Almost Everything You Ever Wanted To Know About Adding A Subwoofer
This article was originally written by Barry Ober, also known as The Soundoctor. There have been minor changes to his original text for clarity. Additionally, some sections were updated to reflect new products in the JL Audio lineup that do not have the same feature set as those mentioned in the original article.
Adding A Subwoofer
Whether you are adding a subwoofer to an existing 2-channel “stereo” system, or building a “surround sound” home theater system, let us 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 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, etc., 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 subwoofer to system is to define everything above the subwoofers 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 subwoofer 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 Hertz signal coming from both the main speakers and the subwoofer(s), and if the subwoofer is 12.5 milliseconds late the two sources will “seem” to be in phase but the subwoofer really will be 360 degrees late. It is the impulse smearing that this affects, but people do not 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 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, 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, (quick history here, a very quick technical blurb here, and the 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 can be, or want to be, bothered with.
So the overall view of adding a subwoofer is this: In essence you are designing and assembling a new speaker system which is effectively a two-way system; the subwoofer is one way and everything else above it in frequency is the second way.
Simply connecting a subwoofer to existing main 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 main speakers affecting the low frequency 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 does not really matter.
Indeed the only thing that does matter is an individual’s happiness with their system, regardless if anyone else thinks it is “right” or “wrong”.
But 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” (5.1, 6.1, 7.1, etc.) systems. For more background on surround sound, you might want to look here.
With home theater, there is relatively little phase correlation between the LFE channel (the true Low Frequency Effects channel) contained on the DVD and all the frequencies above 80 Hertz; all that is really essential for the low frequency part of movie enjoyment is the best coupling of the below 80 Hertz effects to the room and indeed the listening position. There is also the rest of the bass; all the below 80 Hertz information from all five channels that is stripped off and summed together into mono and sent out the subwoofer out connector on every modern home theater receiver. That plus the LFE channel (if it in fact exists on that particular DVD, and it may not) constitutes all the bass managed bass. Therefore the most desirable scenario in a home theater situation is to best couple the subwoofer(s) to the room first and then timing and phase match the subwoofer(s) to the rest of the system.
But let us back up one step. 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.
Impulse response is the holy grail of all of audio. With more pristine two-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 Hertz), its subharmonic (can octave lower; 25–30 Hertz), and its mostly odd order harmonic structure (all the way up to 8 kilohertz 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 main speakers have already correctly addressed the issues of both impulse response and frequency response. So for the purposes of this discussion we will assume that whatever your main speakers are, from a two-way bookshelf to an 8-foot tall floor-stander monster, that within the desired passband of the main speakers the impulse response and frequency response are already correctly handled.
Then there is 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 is an impulse. It is 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 that 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 (now out of print). Imagine we are playing back a well recorded cello; we have the fundamentals of both the strings and the wood starting in the subwoofer (that means below 80 Hertz, and you may wish to refer to a frequency-wavelength chart) 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 three-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 us examine the acoustic “spaces” we might be dealing with. There are three useful separate sizes of acoustic spaces in life:
The inside of a car, where you are essentially living inside the speaker cabinet (the pressure zone).
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.
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 Hertz. You cannot not have these issues in a room this size unless you have a real acoustically treated room with full size, perhaps 32-foot bass traps in the walls and all the correct ratios of absorption versus diffusion especially at low frequencies. This does not mean a couple of pillows in the corners or ineffectual 400 Hertz 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 low frequency 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 two low frequency sources of differing phase relationships and/or timing relationships they will cancel. Period. And if they are “seemingly 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. It is not as “clear” as you think it should be, and it is not as focused as you think it should be. The issue is only timing.
We can call the red and green waves signals from two separate “speakers”, two separate subwoofers, or a subwoofer and a main speaker. Here are the diagrams that show this:
Figure 1: Obviously “in phase”
Figure 2: 90 degrees “out of phase” (the red wave is lagging the green wave by 90 degrees)
Figure 3: 180 degrees out of phase (the net result is complete cancellation)
Figure 4: An example of group delay. This only shows one cycle of many. It is 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 (three complete cycles) etc. shifted in time out of phase.
