almost everything you ever wanted to know about subwoofers


Whether you are adding a sub to an existing 2-channel "stereo" system, or building a "surround sound" Home Theater system, let's examine every detail involved, including all sorts of setup options and adjustments, subtle and not so subtle. Whether you fancy yourself a pristine audiophile, a professional recording / mixing / mastering engineer, a Sci-Fi movie buff, or a professional musician, you have your own rainbow of views, and objectives, and they each have a "scale" whether they are emotional, financial, audiophile, social, practical, experimental... therefore you have to "fit" your purchase(s) and adjustments into this range of ideas you have.

Well, be prepared for a ride.

The ONLY correct way to add a sub to system is to define everything ABOVE the sub's [frequency] range as an entity; clearly define the impulse, phase, and lastly frequency response of this entity; and then make a new "2-way" system where the sub is one way and everything above it is the 'other' way. The parts must be combined correctly so that there are no cancellations and no smearing of time-related musical events.

This CANNOT be easily measured in the frequency domain, because you could have (as an example) an 80 Hz signal coming from both the mains and the sub, and if the sub is 12.5 msec late the two sources will "seem" to be in phase but the sub really will be 360 degrees, or one full wavelength late. It is the impulse smearing that this affects, but people don't measure that because there is no simple "hand held" phase or impulse meter as there is an SPL meter. The REASON this meter does not and essentially cannot exist is that in order to measure impulse response or phase response you need a starting REFERENCE point, (in time) and in a speaker system, since the signal has to travel through circuitry, amplifiers, passive crossovers inside the speaker box and then hit the driver; therefore the first reference point MUST be acoustic.

There ARE computer based impulse response systems such as the TEF, ( very quick technical blurb HERE; full story HERE ) but they are involved, require real instrumentation, are expensive, have a seriously steep learning curve, and they are absolutely not the kind of thing most 'consumers' -- or audiophles, can be bothered with or have patience for.

So the overall view of adding a sub is this: In essence you are designing and assembling a new speaker system which is "2-way": the sub is one way and everything "else" above it in frequency is the 2nd way.

Simply connecting a sub to existing mains speaker (or amp) terminals is the WORST POSSIBLE WAY to do this. EVERYTHING scientific and acoustic about this method is wrong, from the additive delay issues to the back EMF of the mains affecting the LF signal. However there are plenty of people who simply do not understand correctly integrated bass, and they will be reasonably happy simply sticking another box on to their system without regard to timing, phase and frequency issues, and they will think it sounds "ok" or "good" and for those people it doesn't really matter.

Indeed the only thing that does matter is an individual's happiness with their system, whether I or anyone else thinks it's right or wrong.

But I want you to know and understand the truth, so to get purely technical...

There are a separate set of issues for 2-channel stereo "audiophile" and Home Theater systems, which we may call "Surround Sound", i.e., 5.1, 7.1 etc. systems. Later you may want to read my Surround Sound page here: www.soundoctor.com/surround.htm


DO NOT CONFUSE LFE with BASS. LFE is a separate channel in a movie theater (called the 'boom' track in the industry) which is necessary because there is not enough dynamic range (headroom, actually) in the existing film optical sound tracks and their associated playback hardware for additional "Low Frequency Effects". In a movie theater, (as will further be explained below) you CAN have multiple low frequency sources because there are essentially no standing waves of any consequence in that size room. This is also a separate channel on a DVD or BluRay, as explained below.

With Home Theater, there is relatively little coherent phase correlation between the LFE channel (the true Low Freq Effects channel) contained on the DVD / BluRay and all the frequencies ABOVE 80; all that is really essential for the low frequency part of movie enjoyment is the best coupling of the below 80 or 90 Hz effects to the room and indeed the listening position.

There is ALSO the rest of the bass; all the below 80 or 90 Hz information from all 5 channels that is stripped off and summed together into mono and sent out the SUBWOOFER OUT connector on every modern HT receiver / processor. That plus the LFE channel (if it in fact exists on that particular DVD or BluRay, and it may not) constitutes all the MANAGED BASS. Therefore the MOST desirable scenario in a HT situation is to best couple the sub(s) to the room FIRST, and THEN timing and phase match the sub(s) to the rest of the system. This way you will get the mostly sub Low Freq "effects" coupled to your chair, AND the correctly timed musical bass present typically in the L and R channels.

In a living-room sized room, the most desirable setup is to run all the speakers as SMALL, and send the fully MANAGED BASS (NOT 'just' LFE) to the SUB(s). Another important criteria is that you want the best "sound effects" from movies and the best bass from 2-channel sources playing back through the same speakers. There have been PLENTY of complaints about "the movies are great but the 2-channel stuff sucks" or vice versa.

Audio is audio. Correct audio correctly presented is what we are after whether the source is AM, FM, a CD, an SACD, a DVD, a BluRay, a simple analog computer out, simple or complex DACs, esoteric music servers, streaming services, the analog headphone jack out on a computer or iPhone... and so on.


Impulse response (NOT frequency response) really is the holy grail of all of audio. With more pristine 2-channel sound, (and when you are playing music through your Home Theater system) as we approach, want, or expect audiophile quality, the issue is to get the IMPULSE RESPONSE through the crossover region (and therefore both the phase response AND frequency response, which is contained under the mathematical umbrella of impulse response) as smooth as possible, so that IF we were playing back a correctly recorded IMPULSE, for example a well recorded kick drum, its fundamental (50-60 hz), and its subharmonic, an octave lower (25-30 Hz) and its mostly odd order harmonic structure (all the way up to 8 kHz and then some) are presented correctly by the time they arrive at the acoustic summation point which is your ears. This is the basis of "high fidelity".

We also have to assume — and this is a huge assumption — that the manufacturers of our "mains" speakers have ALREADY correctly addressed the issues of both impulse response and frequency response. So for the purposes of this discussion (my entire book isn't ready yet) we will assume that whatever your mains are, from a 2-way bookshelf to an 8 foot tall floorstander monster, that within the desired passband of the mains, the impulse response and frequency response are already well handled.


Then there's the subject of absolute polarity. This has no phase relationship to anything other than ITSELF. Imagine you are standing in front of a nice, large, beautifully tuned drum kit. The drummer obliges us and plays just the KICK drum, perhaps loudly and once every second. So the pedal is a mechanical impulse hammer device which hits the skin on the drummer's side; this pressurizes the air in the drum, and the front skin moves forward.

That's an IMPULSE. It's actually the leading edge of a square wave, with a little slope to it. A square wave by definition has a fundamental and only odd harmonics. A sine wave has only it's fundamental frequency, and a triangle wave is the fundamental and only even-order harmonics. So the impulse of a kick drum is nearly a square wave, with some sine wave fundamental and some even order harmonics, but less than the odd order harmonics present in the square wave part.

