[Aether Audio]

Dear Audio Enthusiast,

The field of crossover implementation is probably the most controversial area of loudspeaker design above all other factors.  A seemingly endless number of research papers have been published on the subject and designers are usually well entrenched in their pet topology.  In one regard, this is fine as it offers the end consumer an almost unlimited number of designs to choose from.  On the other hand, depending on the popularity of a designer or just his or her market visibility, many consumers are mislead into the belief that a certain topology is inherently superior to all others.  If a company’s products are well received by the marketplace and the designer is strongly and outspokenly committed to a certain topology, you can rest assured that many with a limited level of understanding and/or experience will elevate the designer’s opinion to the level of gospel.

Our purpose here is not to advocate one topology over another but rather, to help educate the consumer as to the choices and compromises which every designer faces.  If we are able to convey this efficiently, the consumer will be better equipped to make a decision without falling prey to the opinions of designers that promote their views as the final word.

Crossover design is a subset of Filter Theory which is a fundamental mathematical discipline that applies to every field of physics. Crossover Topology is most commonly defined by the resulting “Order” or “Slope” of the network.  “Order” refers to the largest exponent comprising the Laplace Transform equation that mathematically describes the transfer function of the network.  “Slope” is defined as the rate at which the frequency response decreases in magnitude with respect to frequency at those frequencies beyond the selected corner or “crossover” frequency.  Mathematically, it is the rate of change or “first derivative” of magnitude vs. frequency.  In an idealized design, the slope is of a fixed and constant rate and is usually referred to as some value in decibels per octave (dB/Oct.)or per decade (dB/Dec).

A “First Order” network exhibits a reduction of signal magnitude that decreases by 6dB per octave.  A “Second Order” network exhibits a similar reduction that decreases by 12 dB per octave.  Successive orders decrease the signal magnitude or conversely, increase the “slope” by 6db for each increase of network order.  

Complicating the matter is the response at the corner frequency where the filtering action becomes evident from a magnitude standpoint.  The network can cause the signal magnitude to increase above the nominal level or begin to drop off further from the corner frequency.  This property is referred to as the “Q” of the network and is characterized by the names of the researchers that first identified and popularized their attributes and advantages for various applications.  Butterworth, Bessel,  and Linkwitz-Riley (or cascaded Butterworth) are several of the types most commonly encountered in crossover design.

Do to the vast array of different qualities each type manifests, we will not list them for each order and type at this time.  What we will attempt to do is outline the major differences between the most debated types.  In general, there appears to be two major camps or points of view regarding crossover topology, and each camp has its ardent advocates.  These two groups can be defined as either those: (a) to whom the inter- or intra-driver signal delay characteristics of the system are of paramount importance, or those (b) to whom they are not, or are so, but to a lesser degree than other parameters.

We at SP Technology, if forced to choose, must categorize ourselves as belonging to group (b).  It should be noted that such a classification is rather “loose” at best.  We understand and agree, to a point, with those of group (a).  Then again, there is a limit to this agreement.  What we advocate is a balanced approach that suggests the use of a certain topology where its use is appropriate.  One must consider a multitude of variables and prioritize each performance criteria to its level of audibility in order to come to the best set of compromises. Every crossover topology has its advantages and disadvantages, each must be weighed according to a given application and design objective.  At times, one may find the “First Order” network, with its superior signal delay characteristics, is a proper choice.  At others, a higher order slope is called for.  What we will not do is fall into the trap of philosophical adherence to one approach above all others.

At the heart of the debate between the two camps lies the question of signal delay audibility.  Research has shown that under certain test conditions and applying certain test signals, some individuals report a definite ability to hear rather small variations in signal delay.  This we do not question.  What becomes obvious to us though is that if elaborate testing methods must be employed to discern the audibility of small signal delays, they clearly do not carry the same weight with regard to system fidelity as other, more clearly audible factors.  Loudspeaker performance parameters that are clearly audible and repeatable under controlled test conditions rank at the top of the list with regard to priority, as far as we are concerned.  This short list (somewhat by priority) would include: static distortion (THD & IM), dynamic linearity, frequency linearity, frequency extension, diffraction effects and dispersion, efficiency and signal delay performance.

The previous statement could easily be interpreted as suggesting that signal delay behavior is of no consequence.  This could not be further from the truth.  The fact is that the higher priority parameters must be optimized before delay characteristics can be effectively dealt with.  Also, there is little point in attempting to do so when relatively far greater response errors exist. The sad truth is that most loudspeakers suffer greatly from shortcomings in those areas so attempting to correct for delay issues when they are being masked by gross distortions and other non-linearities becomes an exercise in futility.  On the other hand, ultimate dynamic performance cannot be optimized if excessive signal delays exist in the system.

