Just because it can do a square wave does not guarantee it will sound excellent on music. A nice square-wave response does not reveal many other problems that affect what we hear. However, it is a GIANT step in the right direction.
Roy |
Roy...thanks for the response....maybe i should rephrase my question: "if a speaker is fundamentally accurate(time/phase domain)....and can even reproduce a square wave...does this mean it automatically sounds good? I understand that accuracy is a concern...but should it overide overall musical abilities? I have heard both poor examples(phase correct) and non that sounded amazing..... |
Eldartford-
Have a look at this link- let it load all its photos-
http://melhuish.org/audio/response.htm
You'll see how bad many drivers are at impulse response. You'll see ringing- just count the time period for each "cycle"- the ring is 1/period. You will see corresponding ripples in the impedance curve for each of those rings, and also see cabinet/floor reflections arrive, depending on the test and the enclosure.
Best regards,
Roy |
Phasecorrect-
If you use pink noise or MLS or swept sine waves (common for testing dispersion), most of what we see on the printout does not explain what we hear. Dispersion, as heard on music, depends on some very definite factors:
--the diameter of the driver (not so much its shape) vs. the wavelength.
--does that driver remain a rigid piston in its operating band? Most do not.
--how the reflections from the enclosure's front and sides, and reflections off the other drivers' surfaces smear transients.
--how the crossover disturbs those transients.
The real problem is that we listen to music. Look at a musical waveform on a `scope- what do you see? Do you see any sine waves, or square waves, or sharp, stand-alone impulses? No. You see an ever-changing wave "form" that has more dynamic range than the face of the scope can reveal. It REPRESENTS how the mic diaphragm moved in and out, and how our ear drum is supposed to.
The music we hear- all its tones, rhythmic interplays, harmonies, imaging- our minds interpret from that complex "wave envelope". It is this unpredictable envelope's shape that counts. When a designer focuses on the theoretical "sine wave" components only, then the shape of the envelope has become immaterial to him.
Except to the ear. Which is why time coherence, and lack of cabinet problems, and linear drivers, and fewer crappy crossover parts, and proper crossover points are all important. Those all affect dispersion AS HEARD ON MUSIC.
Phasecorrect- you asked, "if time/phase accuracy is indeed retained...why do all time/phase coherent speakers sound different?" Because they are basing their claims of accuracy upon flawed measurements. The measurements don't pick up on all that we hear.
Ever wonder why we can't often play poor recordings? Everyone blames the studios, but it's the speaker's time-domain problems that are further distorting that distortion, contributing to unlistenabilty. Test: play a poor recording on phase-coherent headphones (Grado, Stax, others) then play it on a high-order crossover speaker just as loud.
Music is about time as much as tone and loudness. If you only test for two out of three, you won't be designing- only shoving parts into a box.
Best,
Roy |
Yeah, the original Ohm had a lot going for it. Magazines replaced square-wave tests with the computerized MLS test, which can interpolate the phase response and any ringing from the MLS psuedo-noise (but not in great detail- as most of what you see are the averages of 20Hz-wide frequency bins).
Some of the first-order speakers currently marketed do well on square waves, but manufacturers see no reason to publish the test, for marketing reasons, so the competition cannot find out easily, and because this test is not the only one to be passed for good sound, as I'm sure you suspect.
What you heard, good or bad, in the Ohm lays far deeper than what the square wave can reveal- for two reasons:
--the square wave is composed of only odd-order harmonics plus the fundamental (any even-order content seen on the `scope is distortion). Thus it only tests certain tones, not all tones.
--the square wave's dynamic range is far different than music- it does not stress the drivers enough, nor last long enough to excite the woofer.
A square wave is a guide- if you can find out where the little departures from a flat-topped characteristic come from, and then fix them, great! However, there are better tests for the problems behind those squiggles, ones which a smart manufacturer is not going to reveal, nor a poor one reveal that they don't perform!
You raise valid points- not a very professional industry is it? Becoming a better listener and gaining some technological understanding seems to be the only way to find something decent!
Best,
Roy |
The original OHM with full range Walsh driver could reproduce an almost perfect square wave... as evidenced by oscilloscope pictures. It delivered an omnidirectional cylindrical wavefront, and the imaging that I heard was superb, as you would expect. To my knowledge no other speaker has ever attempted this test, or perhaps,no one has published the (rotten) results. Unfortunately, the speaker wasn't so great in other ways. Ain't that a shame. |
Also..Roy...could you shed some light on the correlation between a wide dispersion pattern and time/phase integrity?