The net human result, since you are standing or sitting in front of the drum, is you feel and hear this positive pressure wave, and your ears, body, intellect, social acuity, and previous memories of such things all converge and you "hear" this phenomena as a kick drum hit. You see it; you hear it, you recognize it, and it fits your preconceived notions about what a kick drum should sound like. In theory, this sound is then picked up by a microphone. Positive [air] pressure on the diaphragm of the mic produces a positive-going (+) voltage at pin 2 (of the 3-pin connector); then this goes into a mic preamp, the rest of the line amplification, and at some point in the control room of the studio, out to a monitor amp and then a loudspeaker. If all goes well, we then stand in front of that speaker, and listening to the monitor system, we are socially convinced there is a drummer obliging us by playing a kick drum right in front of our face. If the absolute polarity of that impulse is "backwards" — i.e. the polarity of anything in the circuit is changed, such as the monitor speaker is wired out of polarity — then the absolute polarity is not the same as the original and we can hear that. This is one instance where this phenomena is very easy to both set up as a test and easy to discern. Clark Johnson has written a entire book about this called The Wood Effect, available HERE.

Imagine we are playing back a well recorded cello: we have the fundamentals of both the strings and the resonance of the wood STARTING in the subwoofer (that means below 80 Hz, and you may wish to refer to my frequency-wavelength chart here: www.soundoctor.com/freq.htm ) and the harmonics extending smoothly up through the various drivers in the rest of the system.

Being a recording of actual "wood", (and the strings!!!) these harmonics are mostly even order. If we can correctly preserve the exact timing (and therefore phase) relationships of the ratios of the harmonics of these signals, we will preserve the imaging "in space" of this instrument. If we do not do this, then the focus is lost. One part of this assumption is that the instrument is correctly recorded in the first place, ideally with a stereo pair of microphones which therefore ARE picking up the 3-dimensional phase and harmonic structure of the instrument in space.


You CANNOT have multiple low frequency sources of differing phase relationships in a living room-sized room. Let's examine the acoustic "spaces" we might be dealing with. There are 3 useful separate sizes of acoustic spaces in life:

1)  The inside of a car, where you are essentially living inside the speaker cabinet. (the pressure zone)

2)  A large movie theater, amphitheater, or outdoor space where there either are no reflecting walls or the walls are so far away in time that any reflections, partially because of the Haas effect and frequency cancellation effects are essentially of no importance; and...

3)  The inside of a typical living room / home theater room. In this size room you will ALWAYS have standing wave issues somewhere in the bass passband from 20 -125 Hz. You CANNOT NOT have these issues in a room this size unless you have a REAL acoustically treated room with full size, perhaps 32' bass traps in the walls and all the correct ratios of absorption vs diffusion especially at low freqs. This does not mean a couple of pillows in the corners or ineffectual 800 Hz absorbers on the side walls. IF you were to have a room with REAL bass trapping then there would be no bass standing waves because the LF signals hitting the walls would be absorbed before they had a chance to bounce back. (what a concept!!!) Rooms like this are a revelation, (not to mention extremely rare) because for the first time you are actually able to hear the speaker, and not the speaker "in" the room.

But back to "most rooms"...

If you have 2 LF sources of differing phase relationships (that means timing relationships) they will cancel. Period. And if they are "in phase", but 1, 2, 3 or more full cycles (that means wavelengths) shifted, (that means 360 or 720 or 1080 degrees out of phase) then the overall frequency response will not seem bad but the impulse response and clarity and focus will be smeared, and localization and imaging will be lost. This is the main reason measuring in the frequency domain especially in a home-sized room is such an incredible waste of time. Your measurements "seem" pretty flat and yet you don't like the end result - isn't as "clear" as you think it should be, and it isn't as focused as you think it should be. The issue is ONLY timing.

We can call the red and green waves signals from 2 separate "speakers", 2 separate subs, or a sub and a "mains" speaker. Here are the diagrams that show this:

Fig 1.  Obviously "in phase"

Fig 2.  90 degrees "out of phase" (the red wave is lagging the green wave by 90 degrees)

Fig 3.  180 degrees out of phase (the net result is complete cancellation)

Fig 4.  ...an example of group delay. This only shows one cycle of many. It's entirely possible the signals are overlaid so they look like they are "in phase" but they are actually 360 degrees (one wavelength or cycle), 720 degrees (two wavelengths or cycles), or 1080 degrees etc. shifted in time out of phase.
Fig 5.  Group delay drawn another way. The GREEN wave might be coming out of your "mains". The RED wave is coming out of your sub. Notice how at first they "look" as if they are "in phase" but the red wave (from the sub) is actually a full wavelength LATE.

How did the sub get to be 360 or more degrees late? It's the overall physics of how it's built. The only correct way to implement a sub so the frequency response and phase response can be controlled and have it socially acceptable in a living room is to implement a sealed box design, and that means EQ circuitry. Also most of the better brands of subs, JL Audio included, use massive drivers which have a relatively large X-max (that means cone movement). The combination of the air pressure in the sealed box, and the rest of the equalization circuitry necessary equal a mechanical and electronic phenomena which equals an overall time or group delay. Therefore IF the sub is already 8-10 msec late, AND it is placed in the corners further away than the mains (just for example) then relative to the mains it might be 12-16 msec late. YOU CANNOT TAKE THIS DELAY AWAY.

If the sub has a VARIABLE PHASE knob, (and not an incorrectly labelled "phase" switch which is actually POLARITY), then as you turn the phase knob "up" you are ADDING DELAY to the low freqencies going through the sub.

You might enjoy referring to my handy FREQUENCY-WAVELENGTH-PERIOD chart here: www.soundoctor.com/freq.htm


In addition to all the above, there is the complex issue of the "main" speaker you are coupling to. There are essentially 6 types of speakers that exist:

1) sealed
2) port in the front
3) port in the bottom
4) port in the back
5) a dipole, which is a flat panel such as an electrostatic (Sound Lab, Magnepan, Quad, Beveridge, Martin Logan, etc.)
6) an true omnidirectional system such as the MBL or the BEOLAB 5.

Each of these speaker types couples somewhat differently to the room, and certainly to a sub in that room, and therein lie the problems in acceptable integration.

A port is ALWAYS nothing more than a cheap way to attempt to get free bass out of an enclosure and /or driver that's too small. It's a holdover from the 1930's when because of driver inefficiencies (especially when compared to today's units) you had to do everything possible to increase the useable output over the desired range of low frequencies.

At one level, all the guyz want 9 foot speakers in the living room (read "man-cave"). All spouses, of whatever gender, want tiny 3" speaker cubes that disappear, but expect 9-foot results from them. Since they haven't repealed ohms law or any other laws of physics while we were sleeping, the only way to get correct sound is to move a correct amount of air.

Lets examine ported speakers. We'll start with the worst case, the port in the front. At mid bass frequencies, say 50-80 Hz, the LF driver moves IN the cabinet, the air in the cabinet is elastic, and the port air moves out of the cabinet. Because of the frequency at which the cone is moving, by the time the cone moves out (forward) again, the port air is now moving out, so in front of the cabinet the two air pressure sources sum together and you get a fake bass "bump" or "boost".

As you go lower and lower in frequency, at some low frequency the air pressure from the LF driver and the air pressure from the port are exactly opposite each other, so they cancel, and there is no more audio at that frequency: it disappears.

