Gentlemen,
I am glad to know my previous writings were appreciated, and it's become easier to explain since then.
I can clarify some basic technical concepts above, without reference to our products, if you like.
Please let me know.
Thank you,
Roy Johnson |
Hi all,
Sorry to have delayed this post-- unexpected duties arose.
I hope the majority will be served by some words below, along with a close study of the diagram I've posted at
http://s1374.photobucket.com/user/greenmountainaudio/media/TimeCoherenceDiagram_zpse8c92f2a.jpg.html?filters[user]=140737398&filters[recent]=1&sort=1&o=0
For whatever reason, I cannot get this to post as a clickable link here, sorry. Perhaps someone else can! At least Cut and Paste works, so please open this image in a new window/tab and magnify, as it illustrates much of what takes too many words to explain.
Here we go:
When a speaker spits out a brief piece of sound, making just a "Beep" then falling silent, what is moving towards us is a chunk of higherthenlower ("wavering") air pressure. The air itself is not going anywhere.
I encourage you to conceive of this as a traveling packet of sound, silent before and silent after, a packet that contains perhaps six wavelengths (six cycles) of a pure tone just like what is emitted from a tuning fork.
However, that one "Beep", high or low ("Boop"), is not a perfectly accurate example for a 'pure tone'. We must imagine instead that "Beep" stripped of its B and p, leaving only the eee or ooo.
That simple burst of an 'eee' or 'ooo' still conveys useful information- perhaps to warn of a car door ajar. Its message comes from its possession of just two characteristics:
1) It has a unique tone, high or low on the scale,
2) Lasting for a unique period of time.
Thus, to make and then hear any message takes both tone and time. Time is important to our sense of hearing.
Non-time-coherent speakers delay bass tones more than voice range tones, and those tones more than its highs. Another way to state this is their highs always come out first/too soon. This cannot be completely fixed by digital delays nor by stepping the tweeter back from the plane of the mid, because the amount of time delay is DIFFERENT at EACH frequency, which also leads to serious measurement difficulties for most designers.
Perfectly time-coherent speakers do not delay ANY tones whether bass, voice or treble.
Designers of non-time coherent speakers quote studies showing we cannot hear less than a 2 millisecond difference in arrival between the voice range and the high treble.
This means they believe it is OK for the mid's voice range to arrive up to ~60cm (two feet!) after the tweeter's highest tones. And even greater offsets/longer-delays are OK in the bass.
When those tests first came out, I saw they misled by only using tones that do not mimic the complex sounds of music, nor even resemble sounds anyone has grown up experiencing everyday naturally, so untrained listeners instinctively would know how 'it' is supposed to sound.
For myself, after years of being intimately near to the artistry of very many world-class musicians and singers, each for long hours, sometimes for days on end (rehearsals), I think "Who are engineers to say time delays are OK-- that somehow those don't screw up the music?"
I have no doubt everyone has watched even simple music bounce along on the computer screen. Know that we are observing only a miniscule faction of that music's waveform.
When a loudspeaker injects time delay, the shape of that waveform changes. The 'wave envelope' changes.
What is in that one shape, inside that one envelope? A zillion different sounds, each occurring with its own unique loudness, tonality, onset, duration, and decay. In that shape also lies the ebb and flow, the give and take, the emotion of the music, and the texture of each sound, its unique timbre ('tam-burr').
And we only get to see and record that one complex wave-shape per channel. So I think best to strive not to change it and hear what happens.
When this time-delay-as-we-go-lower is progressively removed from a speaker's design, along with the sonic reflections off its cabinet surfaces, then we always hear from any recording more and more the sound of people playing music 'over there'. There are no microphones. Our attention is no longer drawn to "the details" such as "the sound of the bass", "the airiness of the highs" or the sharpness of images.
Instead, even an inexperienced listener soon focuses instead on HOW the bass player is responding to the others (and thus WHY). All of the hi-fi 'details' are still there but now serve to shape the tones, to give each sound precision and purity (or perhaps a wandering pace and a fuzzy tone) RELATIVE to all the other musicians' sounds. Also, each musician remains distinctly separate in space, regardless of the music's complexity.
Ebb and flow, sudden changes- all are part of music, and hearing those makes sense to the ear. A much wider range of music is enjoyed, with little effort. Music FEELS good, just as if you were a teenager again, before becoming caught up in hearing all the very cool and entertaining 'hi-fi details'. You also do not need to turn it up.
By the way, speaker designs have become far more time-incoherent since the 1970's. Marketing pressures combined with the appeal to designers of 'new technology' has led to the use of many drivers that require high-order crossovers to operate.
This has given the majority of audiophiles, reviewers and designers loudspeakers that present "details" instead of music ("I can't stand loud rock recordings from these expensive speakers!"). They do not hear this as being a problem due to time-delays for any or all of several reasons:
a) They have not gone to enough intimate concerts, live theater or recitals, or sat in a living room for hours listening to singing or an acoustic guitar, clarinet, piano, a violin, played superbly. In that intimate environment, such sounds are to die for; far more breathtaking that us mere mortals can imagine- until we hear 'it'.
b) They play only a limited variety of recordings to evaluate gear, much of those recordings electronically manipulated, even of acoustic instruments (I read the reviews).
c) They have collected some non-musical gear and cables, as there is a lot of it out there, some very expensive. Very clean, but sterile, devoid of musical flow.
d) They have been told over and over again that time-coherence does not matter. No editor wants to piss off any large speaker manufacturer that uses only high-order crossovers.
e) They have lived with only very time-incoherent speakers, never with original Quads, nor electrostatic or planar headphones.
f) They do not know how to work the math of speaker design from a time domain perspective, and I don't like it either.
g) They have ears of cloth, for which no amount of exposure to live music can help. Fortunately, I "had to take piano lessons" as a youth, enjoying it and eventually playing much music well enough as an adult, from Mozart to Bartok to Joplin, to know how terribly bad I still was compared to any prodigy! Now I am long out of practice, as the piano needs new strings and new action ($$$)-- an Everett upright grand from 1887, weighing 600 pounds with a bronze frame holding a spruce sounding board. Sorry to digress.
So, we have that "eee" from the speaker coming towards us at 343 meters per second, no matter whether it's a bass tone or a treble tone. It will arrive at our chair in about 10 milliseconds, to begin to push or pull on our eardrums, because we are sitting about 3 meters away. Unless the speaker delayed when this sound came out.
A jet liner up high is cruising at ~85% of this speed. Sound is about ten times faster than cars on a distant highway. It is the speed of that blast of pressure coming at us from an atomic bomb. We could see ordinary sound traveling if the air weren't so clear.
Non-time-coherent speakers create time delays by their choice of drivers, those drivers' locations from your ear, and the type of crossover circuits used.
Drivers have both mechanical time delays and electrical delays, as they are Transducers, which operate in both domains.
A tweeter may be stepped back from the mid, to "put it perfectly in phase with the mid" at the crossover point only. This still does not make for time-coherent operation (study my diagram and my other Audiogon posts in the links others supplied above, thank you).
ALL crossover circuits introduce time delays, but only first-order crossover circuits create time delays that naturally offset each other, when crossing from the tweeter to the mid, and mid to woofer, thus producing no RELATIVE time delay between the drivers, which is the (my) goal.
I hope this helps! For those wondering what the step response or the impulse response indicate in Stereophile, I advise you to study John Atkinson's explanations of them and to remember that, in those tests, nothing nearly as low 'C' above middle 'C' is shown. That takes a large, expensive anechoic chamber or careful measurement outdoors with the speaker up on a pole, far away from the ground, using a very loud pulse, one that likely damages any tweeter.