Figure 5: Group delay drawn another way. The green wave might be coming out of your main speakers. The red wave is coming out of your subwoofer. Notice how at first they appear as if they are in phase but the red wave (from the subwoofer) is actually a full wavelength late.
How did the subwoofer get to be 360 or more degrees late? It is the overall physics of how it is built. The only correct way to implement a subwoofer 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 subwoofers, JL Audio included, use massive drivers which have a relatively large Xmax (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 subwoofer is already 8–10 millisecond late and it is placed in the corners further away than the main speakers (just for example) then relative to the main speakers it might be 12–16 millisecond late. You cannot take this delay away.
You might at this point enjoy referring to this handy Frequency-Wavelength-Period chart here.
Types Of Main Speakers
In addition to all the above, there is the complex issue of the main speaker you are coupling to. There are essentially six types of speakers that exist:
Port in the front
Port in the bottom
Port in the back
Each of these speaker types couples somewhat differently to the room and certainly to a subwoofer. A port is always nothing more than a method to attempt to get free bass out of an enclosure and/or driver that is too small. It is 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 frequencies.
At one level all the guys want 9-foot speakers in the living room (read: “man-cave”). All spouses, of whatever gender, want tiny 3-inch speaker cubes that disappear but expect 9-foot results from them. Since they have not repealed Ohm’s law or any other laws of physics while we were sleeping, the only way to get correct and good sound is to move a correct amount of air!
Let us examine ported speakers. We will start with the worst case: the port in the front. At mid bass frequencies, approximately 50–80 Hertz, the low frequency driver moves in the cabinet, the air in the cabinet is elastic, the air compresses, and the port air moves out of the cabinet. Because of the frequency at which the cone is moving, and the elasticity of the air, by the time the cone moves back out 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”. Note that because of this delay the entire port concept will always smear the impulse response.
As you go lower and lower in frequency, at some low frequency the air pressure from the low frequency 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. A point slightly above this defines the –3dB “cutoff” point of the cabinet in question. When the manufacturer of a speaker cabinet defines the frequency response (for example: 37 Hertz–20 kilohertz +/- 4dB) this is what is defined by the entire arrangement of the port and the air in the cabinet and the driver. It is a system.
You must understand that any driver goes down to 0 Hertz, or DC. If you put a battery across a speaker, the cone moves and stays there. If you were to have a DC coupled power amp feeding a speaker, any speaker, from a 1-inch dome tweeter to an 18-inch rock n roll stage bass driver and you put 4 Hertz into it, it would simply move back and forth at 4 Hertz. Of course in order to actually “hear” the audio it would have to be in the generally accepted passband of 20–20,000 Hertz 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, (and therefore the air back pressure) and many other factors that determines the overall response of that speaker system as an entity.
That means if you were to simply put those same frequencies through the main speakers and the subwoofer (with no crossover; and this is the mistake that nearly everyone makes) you would now have three sources of low frequency energy and differing phase: the main speakers’ low frequency driver, the port, and the subwoofer, all fighting with each other, especially in the time domain. A further corollary is that since the air inside the main speaker cabinet is elastic, the phase relationship of the port air to the driver air is also a sliding one; that means it is “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 “more” 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 low frequency wave might be “in phase” with the front driver but be 360 or even 720 or even 1080 (or more) degrees late; therefore it sounds like the bass frequencies are ok in the frequency domain but the impulse response is now further 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 energy, and at some frequencies you are canceling or reducing energy. You have simply made a physical/mechanical frequency comb filter out of your room that you cannot 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 S.A.L.L.I.E.. This is, essentially, an absorber which absorbs the entire back wave output of the electrostatic panel. Since now there is no comb filtering all, all you are hearing is the front signal and it sounds “cleaner”.
A ported subwoofer 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: each of the low frequency drivers in your two main speakers, each of their ports, the subwoofer driver, and its port. In addition, since it is a bandpass device it cannot go down low enough for serious home theater effects.
In some cases, such as a bandpass subwoofer used in a night club or on stage, you are most concerned with efficiency and directionality and not with getting frequency response “flat” down to 20 Hertz; therefore a correctly set up bandpass box that might roll off at 34 to 40 Hertz 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.