When the manufacturer of a speaker cabinet defines the frequency response (i.e., 37 Hz - 20kHz +/- 3dB) this is what is defined by the entire arrangement of the port and the air in the cabinet and the driver. At some low frequency the port air is exactly out of phase with the driver air pressure and since they cancel, there is NO output from the cabinet into the room. Therefore with a ported cabinet, the entire sloppy concept is this juggling game between the response of the drivers under air pressure, the passive crossover inside the box, the port size and placement.

You must understand that ANY driver goes down to 0 Hz, or DC. If you put a battery across a speaker, the cone moves out and stays there. If you were to have a DC coupled power amp feeding a speaker - ANY speaker, from a 1" dome tweeter to an 18" rock n roll stage bass driver - and you put 4 Hz into it, it would simply move back and forth at 4 Hz. Of course in order to actually "hear" the audio it would have to be in the generally accepted passband of 20-20,000 Hz and the cone diameter would have to be enough to actually move some air in the room. So it is the overall combination of the driver size, the excursion, the box size, (therefore the air back pressure) and many other factors that determines the overall response of that "speaker" AS AN ENTITY.

That means IF you were to simply put those same frequencies through the mains and the sub (that means with no crossover, and this is the mistake that nearly everyone makes) you would now have 3 sources of LF energy and differing phase: the 'main' LF driver, the port, and the sub, all fighting with each other in the time domain. A further corollary is that since the air inside the [mains] cabinet is elastic, the phase relationship of the port air to the driver air is also a sliding one; that means it's "out of phase" — and smearing — over a wider range of frequencies than you might think.

If the port is on the back, again, a cheap attempt to use the back wave bouncing off a wall to give 'additional' bass, you have the ADDITIONAL issue of the transit time it takes for the back port pressure (already delayed because of the elasticity) to leave the cabinet, travel back, hit a wall, and bounce back around the front of the cabinet again; therefore this LF wave MIGHT be "in phase" with the front driver BUT BE 360 OR EVEN 720 DEGREES LATE; therefore it sounds like the bass frequencies are ok in the frequency domain but the IMPULSE RESPONSE is now muddied.

Also, in the case of back ported or (type 5) dipole speakers, since the path length from the back of the speaker to the wall and bouncing back around to the front of the speaker is a fixed physical entity, at some frequencies you are adding and at some frequencies you are canceling: you have simply made a physical/mechanical frequency comb filter that you can't do anything about. Sound Lab's answer to this (for use with their flat panel electrostatic speakers, which are dipoles) is they sell you a "Sallie", which is an absorber to absorb the entire back wave output of the electrostatic panel. Since now there is no comb filtering; all you are therefore hearing is the front signal.


A ported sub for home use is even more wrong than ported mains. Now you would be attempting to acoustically add together in the room at least SIX low frequency sources with differing phase and frequency slope conditions: the LF drivers in your two mains, their ports, the sub driver, and its port. In addition, since it's a bandpass it cannot go down low enough for serious Home Theater effects. (that typically means a real 20Hz or close to it.)

In some cases such as a bandpass sub used in a club or on a modest-sized stage in your local pub, you are most concerned with efficiency and not with getting frequency response "flat" down to 20 Hz; therefore a correctly set up bandpass box that might roll off at 35 to 45 Hz is quite sufficient and also very efficient for the defined purpose. And again, as a point of reference, "flat" response in the frequency domain is FAR AND AWAY the LEAST important phenomena: impulse response in the time domain is the most important, but it cannot be measured with a handheld meter therefore almost everyone simply ignores it. If you're interested in learning about the newest (and evolving) pro sound system / stage methods of "steering" bass, Dave Rat has some very cool videos here:

part 1  www.youtube.com/watch?v=VwLH7zP6Lwo

part 2  www.youtube.com/watch?v=B-3pURYOwfw

part 3  www.youtube.com/watch?v=aSZK9Altvm8

There's a nice article here:

But back to our Home / HI-FI / 2-channel / Audiophile / Surround Sound systems: There is ONLY ONE truly correct way to "add a sub" to a system in an controlled listening room situation: you must correctly cross over the 2 sealed cabinets; and their timing must be correct. ANY other method will lessen the focus and clarity and imaging you have tried so hard to preserve.

I have many clients and customers with extremely exotic high-end 2-channel systems that are all chasing the holy grail of 3D holographic sound imaging, and until they follow my distinct guidelines they are never completely satisfied with the results.


A similar situation exists with home theater setups where the customer THINKS that the front speakers are "full range". Even so...

The BEST overall approach is to seal the ports, operate the 5 channels as "small", crossover at 80 (or even a higher, like 90 Hz, but NEVER lower) and correct the timing issues inherent in all modern subs by setting (in the receiver or processor's setup menu) ALL the distances THE SAME, and to a small number such as 7 feet; then set the sub distance to 12 feet MORE (i.e. 19 feet) and THEN use the variable phase control on the sub to fine tune the relationship at the 80 Hz crossover point, at the listening position.

Some better speaker companies that make "large" speakers (such as B&W) are aware of this port issue and supply port plugs just for this purpose. Kudos to them.

People who have fought with their systems for weeks or years finally email and call me to tell me that for the first time they are finally satisfied — in fact thrilled — with the incredible integration of their JL Audio, MK Sound, SVS sealed, or other fine subs.

All of this discussion (so far) barely scratches the surface of the true complexity involved in flawless integration, so let's continue.

The idea of setting exact speaker distances is flawed from early mistakes made by both receiver / processor manufacturers and the somewhat misconstruing of the acoustic and other technical differences between a large movie theater and the home setup. I cover this in more detail on my surround sound page, here:


On top of all the previous variables we have all the issues, errors, and modern production values and practices inherent in the recording process. It is simply laughable (and pathetic) when I read the magazine articles where the reviewer calls the "soundstage" of a rock recording "palpable". Sorry, but every rock/pop recording made in the last 50 years is composed of a series of panned mono sources that have absolutely no "depth" or "width". They are each separately sent to an echo/reverb device, the delayed returns of which are usually (but not always) panned somewhere in the left to right soundstage 'width'. The combination of the 3 panned signals ("real", "echo return 1", and "echo return 2") then present an auditory fantasy (hallucination, actually) of a "soundstage".

The summation of all the Left-Right panning placement and the summation of all the reverb returns therefore fools you into thinking there is a "soundstage". Sadly, precious few recordings are made with any regard to true stereo or binaural imaging sound in anything resembling a true form; even better classical recordings of large orchestras have morphed into combinations of stereo miking and "some" local more-nearfield mono miking added to the mix to achieve whatever the producer/engineer determine is a suitable balance, perhaps between a soloist and the rest of the orchestra.

There are precious few companies who do pay attention to this; AIX records is one. Chesky Records is another, here: www.chesky.com/content/binaural-series

But to think that any modern, commercial pop recording mix has any true acoustic space (and even uses real instruments!) is, for the most part, sadly mistaken. (There will be MUCH more about this in a long white paper to come in early 2019.)