Secondly...I believe it was Joseph from JA who brought up a valid point in what I assume was in defensive to his designs(which I dont own)...that is...if time/phase accuracy is indeed retained...why do all time/phase coherent speakers sound different? I know you are a busy man...and this is not intended as a set-up question...I am just curious...thanks for all your informative replies... |
Roy great response...I was just thinking about this the other day...the real challenge in a time/phase design is not the crossover...its building the entire darn thing in a proper manner to use a first order network...in short...all your ducks must be in a row...and this can be applied to any "minimal" crossover network as well...a good design isnt enough...it has to be well executed with quality componets,etc...which is why I assume so many DIY speakers sound like crap...it is much more than simply throwing off the shelf drivers in a box and hoping for the best...I am constantly surprised at how many think they can outsmart engineers...even those with hi quality gear that like to perform their own mods.... |
The simplest electrical crossover on a speaker is an inductor placed in series to the woofer, and a capacitor to the tweeter. The amplifier drives into both simultaneously. If they are perfectly equal and opposite in "reactance", then they cancel out, as far as the amplifier is concerned. This cancellation is what makes this the only dividing network without time-delay distortion.
This is a first-order network. Its two components can be used only when the drivers and cabinet designs are "perfect".
Not bloody likely.
More complex circuits are usually required, whether using two or ten more parts. The result can still be a "measured" first-order acoustic rolloff. The extra parts "modify" driver non-linearities and "make up" for cabinet reflections. Which they cannot- but they can fool the microphone. Of course, extra circuit-parts cannot be perfect either. Reductions in transparency and dynamics are givens.
To keep the number of crossover parts to the barest minimum, one has to use the most linear drivers- which are relatively few. However, not that few: specific examples include tweeters from Morel, Dynaudio, Foster, Stage Accompany, Pioneer TAD, and Scanspeak. Certain mids from Audax, Eton, Davis Acoustics, Bandor, Jordan, Foster, Peerless and Aurasound. Specific woofers from Scanspeak, Davis, PHL Audio, Volt, Audax, Peerless, Pioneer and Aurasound. There are others.
And every one of them is far more expensive than the drivers used in most designs.
For a commercial designer who wants to use the simplest crossover, it's hard to find the best drivers under deadline conditions. But by using the most linear drivers, within proper cabinetwork and correct bandwidths, the crossover circuit can be reduced to just a handful of parts, for clarity and for time coherence. The converse is entirely true.
Best,
Roy |
Look at a Vandersteen crossover and decide for yourself. There is a lot more to the crossover than most other speakers. As for being complex, I have always thought the more pieces parts the more the complexity. You can get a glimpse on his website. |
I'm not sure about the Vandersteens, but, I do think some of Jim Thiel's crossovers are quite complex. I suspect his goal of amplitude coherence has something to do with this. I suspect that the complexity of his cross-overs is what accounts for the low yet narrow impedance of his speakers. Have you ever seen the one Thiel used in the CS-5. Some have criticized him for this, complaining that his cross-overs suck the energy out of the performance and add a veil to the sound. Yet others critcize his products for being too "analytical" and "clinical" sounding. Go figure? Of course due the the mechanical cross-over nature of some of his new co-axial drivers, the electrical crossovers have simplified. On the other side of the spectrum I think that the Meadowlark offerings use a different set of priorities and accomplish their results with a simpler crossover. Mind you I enjoy all of the above manufacturers products. In as much as one can readibly identify from whose pen what product came from, I still think they have more in common than not. To me, there is something that just sounds more "right" about them. |
I dont believe Vandy or THiel use "complex" x-over circuitry...these are afterall pioneering first order time and phase correct speaker designers...both Richard and Jim...and as RIchard V has openly stated...there are only a handful of truly time and phase correct speakers on the market...SPica would be the one that fall into a more complex x-over category... |
Cdc- Thiels use a complex crossover circuit to force first-order acoustic rolloff responses from the drivers, because of their driver choices and cabinetry. Dunlavy and to some extent, Vandersteen employ similar approaches.
Your question about a single driver- if there was one with a perfect cone, free of breakup, the end result would be little bass response and very narrow dispersion in the highs. To widen dispersion in the highs and extend high-frequency response, a whizzer cone is attached, driven through a mechanical crossover (the adhesives used) between it, the voice coil, and the main cone. Those type of crossovers at best are 2nd-order, which means there's 1/2 wave-period of time delay at the crossover point. There are cone breakups usually still present. There are standing waves in the whizzer, reflected from its un-terminated outer rim. Those all are responsible for less than pinpoint imaging. At least their designers get to leave off any electrical crossover, which is a good thing, considering how much information is lost in most of circuits. Whizzer-cone drivers are not time coherent, but just less imperfect that most speakers that do use a crossover.