But back to our home / HI-FI / 2-channel / Audiophile / Surround Sound systems: There is only one truly correct way to “add a subwoofer” to a system in a controlled listening room situation: you must correctly cross over the two sealed cabinets (in the frequency domain) ; and their timing (the phase domain) must be correct. Any other method will lessen the focus and clarity you have tried so hard to preserve.
There are many people with extremely exotic high-end two-channel systems that are all chasing the holy grail of 3D holographic sound imaging, and until they follow these distinct guidelines they are never completely satisfied with the results.
A similar situation exists with home theater (e.g. “surround“) setups where the customer thinks that the front speakers are ”full range“. Even so, the best approach is to seal the ports, operate the 5 channels as ”small", crossover at 80 (or even a bit higher, but never lower) and correct the timing issues inherent in all modern subwoofers 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 subwoofer to 12 feet more (i.e. 19 feet) and then use the variable phase control on the subwoofer to fine tune the relationship at the 80 Hertz crossover point.
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 us to tell us that for the first time they are finally satisfied, in fact thrilled, with the incredible integration of their JL Audio subwoofers. All of this discussion barely scratches the surface of the true complexity involved in flawless integration, so let’s continue.
The Recording Process
On top of all the previous variables we have all the issues and errors inherent in the recording process. It is simply laughable (and pathetic) when I read articles where the “soundstage” of a rock recording is described as “palpable”. Sorry, but every modern rock recording made in the last 40 years is composed of a series of panned mono signals that have absolutely no “depth”. They are each separately sent to an echo/reverb device, the returns of which are usually / often (but not always) panned full left and right.
The summation of all the L-R panning placement and the summation of all the reverb returns fools you into thinking there is a “soundstage”. Sadly, precious few recordings are made with any regard to true stereo or binaural 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 determines is a suitable balance, perhaps between a soloist and the rest of the orchestra.
Yes, there are precious few companies who do pay attention to this; AIX records is one. But to think that any modern, commercial pop recording mix has any true acoustic space is, for the most part, sadly mistaken.
Oh, and to touch upon “stereo bass” for a moment… there almost is no such thing. Going back to vinyl, every stereo record cut in the last 70 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 every cutting lathe on the planet has a “compatalyzer” circuit, that dumps frequencies below 160 Hertz 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 what’s the point?) to make use of those sort of tricks. There are some trance / psychedelia / electronica dance music releases where there are bass tracks where there is stereo bass in the form of something like 24 Hertz in one channel and 24.2 Hertz in the other channel; therefore you get an air pressure differential which travels around the room. Cool! In the above example, the “traveling wave” would take 5 seconds to go back and forth around the room. If you are 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. 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 no standing waves, and so your ears (and indeed your whole psychoacoustic receptive system) can process and differentiate 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 subwoofers were more near-field (the pressure zone). In this instance, the closer you put the subwoofers to your body the more you are in the pressure zone, and therefore the less the room itself enters into the equation.
There is a discussion of this in a studio environment, with pictures, here.
About Using Two Or More Subwoofers
So most people’s reasons for multiple subwoofers in a room is “more even coverage”. Let’s examine the instances of multiple subwoofers and what they do. One interesting issue with using multiple subwoofers concerns arrival times.
Here is a hypothetical situation. You are feeding the same signal to two subwoofers. So this begs the question what is your room like? Is it symmetrical? L-shaped? A closed room? A Huge open space? Notice we’re back to acoustics?
Referring to Figure 6, the subwoofers are also equidistant from your body. So the subwoofers each couple to the room however they do. The whole setup is essentially symmetrical.
Now here is another example. Refer to Figure 7. We are still putting essentially the same signal into both subwoofers.
Figure 6: A symmetrical layout
Figure 7: An asymmetrical layout
There might be three ways to do this:
From the left and right of a stereo preamp ( much more on this later)
Using a “Y” cord from the bass managed output from a home theater receiver.
In the case of some JL Audio subwoofers, it might be a Master/Slave setup. (This is not always suggested, because you cannot then have individual control over the separate ARO, should the model have this feature)
The point is, the sound leaves both subwoofers at exactly the same time. Notice in Figure 7 the right subwoofer is closer to your face. Perhaps the left subwoofer is 11 feet away and the right subwoofer is 4 feet away. That is a 7 millisecond differential. So you hear the leading edge of the bass wave from the right subwoofer, then 7 millisecond later the leading edge of the left subwoofer… then the note dies away from the right subwoofer and then 7 millisecond later the note dies away from the left subwoofer.