As has been stated countless times by others, music is not the same as test signals.  This is true at least for the more commonly encountered test signals anyway.  One can construct “packets” of complex signals with multiple overlaid harmonics to mimic those encountered in music though.  If we think of percussive sounds produced by certain instruments with certain fixed note durations as test signals, then we can use them to test the system in question.  What we find when we examine these percussive signals is that they have a beginning with a fast rising edge, a middle part with some dominant fundamental tone overlaid with various harmonics and a trailing edge that decays in amplitude over time.  All of this may be on the order of a few, to many milliseconds.  These packets manifest what we in engineering refer to as an “envelope.”  This envelope should retain its  original form from beginning to end of the record/playback signal chain if it is to be perceived as being a direct reflection of the original.  The degree to which this envelope changes shape will determine the fidelity of the system under test.

One can easily envision the envelope being “stretched out” in time such that the note duration becomes effectively longer.  This is what happens when a complex musical “packet” is passed through a circuit or network (crossover) and has certain components delayed in time with respect to others.  This characteristic delay of the system is what is referred to as “Group Delay.”  Usually such networks delay the higher frequency components less than the lower ones.  As we can see from this illustration, it would tend to degrade the dynamic response or perceived “speed” of a system if the leading edge of a percussive envelope (which is composed of the higher frequency components) were to arrive at the listeners ears significantly ahead of the middle or trailing edge of the envelope.  Such signal delay distortion would change the fundamental shape of the envelope and “soften” the percussive impact.  It could also change the way the different harmonics that comprise the original waveform add and subtract from one another, potentially changing the perceived timbre of the reproduced instrument.

“Ah Ha!!!” -  you may say, we have just supported camp (a)’s position.  Well, yes… and no.  Many other forms of distortion can alter the shape of that envelope, none the least of which are THD & I.M. that are introduced by driver cone break-up modes as well as driver motor and suspension non-linearities.  In fact, these forms of distortion are far more prevalent, greater by orders of magnitude and easier to correct - to a certain degree.  As we have said, without correcting these far more dominant forms of distortion, there is little point in addressing signal delay induced distortion.


Also, with regards to the issue of timbre, there is one caveat to the above statement that tends to support the argument of group (b).  Recent research into human hearing seems to suggest the ear/brain system behaves as though it were a form of Fast Fourier Transformer (FFT).  FFT’s are often used in engineering to analyze the spectral make-up of various complex signals.  Although they can reveal with great precision the frequency content of a signal, over the specified sample period they are “time blind.”  This blindness means that any time relevant information within the selected “time widow” cannot be discretely discerned.  If it is true that human hearing does in fact work this way, then the upshot is that we are limited in our ability to hear small time/signal variations.  It means that the spectral content of a signal is of significantly greater importance than when the information arrives in time.  

In our opinion, the truth of the matter is likely to fall between the extremes.  Time information is undoubtedly important with regard to the perception of dynamic realism, but is limited to within a time window of greater than a few hundred microseconds and less than 1 millisecond.  Obviously, this has yet to be proven but the available research tends to support this view.  In addition, a preponderance of anecdotal evidence suggesting this is true exists in the form of products that have received wide acceptance in the marketplace.  A great many highly acclaimed loudspeakers do not offer linear phase and/or zero group delay performance.  In fact, this author is aware of several highly praised models that exhibit rather poor response in this area.

So it is… that the problems the designer encounters are manifold.  In his attempt to minimize the more dominant forms of distortion, he may actually introduce more delay into the system.  This is most often the consequence of using “higher order” slope crossovers in combination with multiple drivers.  The combination of higher order slopes and multiple crossover frequencies tends to represent the “worst case scenario” with regards to excessive signal delay.  Conversely, the combination of “first order” slopes and fewer crossover frequencies/drivers tends to aggravate the more traditional and dominant forms of distortion.

Camp (a) is now saying, “Well… use better drivers and more of them and you won’t have that problem.” Upon first analysis, this would seem an obvious solution.  Easier said than done.  If one chooses to use more drivers to increase dynamic response when using first order slopes, the already existing problem of poor vertical dispersion is multiplied by each additional driver/crossover.  On the other hand, due to these same dispersion issues, the most optimal application would be to employ a 2-way design.  This is the designer's paradox.  