The detractors of first-order speakers talk of power-handling issues caused by the slow rolloff of the crossover allowing bass to get into a driver, even a tweeter, making it distort or melt. Maybe the mid's cone would ring in the tweeter's range, because of its strong resonance at a high frequency. For us, there has been no problem because we use the best drivers, by anyone's standards.
Detractors also claim there will be off-axis cancellations between drivers, leading to a weird tone balance for a listener off-axis. Not true with proper crossover points, slender cabinets, and a lack of cabinet reflections. Never is the math behind those claims shown, as it never supports them.
To the original poster- thank you for this opportunity, and know that one reason a tweeter is placed behind the plane of a mid is because the sound from that mid emerges first from down deep in the center of its cone, no matter the crossover slopes used nor the mid driver's design.
Best regards,
Roy Johnson
Designer
Green Mountain Audio
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Kiddman, no one benefits from your insults.
If you did the math, or at least read my technical papers and relevant papers in the AES Journals, and above all hear what everyone we know hears, you would agree with me, no doubt.
You could read my letter to six moons describing the problems that measuring speakers presents, and WHY each measurement technique has particular problems. No Roy opinions there-- just scientifically-tested facts accepted by the AES.
The complete sets of measurements we post on our website for our speakers are far more detailed than any others anywhere out there.
You are wrong about being able to make a highly-distorted speaker somehow time-coherent. Its drivers themselves would not even be minimum-phase to begin with (that is, well-behaved) to be able to employ the required first-order crossover.
Best regards,
Roy |
Sounds Real,
For an eyeball estimate, the acoustic center is approximately where the voice coil former meets a cone or dome- the glue joint. But this is true only in the upper-middle range of any driver, whether tweeter, mid or woofer, where each one's frequency response is still flat.
To measure it (within +/- 1/8th inch at best for a tweeter, much more for a woofer), one sends an impulse, a click, to a driver having no crossover.
On a `scope, one examines when that click arrived compared to when the `scope's sweep was triggered by the click electrically.
Now, what we are looking for as markers will not be the top of those two spikes that click generated. We are looking for when each spike just begins to turn upwards from 0.0 at its bottom-- when each just begins to rise up. That is a very difficult transition to judge, which leads to inaccuracy.
Regardless, that time-difference times the speed of sound is your distance from the mic to the acoustic center in a driver's upper-middle range. Compare that to the tape measure distance and you often get close to the eyeball estimate I mentioned above. Of course, the test mic will be expensive, not a $200 special, for those cheaper ones have their own phase shifts in the audible range. Figure $1000 for a proper mic, plus a $1000 wideband mic preamp. Even so, the results will still be rather inaccurate. I was able to find ways around this, fortunately.
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Unsound,
It is not that the two woofers are equidistant from their surfaces but the fact that we have two (four in stereo) woofers rather near two surfaces with you living in between.
Do have a look at that new drawing I posted to get an idea where "the bass source really is", which is my red dot in that drawing. Imagine what standing waves would then occur in between a red dot on the ceiling and one on the floor when the measuring mic/your ear is placed somewhere in between. Double trouble has been my experience.
No doubt about the outer horn-surfaces making reflections. But those reflections would go mostly upwards, and we can apply wool felt or acoustic foam to minimize most of them. I still think the biggest problem to be getting far enough away from the speakers so stand up/sit down differences would not drive us crazy- a large living room, say 30 x 40 feet is probably enough.
I would like to hear a Klipschorn corner horn triamped with time-delays applied to its mid and tweeter, since the woofer is so far back inside (~4 milliseconds) and the tweeter is so far in front of the mid driver (~2ms). Again, one would be stuck with using second-order crossovers on the drivers with the mid driver in inverted polarity.
Best,
Roy
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Certainly, but a lot of other problems would be fixed. |
I forgot to mention a couple of things:
When a driver is being run full-range with no crossover or Zobel, its changing impedance curve has no effect on its tone balance when using solid-state amps, but only on tube amps via interaction with their much higher output impedances. For a tube amp running a 'full range' driver, a voice-coil Zobel circuit on that driver would return its tone balance to 'factory spec'.
When a speaker has a flat impedance curve, that does not indicate if this speaker is time-coherent. From the outside, all we can see is how the many different impedance curves I described above combine into one curve.
Best,
Roy |
Hello to all,
I am happy to answer some questions on design. Before that, I need everyone to truly understand the simple difference between the definitions of phase and time coherence.
Please open that earlier link to my illustration, and study the differences between waves combining. Then consider the following:
1) Time coherence-
Send the speaker a 'beep' near the crossover frequency between mid and tweeter.
Unplug the mid.
The beep coming from just the tweeter arrives at the ear 'X' milliseconds after the signal left the tweeter.
Now, turn on the mid; unplug the tweeter.
If the beep from the mid arrives at THAT SAME INSTANT, the mid and tweeter outputs are TIME COHERENT at that one frequency (perhaps not at others).
2) Phase coherence-
The mid's beep arrives at least one FULL cycle later than the tweeter's. Thus, their peaks and valleys still line up, making the two drivers IN PHASE. Yet they are not time coherent since the two beeps' beginnings and endings did not line up.
Any decent speaker is PHASE COHERENT. If not, cancellations occur at its crossover points. To market that a speaker is 'phase coherent' or 'phase linear' is only a 'feel good', to impress those who know little of speaker design.
Referring to my illustration, note that a time-coherent speaker is automatically phase coherent. It is redundant to write that a speaker is "time- and phase-coherent."
A speaker that is 'phase coherent' or 'linear phase' you can bet is not time coherent. Several speakers companies claim to make time coherent designs, but Stereophile tests reveal those claims to be completely false.
By the way, metal dome tweeters are no lighter than soft-dome tweeters. Visit Madisound.com to examine the specs of the best tweeters for yourself.
There are other misunderstandings I would like to address, but first, everyone must have a clear understanding of what IN PHASE means versus what TIME COHERENT means (hint- the latter always involves a stopwatch). For your own edification, I strongly suggest all of you discuss some examples such as:
Two cars traveling along the highway, one always fifty feet behind the other. As they travel, their RELATIVE phase is UNCHANGING (their phase relationship remains CONSTANT).
Two cars start off at the same instant, and travel along side by side. They are again in phase, since their relative positions are unchanging, and they are also now time-coherent.
===========
Two bicyclists crank their pedals at the same RPM, in the same gear. Thus, they travel at the same speed.
But notice when one rider's pedals are UP and DOWN, the other rider's are always in some other position. Because both riders' RPMs are staying constant, the pedals' RELATIVE PHASE remains constant. But these two sets of pedals are not "IN PHASE" with each other, since their peaks and valleys (their ups and downs) are not happening at the same time.
Time coherence, at its most fundamental, is about beginnings and endings lining up. Phase is only about peaks and valleys of any REPETITIVE cycles lining up. Music has BOTH characteristics.
Hope this helps.
Best,
Roy
Green Mountain Audio |
Again, I look forward to people coming to understand the concepts behind my waveform illustrations. This understanding is necessary to our discussion here, because we then have agreed on the nature of these concepts at hand and also on some vocabulary.
And I wish my answers could be shorter, but that would leave out necessary details- the same ones glossed over/ignored by the press.
Bifwynne,
DEQX seems fine in theory, and certainly makes a positive difference. For me, it has serious limitations because it cannot measure exactly what needs to be corrected. This leads to results that depend on the music being played and sometimes a limitation in one's seating position.