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 subwoofers 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 subwoofer is its own entity coupling into the room with slightly differing standing waves.
So now we have two ways to view the multiple subwoofer 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 subwoofers, that have variable phase adjustments, my suggestion in setups like this is to use either method shown above (Figure 6 or Figure 7), and then adjust the phase control knob on each subwoofer for most accurate transitioning at the crossover frequency. It is slightly tricky, but you will keep the real phase relationship between each subwoofer and the main speakers, and you will keep the arrival time differential that you “love”.
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 110 Hertz 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 subwoofer 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 with surprising precision, and in my years of night club building we used to adjust the steering of arrayed subwoofers so that the bass was correct on the dance floor and much less off the dance floor in the corners of the club.
Understand that the largest percentage (I place it at 70%) of all audio issues is room acoustics. You cannot put a great speaker in a marble shower stall and expect it to sound good. (Do not try this at home. If you do, do not turn on the water.)
Room acoustics itself is a 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 the nebulosity of all the acoustics terminology (not to mention the international linguistic and technospeak differences, 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 an audio non sequitur; you would have to define how many dB, and at what frequencies… and what is the ambient noise level of the area of interest? And so on. So real room tuning is one entire entity, and then once the room is deemed to be as useable as it is 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 items in a row, so you have an end result you like.
Some people say they are going to put a subwoofer 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 coupling at the lowest frequencies because the two walls and the floor are acting like three (3) sides of a horn at those large wavelengths. So it is not that the corner has any gain; it is 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, folds back on itself and cancels out. If you put the subwoofer 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 no bass in the room.
So you have the three options for subwoofer placement:
- Wherever you can, or wherever your spouse tells you to put it.
- Where it is mathematically correct to couple to the most applicable part of the room.
- By doing the crawl-around test and matching up the subwoofer coupling into the room inversely the best to your chair.
For some very enlightening articles about bass, room modes/nodes, standing waves, and room coupling, see Art Noxon’s articles 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 subwoofer 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.
An outline of this methodology as well as a great test CD is available here.
The crawl-around test has nothing to do with the rest of your system. What you are doing is figuring out how to couple one or more subwoofers back to your listening position based on the physics of the room. After you have finished the test, you then match the subwoofers with the rest of your system in the frequency domain (crossover), and in the time domain (phase) mode.
If you do not couple the subwoofer(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 subwoofer you might as well not have bought the second subwoofer 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 is fun.
But back to reality; there is a place in life for subwoofers connected almost any way, where there is just another extra bass boom which impresses some people. To someone who only has experienced a cheap table radio or the moral equivalent in any sort of surround system, any subwoofer, even one poorly set up will “seem” like a revelation.
With JL Audio subwoofers there is a further magic which should not be overlooked. JL Audio offers several single woofer subwoofer systems. The E-sub is available as an e110 and the e112. The Fathoms subwoofers are available as floor-standing models; the f110, f112 and f113 as well as our Fathom IWS in wall subwoofer systems. For the E-sub and floor-standing Fathom products, the woofer size is indicated by the last two digits and the number of drivers is indicated by the first digit. So the “f112” is a Fathom with a single 12-inch subwoofer driver. We also offer two rather special dual woofer subwoofer systems. In the Fathom line we have the f212 and we also offer the Gotham g213. As you might guess, both of these models have dual drivers mounted vertically (one over the other, not side to side).