Phasecorrect- just remember that "phase correct" and "phase coherent" do not mean "time coherent". The converse does. See my first 02/12/03 post above.
Best,
Roy |
Roy...do you have a dealer in the midwest?...I would love to audition the Europas...cheers...or do you have any b-stock/demos available? |
ALso...Spica tc-50s were designed and marketed as a truly time and phase correct speaker...I know there is alot of jargon these days..."phase correct","phase coherent",etc...but the spicas...like Vandersteen, meadowlark,Thiel,etc...were the real deal...with that said...they were not a perfect creation...the highs were rolled off, they had very little bass, and were not very efficient...however...their 3-d imaging&spacious qualities as Roy mentioned were their claim to fame...and why they still fetch 3-$500 used almost 20yrs later...not much more than they retailed for back in the day...a great speaker during its heyday...and a fine value...but a bit outclassed today...especially in the detail department...at even at the $500 level... |
Roy, that's okay to dominate the thread, I think we are all learning a lot. At least I am. One note though. Thiel speakers uses metal drivers and are first order (time coherent?).
Would a single driver be better still than 1st order crossover? I've got a 5" single driver speaker for a week. But it has a plastic "whizzer cone" glued on it. So I'm not sure if this counts as a true crossoverless design. Maybe mechanical, not electrical. Female vocals with acoustic instruments sound very different than my 4th order speakers. There seems to be more ambience to the music. But for some reason the single driver speakers don't have the pinpoint imaging. |
Thanks for the feedback.
I don't feel bad about answering- I just don't want to dominate this thread. And it would be more fun to ask the questions. But you're right, this is my profession. Much of what I learned came through sharing, so I'm glad you appreciate the information.
My words here have not been about reaching perfection- speakers are man-made devices, and I am not going to give away any trade secrets on how to better approach perfection. My intention is really to tell you about certain design pitfalls that could be avoided, except "the math's too hard!".
And out of respect for your and everyone else's intellect, I try not to make unqualified statements- so you see the reasoning behind my logic and of others, and in 20 years come up with a new way to drive the air. Mostly, I would hope any reader leaves with enough knowledge to recognize when "marketing" is disguised as engineering. Like the claims of cones dissipating energy. The lack of educational/technical writing in the press over the last 20 years is one of the main reasons hi-end has become a confusing, poor-value hobby, full of frustration for most participants.
Anyway, thanks again to you and the others for your interest and insightful questions. So I do not repeat myself for related topics, such as why a cone breaks up in the first place, please look at my posts at "the Vinyl Engine" at
http://www.nakedresource.com//yabb/cgi-bin/yabb/YaBB.pl?board=general;action=display;num=1038342561
About the Spica? It's been a long time since I studied some of what he claimed, but I do have several AES papers on similar computer simulations done in the early eighties that do not support that claim of virtually no phase anomalies. Actually his claim could be true- because of the word "phase". Remember though, that "lack of phase anomalies" does not mean "time coherent" nor "no time-delay differences", an important distinction. See my 2nd-to-last post. Phase is not time. Phase is relative; time is absolute.
Chances are that Mr. Bau was among the first to combine the 4th-order filter with the supposed "perfect" second-order mechanical rolloff of that soft-coned woofer, plus combine a 1st-order electrical filter with the natural second-order "sealed-box" rolloff of the tweeter:
That would be 6 x (-45 degrees) shift on the woofer (-180 degrees), and 3 x (+45 degrees) shift on the tweeter, (+135 degrees). Which means they are 315 degrees out of phase- only 45 degrees away from 360 degrees- near the same as from two 4th-order acoustic-rolloff filters. That remaining 45 degree discrepency? If one considers that the woofer's cone will not be breaking up in a "perfect manner", which is true, then that 45 degree differential can be easily "added back in" from the woofer's non-perfect cone breakup. So you come up to a total of ~360 degrees phase shift- which is indeed "no phase anomalies", as they are "in phase". But not in sync.
The main reason Spicas were so spacious has to do with
a) being a two-way with good drivers,
b) the felt-work around the tweeter,
c) the shape of the cabinet,
d) the slant-back of the cabinet,
e) the lack of resonance/echo inside the cabinet,
f) there was only one cap on the tweeter, in the days of poor capacitors.
g) he was among the first not to let the woofer-cone breakups interfere with the tweeter.
Smart guy. An inspiration.
Best regards,
Roy |
Roy, I'm sorry you feel that your the only answerer here. There just really aren't that many people with your expetise, and even less that do who are willing to share it. It is deeply appreciated. Any thoughts on the now defunct Spica claims? |
I should have included Linkwitz-Riley circuits in my post previous to yours. They operate in the same ways as described, but change phase approaching the crossover point in a different manner, still totalling 360 degrees difference at the crossover point. Which makes them in phase for test tones, but out of sync for transients and anything that forms what we hear as "space".