In particular, DEQX cannot see the immediate reflections from the cabinet surface surrounding the tweeter. It cannot correct properly for anything happening below middle C because of floor-bounce effects on the microphone are not the same as they are to our ears on music.
There are other issues, but to me, those are the two largest ones. I find that a much higher level of coherence is achievable passively.
Regarding Treos-- I've not heard them and it's never good for me to comment on the sound of other speakers. I say to trust your ears above all. And you are right to listen backwards and from another room if possible! I can point out Treo, like other Vandersteens and most speakers for that matter, has a terribly complex crossover circuit, made of what I know are not the most transparent parts.
Ngjockey, thank you for your comments. Our experiences with everyday sounds and noises allows us to use them to imagine the SHAPE of a sound, which means its starting and stopping, and what happens in between. The next step up the chain is to imagine the combination of that sound with another, or literally just hear it via programming a synthesizer.
Remember, in the word 'beep', the opening 'b' has its own shape, since the lips are opening. That 'b' is is a CHANGE that happens along a certain TIMELINE, and we recognize its waveform's CHANGING SHAPE as unique to the letter 'b'. The same happens at the end when the lips close, but like 'p' instead. When you imagine hearing only the the middle 'eee', that is exactly what comes out of a sinewave test tone being switched on and then off. Which is exactly what I illustrate in my drawing.
Unsound, I am glad you find my comments useful. Thanks! Using digital EQ to treat room problems seems like a good idea, but again, just know that what you will measure is not what you are hearing-- not to say there will not be some or even a good amount of improvement. If it is used just for subwoofer correction, there are issues in most every sub's design that look exactly like room problems to the measurement microphone.
I think it best to first measure and correct a sub up close with the mic, then use that correction as your basis for further corrections YOU HEAR out in the room, listening to string bass run the scale and to kick drum (the first for tone balance; the second for transient alignment with the main speakers).
Mr. Dunlavy decided early on that driver symmetry about the ear was important, since it was important to his microphone and to his antenna-derived math. Turns out that when you are seated, the MTM arrangement is not important to the ear. In particular, you hear the tweeter's sound come from the mid when just one tweeter and one mid operate time-coherently without cabinet-surface reflections.
An MTM arrangement, including the infamous D'Appolito arrangement, always places one mid above our heads. This causes the image to be unstable with small head movements, and just plain poor for anyone off center. Why would this be? Because we have a head between our ears and a chest below them. Thus, with a small head movement to the left, much more middle-range sound literally leaks over the TOP of the head to the right ear than it does from the mid placed below the head, so the image jumps to the left speaker. This is also true for any large sound-source, such as a panel speaker or a so-called 'line-array', for the same reason.
When only one mid is used time-coherently with one tweeter, and then placed right in front of us, that single source of sound then leaks over the top of the head by the same amount no matter how much we rotate, move sideways or even stand backwards! So the image remains stable, even when far off-center, if and only if the speaker baffle is also narrow and reflections are not allowed from around the tweeter and mid.
MTM also leads to room placement issues, since sidewall reflections throughout the voice range are more complicated than when only one time-coherent tweeter and mid are used.
Time-coherent Coax operation ala Thiel would be best, except there's no way to avoid the intense tweeter reflections off the mid's cone. Also, a terribly complex crossover is required to get the tweeter's timing right. There are other limitations.
Finally, with two woofers placed high and low, for a WMTMW alignment, the bass response in any room becomes unpredictable, since you are driving bass near TWO boundaries, with your ears trapped in between.
Since everything is a compromise, a one-woofer arrangement works best when the woofer is a certain distance from the floor, in medium-size rooms, with a certain crossover frequency. But in those rooms, the bass output will then be predictable, which helps me. Nothing wrong with having the extra 'slam' from two large woofers- it just requires a very large room to make them perform as one. Then again, a very large room I find uncommon.
Speaker placement/spread is similar for very many speakers using slender front baffles, regardless of their crossover design, when these speakers are placed in 'good' rooms. This is because we need to hear a certain amount of crosstalk for the image to be continuous.
Sidewall reflections and reflections off all the fancy gear piled up between the speakers affects the final spread and the toe-in. Speakers having a large amount of reflections off their fronts are sometimes used with less toe-in, so those reflections are not shot as directly into one's ears. When there are many center-reflections (from that gear or off a video screen), toe-in is reduced. When a speaker is not time-coherent, its particular phase shift may mean those speakers sound best placed close together, pointed nearly straight ahead.
Sealed box is the best for woofers, but the market prefers more efficiency and compact enclosures, so our woofers are smaller, requiring a port. Our new three-way coming out uses twin 6.5-inch woofers, each ported at 40Hz in its own enclosure, for a sensitivity of 91dB with the same cone area as one 11-inch woofer. A single ten-inch sealed-box woofer would be in a cabinet half again larger, with only an 88dB sensitivity (requiring twice the power). The mid and tweeter would also need to be turned down by 3dB -not a great solution.
Again, I hope this helps! I realize other questions still remain, posed earlier in this tread, but I thought it best to get these out of the way right now, so I can look forward to folks' thoughts on my waveform illustrations. I will endeavor to cover the other questions soon.
Best,
Roy |
Hi to all,
Bombaywalla, you ask-
"So the question then becomes: Doesn't the presence of that inductive component of the driver impedance (especially in the case of the tweeter) cause a deviation from first order 6 db/octave behavior? And if so, to a degree that may audibly compromise phase and time coherence? And if so, is that or can that be compensated for in other aspects of the speaker's design?"
My answer is YES, but we and others use simple Zobel networks on woofer, mid and tweeter. These offset the changes in impedance at high frequencies.
Normally, the impedance of a woofer, mid or tweeter (= 'driver') becomes 9, 12, 30 Ohms as we go higher and higher up the scale (= "inductance"), instead of staying constant at say, a steady six Ohms, which is what any type of crossover circuit wants to see-- a flat impedance 'curve', so that all of its capacitors and inductors do what they are supposed to do. Any change in impedance literally turns some of those circuit parts off, with the result 'not measuring right' to the microphone.
A Zobel circuit offsets a voice-coil's rise in impedance with increasing frequency, and is quite simple to construct: a capacitor is connected to a resistor (= 'in series'). Those two are then placed in parallel with the driver's +/- wires, the capacitor connected usually to the "+" and the resistor to the "-". I hope that is clear!
With those two Zobel parts placed 'across' a woofer, mid or tweeter, the result is literally a 'Y-adaptor' to the signal coming from that driver's crossover, because two paths now exist for the signal. One goes through the driver back to the crossover as usual, and the other through that Zobel cap, then its resistor, and thence back to the crossover.
That woofer, mid or tweeter's impedance is still going up and up the higher up the scale we go, but our Zobel's capacitor has an impedance that is going down and down by the same amount/at the same rate. This is its 'correction', with the resistor limiting/shaping the amount of correction provided.
When those two paths are made to change by 'equal and opposite amounts', that driver's crossover circuit then sees 'no change in the impedance at any frequency', so goes the theory.
Where most every designer goes wrong is by making the assumption that the electrical impedance curve of the voice coil is what one is measuring and correcting. Not so, unfortunately. There are many other impedance curves that overlay, thus hide, the real electrical curve one is looking to flatten. These other impedances include:
- the mechanical impedance of the driver's suspension and any ferrofluid used.
- the acoustic impedance from how each driver is coupled to the air in front of it, and behind.
- any cone/dome flexing (= mechanical impedance changes).