Oh, and to touch upon "stereo bass" for a moment... there almost is no such thing. Going back to vinyl, every stereo vinyl record cut in the last 60 years has mono bass. It has to. If the bass were 180 degrees out of phase L and R then there would be vertical modulation and the stylus would jump out of the groove. Therefore most cutting lathe electronics have a "compatalyzer" circuit, that dumps frequencies below 160 hz into mono (typically a single-order filter, therefore 6dB/octave). You MAY have out of phase bass (i.e. "low frequencies") on a CD, but precious few producers/engineers are savvy enough (or care enough to even bother, since, typically, what's the point?) to make use of those sort of tricks. There are some EDM dubstep dance trance psychedelia eurotrash electronica club music releases where there are bass tracks where there is stereo bass in the form of something like 24 Hz in one channel and 24.2 Hz in the other channel; therefore you get an air pressure differential which travels around the room. Cool!  In the above example, the "traveling pressure differential wave" would take 5 seconds to go back and forth around the room. If you're a really bored or obsessive techweenie you can have a lot of fun with this - we played with this phenomena at Moog Synthesizer as far back as 1969. Expect to either make your listeners nauseous or to watch their heads rotate on their bodies not unlike the effect in the movie The Exorcist.

As far as PLAYING BACK signals like this goes, as mentioned above, in a large theater or outdoors you CAN have multiple bass sources of differing phase because there are essentially no standing waves, (and if there are, they are so delayed in time that they are of no cancelling consequence in the audio passband) and so your ears (and indeed your whole receptive system) can process and differentiate and accept all the phase issues. In a much smaller room like a living room, it is more difficult but you might be able to pull it off if your subs were more nearfield (the pressure zone). Perhaps if you have a large room, with too much low freq reverberation, you could put the sub(s) right next to your listening chair and adjust their phase appropriately. This would tighten the whole system up. If you invent something new, let me know. Bass is fun!


Fig 6.   A symmetrical layout.  
Fig 7.  An asymmetrical layout.  

So most people's reasons for multiple subs in a room is "more even coverage". Let's examine the instances of multiple subs and what they do. One interesting issue with using multiple subs concerns arrival times. Here's a hypothetical situation. You are feeding the same signal to 2 subs. So this begs the question what's your room like? Is it symmetrical? L-shaped? A closed room? A Huge open space? Notice we're back to acoustics.

Referring to Fig 6, the subs are also equidistant from your body. So the subs each couple to the room however they do. The whole setup is essentially symmetrical.

Now here's another example. Refer to Fig 7. We are still putting essentially the same signal into both subs. There might be 3 ways to do this:
1) from the L and R of a stereo preamp (MUCH more on this later)
2) using a "Y" cord from the BASS MANAGED output from a Home Theater receiver and
3) In the case of a JL Audio Fathom/Gotham series sub, it might be a Master/Slave setup.

The point is, the sound leaves both subs at exactly the same time. Notice in Fig 7 the R sub is closer to your face. Perhaps the L sub is 11 feet away and the R sub is 4 feet away. That's a 7msec time differential. So you hear the leading edge of the bass wave from the Right sub, then 7 msec later the leading edge of the L sub... then the note dies away from the R sub and then 7 msec later the note dies away from the L sub. What have you accomplished? Here comes the magic: YOU HAVE FATTENED UP THE LOUDNESS ENVELOPE IN TIME !  This is the magic that humans love. This is why someone says, "OMG, two subs are SO much better than one!" So you have a combination of the arrival time differential, and to a certain extent you have the separate room coupling issues such that each sub is its own entity coupling into the room with slightly differing standing waves.

It's NOT that the two subs are louder than one, since typically you would adjust both to have the correct desired loudness leval AT the listening position. A quick word about acoustic summing: In theory, 2 speakers in the same room will sum so the result will be 6 dB louder. However they will only do this if the phase relationship is the same. Therefore because of the standing waves involved, if two subs are right next to each other (or one is on top of the other) they will essentially sum at about +5 dB. If they are apart (say, placed next to each main) then they will essentially sum about 3 to 4 dB in the room.

So now we have 2 ways to view the multiple sub issue: as a method of attempting to get better coverage over a larger seating area of a multiple-seat Home Theater room, or as a method of fattening up the bass presence for one or two listeners in a sweet spot.

In the case of better subs, that have variable phase adjustments, my suggestion in setups like this is to use either method (1) or (2) above, and then adjust the phase control knob on each sub for most accurate transitioning at the crossover frequency. It's slightly tricky, but you will keep the real phase relationship between each sub and the mains, AND you will keep the arrival time differential that you "love".

Here's the magic trick if you have one sub in the front and one in the "back" (as shown in Fig 7, above.)

Turn off the back sub.

Align the front sub using the out-of-phase nulling setup on my TEST CD page, here:

Now turn the MAINS off.

Flip the FRONT SUB's POLARITY SWITCH to the opposite position from wherever it is.

Turn the BACK SUB on. Play a sine wave at the crossover freq.

Null the back sub to the front sub AT THE LISTENING POSITION by adjusting the back sub's phase knob and level control.

When finished, put the FRONT SUB's POLARITY SWITCH back to where it WAS.

Now the subs are both level matched AND TIMED correctly AT THE LISTENING POSITION. If you accomplish this correctly if you are playing a drum solo (as an example) you should perceive the lower drum freqs (like from the kick drum and floor toms) to be coming from in the front of the room, as you would expect.


As a separate discussion I should touch on this. In SOME instances, let's say in certain Home Theater setups, you might have the option of "homogenizing" the room, that is, making most of the seats sound "the same". Depending on the number of seats, you might want to make ONE seat as good as possible (your seat...) and for the rest, let the chips fall where they may.

Here is why I actually suggest this: Everyone wants something "different" from the installation. The audiophile dude wants magic. The wife hates audio. The mother-in-law likes movies but hates bass. (she is going to move her chair around until she finds a bass null, believe me) And the little kid wants to climb inside the subwoofer cabinet, and the teenage son wants everything at 118 dB. If you decide this scenario applies to you then you can use the room anomalies (perhaps a bass null) to your advantage.

Perhaps the room is for just two people; Then, typically, the chairs would not both be centered; rather you would have to average out the sweet spot. You decide, carefully.

Some people think that "bass is non-directional". That is a mis-statement. The reality is that as you go lower and lower it becomes less localizable by your mechanism of hearing; above about 100 Hz you can start to localize it and the precision of the localization depends on the rest of the frequencies playing (or not); and the standing waves in the room at the frequency you are trying to determine. Feeding two subs with the same sine wave from a test oscillator or test cd and adjusting the phase knobs separately will show you just how directional it can be. It can be steered around the room with surprising precision, and in my [15] years of night club building we used to adjust the phase steering of arrayed subs so that the bass was correct on the dance floor where it belongs and much less off the dance floor and in the corners of the club.