Now imagine you had a pair of f112’s. You could put them in the corners, 29% down the long wall as Art Noxon suggests or where the crawl-around test tells you to put them and so on. But notice in every one of these instances they are on the floor! You have not taken advantage of the height placement which changes the vertical modal coupling to the room! This is the magic of the f212 and g213; not that they are “just” a pair of f112’s or f113’s, but that the inherent design means the lower driver being near the floor will couple the lower frequencies (LFE noises in movies: 22–35 Hertz) better to the room, while the higher driver (at approximately 29% of the room height) couples the higher bass music frequencies (35–70 Hertz) tighter to your body. This is where the coupling ability of these subwoofers across a wide band of frequencies shows its magic! And if you own two single subwoofers, such as the f110, the f112 or the f113, you can do a coherent test yourself to determine the perfect modal room coupling for your particular setup. It is work but it is easy. Go to a home improvement store (Lowes, Home Depot, etc) and get four 8-inch cube cement blocks they are about a dollar. Now once you have found the best lateral placement for your room by doing the crawl-around test, you can experiment with vertical mode coupling by raising the subwoofer 8 inches at a time and learning the result. You may find that for your room and the type of program material that you mostly play, that this last step in fine tuning really makes your system perfect. Notice that there are some manufacturers such as Acoustic Sciences who make subwoofer stands; you should have at least 8-inches in order to make a difference in vertical modal coupling.
Audiophiles? Computerized Room Setup?
Do not fall into the trap of having a home theater receiver / processor with a “computer” inside and any JL Audio subwoofer with it’s ARO (Automatic Room Optimization) inside 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 two watches who is never really sure exactly what time it is…
Until there is a real holographic computer system which can sample the room in a three dimensional grid (for example in 36 or 48 places) and build a ray-traced image in our computer… the best we can do right now is 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 cannot necessarily determine what is real, what is reflection, what is standing waves, and so on, and it simply will not 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 or EQ. 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 four 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 do not hold your breath. An even worse phenomena is that if you adjust the speakers so much in an attempt to fix the net result in the room, then the direct sound coming out of the speakers becomes objectionable.
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 using the JL Audio ARO 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. The JL Audio ARO does not correct issues in the time domain. It only attempts to correct one frequency peak anomaly and flatten that out if it finds one. Why would it find one in the first place? Standing waves in your room! And if you have two ARO-equipped JL Audio subwoofers (or more) and have followed the rest of the procedures, then I suggest running each ARO separately, because the test microphone will be listening to that subwoofer only coupling into the room.
Some people incorrectly use a Y cords to feed both inputs of a subwoofer. This is or should be completely unnecessary; all it does is the same thing as turning up the level 6dB. And if you happen to have two subwoofers you should actually wind up turning each one down perhaps 3dB, so you wind up with the correct overall net level in the room and you have gained 3 dB of headroom in each subwoofer. If you were to leave each volume at its “zero” reference level you might find that it is easier to turn down the Subwoofer Level in the setup menu of your home theater receiver/processor.
ELF Trim And Boundary Settings
On the JL audio Fathom and Gotham subwoofers, the ELF trim (not a haircut for a small, mischievous fairy, but Extremely Low Frequencies) is an equalizer operating in the 25 Hertz region which can compensate for the [apparent] bass buildup if you are placing the subwoofer in the corner. (See the paragraph on room acoustics, above) Typically if you placed the subwoofer in the corner you might want to turn the control down. if for some reason you had to place the subwoofer at the middle of a wall or in another less than desirable position, you can add up to 3dB at low frequencies. Remember +3dB uses twice the power!
Some receivers / processors have THX and other settings for “boundary” effects, and these are similar to the ELF trim on the JL Audio subwoofers.
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!
About Group Delay And Impulse Timing
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 is typically 66% of the speed of light. 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 millisecond. That is 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 is all math and it is a ratio, not a division… but I digress…) 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 “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 responsible for instances where the picture and sound are “out of sync” in modern equipment. Since you cannot 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.
Figure 8: The same signal applied to both the main power amp and the subwoofer are delayed going through the subwoofer. As shown, the delay of the subwoofer would be 1 wavelength at 80 Hertz, or 12.5 millisecond.
Figure 8 shows the incorrect method many people use when connecting a subwoofer. It pains me to even have to use this diagram it is so wrong. 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 subwoofer, where the subwoofers own Low Pass / High Cut filter is engaged.
Here is the clincher: since the subwoofer is always at least 8, 9, 10, 11 millisecond late, the phase relationship can never be correct. It can be corrected in one of two ways only: you can use some electronic means to add the same amount of delay to the top (mains); or you can move the subwoofer(s) closer to your body the correct number of millisecond. You can not match the phase of the subwoofer to the main speakers because you cannot use the phase control on any subwoofer to remove delay; you can only add delay.