- the mechanical impedance changes caused by the size of the boxes used for woofer and mid, and a tweeter's rear-chamber.
- what happens in the various types of fibers placed behind drivers to absorb their rear waves.
Then (!) most of those change with loudness, especially with 'average' drivers. Some of those also change when a voice coil is moving inwards versus out, again especially on 'average' drivers. Visit the Klippel company's website to see some of their measurements for these problems, now done automatically by their unique computer and programming- such a smart designer! I had to perform them manually, darn it. On our website, I describe much of what can be done to minimize or avoid these last issues.
Finally, a few years ago, I found a way to use the values of cap and resistor in that simple Zobel circuit to perfect our final acoustic phase down to near Zero at all frequencies. While I still must update our website about this, it's never been accomplished by anyone, as seen in the values they still use in their Zobels. If we sought a patent, I'd have to reveal how to come up with 'the right numbers'.
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FYI, it is a big mistake to flatten the low-frequency change in impedance caused by a woofer or mid's box-size and a tweeter's rear-chamber size, by using a different type of Zobel circuit. Those who do this type of correction do not understand what these impedance rises at resonance actually represent electrically, mechanically and acoustically. The result is poor sound, and usually a very difficult speaker to drive in the bass.
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Addressing another question of yours right above: Thiel/Small parameters are a guide only to box tuning, nothing else. However, those equations turned out to give, at best, only an approximation of the correct box size, because the impedance values plugged into it are not 'right' because we have left out all of those other impedances I just described.
Box-modeling software relies on those simple equations, so they cannot give you the exact box size for a woofer or a mid. One must build several test boxes to determine the actual 'best size'.
FYI, Seas' best drivers are their Excel line. Even so, have a look at the high-frequency cone-breakup resonance in their best metal-cone woofer. Designers believe a high-order crossover and a notch filter 'fix' that problem. Not true, as that cone resonance is also triggered by lower-frequency sounds and 'noises'.
Example- your car's dashboard buzzes from the low-frequency 'thump' of a pothole. This concept and its math are taught in high-school level physics, which is what most speaker designers never study. It is why those metal-cone drivers still sound 'metallic'. Stereo magazines and reviewing websites never mention these facts, but then again, it makes sense how they do not want to upset any advertisers!
For tweeters having a strong ultrasonic resonance from their metal dome breakup, the same thing still happens,with the ultrasonic HF resonance ringing out. However, what is heard instead is its effect on the audible-treble tones. That is a 'modulation distortion', and sounds like perhaps a 'zing' to the treble, or again, a metallic sound. These are all factual statements supported by physics theory and math, and by measurements. They are not 'Roy's opinions'.
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Omsed,
The phase shift of that one inductor used on a woofer with its Zobel produces -45 degrees of phase shift (= time delay) at the crossover point. The single capacitor used for the tweeter's first-order crossover gives the opposite shift of +45 degrees (= time 'advance'). The difference between these two is 90 degrees.
When a website or text makes this mistake, that writer had never looked at the simple math involved, which any competent electrical engineer should have learned in their first Filter Theory class. Only hearsay is being passed on to you, including the non-existent 'downwards tilt to a first-order speaker's radiation pattern'. A totally bogus claim. There, the math was completely mis-interpreted.
The important aspects of this 90-degree DIFFERENTIAL produced by a simple first-order crossover, proper Zobels and really good drivers are
a) it remains a CONSTANT 90-degree difference between those two drivers as we go up or down the scale, and
b) that constant difference of 90 degrees allows the sonic outputs of those two drivers to always add up to the one original wave, having no added time delay, which is totally non-intuitive.
A 'perfect summation' happens because those two drivers are operating 'in quadrature' (90 degrees being one-fourth of 360) at every frequency. The math involved shows their outputs, one lagging, one leading, really do combine to make only the one original wave having neither lag nor lead. Weird.
No higher-order crossovers can maintain this CONSTANT phase differential, so they produce a time delay, a phase shift, that changes with frequency, perhaps 'linearly' but always changing.
This varying time-delay is what DEQX-type components are trying to correct, and what regular digital crossover circuits never attempt to correct (offering only fixed time delays, such as one millisecond). To correct the varying time delay, a heck of a computer is required, hence the high cost of DEQX type of gear.
Measurement issues and limitations still confuse DEQX type of gear, for two reasons- we cannot (yet) program that computer to how we actually hear on music, and that a measurement microphone cannot resolve the (countless) reflections off the front of a cabinet. If I had spent money on a DEQX, I would first place an "F-11" pure wool felt all around the tweeter, and then run the calibration routine.
Best,
Roy
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Sounds Real-
You are indeed right about 'just tilting back the front face'. That can be enough to line up the acoustic centers of woofer/mid and tweeter, so that the drivers are possibly in their best positions to combine properly at your ears, no matter what crossover design is used. The high-order crossover circuits then put more and more time delay on the signal the lower and lower down the scale we go. That cannot be fixed.
And then to make the first order crossover work correctly, one must choose the correct drivers to begin with. I hope this clarifies a bit more for you.
Omsed-
You ask "though the difference between the woof and mid remain constant, there is a difference, yes? And that means that the wave launch of a transient will not be the same for the 2 drivers, correct? The are not time aligned, it would seem. Even if the sum of the outputs through the crossover point remains correct, are we not stuck with the constant time differential between the 2 drivers?
Could you tell me what I am missing? "
Yes, I agree. Again, with the right drivers, Zobels, and first-order crossover design, there will always be a time difference created by that constant 90-degree differential, a constant difference 'in degrees only' at every frequency we examine.
90 degrees is one-fourth of any sinewave's period. At a 3kHz crossover point, that wave's period is 1/3000 of a second. One fourth is 1/12000 of a second. This is the time-difference between the mid and tweeter at this frequency. If we choose 1000 Hz instead, the time difference would be three times longer, 1/4000 of a second.
I can only tell you that the math of "two waves of the same tone traveling out of phase with each other by 90 degrees" will measure and sound like one wave having no time delays. Perhaps you must do the math yourself to see this-- I certainly understand that feeling! Again, the key words to look up are "operating in quadrature".
Bombaywalla-
I apologize if I gave the wrong impression. T/S parameters are quite important, as they tell us a great deal about how the driver will perform in any box.
They just do not give the exact box size, which was the hope. The error can be 10 to 20% off of the correct box volume.
A real test box's performance is determined by listening and then measuring its impedance curve and resonant frequency, to find out the Qts and Fs. That tells us how close we came to meeting the T/S ideals with that test box. Then build another...