One noted designer that has recently posted on AC has even suggested that the way to overcome the dynamic limitations of first order networks is “to use better drivers.”  If this were truly the answer then there would be no other type of loudspeaker on the market, as every high-end manufacturer would just “use better drivers” with “first order” slope crossovers and be done with it.  Yes, a true first order network can be difficult to achieve in practice, but competition would force the manufacturers hand.

The fact is, even so called “better drivers” have their limitations and driver design is limited to the materials available.  No manufacturer has a monopoly on these constituent materials either, so the problem must lie within the limitations of the sub-components.  As long as drivers are made with wire that does not exhibit room temperature super-conductance, their magnets have flux limitations and their cone materials lack the rigidity of diamonds in sizes large enough to produce bass frequencies, they will forever be limited in their ability to transform electrical current into a distortionless acoustic waveform.  

To be sure, better drivers will always help and are called for universally to reduce every form of non-linearity.  But to the degree of performance that is being asked of them, we have yet to develop ones that will overcome the demands that “first order” slope crossovers place on them, especially when used in 2-way designs.  Of course, the above statement is based on the assumption that traditional forms of distortion are not to be tolerated.  If you are willing to sacrifice this and accept higher conventional distortion to reduce a less offensive type, then of course there are products out there that offer this. We have not even discussed in any length the real world disadvantages that the poor vertical dispersion produced by  first order networks cause.  Even if one is willing to accept the seating limitations that result from their use, the erratic room reverberation that is another consequence degrades from the one advantage they do offer.  If you do not listen to wide dynamic recordings at anything other than moderate levels, you only listen in the “sweet-spot” and you already have a very dead room, then the simple 2 or 3-way “first order” network design may be for you.

Just to throw one more monkey wrench into the whole first order issue, we have also read recently where a designer devoted to first order designs suggested that the difference in signal delay of 1 microsecond was clearly audible.  We will not take issue with this as the research into human hearing is still ongoing and anything is possible.  But if this is true, then such diehard advocates of the “first order” topology are in as much trouble as the rest of us.  While we cannot disprove this statement, it can be easily proven that the drivers these designers are forced to use are a very source of signal delay distortion that no simple “first order” network can correct.  

You see, many that have come before us have shown in various published papers that the acoustic center of a driver “wanders” throughout the operational range that it is used in.  This effect occurs without any crossover network whatsoever connected to it.  Such delay shifting is the source of the common question among designers, “where is the acoustic center of the driver?”  The answer is somewhat ambiguous and most designers “tweak” the crossover in the lab to minimize this effect the best they can.  Let’s just put it this way, the effect is not easily modeled by the various design software we all use.

To be specific, the types of drivers that manifest such behavior worst/most frequently are those bass/midrange units that are operated up to around 3KHz.  If they are to be used in a “first order” design and with any form of “conventional” tweeter, then they must have a flat response out to at least 5 or 6 kHz.  This is because most tweeters, even very good ones, cannot be operated much below 3 KHz due to power handling limitations.  Even in designs that crossover a little lower, the woofer must have a flat response out to at least one octave beyond the crossover frequency.  That requires the woofer in a 1.5 KHz design to extend out to at least 3KHZ.

In any such case, by default the woofer/ midrange driver will exhibit a form of designed-in “mechanical crossover” that permits the center of the cone to operate somewhat independently from the rest.  Due to this “controlled break-up,” as frequency increases the acoustic center moves toward the center/rear of the driver.  The amount of signal delay then changes correspondingly.  This delay will be on the order of maybe less than ten (smaller midrange driver) and possibly up to 100 (larger woofer) microseconds, depending on frequency extension and other variables.  This is well beyond and in excess of the 1 microsecond delay that is claimed to be audible.  It is also uncorrectable without deviating from the minimum phase crossover topology.

If phase/group delay variations of such small magnitudes do indeed represent a significant source of error, we are all going to suffer from a considerable lack of sonic fidelity for some time to come.  Also, if the designer making this 1 microsecond audibility claim chooses to tackle that problem, it would appear that he has his work cut out for him.  May the Force be with him!

As a side note, designers of these type drivers know full well that at any time one chooses to develop such devices, they are automatically “building-in” a significant source of distortion.  The controlled break-up of the cone that permits the driver to operate beyond its “mass cut-off frequency” will also open the door to undesirable break-up modes that generate all sorts of odd-ordered harmonic and intermodulation products.  

What few companies will tell you is that this source of distortion is a far greater detriment to fidelity than even the worst cases of excessive phase shift/group delay due to crossover design.  If you think about it, this fact becomes obvious to even the most unfamiliar individual.  All you have to do is imagine the sound of a drum being struck.  Drums are very discordant, so much so that it is difficult to mentally associate a particular drum sound with that of a specific note in the musical scale.  The cones (or membranes of any driver for that matter) have more in common with the head of a drum than not.