Best regards,
Roy
Best, |
B&W claims 4th order crossovers even in their mid-level 600 series speakers. (It's in their 600 Series 2 sales brochure.) Believe I read in another manufacturers sales brochure that use of 4th order Linkwitz-Riley crossovers insured phase and time coherence. That was news to me having "grown up" with the idea that only 1st order was phase/time coherent. I don't know enough to comment on 4th order....just an observation that this seems to be out there as truth. |
Cdc, it is time to dispel the myth hidden inside their statement, as many other firms say exactly the same thing.
You say Revel states, "The crossover networks . . . maintain a 24db per octave, 4th order acoustic response..."
FYI: Acoustic response- this is the actual frequency vs. amplitude response a mic would measure, freefield at one meter on swept sine wave tones (no floor bounce). This is the right way to describe any crossover- by its final acoustic response (assuming you even listen at one meter... which is a whole other problem).
". . . the steep filter slopes ensure good acoustical behavior in the crossover regions, with a minimum of acoustical interference,.."
This is true -no question- when measured by swept sine waves or pink noise, w/o the floor bounce. FYI: "good acoustical behaviour" means there are major no peaks or dips in the frequency vs. amplitude response on swept tones or pink noise.
"along with low distortion and wide dynamic range."
Yes, because 4th-order rolloffs keep the drivers protected really well from low frequencies- the upper-range cones/domes don't even wiggle on a bass drum.
"The somewhat steep 24dB per octave slopes also provide the benefits of keeping ALL DRIVERS IN PHASE AT THE CROSSOVER POINTS.."
Yes they are in phase, which is a benefit, but THEY DID NOT START AT THE SAME MOMENT, NOR WILL THEY STOP AT THE SAME MOMENT. THEY ARE NOT TIME COHERENT.
With a tone generator, feed that woofer and mid a steady sine wave at the frequency where the crossover occurs. Put a mic out front- look at the combined single sine wave from the woofer and mid on a `scope. You cannot see the beginning or end of this steady tone- it's just a sine wave going up and down.
Then unhook the mid- look at just the woofer. Then look at just the mid. THEIR WAVE'S PEAKS AND VALLEYS LINE UP- they are in phase. But if you could see the beginning or the end, one starts A FULL CYCLE LATER THAN THE OTHER. And when they stop, ONE STOPS A FULL CYCLE LATER THAN THE OTHER. You could even say their combined output rings at that crossover frequency.
ALL of that is in any second or third-year electrical engineering "Filter Theory" book in plain English- that is the behaviour of the 4th-order filter. In phase, yes, but 360 degrees out of step.
Can you hear this? Yes. Just listen to a speaker (or headphone) without that time delay. How much time delay was imposed? Exactly one full period of the crossover frequency. If that was 400Hz, then the time delay between the two drivers is 1/400th second, or 2.5 milliseconds, which is ~32 inches for time of travel, acoustically.
You just smeared the guitar spatially by ~32 inches, front-to-rear, and transiently by 2.5ms, across its 400Hz range (just above middle C). It sounds like the upper strings of the guitar are "leading" the lower strings, or that there is more pick on the string. It also changes the wave envelope, which means a loss of clarity. And all that means it changes the musical message.
But how would you know unless you've heard the real thing? All you really know is that there are many "poor recordings" you can't play, can't enjoy. Many performances "you just don't get". Why? Because that phase distortion is distorting everything that comes into it- including any recorded distortions- so you hear distorted distortion- which is multiplicative, not additive.
You are being presented a speaker that warps not only the image of the guitar, and its dynamic attack, but mis-aligns the individual tones that go into shaping its harmonic envelope, which is its timbre- the very nature of why it sounds like a guitar.
"...The steeper fourth order slopes, however, avoid the power handling problems associated with first order crossover networks."
Yes, if you use less than the best drivers.
Cdc, you remark, "and comments made above about bad in-room frequency response (some Revel is very good at) with slow roll-off x-over design 4th order sounds very convincing."
I am not sure what you mean by "(some Revel is very good at)", but all you have to do is listen to end the debate. Music waveforms have little to do with the math Revel and others use, math based on sine-wave arguments, "proven" by sine wave measurements.
The standard response to all that I've stated above about the drivers not being in sync is, "Well, you can't hear the phase errors, anyway. There are tests that prove that."
The tests don't prove that- they are ambiguous at best, because they were based on clicks and other unfamliar sounds.