Understand that the largest percentage of all audio issues is room acoustics. You cannot put a great speaker in a marble shower stall and expect it to sound good - it will sound like a speaker in a marble shower stall. Room acoustics itself is a very complex set of interactions of physics and perception.

Sadly, there are many instances where manufacturers or individuals skew the relevant terms and confuse people. For example, beware of (and be aware of) the dangerous term "Room Tuning". You CANNOT tune a room using an "equalizer". You are tuning THE SOUND SYSTEM with the equalizer - the room is still the same. REAL room tuning means anything from sticking pillows in the corners to rebuilding the room (perhaps correctly) from scratch, incorporating a set of acoustic devices and parameters which sometimes seem nebulous but get a desired result. Because of this nebulosity of all the acoustics terminology (not to mention the international differences in measuring techniques, terminology, and 'scales', which are substantial) it is often difficult for an end user (and many audio professionals, for that matter) to be able to mentally visualize just what a room without standing waves will sound like, or a room which is so rolled off that the high frequencies seem to "fall to the floor".

To make matters even worse, a term like "soundproofing" is essentially a meaningless audio non sequitur; you would have to define how many dB, and at what frequencies... and what is the ambient noise level of the areas of interest to begin with? And do you mean sounds coming IN or sounds LEAVING an area? And so on.

So real room tuning is one entire entity, and then once the room is deemed to be as useable as it's going to get, then we enter the realm of SYSTEM TUNING. The big trick of course is getting the correct balance of all of these ducks in a row, so you have an end result you like.

Different frequency ranges have VERY differing interactions with a room, AND with the speakers.
At LOW freqs, 20 - 120 Hz or so, you have signals cooming from a sub or subs, and the all-too-real issue of standing waves and very distinct cancellations because of these standing waves.
At MEDIUM freqs, perhaps 125-2k or so, you have more issues with first order reflection.
At HIGHER freqs, perhaps 2k - 15k, you might have flutter/ reverberation issues.

Some people say they are going to put a sub in the corner because of "room gain". Another misnomer! There is no gain; there is no amplifier attached to the room! What is taking place is the corner of a room has the MOST EFFICIENT output coupling into the rest of the room at the lowest frequencies because the 2 walls and the floor are acting like 3 sides of a huge linear-sided horn at those large wavelengths. So it's not that the corner has any gain; it's that everywhere else in the room has apparent loss. The middle of each wall has the most apparent loss, because the sound leaves the driver, goes in all directions, reflects / folds (bounces) back on itself and partially cancels out. If you put the sub in the middle of a wall left and right and ALSO placed it in the middle of the wall floor-to-ceiling you would get essentially NO or very little bass in the room. Instead of thinking about it as "room gain", think about it correctly as "room loss". That will help to focus your thinking on where to best put the sub(s) for the best coupling back to your chair.

So you have the 3 options for sub placement:

1)  where you think it / they should go

2)  where your spouse tells you to put them

3)  where they ACTUALLY belong...

For some very enlightening articles about bass, room modes/nodes, standing waves, and room coupling, see Art Noxon's articles HERE . And for an in-depth listing of acousticians, acoustic materials, design/build companies, and so on, see my links page on the Boston Audio Society HERE.


So AFTER you have addressed the issue of room acoustics to the best of your ability, and this means you have decided if you have a 2-channel system, a Home Theater system, (perhaps both, even perhaps separate!) what your seating priorities might be, and the rest of your decor, you might have decided to make the sub placement a priority. Or not. IF YOU ARE ABLE, here is generally the best method: THE CRAWL-AROUND TEST. While it might seem funny or silly the end result compared to hours or days of computer analysis is usually spectacular. And often better.

The methodology is outlined on my test CD page, here: www.soundoctor.com/testcd. The crawl-around test has nothing to do with the rest of your system. What you are doing is coupling one or more subs back to your listening position based on the physics of the room. AFTER you have finished the test, you THEN match the subs with the rest of your system in the frequency crossover mode, and in phase and absolute timing mode.

If you DON'T couple the sub(s) to your listening position or area as well as they might be, you could be throwing away "a few" dB in coupling efficiency. If you are "throwing away" 3dB PER SUB you might as well not have bought the 2nd sub in the first place. Remember 3dB is twice the power, and 6dB is four times the power.

Most people who are NOT used to audio tend to equate 10dB (10 times the power) as "twice as loud", while engineers who are all too familiar with the financial issues of trying to make something louder have learned that 6dB is, in fact, twice (or half) the loudness, or Sound Pressure Level. Actually there IS NO SUCH THING as "twice as loud". Your brain and senses operate on a 20 log scale, and you should learn how that equates to real life. It's fun. Similarly, there IS NO SUCH THING as "50% louder". How do you learn this? Get a good SPL meter or app for you phone (check out SPLnFFT) and watch it all day, in a quiet room, in a loud room with audio blasting, at a party, in your car, etc. Get a feel of you YOU perceive the certain SPL's and how they relate to your spaces.

But back to reality: there is a place in life for subs connected almost ANY way, even where there's just another extra bass boom which impresses some people. To someone who only has experienced a cheap table radio or a the moral equivalent in any sort of surround system, ANY sub, even one poorly set up will "seem" like a revelation!


Do not fall into the trap of having a home theater receiver / processor with a "computer" inside and your JL Audio sub with it's ARO inside (or other fine sub) and think you are going to run these two computers and your life is gwanna be great: you might be in for a rude awakening. You will more than likely be like a person with 2 watches who is never really sure exactly what time it is...

Your room is at least a 5 dimensional system:
Height x Width x Depth; and Frequency and Time, which includes reflections and their subsequent cancellations.

Until there is a real holographic computer / Lidar correlation / deconvolving system which really can sample the room in a 3 dimensional grid (for example in 36 or 48 places) the best we can do right now it to attempt to approximate the net results in a room at a few (1, 2, 3, or 4) places. In SOME setups like this the results can be great. But here is where it sometimes falls apart: If the room is so bad that you really "need" a setup computer in the first place, it can't necessarily determine what is real, what is reflection, what is standing waves, and so on, and it simply won't work as you expect. Imagine trying to adjust a sound system in the aforementioned marble shower stall. You cannot fix or change the room reverberation or standing waves no matter what you do with a computer. Someday there will probably be computers powerful enough to do subtractive room decorrelation, and they will probably work by scanning the room with laser interferometers first, then build a 5 dimensional graphic of the room, (by then probably in n-dimensional space, but I digress) then correlate all the standing waves at all frequencies, calculate all the Rt60 times at all frequencies, then adjust the output of all the amps to decorrelate all this... (hear that, Darpa?) but don't hold your breath. It will initially probably be very, very sloppy.

My suggestion is to follow the necessary steps separately AND MANUALLY, and in the correct order; learn the equipment, and then experiment with ONE "computer" at a time (I would suggest the JL Audio ARO / DARO first) and determine if it helps you. If not, try something else. The only way you can determine if something works is to make one change at a time. Remember, the JL Audio ARO / DARO does NOT correct issues in the time domain. It only attempts to frequency anomalies and smooth that out. You should make every attempt to correct the overall timing FIRST. And if you have two JL Audio subs (or more) and have followed the rest of the procedures, then I suggest NEVER running them in master/slave, because the ARO / DARO results from each placed in the room will be different. They can only be the same if the room is TRULY flawlessly symmetrical or if the subs are RIGHT NEXT to each other. Even if one sub is on tho of the other sub it will be different because the top sub is now coupling modally different into the room.