To be sure, driver design is far more complicated than merely developing a crossover network for a loudspeaker system.  If you doubt this, then answer this question.  Why do you see many amateur speaker builders assembling their own “home brew” designs but very few attempt to build their own drivers?  Very few speaker companies even attempt to build their own drivers for that matter.  Why? – Because it’s hard to do!!!  Virtually anybody can slap a couple of off-the-shelf drivers in a box, add a few crossover parts based on some formula and get decent results.  It simply doesn’t work that way with designing drivers.

Due to the fact that designers generally have had little recourse, decades of research have gone into trying to tame these “cone generated” side effects.  That is why many different cone shapes and formulations exist in the first place.  Everything from “mineral loaded” polypropylenes  -- to carbon graphites -- to various composite materials... all have been tried.  A very large amount of money and time has been invested in order to extend driver bandwidth while reducing the inevitable distortion by-products.  All of this could have been avoided if the decision to deviate from true “piston behavior” was never made in the first place.

On the other hand, drivers that do behave as true pistons do not suffer nearly as much from these cone-based distortions or from the “acoustic center shifting” effect - to any significant degree.  They would be ideal candidates for first order designs except that they do not have the frequency extension needed.  It would seem there must be an acceptable compromise.

 One solution would be to use such a woofer/mid up to the highest frequency possible and cross over to a tweeter that has its low frequency response extended in some manner.  Such a limited upper frequency response of the woofer along with the lowering of the tweeter’s response would undoubtedly require using a higher order crossover slope.  What one would do is include the woofer’s natural roll-off above its mass cut-off frequency in the final crossover transfer function.  If this woofer were of exceptional design, its lower frequency extension/excursion could be optimized as well.  Such low frequency optimization would have little effect on its higher frequency performance other than reducing its efficiency/sensitivity.  This would permit one to design a simple 2-way system, thus avoiding the pitfalls of multi-way design.

Although such a design would not offer true linear phase performance, its total group delay would still be quite low.  This would be due to the fact that only one crossover is used to begin with and acoustic center variations are non-existent as well.  One could also employ a method of recessing the tweeter from the front baffle of the woofer to help reduce the group delay error.  From a time response standpoint, signal envelope distortion would still exist, but it would be greatly minimized compared to most other designs.  The possibility of modifying this design at some future point to incorporate a simple delay line that would completely correct for the remaining error exists as well.  Alteration of the crossover would be required but the physical assembly would not need to be altered.

Even though such a design would not be technically “perfect” from a phase response standpoint, it could still, never the less, become a world-class reference system.  Especially if such a combination could provide the dynamic headroom needed to achieve the full 120dB dynamic range of human hearing AND significantly lower all other forms of distortion.  In the end, the system’s group delay would still be minimal while providing the best possible set of compromises.  If this “hypothetical” system also offered significant improvements in other performance areas such as dispersion, diffraction, and frequency linearity, that would be one serious improvement over the present state-of-the-art!

If the above essay seems to make reasonable sense to you, then we believe you are better equipped to make an informed decision when purchasing your next loudspeaker.  We also believe you will be better equipped to judge the value of our products.  Remember: When the debate remains unresolved and you're tempted to accept status quo, a paradigm shift is in order.  Never accept status quo.

We at SP Technology do not accept the status quo of the industry’s imposed constraints on performance.  We do not accept the here-to-fore paradox of choosing between every other significant factor and excellent time domain performance in our designs.  We will not be constrained by ideology.  What we will do is use the “first order” or any other crossover topology as well as any other method when the guidelines of NATURAL LAW dictate their need.  By so doing, we will ultimately change the course of recorded music.

Bob Smith – president
SP Technology Loudspeakers
04-27-04

[infiniti driver]

Excellent article Bob!

One factor that is most impossible for a loudspeaker engineer/developer/company to figure in is that of room calibration. Typical boundrys in rooms are the culpret for many fine loudspeakers offering shortcomings in the performance arena. EVERY loudspeaker made will have a bottom end boost (35 to70hz) due to the room and the boundries..not to mention what certain shapes of other articles can do to the group delay patterns AFTER the sound has left the enclosure. In my mastering, I look at powerband response. A properly positioned set of reference loudspeakers that have a clear shot to the ears and away from boundrys will give a more accurate display of how the recording was calibrated and rendered. Since the art of mastering encompasses compromises to deal with the majority of music lovers and their systems ills (and rooms) one would say that what I do and why I do it is unorthadox. True bottom end response, compensation for the room is something I do not do. True bass response compensating for Fletcher-Munson effects is another area I do not dabbel in either, due to the fact that everyone seems to have a "different sound pressure level of enjoyment".