Best,
Roy |
Hi Unsound, Thank you for your thoughts. The use of multiple subs does smooth out standing-wave issues. The math used for the theory behind that is formed from adding together the simple sinewave/wavelength equations for standing waves you have seen for bass tones and room modes before. That is fine for long-running test tones, for movie sound effects, and certainly for a pipe organ. The test tones used to adjust those multiple subs are long-running, and not found in music. When the time-arrivals at the ear between multiple subs are 'excessively different', you would think we'd hear stumbling or mumbling on string bass, drum kits and perhaps even cello. But if those subs are not allowed to go above ~40Hz, those issues are bypassed. === WMTMW bass problems arise from both woofers being close to the bottom and top surfaces of our room. This is a 'very symmetrical' situation, which always produces the strongest standing waves. Another 'very symmetrical' layout would be subs placed in every corner. Have a look at this drawing: ReflectionsAlso, do note that WMTMW woofers operate to 150 or even up to 300Hz, which is above middle 'C' on the piano. In these upper ranges, changes are very audible standing vs. sitting vs. walking into the kitchen. === You ask about the over/under head effect of an image jumping when hearing live sound from vertically-large concert speakers. Good question. I can say I've never heard that problem, including from long line-source speakers. Remember, most concert sound systems are mixed close to mono, so everyone hears everything. And in most live situations, sound from a tall concert speaker comes to you from a narrower vertical angle than when at home listening to a six-foot tall speaker ten feet away. Also, I probably did not make it clear enough before that the over/under head leakage of sound to the opposite ear is caused by the WMTMW use of double mids, not double woofers, because of those shorter wavelengths vs. the size of our skulls. === We get reflections off any hard surface-- it matters little that a Thiel's mid surface might be flat or corrugated around its coax tweeter. This is because any 1" tweeter, without a several-inch deep horn around it, is omnidirectional below 5kHz. That means it pushes waves between ~1kHz and ~5kHz across the face of the cabinet, since they cannot escape to the rear. So those pressures escape to the front as they move across the face of the cabinet. Hence, reflections. === Putting the measuring mic for DEQX up close to a speaker is pointless (except for fixing up a subwoofer), as what the mic would then be hearing is coming from drivers at much different path-length-differences to the mic compared to the path-lengths to an ear ten feet away. We all know how walking up to a speaker changes everything we hear. Perhaps they are suggesting this for fixing one driver at a time. That has problems too, because any driver's tone balance is different at ten feet away vs. ten inches away. === Horn speakers can be made time coherent, but our best technology leads to that speaker being at least a four-way if not a five-way design, to stay far enough away from horn cutoff points on the low-end of each driver, and the high-frequency breakups which come from running a large mid high into the upper voice range, and a compression driver with a 4-inch diaphragm into the high treble. Also, with 4 to 5 horns stacked up, their vertical height would make for very strong changes as one stood up or even just sat higher. The nicest sound I ever achieved on horns was to use the lowest order of electronic crossover possible (12dB/octave, 'second-order') on a three-way horn system. The tweeter horn was moved far back on top of the mid's horn, and mid horn `way back on top of the woofer's folded horn, to equalize the driver-to-ear distances for people twenty+ feet away. This describes a system I put together for Taj Mahal. I had to add a small amount of EQ to smooth the mids, boost the ultra-highs, and for flat output to 40Hz. Of course I had to reverse the polarity on the mid horn because 12dB/oct. crossovers need that to avoid cancellations at the crossover points. Since everyone was 20 to 70 feet away from either the left or right speaker (mixed to mono), everyone heard a smooth blend from a speaker whether seated of standing. Sure there was phase shift from those speakers, but it was far less severe than any higher-order crossovers would have been. I received very many compliments on the ease and clarity of the sound. === I hope everyone sees my answers are lengthy because I include WHY something is audible or will measure a certain way, so you finally get a proper technical perspective on the VARIABLES that must be considered, and also HOW they must be considered. Magazines and reviews leave out all these variables-- make of that what you will. Best, Roy |
Ngjockey,
I have looked at this site for very many years. The Soundwest site has enough errors to mislead someone relying upon it for 'basic information' and a bit of the math.
Specifically:
In its Section one, the author does not understand a tweeter is still not time coherent when its wires are flipped over to invert its polarity (paragraph 3). He goes on to mis-represent the amount and degree of cancellations between mid and tweeter when the tweeter is not in the right position (below Fig. 5). What he presents instead is a graph showing TWO IDENTICAL, PERFECT, FULL-RANGE DRIVERS interfering, not a graph of one mid crossing over to one tweeter.
In its Section two, the information in the paragraph below Fig. 10, about phase shift and its audibility on square waves, is just plain wrong (even stating we can't hear it, then giving real examples of how we can hear it).
In Section four, on the audibility of phase distortion, not only is he wrong about its audibility, but he goes on to present an argument based on sound coming from live instruments.
He does not get it that we want to PRESERVE whatever phase relationships exist in the music, no matter where we sat, no matter where the recording microphones were placed. Can you spot the big flaw in his argument based on hearing live music? I have seen this exact bad-logic presented on many other forums as the main reason not to bother with making speakers time coherent.
In his Conclusions, he claims the room acoustics and bad recordings will hide much of what should be gained from making the speakers time coherent. To me, that makes it obvious he's never lived with time-coherent speakers for any length of time.
He mentions how a little pair of speakers in his workshop will reproduce a square wave at one frequency if he holds the mic in just the right place. I can see he does not recognize those speakers likely still have a phase shift of 360 degrees at some frequency, and how that will make a CONTINUOUS square-wave signal still appear square.
He does not remember that 360 degrees of shift at some frequency means the previous square-cycle is then projecting/delaying some of its frequency-components INTO THE NEXT CYCLE, and so on. He should have been examining only the first half of the very first square-wave cycle-- its first up-and-down only, to figure out what a speaker is doing.
===
His are the answers I find quite common on the web, but not in most of the professionally-reviewed papers published by the AES. Their important papers on speaker design can be purchased by anyone as their three Audio Anthologies books. There are still errors in too many of those, but one must know calculus and physics quite well to find them.
I think the general public should not take a writer's claims about audio design for granted, unless the writer also presents the scientific concepts and logic behind those concepts, and WHY those have to be correct. Which is what I've endeavored to do.
Best regards,
Roy |
That's how I was presented this subject. A good link, thank you. Made me flash back to all the horrible homework involved. And then, as the math of physics became ever more advanced during grad school, one wound up using this math daily...
