz the 50Hz woofer would exhibit ~4/100th of 1/150th second delay, ~.25ms delay.

And since time is proportional to distance traveled, then for sound, 1ms is worth about 13.5" of travel. Thus at 50Hz, the 5ms delay is about six feet. At 150Hz, 0.25ms delay is ~4".

This means the lowest bass tones are heard as emerging from another "woofer" 5+ feet behind the real woofer's upper bass/lower mid location. What that does to the formation of a sharp image or to any transient or harmonic relationship, one can imagine, but we do often mistake that 5ms+ delay for room problems (which have similar 5ms+ delay to/from the walls).

The time delay gets even longer when that woofer is equalized (like most powered subs). Throw in the sub's crossover and it only gets longer still (and rings)- which is why you see someone drag a sub all over the room until it "blends".

If a woofer is underdamped, it resonates on for several extra cycles, which again is nearly always mistaken for room problems because those extra cycles arrived late- from the cone, and from the walls. That underdamped cone also reaches its full stroke later- so it sounds sluggish or "behind the beat". Then the walls get those delayed sounds and put their own delayed reflections on top of them. Then add on the effect of multiple woofers headed to you and to the walls, as they are all differing distances too!

That was for sealed woofer designs. Perhaps the panel speaker designers would explain what they have to deal with.

If the woofer is ported or is mounted in a "transmission line" (a big mis-nomer), or loaded by a horn in the front or rear, you still have the same sealed-box-woofer time delay relationships for the sound from the front of its cone.

You also have the same time-delay relationships for the sound emerging from the port- as it's just a different mass bouncing on the same spring. And its motion is also inverted in POLARITY (not "phase") compared to the motion of the front of the cone.

When we measure the combined port/cone output using pink noise, or by MLS, or using steady sine-wave tones, what we "see" is 180 degrees of shift at the frequency which coincides with the port's max output. At a 50Hz port tuning, that would be 180/360 (=1/2) of 1/50th second = 1/100th sec = 10ms delay.

That is not what we hear.

We hear two different sources separated by the time delay from the extra distance over to the port, and by the extra time it takes to start to move the air at the end of a long "transmission line". And we hear that one of those sources is inverted from the other.... all which means less definition.

We wonder why speakers aren't perfect!

Dr. Butterworth developed the math predicting the response and phase shifts of electrical filters, and Dr's Theil and Small first applied that math to the moving system of speakers.

Wayne asks, "What effect does this phase shift have on bass definition?" Well- studios have to listen to it too, from their monitors- and thus mix for "it" on their pop/rock/jazz recordings. Only on classical, or other recordings where things are left alone (sheffield, telarc, delos, chesky, etc) can an approximate standard of reproduction in the bass be obtained.

We reduce phase shift/time delay at 50Hz ONLY by lowering that woofer's resonant frequency- by adding mass to its cone. Or one can change to another woofer having the same mass but higher compliance (softer suspension), a larger magnet and cabinet. Or use a larger diameter (= heavier) cone that has the same or higher suspension compliance, with a bigger box.

No matter how we decrease the resonant frequency, we hear tighter midbass, "faster bass", more spaciousness (ambience), easier room placement, a more realistic bass image, and better voice and highs. All but the latter two are from creating a better time alignment between the harmonics and their fundamentals. The last two are effects from bass output that reaches farther, as loudly, down the scale.

Of course, using a heavier cone with a longer-stroke (heavier) voice coil decreases efficiency. Which means we turn up the volume. That extra power means an even hotter voice coil on any peaks or sustained loud bass. The heat increases the voice-coil's resistance momentarily, which means less amplifier power is delivered, which means more "power compression". This sounds and measures like a softening of peaks and of any loud, sustained bass.

Some drivers have extra-large diameter voice coils that heat up less, as they are larger radiators of heat. Yet a larger diameter coil is heavier, which reduces efficiency. Unless the winding length is shortened to keep mass the same- which reduces stroke. Also, with a large voice coil there is more surface area in the voice coil gap over which to spread out the field from the magnet, so efficiency decreases, unless you use a huge magnet. A large coil also means less room for pleats in the important rear spyder suspension- and so the cone rocks more. The voice-coil gap has to be widened to prevent that coil from rubbing or jamming, which reduces magnetic field strength, thus efficiency, even more. But at least the large coil doesn't burn out, and it makes for good advertising...

Higher efficiency woofers are more efficient because they have they have less mass- usually via a shorter voice coil. But that means they run out of stroke, and won't play loud. And if you could reduce the cone mass so that one could keep the longer voice coil, then you often have other problems from the lighter cone. Also, for a truly lighter-weight cone/voice-coil combo, one must use a VERY compliant suspension to keep the resonance down at that same low frequency, and to keep the high-end response from tilting up, like a PA speaker's woofer (mid). Yet suspensions are already as compliant as can be made consistently.

Wayne asks, "Has anyone heard phase coherent bass?"
Yes- we all have- but only from any live instrument without a PA system, and from any live voice. Which is why we can hear benefits when we reduce phase shift- it sounds closer to the real thing.

Wayne asks, "If there is a phase shift and you can not go back in time, is it possible to phase shift the good guys to equal the sluggards?"

You bet- digitally time delay everything else in the speaker by the appropriate amount, so that the low bass is not "behind" anymore- and it does help a lot! But the correction machines cannot read the subtle time delay problems that occur higher up the scale- from bad cones and from cabinet reflections, so we wind up "tweaking" by ear. Sigtech-type devices also cannot separate the woofer from the room at the lower octaves, nor "fix" the port inversion problem. In fact, these devices "cutoff" in the midband, before the room starts to confuse their measurements.

If you are going to do the digital delay PROPERLY for a speaker, you have to do it for ONLY the sound which comes right from the face of each cone- no cabinet reflection correction would be allowed.

But ask where does one stop the correction? Besides the speaker's woofer, we have bass time delay from your phono needle, phono stage, any coupling transformers/caps anywhere in the chain, and from the analog master-tape copies, from the original analog master tape, and from certain mics.

Digital mastering (DDD) threw out all that bass phase shift except for three: your woofer, any transformers/caps in the chain, and the mics. This is one reason digitally-recorded bass is better in many ways than analog.

Good questions, Wayne. Complicated answers, sorry- but that's why you do not see "time delay" or phase shift properly covered in the press. The above is taken from what is being prepared for our website.

Best,
Roy
DIY inspired by Richard "The Doc" Dunn RIP

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