On the Evaluation CD, I have 3 test tones. One should calibrate their system so that these tones are at about 90dB in order to have a good understanding of the level that is "pretty uniformly" acceptable. These tones are "set" to a reference -12 dB. Many fine recordings will not even get to -12dB on average and they shouldn't. An average reference level for even a "POP" CD should be around -17dB , although with the loudness wars that are raging in the recording and mastering industry, I have found some works that run in the -6dB rms range and have little or no dynamics. Peaks dependant on the music should be allowed to 0dB but only for a few milliseconds. What you set your level, in your system for your music, is strickly dependant on comfort level, max power headroom and the music itself.

When a loudspeaker manufacturer calibrates their loudspeakers to a "recording" or a "room" is when the litteral balance of the recording arts gets disrupted. I am sad that the SACD format has some shotty mastering. It seems that what I am hearing is "all over the place" in terms of "powerband balance". Hopefully more studios will select SP products in the future so at least they will get the lower registers and the 2 to 5K range back to where it belongs. Music is to be enjoyed, not to be suffered. Establishing a new paradigm is something that takes time since we also do not hear the same way, many people use all kinds of tweaks and their simply are no uniform standards for rooms, wire, amplifiers, recordings or mastering. All we can expect to do is at least eliminate as many barriers that have been posed with the entire signal chain and hope for the best. Loudspeakers and rooms are one of the large areas of oppurtunity. Getting mastering engineers to sing off the same sheet is another large oppurtunity. One thing about paradigms, their is a shift and it is not "same old stuff" or "standard issue". A paradigm shift is one that comes with a whole new game of how to do things. Tweakers do what they do to raise the bar. When the bar is raised high enough, tweaks are something that can also deteriorate performance instead of enhance it. Sometimes it is a good idea to "detweak" your system and get back to meat and potatos without all the trimmings. You may find that nirvana that lurks out there waiting to be discovered!!!

Good common sense is the key. Put the loudspeakers where the room can enhance the sound (decalibrate) and you will find some unlistenable music. A fine balance of room interaction minimizing and good placement at ear level with sensible short run connections and high grade source material and quiet electronics, it gets you on track.

[JoshK]

Boy did you ever speak to my mathematical sensibilities!  Great reference, you should sticky this thread.

[JoshK]

One question.  What is your feeling as to active versus passive crossover implementation.  Does it just trade one set of problems for another or can it actually have very beneficial results, assuming it is implemented properly?   I know you need to have more amps, but for the moment lets look past that issue.

[KevinW]

Bob,
Nice writeup.  Since i'm using 1st order slopes in my speakers, my only comment is that I think the phase-correct alignment is important to get truly world-class imaging.  Hearing in stereo requires phase information from ear to ear, and thus phase is a factor.  You are correct in saying that it is not necessarily the most important factor, and it is a designer's decision just how high of a priority to make it.
 
Cheers,
Kevin

[Aether Audio]

Dear JoshK,

Thanks for your support!  After reading a few other posts here on AC concerning crossover issues and what I perceived to be somewhat misleading remarks, I got all fired up.

To answer your question the best I can, lets just say that there are obvious advantages and maybe some not so obvious disadvantages.  Let's list the advantages first.

To begin, every amplifier exhibits to some degree the property of "Damping Factor."  This quality is dimensionless and can be thought of as the inverse of output impedance.  So...output impedance is equal to the load impedance (resistance) that the amplifier can drive wherein half of the output voltage is dropped across the load (speaker or resistor) and half is dropped across the output devices (transistors or output transformer) of the amplifier.  Realistically, if that condition lasted more than a second or two, most amplifiers would go into their self protection mode or burn up (or pop the circuit breaker).

Anyway, output impedance reflects how much current an amplifier can source.  It is also an indirect indicator of how solid its power supply is.  The lower the number (in Ohms) the better.  Conversely, if Damping Factor is the inverse of output impedance, then the larger the number the better.

Damping Factor then is a reflection of an amplifier's ability to absorb or sort of "soak up" any signal that is fed backwards into its output stage that originates from the load (speaker).  When the amplifier is putting out a signal we say it is "sourcing" current, which is directly related to its output impedance.  When it is absorbing a signal that is not supposed to be present on its output, we say it is “sinking” current.  This ability to sink current is another way of saying its ability dampen or dissipate an erroneous signal.