Best,
Roy |
Here are my answers to important questions posed earlier, and some clarifications. To the OP: Psag, you originally asked if a sloped baffle is important. Speaker designs that avoid this are instead using the phase shifts of their crossovers to make sure there are no cancellations/suckouts in frequency response. That is about all their designers look for/measure during the design phase, since they do not make any measurements in the time domain. I think those designers would have an easier time developing their high-order crossovers if their drivers were first stepped back from each other, as on a sloped baffle, and they got rid of the sonic reflections off their front surfaces. === Bifwynne, at the beginning, you asked "perhaps someone could explain in layman's terms what causes speaker to operate out of phase. Does it have something to do with the use of caps and chokes in the x-over? Or perhaps the attribute of a dynamic speaker creating its own back EMF by reason of the voice coil moving in a magnetic field?? Incidentally, do all these electrical dynamics operating in tandem cause the electrical phase shifting that gives most amps a headache? " Let us begin with the phase definition. If a speaker's woofer and tweeter were out of phase more than 'a bit', they would show a dip or even a complete suckout in frequency response, at or near their crossover point, with the microphone placed where your ear would be. As we see from Stereophile's tests, most speakers do not have this issue. So all of those must be "in phase", "phase coherent", "phase linear", or "phased aligned" As I explained earlier, that does not mean they are time-coherent speakers. As a reminder, the opposite IS true: time-coherent speakers are always phase coherent. What makes the phase go weird? -- In the speaker cabinet, it is from the drivers' locations/no stepped baffle, and having too many drivers per frequency range. -- Any crossover circuit's inductors and capacitors delay the signal or advance it, respectively. Resistors do neither. A simple first-order crossover circuit has an inductor going to the woofer, and a capacitor on the way to its tweeter. At their crossover point AND ALL other frequencies, the time-delay created by the woofer's inductor is precisely offset by the time-advance created by the tweeter's capacitor. This is not possible with higher-order crossovers, because the values of their more-numerous inductors and capacitors cannot offset each other. -- The back-emf from any driver is also a contributor to time-delay in its lower-range, whether woofer or tweeter. Thank you for pointing this out. I should have mentioned this earlier. That back-emf situation is altered by the type and size of the cabinet behind a woofer, and the size of any rear-chamber on a tweeter, and from ferrofluid in its magnet gap. -- Any cone or dome breakups change the arrival-time as we go up the scale, but mostly we would hear ringing, sibilance, maybe 'dirt' being added to the music. Regardless, the best cones will not show a loud ringing at some frequency (as with most metal cones available in 2014) nor have a ragged frequency response in their upper ranges. -- And yes, all these phase shifts will talk back to the amp. However, the crossover circuit's design is the primary cause of large swings in a speaker's impedance curve, above 100Hz. Those variations are 'electrical phase shifts' only. These swings in impedance do not reflect the acoustic phase at one's ear- no direct correlation. The amp gets a headache because large swings in impedance means its output voltage (the pressure it puts on its electrons) is no longer sync'd up with WHEN those electrons are allowed to move by the crossover parts (inductors and capacitors). When the values of those caps and inductors do not offset each other, the result is exactly like pushing a child on a swing at the WRONG time. === Bifwynne, on the first page, you speculated on the effects of mics, of recording and mastering, processing, playback, etc. Each of those areas has unique problems, which do not sound like phase shift from a speaker. Each process produces a time delay in the highs and sometimes the lows, but only a speaker can put phase shifts (plural) across the main tone range. Also, whatever that signal is, I see no reason for home- and studio-speaker designs to distort it more. On that same page you asked "How are small speaker manufacturers able to design speakers without the benefit of the R&D budget, engineers, and testing facilities that some of the larger manufacturers have at their disposal?" For me, it's been knowledge, education, and longer, much wider experience. My talent seems to have been expressed as an ability to make the cognitive leap between seemingly unrelated factors, which then made one more link to hearing vs. measurement. All of this has led to me not needing an anechoic chamber (I can always go outdoors for that). I also found the fancy digital test gear gave misleading and often incorrect numbers, compared to analog test gear. When a designer does not really understand the fundamental physics of how and why drivers move and respond as they do, nor how crossovers delay the signals, then their only recourse TO IMPRESS their board of directors, is the anechoic chamber/digital route, for that is what the AES and any university would also advise those board members responsible for hiring 'a great designer'. Such a designer then blames the sonic differences between his and other speakers as 'we all hear differently'. His board of directors and all reviewers and editors gladly go along with that bullcrap. We all certainly listen for different things. But here we have found, that as a speaker is made more and more time-coherent, everyone AGREES on the sounds heard in each and every tome range. They all hear 'the bass' in the same way, etc. === Ohlala, on page one, the possibilities of off-axis cancellations you mention turn out to be non-issues on music, especially when the cabinet is not large, and has little sonic reflections from its surface. === Timlub, on page one, your speaker design is only phase coherent at its crossover point, not time coherent, as you may know. Your electrical crossover slopes work well because they are combining with the phase shifts of your particular woofer and tweeter, which I am sure you suspect. Thank you for sharing your experiences! Appreciated. === Bifwynne, you ask too many (good) questions! On page one you ask, "here the ultimate Q. How can one tell whether a speaker is time and phase coherent? Critical listening? Reviewer comments? Bench test? If critical listening is that important, the real challenge for us is, as many have written, that it is not easy to meaningfully audition speakers. So what's a person to do? I'll ask again, how important is time and phase coherence? FWIW, ... really more as an FYI, ... Paradigm's web site states that its 'speakers have phase coherent crossovers designed so that the summed output of the drivers is completely and accurately rejoined.' Is that hype? It is true at all frequencies?" On my website, I have suggestions on how to audition speakers. I know these work. They are simple, taking only time and effort. The time-coherence part of the audition sounds like clarity and depth, and when time-coherent speakers are designed with the best parts, the musicality is greatly improved. With the very best, you find yourself never, ever thinking about 'the sound of the bass' or 'the highs'. Instead, you subconsciously always focus on the music and how it is being played, and its emotional and physical connection to you. When a speaker is time-INcoherent, the music is fragmented, leaving you to hear only 'the details' and 'the soundstage' or 'the air', or 'the impact'. Right now, I see only Green Mountain Audio, certain models from Thiel, and Vandersteeen as making time-coherent speakers. The Audio Machina company is part-way there. With any others claiming time-coherence, I've seen no proof on their websites, or in Stereophile tests. === Ivan_nosnibor, I appreciated your thoughts, thanks. However, the time delays in your digital crossover circuits are fixed time delays for each driver, when the real problem is the amount of time delays are different at each frequency. You remark on hearing perhaps the highs 'imaging closer to you' on non-time-coherenet speakers, with the mids 'not projecting as far into the room', and so on. I have found instead it is about the lack of depth in the highs, caused by the smearing of a late-arriving mid, and so on down the musical scale. WHEN the highs arrive is not WHEN you hear the image, but only a portion of that image. One example is hearing the esses and tees of the singer's voice arrive from the tweeter's location above the mid, not from the mid driver's location, where the main part of her voice comes from, listening with eyes closed. That is one sound of a tweeter arriving too soon. It can also sound like the band is leaning forward, for want of a better word. It can sound like the rhythm section is behind the beat (as they would be in those speakers). === Almarg, Your described a square wave as "the summation of an infinite number of sine waves, one being at its 'fundamental frequency"' (the frequency with which its pulses repeat), plus others at every odd multiple of that frequency (i.e., the 3rd, 5th, 7th, etc. harmonics). The amplitude of each harmonic decreasing as its order (i.e., its frequency) increases." This is all true, but only of an ongoing series of square waves. The analysis is somewhat different when we examine just the first up-cycle, without even the first down-cycle following it. Just an FYI, seemingly never mentioned on the internet nor in textbooks. === Mofimadness, the Loudspeaker Design Cookbook is generally excellent, but all previous issues got the concepts of phase time-coherence somewhat wrong. It has been awhile since I looked over a copy, so I can't remember where the problems showed up. I advise to take its advice with a modicum of salt. === Bfwynne, the Revel 2 and Magico have oodles of phase shift, mostly from their crossovers. What you are seeing in the Stereophile tests is just as John Atkinson says- the mid and woofer take longer for their sounds to arrive. What is not readily apparent is how the phase (time delay) is changing at EVERY frequency. Otherwise, one could fix the Magico and Revel 'problems' by moving their tweeters back, etc. Actually, Almarg gave you a very excellent answer. === Usermanual, you ask about us proving we are time-coherent. 1) This would not change our sales. 2) It cannot be done in a singular graph or 'scope image useful to a layman, by anyone including us. This is not a case of sour grapes- please read my letter to sixmoons regarding the issues with measurements. Note some of my graphs do not line up correctly with my text on their website. In the 1994 Stereophile test on our Diamante three-way, remember JA always measures at 50 inches, right in front of a speaker's tweeter. That makes ANYONE'S mid and woofer too far away, relative to the tweeter. JA then moved his mic straight down, to get farther from our tweeter, closer to the mid and woofer, looking for our claim of time coherence. You see our step response get sharper, more compact. But our frequency response goes to heck because he is now going VERY far off-axis of both mid and tweeter. Again, this test was done in 1994. In the intervening twenty years, every aspect of our sound, and of any measured performance, has improved. Above all, trust your ears more than measurements and reviewers. My letter to sixmoons shows why this has to be so. === This covers page one, I think. Perhaps page two will be much, much shorter. Best, roy |
Good questions.