Be aware that on the JL Audio series of Fathom and Gotham subs, the V1 ARO uses one band of determined EQ freq and attenuation and the V2 DARO uses 18 bands of 1/6 octave EQ and attenuation.


Some people incorrectly use a "Y cord" to feed both inputs of a sub. This is or should be completely unnecessary; all it does is the same thing as turning up the level on the sub (or the send level from the receiver/processor) +6dB. And if you happen to have TWO subs you should actually wind up turning each one down 3dB, so you wind up with the correct resultant level in the room and you will have gained 3 dB of HEADROOM in each sub. If you were to leave each volume at its reference level you might find that it's easier to turn DOWN the SUBWOOFER LEVEL in the setup menu of your Home Theater receiver/processor.


On the JL audio subs, the ELF trim is an equalizer operating in the 25 Hz region which can compensate for the [apparent] bass buildup if you are placing the sub in the corner. (See the paragraphs on room acoustics, above) Typically IF you placed the sub in the corner you might want to turn the control down. If for some reason you place the sub at the middle of a wall or in another less than desirable position, you can add 3dB. Remember 3dB is using twice the power!

Some receivers/processors have THX and other proprietary settings for "boundary" effects, and these are similar to the ELF trim on the JL Audio subs.

A further discussion includes crossovers, whether passive, active, tube, solid state, analog, digital, balanced or unbalanced; and proper methodology of both measuring and correcting the inherent group delays in modern equipment to fine tune the impulse response. We're getting to that !


So now let's examine the aforementioned group delay. It takes time for a signal to go through a circuit. Inasmuch as everyone thinks electricity travels at the speed of light, that's not quite true. Electrons going through a wire, which we can call a transmission line are slowed down by a certain amount. For some types of cables this is called the velocity factor, and it's typically 66% of the speed of light. (Not that that's slow!) It also takes a certain amount of time for the signals to get through each piece of equipment, although relative to other human events, this is quite fast: it might take 5-50 microseconds for the signal to go through a power amp, because there are no mechanical devices in the way. Once we get a signal into a mechanical device such as a speaker, whether it is passive or active, we now have the sum total of all the electrical plus mechanical phenomena to take into account. The typical group delay through a modern, sealed box subwoofer, is perhaps 8 to 15 msec. That's milliseconds, not microseconds.

In the digital world, delay issues are often called latency. Specifically this refers to some circuitry where the signal starts out as analog, goes through an A:D converter (not an A/D converter as incorrectly stated in much literature; it's all math and it's a ratio, not a division... but I digress even further...) then gets processed digitally in some fashion, then goes through a D:A converter, and then we hear it as an analog signal. This is a HUGE issue with modern recording studios and live venue "digital" mixing boards and everyone is continually fighting against seemingly impossible odds...sometimes there is so much latency when devices are used in series with each other that the musicians hear themselves as an echo and this makes it nearly impossible to play. The entire premise of the "convenience" and "power" of "digital" is sometimes negated by these latency issues and the difficulties in "fixing" them.

This is also an issue inside Home Theater receivers/processors, where the purely digital HDMI signal is stripped apart and reconverted back to analog. Collectively, this mess is partially responsible for instances where the picture and sound are "out of sync" in modern equipment. Since you CAN'T get rid of the delay, the only answer is to delay something else so it all "matches up" in the end. In the analog world it still takes time for a signal to go through a circuit, and although the phenomena should probably be called transit time, group delay is what has stuck; a holdover from the early telephony days, when the concern was the delay of the audio frequencies, not the DC control or bell ringing signals (all carried on the same lines), and the term meant a "group" of frequencies we were concerned about.

Let's start with a 2-channel (stereo) setup and look at this block diagram:

Fig 8. The same signal applied to both the main power amp and the sub are delayed going through the sub.
As shown, the delay of the sub would be 1 wavelength at 80 Hz, or 12.5 msec.

Fig 8. shows THE INCORRECT METHOD many people use when connecting a sub. It pains me to even have to use this diagram. NO crossover is shown. The full range signal goes through the power amp and into the mains; and the full range signal goes into the sub, where the sub's own LOW PASS / HIGH CUT filter is engaged.

Here's the clincher: since the sub is always at least 8, 9, 10, 11 msec late, the phase relationship CAN NEVER be correct. It can be corrected in one of 2 ways only: you can use some electronic means to ADD the same amount of delay to the top (mains); or you can move the sub(s) closer to your body the correct number of msec. You CANNOT match the phase of the sub to the mains because you CANNOT use the phase control on any sub to remove delay; you can ONLY ADD DELAY.


Crossovers are always a slippery issue. Many 'audiophile' dealers don't necessarily sell them because (go ahead: squirm) they don't really understand them, and they require a lot of handholding therefore they can't make any money on them... and most speaker manufacturers won't admit or suggest that their speakers need a sub because they don't (or may not) make a sub; therefore they port their speakers in an attempt to get extra "free" bass and therefore the coupling and delay timing issue is made ever so much more complicated. Many customers that I talk to simply buy a sub (or two) parallel ("Y") the output of their preamp into the main amp and the sub, and are then unhappy with the results. They think that because their speakers go down to 38 Hz that they ONLY want to use the sub between 20 and 40 Hz... it simply doesn't work like that, because of the incorrect port, and the fact that the sub is simply not matched to the mains. The results are muddy, indistinct bass, and users who incorrectly attempt this setup often wrongly blame the sub.

One brief word about all the terms being bandied about: yes, a LOW CUT and a HIGH PASS are the same thing. It is MOST USEFUL to use the terminology so it fits the use of the situation. In one example, we have a filter in a recording studio Microphone Preamp. Of course WE KNOW THE AUDIO GOES "THROUGH" the thing; what we want to know is what we are doing - what "change" we are going to hear when we click the switch! We are CUTTING THE LOWS. In this instance the correct terminology is LOW CUT FILTER. In the case of "filtering" a signal that's going to our mains, yes, of course we are "letting the highs through" and we are also "blocking the lows". So the typical useage for this would be "HIGH PASS" filter. Technically and mathematically, either is correct. But it's always a good idea to use the term which will yield the least confusion, especially where people are concerned who don't necessarily have audio as a first language. Manufacturers, pay attention...

Be aware that there is very annoying current marketing/sales term where some manufacturers say there is a "crossover" in the sub and it is only a low pass filter. THERE IS NO HIGH FREQ OUT to go back to your amp, so it is a lie, plain and simple. There ARE, however, subs with real crossovers in them. For example the JL Audio E subs HAVE a real crossover in them, with HF OUTPUTS which then go back to your power amp. The Fathom series does not; it has a low pass filter.