Why is this damping effect important?  Because speakers operated backwards will generate a signal.  You can even use a speaker (conventional dynamic type) like a microphone.  Just hook the speaker’s output to a little microphone mixer or preamp and then talk into it.  It may sound sort of crappy but it will work.

Speakers work backwards and generate a signal when they are fed a signal from an amplifier and then that signal stops (imagine some sort of impulse).  Because the cones have mass, they can’t stop instantaneously and will keep on vibrating back and forth, generating a signal that gets fed back down the speaker wire to the amplifier.  All speakers are self-damping to some degree though.  This self-damping (to some degree) is a good indicator of the strength of the magnet/motor system of the loudspeaker drivers.

Regardless of how good the speaker system is self-damped, it will never be perfect.  To the degree these unwanted vibrations are not damped, the speaker will produce distortion.  If the speaker is driven by an amplifier that has a high Damping Factor, the amplifier will greatly assist in stopping these vibrations.

Now, if you put any resistance between the amplifiers output and the speaker, that resistance will degrade the damping effect of the amplifier.  This undesirable resistance acts to sort of  “decouple” the speaker from the amplifier.  This is why we try to use large gauge/high quality speaker wire in our high-end systems.  Larger gauge wire has less resistance.

The above explanation may have been somewhat redundant to you, but others may benefit from it.  Anyway, the passive components (capacitors, coils, resistors, etc.) in the crossover network built into any speaker will also add their own amounts of resistance to the path between the amplifier and the drivers.  These components then will degrade the effective damping factor of the amplifier and hence, the damping of the drivers.

If we connect the output of the amplifier(s) directly to the drivers as in bi-amping, we will improve the overall system damping and thereby (hopefully) reduce distortion.  If we also place the amplifier(s) as close to the speaker as physically possible, we will use less wire and that will further improve things.

The other benefit is that certain passive components, notably iron of ferrite core inductors, will become non-linear and produce distortion if over driven.  These components will not be in the signal path if one chooses to bi-amp, so there is another source of potential improvement in performance.

One final advantage is that the amplifier power is used more efficiently as the losses in a passive crossover are avoided.  This is usually not a major factor in home use though.

Now the disadvantages.  As we all intuitively know, every component that the signal passes through is another potential source of distortion.  The active implementation of a crossover requires quite a number of both active components (transistors, op-amps, diodes, tubes, etc.) and passive ones (capacitors, resistors, transformers, etc.).  If the active crossover is not constructed of the highest quality components and of superior design, it can significantly degrade the signal before it ever gets to the amplifier/speaker.  

Since active components are inherently non-linear and must be forced to operate in a linear mode, they represent a significant potential for signal degradation.  The effect is usually one of imparting an “electronic” sound to the music.  This can be far worse than the components in a passive crossover network, as seeing that they are passive, they are (sans non air-core inductors) inherently linear to begin with.

As you can probably tell by now, I don’t have a solid opinion one-way or the other.  In Pro Audio, active crossovers are used all of the time.  I’ve used them many times and with excellent results.  This has more to do with convenience, increased power efficiency and flexability than actual fidelity though.

I hope this answers your question and I want to let you know we appreciate yours and everybody’s questions.  We’ll always try to do what we can to help lift the veil of mystique that permeates the audio world.

Take care,:D
-Bob

[JoshK]

Bob,

Thanks for taking the time to answer my question.  Your answer was clear and comprehensive.  

Somehow I had the opinion that active crossovers would (assuming both active and passive were designed well) be better, but I couldn't quite explain why I thought this.  I just thought that passive components were less efficient and caused IM distortion.  But I guess if the active crossovers were to cause non-linearities then this would moot all gains.

[Aether Audio]

Dear Kevin W,

Thanks again for your comments as well.  The only response I have to your comment about imaging will be short.  I understand your point but I believe good imaging has more to do with the relative phase between the left and right speakers, low distortion, diffraction, dispersion and both linear frequency response and matched frequency response per speaker pairs.

The reason for my belief is that as long as the left and right speakers track each other in their inherint phase errors, there will be no difference between them and the error will cancel out.  It is the phase differences between the left and right channels of a system that impart the stereo effect.

Besides, the spacial cues (delays) that the brain uses to discern image location are orders of magnitude larger than the typical inter-driver delays imparted by crossovers.  I will limit this comment to higher order crossovers being implemented above 500 Hz though.  I admit that the larger delays exhibited when higher order slopes are used at lower frequencies do lie within the range of imaging cues used by the brain.  