If a designer has never built a speaker that is minimum phase (the standard term for any time-coherent system), he has not even tried to hear the difference.
A 4th-order acoustic rolloff is used because:
--the designer believes the sine-wave arguments,
--it allows high-tech-looking metal cones to be used, as most all of those have a severe (>10dB) peak at the cone breakup frequency, along with 5-10% distortion caused by that peak. Look at the SEAS and others' metal-cone driver frequency responses- readily available. The 4th-order rolloff chops them off.
--a 4th-order rolloff lets the designer lower costs by using less rugged drivers.
--a 4th-order rolloff is often computer-aided in its design. Sounds high-tech, and good for advertising.
--the designer doesn't have to learn time-domain math, which is quite difficult.
All the math of acoustics and physics and music supports my claims, fully. What's known about how the ear uses the time domain for image formation and timbre retrieval supports my assertations. The actual times and timing of music's transients support them. The method behind how any timbre-containing wave envelope forms as time evolves, supports them.
As far as off-axis interference? 1st-order drivers overlap a lot. So those drivers must be really good- and it's hard to find "the best". However, they are overlapping TIME COHERENTLY, `most everywhere in the horizontal plane, so there is no interference or lobing. Period.
If you stand up- yes, they start to go out of phase by many degrees. But the 4th-order circuits are already HUNDREDS of degrees out of phase, no matter where you stand or sit.
Some would claim this time-coherence stuff is not a question of right or wrong, but a matter of taste. I would remind you that keeping the speaker a minimum-phase design (not possible w/4th-order) is the ONLY way to preserve the actual shape of the musical waveform received by the mic. For inside that "shape", that irregular, non-sine wave envelope, lay all the tones of the music, all the dynamic changes, all the musicality, and all of the musical message. Why disturb it?
I welcome any attempt to refute my points on any grounds.
Most of all- just listen. Non-time coherent speakers sound like a wall of sound -no depth- as the time domain is scrambled. Time is distance. Time delay is depth. Time is transients coming and going, and tones building then decaying. Scramble the timing between bass and treble- you lose the depth. You lose the timbre. You lose the musical intent. You lose access to many, many recordings.
Time coherent behaviour can be heard in any decent headphone- even a $30 Walkman headphone, let alone the Grado or Stax. Just listen, especially to music that has energy near the crossover frequency.
Thanks. I don't want to be the constant "Answerer" here, but I really hope this helps.
Roy |
I believe SPicas used a 1st order/4th order x-over configuration in their tc-50...with the 4th order on the mid-bass driver...designer John B. claimed through computer simulation tests that the 4th order had virtually no phase anomalies...I do know this...in terms of 3-d imaging...the SPicas are still one of the best...I also believe B&W uses a 4th order design in their higher end models... |
Revel literature is interesting:
"The crossover networks . . . maintain a 24db per octave, 4th order acoustic response . . . the steep filter slopes ensure good acoustical behavior in the crossover regions, with a minimum of acoustical interference, along with low distortion and wide dynamic range.
The somewhat steep 24dB per octave slopes also provide the benefits of keeping ALL DRIVERS IN PHASE AT THE CROSSOVER POINTS - A BENEFIT SOUGHT IN THE MORE COMMONLY USED 6dB per octave crossover designs from other companies. The steeper fourth order slopes, however, avoid the power handling problems associated with first order crossover networks."
Between Revels claim of maintaining phase coherence with 4th order x-over and comments made above about bad in-room frequency response (some Revel is very good at) with slow roll-off x-over design 4th order sounds very convincing. |
Cdc/Unsound you are too smart for your own good...
Cdc asks:
- Is time / phase coherence maintained through the whole recording / playback process so the speaker is the only thing messing up phase and time coherence?
The answer is no, but... ALL gear and speakers and mics add time delays at the high and low ends of their operating bandwidths. Those are (fortunately) gradual changes in the phase vs. frequency. ONLY speakers (and certain microphones) will jerk the phase around abruptly in the middle of the audible band.
- What if a driver naturally rolls off greater than 1st order. Does the crossover have to boost the driver? Is this bad?
Any driver is rolling off because it has a "mechanical crossover", which is usually > 2nd-order. It rolls off because it has moving mass and compliance. The compliance is the flexibility of the hinge-points on the cone as it breaks up. The moving mass is varying with frequency- the outer portion of the cone coming to a standstill.
With less effective moving mass for that voice coil to move, the frequency response peaks before it rolls off- you can see how the wiggles in a raw-driver's impedance curve correspond to the breakup modes, and to peaks and dips in the frequency response.