Some audiophiles don't want to introduce yet another active "thing" in their precious signal path, not realizing that adding the crossover is very much the lesser of two evils.

Actually adding a crossover is really a WIN-WIN situation:

WIN # 1)  Since you are now NOT putting in 20 Hz - 80 Hz into the mains you are not using up the available LF cone movement with bass, so the LF cone in your mains is able to play its higher freqs (up to IT'S crossover point) much more cleanly. You get an apparent 6dB or more dynamic range. You can play your system LOUDER, and also with less compression distortion in the LF driver when you're having that Saturday night dance party and you're playing urban bass technopop at 110+ dB. Really.

WIN # 2)  Since you are not putting bass into that same driver you are not Doppler modulating everything between 80 and 600, or whatever the next crossover point is. This means cleaner mids. By far.

WIN #3)  You are not sucking current out of your main power amp at low frequencies, so there is more current reserve to play those highs louder...

WIN # 4)  Since the cones aren't moving as far at the low freqs the driver itself is not generating as much back EMF therefore the damping factor and all of its issues are greatly negated. And you don't need to run silver plated cold water pipes to your mains as speaker wires because there is less current draw by the speakers.

WIN # 5)  Freqs below 80 are now NOT causing transient intermodulation distortion with the higher freqs (and vice versa) in your power amp. Cleaner still.

So let's start with the simplest method: a passive "filter" that blocks below 80 Hz from going to your "mains", and PASSES the highs to your mains:

Fig 9. Here's the Marchand XM46SB PASSIVE Filter.

Here's how it's connected to a typical 2-channel system.

Fig 10. The passive filter used as "half" the crossover

So you roll off the mains at some frequency, such as 80 or 90 Hz, 24 dB/octave; (you have to purchase the frequency you want, since it is custom made, and I HIGHLY SUGGEST 90 Hz, 24 dB/octave) and you set the low pass filter in the sub the same way. If you want (for some reason) to only use passive capacitors and inductors in your system, this is one answer. Overall I do not necessarily recommend this though. More modern solutions are FAR better. I only want to show the option. Please be aware that your precious audio signal has gone through MANY THOUSANDS of opamps from the microphone through a myriad of 'stuff' in and out of computers or tape recorders or both, and then THAT signal has gone through further opamps in the mastering process, and so on. Using an active filter with a few more opamps is not going to destroy your 'precious' audio.

To use an active filter (if it has 2 or more sections we can now call it a CROSSOVER), there are many choices some of which are each explained below.

There are the Bryston crossovers, very handsome, built like a tank, and with a terrific warranty... except the ordering options are quite complicated and many people wind up getting the 10b 'standard' (which does NOT have a 24 dB/octave setting) when the better choice would be the 10b SUB, or LR. Then the crossover winds up on Audiogon or Audiomart, because the user is frustrated. If you order the SUB version or the third LR version then you must order separate plug-in parts for different frequencies. Be very careful reading their very complex user manual.

Many versions of the Marchand (solid state, tube, balanced, unbalanced, 1 way, 2 way, rotary knob, precision stepped attenuator...) are available.

Fig 11. One variation of a Marchand crossover (XM9) showing stepped volume control knobs.

Another choice for simpler experimentation and budgetary concerns is the dbx 223 series, here: https://dbxpro.com/en-US/product_families/crossovers

Fig 12. The dbx 223 xs crossover

Note the dbx has separate models for use with XLR or Phone/RCA connectors. BOTH models are balanced (but may be wired unbalanced) - only the connectors are different. If you are intending to use UNbalanced RCA's then you must get these RCA to 1/4" TS (Tip/Sleeve) adapters: (you will need SIX).

Fig 13. RCA to 1/4" Tip/Sleeve adapter

Here is the dirt simple front panel setup for the dbx units:

Fig 14. Dirt simple dbx XO setup for 90 Hz

Putting the active, 2-way crossover in your system is done like this:

Fig 15. Showing a typical active crossover in a "typical" system

Since ALL the filtering is done IN the crossover, you turn OFF the [low pass] filter in the sub. For fine level matching adjustments you typically have HIGH and LOW output knobs on the crossover to play with.

There's also, at the higher end, the Pass Labs XVR1

Here is the very best cut-to-the-chase analog answer: The JL Audio CR1 Crossover. It is VERY comprehensive and the cleanest device there is. It was my concept/idea while at JL Audio and it took the amazing engineering department and I nearly 4 years to finalize the design, development, and production.

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Fig 16. The JL Audio CR1 Crossover. (Click on any of the pix for a larger page)

Here are some of its UNIQUE features.

1) In the crossover mode, you can use either/and/or RCA, Balanced XLR, or balanced or unbalanced 1/4" (Tip/Sleeve Tip/Ring/Sleeve) input or output connectors at any time.

In the [hard] bypass mode (see the color diagram above) of course the RCA input connectors connect directly to the RCA output connectors, and the XLR input connectors connect directly to the XLR output connectors.

2) The frequency controls are SEPARATE - that is, you can set them the same, or overlap or underlap to accomodate ANY preferences you might have.

3) The Bypass/On switch will give you the PERFECT A/B comparison: in the "on"/operate mode the sub is crossed over (as it should be); and so are the mains. In the bypass mode, the mains are operating full range and there is no sub, so you can FULLY and IMMEDIATELY appreciate exactly the benefits of what the crossover (and sub) is doing in your room. NO OTHER DEVICE is capable of this - and it is this A/B demo feature that blows everyone's mind.

4) The bypass switch may also be used for Home Theater bypass.

5) Separate HF and LF Damping controls give you subtle and desirable control. This becomes the final "I gotta add salt and pepper to the chef's creation" buttons, because everyone really wants one more knob to turn! Or not.

6) The "balance" control enables you to have less or more sub or mains for late night listening, or to assist with mixes which could use a bit of "help". It has a zero reference detent in the middle to easily return exactly to normal.

7) MUTE buttons enable you to discern anything - such as muting the mains to hear if the subs are vibrating something in your room, etc. The separate L and R mains (satellite) and sub mute switches also greatly assist in setup using my unique TEST CD, here: www.soundoctor.com/testcd

8) yes, you can have separated "stereo" subs if you want.

9) Note for you pristine analog fanatics: There is NO "digital audio" in the CR1. The audio path is completely analog.

...and there's much more! Here's the User Manual 3Mb.

What I have determined is that sometimes, a customer might be reluctant to purchase such a device as the JL CR1. But here's the easy path. First get something very simple (and inexpensive) such as the dbx. Experiment with it for a bit! Once you learn the benefits of correctly applying a crossover to your system, you can sell the dbx in a heartbeat and get the CR1 you really want and deserve!


Some people WANT to get more detailed and be VERY involved with complex and comprehensive setups, and want to turn into an engineer. Not everyone does. But in case you do, you can do everything yourself with a computer, test microphones, and products like the ones below. You'll never want to come out of your room, your spouse (ex-spouse, by now) or various buddies will have to throw in cold pizza, warm coke, and an occasional piece of raw meat into your room, but as a now devoted for life audiophile engineer, you WILL be able to control the world! Onward!