I do agree with what Bombaywalla just posted- knowledge and experience in many different areas is required. I know of no way out of that, to simplify a home-designer's life.
Driver selection is by far the most important factor. If all we care about is making the best sound, instead of spending money on the newest technology (usually inferior, I find), then here are the important questions to ask before selecting any drivers:
- How far away will I be from the speakers?
- What kinds of music will I play most?
- How loud will I play, even if only on occasion?
- How large is my room?
- How low in the bass do I want the speakers to go? Here, it is best to use 'body feel' as your guide. If you want to shake the house and your lower pants legs on electric bass, then the speakers need to have good output to 40Hz, but not any lower.
Listening at ten feet away in a room that is not entirely open into the rest of the home, this amount of low-bass output requires a low-distortion eight-inch woofer with a large-diameter bass port tuned to ~40 Hz, or a sealed-box ten-inch woofer, flat to 40Hz (good luck finding that in today's marketplace), at the minimum. There is no reason to use multiple 8 or 10-inch woofers per cabinet.
Which means this will be a three-way design to be able to use a first-order crossover, since no 8 or 10-inch woofer can meet a tweeter.
On the top end, choose ~1" dome tweeter, not one made of metal nor of 'ring radiator' design. That means ~3kHz crossover point. The eight or ten inch woofer means ~300Hz crossover point, or slightly higher. And that means using a 4 to 5-inch mid driver showing no cone breakup nor the HF resonance of metal-cone drivers.
All these drivers need very flat frequency responses. Avoid drivers with impedance-curve wiggles, as those indicate resonances and cone breakups. Avoid molded plastic cones and metal cones.
Sorry- got carried away. I cannot put out my version of the Loudspeaker Design Cookbook here.
Do know that, by careful manipulation of the Zobel parts in my passive crossovers, I can fine-tune the time-coherence between drivers (their individual phase responses), for a better blend. This cannot be achieved digitally without custom programming and the consequent extra signal processing (assuming the right measurements can be made, which is not likely).
But you can always listen to your adjustments, and for that process, I recommend you listen to only your left speaker, but not in mono. Start with getting that speaker's voice range right, such as on a older Diana Krall recording. And get rid of cabinet reflections with wool felt for at least the tweeter, or you are screwed from the beginning.
For a home designer, the results with a simple passive crossover with Zobels or with a digital first-order crossover/EQ/time delay setup will be satisfying on most music. However, the sound would still 'not be quite right' on enough other music to make you think there's something wrong with your source or room or cables or amplifiers.
That turns out to be the residual phase shift of the speakers, which is what I finally fixed .
I will continue to think about questions Bfwynne and Lewinskih01 posed and get back to you.
Best,
Roy |
My goodness, I just glanced through the XO paper by Dr. Brüggemann. With all due respect, he is not right in many ways about how crossovers work!
The technical details are far too lengthy for here, but I will point out that, in Fig. 8 on his page seven, he described 'lining up the peaks' from a woofer, mid and tweeter. Instead, what must be done is to line up WHEN each driver's pulse JUST BEGINS to turn upwards from Zero. That's a point easily judged for the beginning of a tweeter's spike, but not on a woofer's slow rise (hence a measurement problem). Thus I advise not bothering with his paper, sorry.
The diagrams from Bombaywalla on his Picasa page DO get that starting alignment correct, although I see some problems:
- The scale used shows a definite starting point to the woofer's pulse. That point is not well-defined when the horizontal scale is expanded.
- The loudness of the mid driver seems low, but I could be wrong.
- The summation pulse is not close enough to the ideal.
But it is late now, and no one is paying me to analyze what may be wrong there- just wanted to point out some suspicious items.
Best,
Roy |
You are quite welcome.
I know what I write doesn't pose questions to you all. Instead, I've mostly laid out the facts and some science. It's up to you to use those to develop your own questions. This is how I proceeded back in the early 1970's, by reading all of the AES papers and many others on speaker design in old and current magazines, on acoustics, studied basic physics, calculus, and psychoacoustics. Later, I returned to university to master all the math, and to learn more about how materials behave when vibrations exist and when electromagnetic fields pass by/pass through.
Sometimes I would find an error in the logic or math of someone's research paper. Usually, I used a paper as a springboard, expanding upon the author's thoughts and test methods, to better look at 'something' in detail.
To choose that 'something' to examine, to fix, or even to ignore, I first had to understand the very basics of WHY and HOW that 'something' would be important to what we hear, and then learn WHY and HOW 'it' occurs. This included how and why cabinets vibrate, cones break up, critical damping is achieved, a tweeter can fail to move on very tiny sounds, the air itself distorts... countless questions.
The most important ones are addressed in the Audio Engineering Society's Audio Anthology 3-book set.
Also, one should get The Audio Cyclopedia, even a twenty-year old copy. It is full of important info on acoustics, speaker design and recording methods, found nowhere else. Make sure you get one that's not falling apart in its binding.
Another book, out of print, is Elements of Acoustics by Temkin. You need to know calculus to get the most from it, but it's readable without that.
Finally, the Theory of Sound by Rayleigh, from Dover Press, is exactly like reading Isaac Newton's original papers. Get both volumes one and two, first published in the 1880's.
If you are interested in design but will never build your own speakers, these books are full of the very best information found nowhere else, and are written well enough to make for good, casual reading.
In these books, you get to see how others approached issues and usually find out WHY they did, along with what had been tried before then and WHY.
Knowing WHY is the most important factor in making better speakers. I can tell you most current speaker designs say to me that their designers know no more than what was mastered by 1979. If you read over the topics presented in those AES books, you'd see this for yourself, darn it.
At this point, I see nowhere on the internet any guidelines on how to select the proper woofer, etc. While I cannot help you directly with that, I can point out the principle differences in the drivers you selected, and leave you to have a good weekend!
- The Classic Scanspeak woofer has ALL of the right numbers for a sealed box. I wish it were more efficient.
- The more expensive Scanspeak woofer will not go as low in its proper sealed box. And unless you are stroking the heck out of it (not likely), it has no less bass distortion than the less expensive Scanspeak. However, it would be very slightly clearer in the lower-voice, high bass range. But then it goes nuts above 1kHz, all from its harder cone. Its first resonance at 1kHz is from its heavier rubber surround bouncing back, like a ripple in a flag, and then vibrating the cone running around its rim, like a church bell's 'first mode' of ringing `round-the-mouth vibration. The big spike above 1khz is its harder cone ringing like crazy.
- The Accuton woofer is a lot of $$, has high bass distortion, and will not go as low as the Classic Scanspeak.
- The Accuton mid driver has many wrong numbers and is not quite efficient enough.
- The Scan mid has the right numbers, its cone breakup is under control, and it has a vented suspension like the Scan woofer. Cross it over at ~300Hz. Read my Continuum 3 and Calypso speaker design papers for more info on using a mid.
- The only ribbons worth using, for sonic quality and which will not break for our purposes, are from RAAL. Excellent products, the best by far. You will need to create a Zobel to offset its inductance. Cross it over at 3kHz. Use their smallest model, for the best highs.
- I advise you fade in the subwoofer(s) below 40Hz, leaving the main three-way to run 'full range'.
So now you face a zillion other questions. Get the AES books above and the Audio Cyclopedia at the minimum for both guidance and answers, compared to the Loudspeaker Design Cookbook.
The Acourate approach is not right. I advise anyone hopefully learn what 'the numbers mean' for any driver, then use the parts I like above to fine-tune your own passive crossovers, with woofer mid and tweeter in their own boxes so you can move each one back and forth.