Cone breakup is a "dirty" or "ringing" sound at worst. Breakup modes are always excited by the fundamentals down at 1/2, 1/3, 1/4, 1/5... of each breakup frequency. A driver should not naturally roll off `till well beyond the actual crossover point. Any breakup modes should be well-damped so they don't ring.
Cone breakup can be disguised with very high damping- such as with the woven Kevlar and carbon fiber cones, and the mineral-filled poly cones. It keeps those drivers' frequency responses smooth, but crushes the dynamic response as the cone structure is absorbing the energy, instead of moving the air. So why use those cones?
a) They don't have the "peakiness" of poorly-designed paper cones.
b) They look hi-tech.
c) Making proper wood-fiber cones is a dying art.
d) Plastic cones are good for sales in tropical environments and cars.
So we lose dynamics, and inter-transient silence, and detail- who cares?
- Would the extra power handling of the voice coils in 1st order x-over degrade performance in other ways like higher inductance, mass, and hysteresis?
Inductance- nothing we can't adapt to.
Moving mass- higher means less efficiency (see my post today on the High Efficiency thread) and eventually more power compression.
Hysteresis- depends on what you're talking about. The longer voice coil does overhang into more of the fringe-magnetic field created by a poorly designed magnet structure. There are others, such as inadvertently changing the center of gravity of the cone/voice coil assembly- which will lead to unwanted rocking motions, which puts really strange eddy currents back into the pole piece, which...
Unsound,
Your observation that "active speakers don't necessarily need to be self contained" is true. It would have all the advantages you list.
But if a speaker manufacturer merely supplied or recommended a certain electronic crossover, then the customer must purchase several "matched sound quality" amplifier channels and interconnect cables and speaker wires. And those are very strong points for putting all that stuff back into the speaker cabinets.
Either way, my main problem: whose gear do we choose- and will they let us? I've spent a lifetime working with speakers, not amplifiers- the only amp I could design would be from a cookbook recipe, or I'd have to mimic one already out there. And we'd have to service it! Egads..
Thanks for the insightful comments and questions. I wish the press would talk about this stuff. Why do you suppose that is? (serious question for another thread)
Best regards,
Roy |
I think we need to remember that active speakers don't necessarily need to be self contained. One could even argue that isolating cross-overs, amps and drivers have advantages. Even in a self contained active speaker, the various components could be modular in design, so that one doesn't have throw the baby out with the bath water. |
Good points Roy, Thanks. Keeping phase coherence sounds liek a good idea but I had a couple of novice questions on phase coherence, if you have a chance to answer. I don't think they were answered above.
- Is time / phase coherence maintained through the whole recording / playback process so the speaker is the only thing messing up phase and time coherence?
- What if a driver naturally rolls off greater than 1st order. Does the crossover have to boost the driver? Is this bad?
- Would the extra power handling of the voice coils in 1st order x-over degrade performance in other ways like higher inductance, mass, and hysteresis? |
Good question! Active x-overs/built-in amps are great!
Of course, the cost of those parts and the extra labor multiply into a higher retail. Existing powered-speaker companies offset some of that by using far less than the best woofer, tweeter, cabinet, wire, terminals, amplifier design and electronic parts, and via quick-assembly techniques- like push-on connectors to the drivers.
The powered monitors sound better than their competition, but this only reflects on how bad their competition is, and not on what can really be achieved with proper drivers, cabinets and passive crossovers for the same total cost, or less, including the outboard amplifier- which can be changed that day if it goes bad, as opposed to sending the powered speakers in for repair. Talk to a repair tech about the bright idea of building a VCR into a TV set. Makes them curse...
Home users would have to scrap their existing amplifier to use powered speakers- not likely.
Whose amplifier design do we put in, assuming they would even sell their "best design" to a speaker company? Rowland, Edge, Audio Note, 47Labs, Bryston, Wavelength, Manley... none are perfect. And they always get better (almost always) every year or two.
Those are the reasons which have kept us from putting in active electronics here.
Except for the compromised powered monitors out there, or cost-no-object reference speakers, we would seem to be stuck with passive crossovers.
Best,
Roy |
So why is audio staying in the dark ages with passive crossovers? Even a listen to Mackie HR624 active monitors?Shows what active croosovers are capable of. Something to do with not being able to pick your own power amp?
I will admit that passive ATC 10's with ATC $3,500 integrated may have sounded better than active 10's with built in ~$500 amps so the power amp must be very good to see the improvement. Obviously a single $3,500 could be better than 4) $125 amps and this could explain why. |
Roy, your a prince. Thank you. |
No. Yes. No. They could be with appropriate digital time delays (delays that are different for each frequency) built in. Those are available as studio-oriented x-overs with built-in adc/dac's.
