Time coherence - how important and what speakers?


I have been reading alot about time coherence in speakers. I believe that the Vandersteens and Josephs are time coherent.

My questions are: Do think this is an important issue?
What speakers are time coherent?

Thanks.

Richard Bischoff
rbischoff
This might help: a link to my postings on "The Vinyl Engine".

http://www.nakedresource.com//yabb/cgi-bin/yabb/YaBB.pl?board=general;action=display;num=1038342561

Best,
Roy
Phase Coherent Base:

Thiel claims phase coherence of +/- 10 degrees. However, a review (Soundstagemagazine.com/measurements/thiel_cs16/, July 2002) of the Thiel CS1.6, shows that the phase angle varies +/- 45 degrees between 50 and 500 Hz. As I understand it, the compression in the box and the inertia of the woofer causes the phase shift.

OK! No box. An article on the Magneplanar MG 1.6/QR shows a +/- 40 degree phase shift centered at the crossover frequency of 600 Hz.

The phase shift includes serious changes to the 1st, 2nd ,and up, harmonics. The harmonics give the richness to the instruments. Drums may lose their impact, except in the fundamental.

My observation is that I have never heard a speaker match the live performance of an acoustic solo Trombone or kettle drum.

Question: What effect does this phase shift have on base definition? Is there a manufacture that has a speaker that is phase coherent in the base? Has anyone heard phase coherent base? 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?
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
Roy, thank you very much! Perhaps you can discuss the pros and cons of systems using "motion detectors" such as the ones used in Velodyne (not to single them out) products. I'm also curious about what you mean by "PROPERLY" when discussing digital correction and limiting it's range of correction. TacT (again not to single them out) seems to approach this by using super light cones and digital correction and very fast(?) digital amplification with a much higher than usual crossover point, any thoughts on this unique appraoach? As you seem to be saying that our ears can easily confuse room interation with actual direct sound, are you suggesting that bass output (woofers and/or sub woofers) might be best placed well into the room ala' Audiophysics (you guessed it, not to single them out) speaker placement suggestions? Thank you again for your enlightening response.
Thanks!

Regardless of one's opinion about motional or positional feedback applied to a sub, both suffer from the limitations of the accelerometer used, the second voice coil being 'read'. Both are transducers, thus with their own dynamic range limitations, frequency response, resonances, distortions.

Also, the sensor often cannot not pick up motion from more than one direction/dimension- it can't see the cone "rocking" very well, for example- it only sees that it has not stroked far enough. However, a 1-D sensor is cheaper. And marketable. It can help a poor woofer or bad box design.

It's far better to design the best possible woofer and put it into a proper cabinet (not easy), and get the phase accuracy of the electronic crossover correct, by choosing certain slopes of the filters:

Use a 24dB/octave slope on a sub and 12dB/octave on the speakers, if they are sealed or ported, and if the sub is not a resonant bandpass design (THX markets this standard pro-sound crossover as something unique). The crossover point belongs near the impedance peak of the sealed box, or near the upper impedance peak of a ported box.

Other option: If they are panel speakers/Quads- don't give them a crossover. Just put 6dB/octave electrically onto the subs only (must be stereo subs for this).

Either way, you get better phase alignment all through the crossover range- which SOUNDS LIKE far less room problems.

To fine tune either: listen with the sub at the same tape-measure distance from your ear as the main speakers' woofer centers. Then move the sub +/- 12" front-to-rear off that plane/arc. As you do so, listen for dynamics and then for tonal uniformity (separate tests).

Listen to something steadily percussive at the crossover point- a kick drum for 50Hz, a floor-tom drum at 80Hz. You will hear the best dynamic "alignment" as you change front-to-rear position and fuss (a little bit) with the crossover point.

Then listen to a string bass run the scale- try Christian McBride on his "Gettin' To It" album, or use a celloist's solo. This lets you find general tonal weaknesses/boosts most easily. Changing the crossover point may not help much, but try. Usually if there is a drop/boost in output in a range of frequencies, that's the room, as that is how rooms behave on music. To address this, pick up the entire speaker, sub and chair layout, and move it out further into the room by 18 inches and hear what happens. Don't touch the volume control or crossover settings- one variable at a time.

Your question about "proper" digital correction:

Walk around a speaker and ask what sort of things need correcting? Can you point to them?

Do we wish to correct digitally for cone breakup? But that breakup "nature" varies with the music's dynamics and tonal spectrum, and that we cannot measure with a test tone!

Do we wish to correct for a floor reflection? That is heard differently by ours ears than by the mic, so the mic's signal is not "accurate"- hence the LF cutoffs of digital correction. This explains a little of the higher crossover point choice: it narrows dispersion and thus reduces cabinet and floor reflections in that higher tone range.

Is it a "splash" reflection of many simultaneous tones, coming off a big, curved front face? If so, ask what is the 1/4-wave dimension of the panel? Because from that tone on up, you get the "splash"- a new bubble of sound launched off the face- and that happens all the way right down next to the dome of the tweeter. This is what the digital units try most to correct for in the direction of your chair only. But that leaves many driver non-linearites uncorrected, as to the mic, those were swamped by the splash. The mic cannot separate them from the splash the way our ears can.

With digital correction, what is going to happen to the sound everywhere else in the room? Not just what others would hear, but what will be coming back to my chair from the walls? Whatever that is, it can't be zero. And it's definitely not "corrected".

One has to ask ALL the right questions of the digital correction situation. Make the speakers better in the first place is my answer. Costs less, sounds better. Then try correction.

As far as placing the speakers out into the room? Yes, if they were designed from scratch for that location. You cannot have it both ways- i.e., a speaker design accurate when placed within a few feet of the wall, vs. many feet out. As that distance from the back wall is varied, the perceived output below 300Hz changes, unless the speaker is the size of a `fridge. This cannot be balanced out with a switch on the back of the speaker. And in my opinion, it cannot be corrected properly by digital means.

Actually, it's not my opinion, but what physics says we are still going to hear come off that back wall (even with digital correction). To any speaker designer, the first question after, "What kind of room do we have to work in?" is, "How far out from the walls can most users live with these speakers, so they can actually hear the depth of the image?" The third question is always, "How high up the scale is the woofer to go?"

There IS a distance from the back wall which is "far enough"- the Hass effect is an indicator of that distance.

Sorry I could not respond sooner- too much work. Website coming soon.

Best regards,
Roy
Green Mountain Audio

A side note: Anyone recommending a really wide-set speaker placement is throwing a bigger acoustic shadow on your opposite ear. Why do that? When they are set up in the normal 48-53 degree spread (the limits of our peripheral vision by the way), their image is unstable as you turn your head- usually because of double drivers, multiple drivers, or line sources.

As you turn your head, there remains a constant acoustic field UNDER YOUR CHIN, because you have a chest making lots of nearfield reflections. But for sound coming over the top of your head, the field is determined only by the direct sound from the speakers. And if there is sound being launched from higher than your head and also from the same height simultaneously, then the opposite ear hears more of the "higher" driver location as you turn- and the image jumps to that speaker. If the source of the sound is more compact than your head, the image does not jump.

Listen to a solo voice with your eyes closed, no eyeglasses, no coffee table in front of you, nor footstool, in a very quiet room, to identify any image jump most easily.