Never the less, I still suggest that as long as these delays are mirrored in both speakers, there should be no effect upon imaging.  Don't ask me what effect longer delays have on transient response and dynamics though.  Those suckers are bad news when it comes to that issue and I despise them!  Please don't be offended, but I believe that your view is partially a byproduct of a lot of mis-information that has been promoted by designers of first order systems.

Basically, I suggest that if the total system group delay is less than 1 milllisecond from 100Hz on up, no degradation in imaging will occur.  Both of our present models (Timepiece 2.0 & Continuum A.D.) exhibit approximately 500 microseconds (1/2 millisecond) of inter-driver delay at the crossover frequency of 950 Hz.  The total group delay increases to 1 millisecond at 100Hz and is a residue of the systems low frequency response not extending to DC.  Specifically, it is a result of the high-pass function of the woofers low frequency rolloff below 30 HZ.

If you doubt my claims (which you should) then check with a couple of useres that have posted here on AC.  Try PMing horsehead, reefrus, infinity driver or tasos and ask them about our products imaging performance.  You might be surprised!

Take care,
-Bob

[JoshK]

Ever think about teaching?  You are a great teacher.

You definitely caught my attention when you started talking Laplace and FF transforms.   I am an applied mathematician you could say.   I wish I could read up on all audio with that sort of explanation.

[Daniel]

Oh fer cryin' out loud.  If yer gonna kiss the teachers butt at least do it in a PM.  Smoooooooch!

[DSK]

Now, if you put any resistance between the amplifiers output and the speaker, that resistance will degrade the damping effect of the amplifier. This undesirable resistance acts to sort of “decouple” the speaker from the amplifier. This is why we try to use large gauge/high quality speaker wire in our high-end systems. Larger gauge wire has less resistance.

Bob,
Thanks for taking the time to share your knowledge with us, though I must admit my eyes glazed over and I probably looked like a stunned deer after a few paragraphs.

I was interested in your comments re the need for large gauge speaker cable, as there seems to be a growing trend toward thinner gauge cables. For example Audience's Au24 is highly touted, often over high end cables that are many times its size.  Audience's website claims the following...

There is a common misconception that loudspeaker cable must be large in diameter and have a low DC resistance in order to provide good bass response. DC resistance is relatively unimportant. What really matters is the characteristic impedance (AC resistance) of the cable. Music is an AC signal after all. Most of these large diameter/low DC resistance cables have excessively high characteristic impedance anywhere from 100 to 600 ohms with some measuring in the 1000’s of ohms. The Au24 Loudspeaker Cable is only 4mm or 1/8" in diameter. Although the DC resistance may be slightly higher than the garden hose variety speaker cables the characteristic impedance is only 16 ohms. Musical signals from the bass to the overtones pass through this cable with less actual impedance than a cable with a lower DC resistance. The Au24 Loudspeaker Cable provides nearly perfect bass timbre and extension. Plus, the low eddy-current resistance means the cable is very quick and linear in the time domain. Voices and instruments sound strikingly real with this loudspeaker cable.

Just wondering what your thoughts were?

Cheers,
Darren.

[infiniti driver]

This is my observance of thinner gauge wire for loudspeakers. When the wattage is peaking over 400 watts per channel, the wires tempurature raises. Hotter wire = more resistance. I use 8TC Kimber I find it is not only sonically fine, but also can deal with large output amplifiers.

Bobs take will be interesting. Remember, the loudspeaker cables make up part of a CIRCUIT and cannot be judged alone until one knows all the complexities of said circuit and all of the ways it is to be used. Loudspeaker cable can be very dependant on the amplifiers/Loudspeakers combination. Boutique cable of one type can be good or bad depending on how it is implemented.


More..EDIT..

Cable Histeresis. this is important to all the cat 5 users. Most Cat 5 users are running SET amps with output transformers and no more than 3/4 to 8 watts max. Typical SET user is running 3 watts.

Typical audio system for regular listening does not produce (usually) more than 1 to 3 watts for typical levels with peaks of 15 to 30 watts. Beleive me, CAT 5 has many problems running in the 100 to 200 watt peak level and above that, even 24 gauge, well it gets totally non-linear.

I think the cat 5 craze was an attempt to match output transformer wire gauges to the loudspeaker and some of that may be valid.

More talk here is welcomed.

[Tuckers]

I've had so many speakers over the years of many stripes.  But I was always dissapointed with the sound over time.  I began to be educated and hear a few time and phase accurate speakers out there, and there was something truthful about the sound I liked over other designs.