Roy |
Pardon my ignorance, but can one mainatin correct time and phase with 12 and 24 db octave slopes? Many seem to say the only way to do this is with 6db slopes. Are active cross-overs different in this regard? Can active cross-overs be time and phase accurate? |
If you mean digital "tricks", I think the best answer is that the value and limitations of any digital manipulations could be more easily ascertained if they were not trying to correct for things they should not be- like cone breakup, cabinet reflections, gross phase shifts, interior resonances- non-linearities which should not have been in the speaker design in the first place.
If you mean "servo control" of the subwoofer, I have not seen that applied yet to anything close to an already linear design (a well-behaved subwoofer driver and rigid cabinet). I also have never seen anywhere close to the best accelerometers being employed... so expensive! Again, we really can't hear all the benefits and the ultimate limits of servo technology.
Maybe someone who designs servo subs for a living can come forward and explain more on that subject. Anyone out there up to that?
Best,
Roy |
Roy, once again thank you for sharing your expertise. Am I correct in assuming that speakers well executed from the get go can still benefit from all these "tricks" if the tricks are well executed as well? I'm sure I'm not alone in looking forward to your web site. Good luck! |
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. |
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. |
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 |
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? |
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 |
I have owned a couple pairs of Vandersteens in the past and I thought they were great but now own a pair of Tyler Acoustics Lynbrook monitors and they are the best speaker I have ever heard under $10,000 and I have listened to most big name models. Designs are more marketing ploys to get you believe their design is the best. There are many great speakers out there and it is the execution of the speaker builder with all the parts, cabnets ect., that make them either great or mediocre IMHO. |
I tend to like Brit monitors...such as Spendors,Proacs, and my Quad 12Ls...and all of these companies are rather "secretive" on disclosing exact crossover componets,design schematics,etc...in short...they dont overly advertise being time/phase correct...but Im sure they incorporate some of these traits....in short...I feel "phase coherent" has become a marketing slogan...and really detracts from the few companies who are actively applying this viewpoint...and even after listening to Vandersteen, Meadowlark,etc...I still opted for the imaging precision,transparency,soundstaging,and larger sweet spot of a full range monitor...the "lobing" effect of 1st order designs...in my humble opinion...far outweighs the benefits... |
Greenmountainaudio.com is almost done.
We leave to display our speakers at the T.H.E. Show in a couple of days. Don't know if we'll get the site up before I leave, or while I'm gone, but soon...
Best,
Roy |
Hi Roy. How's the web site coming along? |
von Schweikert designs are not time coherent.
Yet, Albert makes very nice sound because:
1) He knows how to get the most out of the 4th-order approach (the only one that makes sense if you have to/choose to give up time coherency).
2) He makes a good cabinet- with proper placement of damping materials inside (it sounds like), and they are strong, fairly slim.
3) He chooses linear drivers, unlike most of what is out there.
4) Really a part of (1)- Albert is one of a few designers that knows mathematically and empirically most of the factors that influence the selection of the actual acoustic crossover points. Not too many do, as it takes years of homework in Physics, and then years of listening, measuring and designing to prove to your ears what is right or wrong with that Physics-based math. Electrical engineers are not taught this math (since the world went digital)- which is why there are so many non-musical speakers out there.
Recommended recording- Hugh Masekela's album "Hope" (CD). Stunning dynamics and excellent musicianship! The Burmeister sampler uses one of the cuts found on this.
Hugh M. is a horn artist, so this disc is not pretty at nightclub levels on non-time-coherent speakers that have a crossover point in the 2-3kHz range. We'll play it in `Vegas on our Continuum 3s, at the THE Show.
Remember, Mr. Bischoff, that there are many physical, electromagnetic and acoustic factors behind achieving time coherence. Those specific design factors are a checklist of well-known concepts, any one of which ignored means no coherence.
They include factors most of us have heard about:
uniform diaphragm motion,
small diaphragm/cone/dome size vs wavelength,
diffraction reduction,
crossover time delays,
time delays caused naturally by the enclosure tuning.
What a layman lacks is a rule of thumb, some numbers, for when "uniform" is "good enough" (as nothing's perfect), or a diaphragm size is "small enough", or diffraction low enough, or what the actual acoustic delay times are vs. frequency.
Those give enough information for a layman to choose music that will reveal a speaker's flaws immediately- some of which he probably can live with. This info will be on our website shortly, along with a "recommended recordings" list that makes sense to challenge a system with.