There's REW (Room EQ Wizard): www.roomeqwizard.com , and there is also integration with the Roon player, here: http://blog.roonlabs.com/digital-room-correction .

The DEQX models are here  www.deqx.com/products

The DATASAT  www.datasatdigital.com

DIRAC is here: www.dirac.com

...and the MiniDSP collection of products, which include Dirac   www.minidsp.com

There's SONARWORKS  www.sonarworks.com

and TRINNOV  www.trinnov.com

The AudioVero products include ACOURATE and The ACOURATE CONVOLVER, here :www.audiovero.de/en/

The JUICE Audiolens is here  www.juicehifi.com

I have posted many more links HERE: www.bostonaudiosociety.org/links3.htm#SOFTWARE


There are two other annoying problems that have to do with so-called "Integrated Amps". MOST of them sadly do not have a method for connecting a crossover between the PREAMP OUT and the POWER AMP IN. The terrific Bryston B135 (and also their B60r) DO HAVE the availability to correctly do this. A few (very few) other brands do have this feature.

Fig 17. The Bryston B135 Integrated Amp with PRE OUT - PWR AMP IN connectivity

Then there's the issue of all the other Integrated Amps which do NOT have the abovementioned loop, but almost all of them DO have a "Tape Loop" - a holdover from the days when people connected cassette decks... So the issue there is the TAPE OUTPUT is at a full, fixed level, taken off at an earlier stage, before the volume control. So you CANNOT connect a regular crossover in that loop because the low freq outs of the crossover will be at a fixed level (full), while the high freqs only will be adjustable. So for the over a thousand (!) people who contacted me to ask about this, there is really only ONE WAY to accomplish the nearly impossible: The Marchand XM9 or XM44 crossovers, which have SEPARATE LOW and HIGH frequency level controls. And for wonderful convenience, these are available with precision stepped attenuators with repeatable click positions.

This is what the front panel with the separate click stepped level controls looks like.

Fig 18. One variation of a Marchand crossover (XM9) showing stepped volume control knobs.

This is how you would connect it - you MUST use a crossover with separate high out and low out level controls. The issue is if you try to use a crossover with "regular" non-stepped controls, (that means a plain old potentiometer) you will be frustrated trying to always match and fine tune the levels.

Fig 19. Connecting a Marchand crossover in the TAPE LOOP

Now you set the maximum level you want in the room by putting the four crossover levels all the way up (L high, L low, R high, R low) and adjusting the main volume control on the amp. Then you use the 4 controls on the crossover to attenuate to the volume in the room to what you want. This the ONLY way to accomplish this if you have a tape loop only and can't insert a crossover between the premp out and power amp in.


So all of this crossover setup so far seems moderately easy (you just... plug it in...) and yet with ANY of the passive or active crossovers we have not YET addressed the critical issue of the group delay in the sub. SO even though we have made everything lovely in the frequency domain, the INHERENT delay in the sub is still there. What are our options?

We CANNOT change (or fix) the inherent / intrinsic group delay in modern subs. That leaves us with two choices IF WE ARE INTENDING TO BE FANATIC !

OPTION 1)  We can move the sub closer to our body about 9-10 feet or so, and then use the phase control on the sub itself to fine tune the match. This is not necessarily as crazy as it sounds. We do this successfully in studios all the time. Of course this might not work in your particular room.

OPTION 2)  We must introduce an equivalent delay TO THE TOP (mains) to match the inherent delay in the sub; then we can super fine-tune the match by using the phase knob on the sub.

Some notes about phase knobs: If you have a (toggle) SWITCH on a sub labelled "phase" that is wrong. It is not phase; it is POLARITY. Phase is ANY NUMBER of degrees shifted, from 1 degree to 360 degrees to 3600 degrees and so on. Polarity is either 0 degrees or 180 degrees, period. (see Fig.1 and Fig.3 above)  If you have a phase KNOB on a sub, the circuit is usually designed to only ADD DELAY. You cannot take away the inherent delay in the entire electro-mechanical physics of the sub, but you can ADD further electrical delay. Some subs are calibrated in electrical degrees of waveform at 80 hz, because 80 hz was the original suggested crossover freq for Home Theater/Surround Sound systems. Therefore IF the knob says 180 degrees it is actually adding 6.25 msec of delay to the sub signal; this is the equivalent of moving the sub 7 feet FURTHER AWAY.

So how do we add delay to the "top"? We would have to introduce a real processor to do that. The options are a device like the DEQX, the Lyngdorf, or the Mcintosh version, here. (see a longer list above) All of these are audiophile grade devices. That means that UNLIKE all the "digital" speaker gadgets intended for use in nightclubs and rock n roll systems, these do NOT operate at 44kHz, (or even 48kHz) and you will NOT be disappointed with what the "processing" has done to your precious highs. Many of the so-called "professional" units are perfectly suitable for a noisy bar or a rock touring PA system but you might be very disappointed if it is your intent to use them in a critical audiophile listening/monitoring situation. That means beware of $99 - $299 processors. But even if you DO get a very inexpensive processor, say on ebay or Craigslist, etc., the learning experience is well worth it, if you have the patience.

In the instance of Home Theater processors, there is an easy method. We can take advantage of the somewhat flawed concept of "speaker distance settings" to perfectly fix the sub timing issues. Simply set ALL the top speakers ( L C R Ls Rs) to 7 feet where they belong, and set the sub distance to 18-19-20 feet. Now, because all consumer equipment operates backwards (!!!) you are introducing 10-12 msec delay TO ALL THE TOP SPEAKERS. Now you can fine tune the phase control on the sub to add a bit more delay to the sub to perfectly match the mains and the results should be spectacular. My test CD and the two different procedures to accomplish this are all carefully explained here: www.soundoctor.com/testcd.

Once you correctly place the sub(s) in your room so they correctly couple to your desired area, cross over the mains to the sub correctly, and fix the timing issue your results will be everything you hoped for.

IN SUMMATION  (pun intended... )

I am therefore not suggesting that everyone force themself to be so fanatic an audiophile, or to necessarily get this crazy when setting up a sub. But I AM SUGGESTING that you should know ALL the possible options and then you can decide just what is best for your particular situation. Is it overwhelming? Yes. Is it a lot of work? Yes. Did you just spend "a lot" of money on a subwoofer or two and expect bang for the buck? Yes. Is it going to adjust itself? Sorry, no.


Even if you CAN'T get the timing of your sub to match your mains as closely as it can be done, there IS a saving grace: re-read the paragraph above about using 2 subs. Notice that humans actually LIKE the (slight) fattening up of the bass loudness envelope in time. Therefore even IF your sub is 12 msec late, and you are one wavelength off, as long as you get that delayed wavelength to line up with the bass coming out of your mains, your frequency response will be pretty good and you won't have any awful objections, again, assuming you get as much else right as possible.

Enjoy your audio journey! And let me know your results!

There is no plug-in for experience...
SOUNDOCTOR                  BARRY OBER                 EMAIL: barry@soundoctor.com