If I were to use a trumpet as an example sound, ordinary cone speakers may sound more detailed and you might hear more room effects etc. at first, convincing an audiophile that the speaker is revealing more information.  You would hear the throat of the instrument distinct from the bell, and the note floats out in front of the instument, and you hear distinct hall sound. To an audiophile, this initially sounds very exciting, lot's of detail, micro imaging, reverberation.  

When I hear the same recording on a well-design time and phase aligned speaker, the same trumpet usually is in a much tighter imaging space, usually farther back in the soundstage.  Those individual pieces of the trumpet are replaced by a smaller but more solid and coherent instrument.  All the sounds come at you together, you can still hear the parts but they are not presented in an exploded view.  

The other thing I have noticed is that when a speaker is time and phase accurate, it sounds good pretty much everywhere in the room. That is you can still hear soundstage, image density, transparency.  Things continue to sound more coherent  than non-time and phase accurate speakers.  What does happen though, is that the treble will either roll off a bit or tip up a bit depending on location (is there a speaker that doesn't really do this though?).

You pose the question as to why more of the big expensive audiophile speaker brands aren't phase and time aligned?  And conlcude that it's because it's not that important compared to other parameters.  But actually I believe it is a few other reasons.

The first is that to design a time and phase accurate dynamic speaker system requires so many compromises on design choices etc.  It really is a kind of boring-looking and unsexy concept next to the competition.   There's not a lot of mumbo-jumbo you can throw into a speaker like that, a simple crossover, very-well made but not exotic drivers, limited cabinet choices.  

Second, time and phase alignment is a long-term satisfaction goal that doesn't sell like the sizzle of titanium or berylium drivers, and hyper-detailed and initially exciting speakers.  In my opinion, one of the reasons audiophiles buy and try so many speakers is that they are looking unconsiously for the wholistic presentation that phase and time alignment can help provide in the context of a special system.  The big flashy speakers provide short-term fun, but long-term disatisfaction.

Sorry for waxing on about this in your forum, but you brought it up and I wanted to voice another opinion.

[amitm]

Great explanations! However, I have never understood one thing. Why can't crossovers be implemented digitally? Crossover order is not an issue, neither is compensation for the group delay through a digital delay line. The only downside that I see is the need for one amplifier per driver, but that is not worse than bi-amping in a two way system.

--amit

[Carlman]

About your waxing...

Second, time and phase alignment is a long-term satisfaction goal that doesn't sell like the sizzle of titanium or berylium drivers, and hyper-detailed and initially exciting speakers. In my opinion, one of the reasons audiophiles buy and try so many speakers is that they are looking unconsiously for the wholistic presentation that phase and time alignment can help provide in the context of a special system. The big flashy speakers provide short-term fun, but long-term disatisfaction....

I really think you nailed why this design isn't popular with your comments.

What's more, we're all gear heads and gear excites most of us... to some degree or to some angle.  I have finally found the sonic characteristics I love in speakers.  I know what I like now better than ever.  However, I also am allured by shiny new speakers, sources, projects, etc.  I like technical accomplishment.  I respect good design.  You can't turn it off...  It's like marrying the love of  your life... you still look if something pretty walks your way...

[lonewolfny42]

It's like marrying the love of  your life... you still look if something pretty walks your way...

Would that be... "lust in your heart ?"........  http://www.evangelicaloutreach.org/lust.htm  

[Carlman]

It's like marrying the love of  your life... you still look if something pretty walks your way...

Would that be... "lust in your heart ?"........  http://www.evangelicaloutreach.org/lust.htm  

Do I really deserve a link to a lecture for this one?  Sheesh you're heavy handed..    Of course, you can say whatever you want since you're buying a pair...

[amitm]

Great explanations! However, I have never understood one thing. Why can't crossovers be implemented digitally? Crossover order is not an issue, neither is compensation for the group delay through a digital delay line. The only downside that I see is the need for one amplifier per driver, but that is not worse than bi-amping in a two way system.

See the thread:
http://www.audioasylum.com/audio/speakers/messages/162696.html

Some hype is in there. A good filter is necessary.  

What people do not mention there is that voicing of the final speaker resulting from it will still depend on the designer -- voicing of NHT speakers is never to my liking. Plus, cabinet design etc. -- the traditional speaker design issues -- are still valid, although to a lesser extent. Diffraction cannot be accounted for by DSP processing! Wonder how much equalization control they will allow, and how user friendly it will be.

Sorry if it distracts from the main discussion here. This will be my last post on this in the current thread.

--amit