In the long run, Mr. Bischoff, you would find that phase error in speakers affects dispersion vs. frequency. It also affects dynamics vs. frequency, and imaging vs. frequency. It affects hearing the rhythmic pulse, the emotional inflections. That is, if one plays a wide variety of music. But if a listener sticks to tried and true recordings only, then he is not investigating very far into any problem with the system or speakers. Using a wide range of material with which you've become familiar speeds up speaker evaluation and system tuning.
Anyway, thanks to all for their kind comments, and Happy New Year to everyone at AudiogoN!
Wish for Peace in the new year.
Best regards,
Roy Johnson
Green Mountain Audio |
No dealers very near to you right now, although a lot of owners. Dealers down in Orange County and Central Coast area.
Call me if ?? 719 636-2500
Thanks,
Roy |
Green Mountain Audio -
Hello Roy, nice meeting you.
Where can I hear your speakers ?
I live near San Jose, California.
thanks,
Mbonn |
I'm surprised nobody has mentioned Von Schweikert speakers in this discussion of time and phase coherent speakers. Anybody have any opinion on Von Schweikert speakers as they fit into this discussion? BTW, I asked Albert Von Schweikert how he achieved time and phase coherence with a 4th order crossover. He gave me a very lengthy response to my e-mail, which was very interesting, but I am not sure he answered my question. Can anybody help ? |
I actually like some of my competitors! (even if I have differing ideas regarding design execution) I've met Roy many years ago at the Florida show and he struck me as a very nice fellow. Same for Pat McGinty.
David & Sheryl Lee Wilson came by and sat through my presentation during the last show, and David stayed and shook my hand afterward. Peter McGrath dropped by shortly thereafter. I've called upon guys like Ken Kantor and Michael Kelly for their advice, and they've always been gracious and accomodating. While the product is very important, a speaker company must be more than the product. The most sucessful companies know that you must take care of the customer after the sale. Ya gotta admire the people at Vandersteen, Thiel & B&W who take care of their consumers and do right by them. It's no small task, and they have set a standard of service that we try to emulate every day. |
I want to make an empirical observation about my Ohm Walsh 300's. They sound good to me no matter where I am in my apartment. I give their coherent sound credit for this. The only complaint that I have is that they won't play loud enough for some heavy tunes that I occasionally listen to. Roy's observations sure seem correct. I can't do this now but someday I would like to add a dedicated subwoofer to handle the lows. I think this will solve all problems. |
Well, Roy you share a quality with Jeff that I truly admire. You speak your mind. You also seem to share the same fault. You condemn those who take a different path. I do think that you seem to be a little overly critical of the "corporate entities". Jim Thiel and Richard Vandersteen both started their corporate entities out of their garages. Those who belittle success doesn't hold much promise for themselves. The designer who lacks 6th grade math skills (and we all know who you are referring to) seems to have fooled his university professors into giving him an engineering degree. This same designer seems to be aiming at the same target you are with his co-axial drivers (the mirror image of a microphone and it's pin point radiation pattern). You find fault with his approach. That's fine, your entitled to your opinion. It's also interesting that his success has allowed him to experiment with designing new drivers, a luxury many competitors don't enjoy. Attacking his education or intelligence doesn't seem like such a good marketing concept. I either don't remember learning this elementary math or never learned it in the first place. I'm sure I'm not alone. You may have just insulted a large percentage of potential customers. Don't worry I'm not that thin skinned. I can't help but feel that one can be opinonated and back up one's position with mathmatical evidence and still be diplomatic. See, I really do want you guys to succeed. I applaud both of you for sharing the evidence that brought about your design philosophies. I look forward to more from both of you and hope you lead others to be so frank. |
Karls- you got it right.
If you wanted to know the actual % modulation distortion, you'd have to know the stroke and frequency of the mid's vibrations that are affecting the tweeter's sound. Which are random, as far as the tweeter is concerned. Which means the modulations are unpredictable on music- so all we can say is that they should probably make the sound hazier or dirtier.
Jeff's crossover is probably the only one that could make a co-ax design work well. Even then, the tweeter dome would require a modest horn around it- a waveguide to keep the tweeter's sound from bouncing off the mid's cone. Of course, the mid's sound will bounce off that waveguide's exterior...
The "horn-loading" coloration is a common term- what we're hearing are the quasi-transverse reflections from the sidewalls of the horns, and the reflection from the mouth of the horn back to the throat.
At the mouth of the horn, the sound pressure goes from travelling in a high acoustic impedance to a low acoustic impedance. Thus, from the un-equal impedances, a reflection/standing wave takes place inside the horn. Then there is the matter of a horn's possible throat-compression ratio that boosts efficiency and distortion (love them PA horns, don't you!). You can't compress/rarify the air more than 1% or you get harmonic distortion from the air itself.
Roy |