- You only need to build one speaker, as I posted before.
- You need a $100 voltmeter, a $200 fairly-low-distortion sinewave generator, a decent measuring mic with preamp, to run into some kind of third-octave spectrum analyzer for looking at pink noise.
- And a pocket calculator (scientific), especially to calculate real "L-Pads" for mid and tweeter using the best wire-wound resistors.
That's about it for tools, IF you go through the AES books.
When 'designers' do not understand in depth the extensive research from the past, they rely upon digital test gear. And then get many wrong answers since they do not understand 'the basics'. They have purchased an expensive tool that does not help solve the real problems. But they don't know-- they just stick a mic up in the air and tweak their crossovers to 'get the right curve' for each driver, which is soooo wrong.
And then they hear something 'not quite right', to then tweak the circuits by ear, so their favored recordings sound 'right'. And of course then brag about how carefully their gifted designer listened, how much money they (Revel/Harman) spent on a robotic speaker-comparison room or anechoic chamber (Paradigm/Canadian government). Hey, this isn't the space program where people get killed. This is an unsupervised field of endeavor, with no university program for it, requiring money more so than any real technical education. They always claim, "Well, we all just hear differently." Pooh.
And do get rid of/prevent any cabinet reflections for your mid and tweeter (get the mid's box away from the woofer's and tweeter's boxes, vertically). Put wool felt near the tweeter's dome.
Hope this gives you food for thought!
Best,
Roy
|
You are looking to reduce your time incoherence, is how I would say it. And yes, moving the tweeter closer may have increased your Paradigm's incoherence. But the only way to tell is to have a friend help you swing, quite literally, an arc between where your ear is and the location of where each driver's cone or dome meets its voice coil. Those should lie along the same arc.
Because you must keep the string or tape measure pulled tight, you would find you cannot just hold that string against your ear. I recommend you tape a dowel rod to a camera tripod, to mark your ear's location.
Also, get out your calculator to find out how far you are off axis. However, do not listen for tone balance, but for 'depth', for each instrument and voice to appear more and more whole, right there in front of you. The opposite is the tweeter and woofer becoming audible on their own, audibly separated away from the mid. The mid's tone range must be our reference point for someone's location, because that's the main tone range we hear every day.
It has been proven to very many people's satisfaction that the ear is not as sensitive to variations in frequency response as we would like to believe- not to say a flat response is unimportant. However, this must be true, as we never get to hear 'the best frequency response' from any source in real life, because we are never in 'the perfect spot'.
However, when you do get the Paradigm speakers into the right tape measure position/arc, the sound may be worse, because that is not 'the position' they intended. So again, always trust your ears.
In that case, have your friend tilt your left speaker back and forth while listening to Diana Krall's voice on just that left speaker. But not in mono. Her well-recorded voice is already in mono, because she and her piano were panned to the center, which means she and the piano are equal in left and right channels. You do not want the distractions of left-right information, but only the depth info and to hear a sharper focus on her voice, which one speaker can deliver.
Best,
Roy |
Hi Lewinskih01,
Thank you for the questions.
1) The Classic woofer would go naturally to ~40Hz, then you would just fade in the subwoofer using its own its built-in crossover. There would be no crossover on the main speakers, which is a good thing. Also, you would then not have as much phase shift above 40Hz, compared to the main speakers needing a sub up at 60 or 80Hz- there you would hear the sub all of the time. This Scanspeak woofer would not have "an easier time" unless you are going to blast your music screaming loud.
2) on the Acourate approach, the first claim on their home page is
"The powerful software enables you
- to measure your audio system."
Yes, as do other measurement programs. None are doing anything wrong, but their measurement techniques do not match what we hear. A user will be misled by the limitations of its measurement techniques, unless he studies in detail the subjects I touch upon in that measurement-letter I wrote to sixmoons. No calculations can be right when they rely upon measurements that are wrong. An analogy is measuring a car's straight-line performance to tell how it corners.
On the other hand, I do know that after each 'good driver' gets a 'good Zobel' from you, a pocket calculator can then design your crossover. You verify its -3dB points on pink noise with spectrum-analyzer software, by measuring each driver up close. It does not matter if your microphone curve is weird, from your mic being so close, because you are only looking for what happens with and without your crossover.
The Acourate home page also claims you can use their software:
"- to display, interpret and process measurement data."
A novice user will not know how data is to be interpreted, compared to what is being heard.
It also claims
"- to establish correction filters for speaker drivers and the listening room"
For any 'correction', the software will be relying upon measurements having large flaws, as I explained in an earlier post. This includes it not being able to measure cabinet-surface reflections around the tweeter, and not being able to measure the floor reflections between you and the speakers in the same way as you perceive them.
Furthermore, no measurement made at your chair will be accurate below 500Hz, because of room reflections from your floor, the sidewalls, the wall between the two speakers, in that order. And since 500Hz is nearly an octave above middle 'C' on the piano, you are not measuring accurately much of the musical range.
I cannot see the need for expensive measurement software that gives inaccurate results, compared to how we hear. You will get far more use out of the analog test-gear I mentioned above, using less-expensive computer software as your spectrum analyzer, such as software sold by PartsExpress.com
Best,
Roy
|
Hi Bruce,
A Zobel circuit for any driver makes its crossover circuit perform more to 'spec'. Zobels can result in a flat impedance curve, making life easier for an amp, but this does not always happen.
The Zobel circuit for any voice coil is just a capacitor and a resistor placed in parallel with the driver, before any crossover is added. It is there to make that driver's impedance curve appear flat to its crossover parts, so that they work as you would want them to, in terms of 'rate of rolloff' and for your actual -3dB crossover frequency.
To determine the values for its cap and resistor, you can use an inaccurate pocket calculator equation, or you can measure the impedance curve of the driver as you try different values. This takes a sinewave signal generator and a good voltmeter. The driver under test is not in its cabinet nor hooked up to its crossover.
There are likely some internet sources for how to hook up the voltmeter and sinewave generator to measure the impedance of the driver + Zobel at each frequency. You can either plot the values on graph paper or in a spreadsheet, or just write them down.
The value at which that impedance levels off is what you then plug into a crossover-parts calculation as 'your driver's impedance'. Despite how carefully you measure, that impedance will be wrong to some degree.
That error happens because your Zobel circuit was used to flatten what you thought was that driver's electrical impedance, but you've been measuring instead its electrical + mechanical impedance(s). Therefore, you must adjust any Zobel to get what you want.
It would not affect your speakers' 20 Ohm peak because that was created by a higher-order crossover. Your speakers may have Zobel circuits built in to their crossovers, perhaps not visible in a schematic, as they may have been 'wrapped in' with the values of other crossover parts, through 'computer modeling' of that crossover.
If you added Zobel circuits to an existing speaker, most all of its crossover parts would then need to be changed. The end result may lower the impedance presented to an amp, but not enough to be of any concern. The speaker can become easier to drive, since the amp could see a more resistive load at all frequencies (= a less 'reactive' load that stores energy).
But then again, using any high-order crossover circuit in that speaker will more than negate this, because these crossovers make their own impedance curve. Smart designers can add more parts to make the final impedance curve look flat to an amplifier and to a magazine reviewer, but that's an illusion, as a complex crossover still lays between the amp and drivers.
The cost per Zobel is 'not much' and there is no loss of efficiency. No penalty at all comes from using Zobel circuits. Distortion is not increased if you use the best parts you can afford. A Zobel is 'all good'.
Best,
Roy |
Thanks, Tim. Good points.
Best,
Roy |