Crossovers are often a complex and slippery issue. Many ‘audiophile’ dealers do not necessarily sell them because (go ahead: squirm) they do not really understand them, and they require a lot of setup and handholding therefore they can’t make any money on them… and most speaker manufacturers will not admit or suggest that their speakers need a subwoofer because they do not (or may not) make a subwoofer; 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 subwoofer (or two), parallel (“Y”) the output of their preamp into the main amp and the subwoofer, and are then unhappy with the results.
They think that because their speakers go down to 38 Hertz or 32 Hertz or 27 Hertz that they only want to use the subwoofer between 20 and 32 Hertz… it simply does not work like that, because of the incorrect port, and the fact that the subwoofer is simply not matched to the mains. The results are muddy, indistinct bass, and users who incorrectly attempt this setup often blame the subwoofer. It is not the subwoofer! It is the incorrect matching!
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 low frequencies. 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 usage for this would be “high pass” filter. Technically and mathematically, either is correct. But it is always a good idea to use the term which will yield the least confusion, especially where people are concerned who do not necessarily have audio as a first language. Manufacturers, pay attention…
Some audiophiles do not 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 if implemented correctly.
Win 1. Since you are now not putting in 20 Hertz 80 Hertz into the mains you are not using up the available low frequency cone movement with bass, so the low frequency cone in your main speaker is able to play its higher frequencies (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 low frequency driver when you are 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 are not moving as far at the low frequencies the driver itself is not generating as much back EMF therefore the damping factor and all of its issues are greatly negated. And you do not need to run silver plated cold water pipes to your mains as speaker wires because there is far less current draw by the speakers.
Win 5. Frequencies below 80 are now not causing transient intermodulation distortion with the higher frequencies (and vice versa) in your power amp. Cleaner still.
So let’s start with the simplest method: a passive “filter” that blocks below 80 Hertz from going to your main speakers:
Figure 9: Here is the Marchand XM46SB Passive Filter.
Here is how it is connected to a typical 2-channel system.
Figure 10: The passive filter used as “half” the crossover
So you roll off the mains at 80 Hertz, 24 dB/octave; and you set the filter in the subwoofer the same way. To use an active filter (if it has two or more sections we can now call it a crossover), you have choices like the Bryston, many versions of the Marchand, (solid state, tube, balanced, unbalanced, 1-way, 2-way, rotary knob, precision stepped attenuator…) and some others.
Putting the active, 2-way crossover in your system is done like this:
Figure 11: The active crossover installed.
Since all the filtering is done in the crossover, you turn off the filter in the subwoofer. For fine level matching adjustments you typically have a high and low knob on the crossover to play with.
We now have the JL Audio Crossover:
Figure 12: The JL Audio CR–1 Precision Active Crossover (front and rear view)
So all of this seems easy and yet with any of the passive or active crossovers we have not yet addressed the seemingly critical issue of the group delay in the subwoofer. So even though we have made everything lovely in the frequency domain, the inherent delay in the subwoofer is still there. What are our options?
Fixing The Group Delay
We cannot change (or fix) the inherent group delay in modern subwoofers. That leaves us with two choices if we are intending to be fanatic!
Option 1: We can move the subwoofer closer to our body about 9–10 feet or so, and then use the phase control on the subwoofer 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. If you have a pair of Gothams it makes them look even more impressive, since they are closer.
Option 2: We would theoretically introduce an equivalent delay to the top (main speakers) to match the inherent delay in the subwoofer; then we can super fine-tune the match by using the phase knob on the subwoofer. This is what we do with hHome theater setups using the “speaker distance” controls in the home theater processor.
Some notes about phase knobs: If you have a switch on a subwoofer labeled “phase” that is probably wrong. It is not phase; it is polarity. Phase is any number of degrees shifted, from 1 degree to 360 degrees to 720 degrees to 3600 degrees and so on. Polarity is either 0 degrees or 180 degrees, period. (see Figure 1 and Figure 3 above) Further, polarity does not change the delay! Flipping the polarity switch back and forth and rotating the phase knob to equal 180 degrees of phase shift are not the same! The polarity switch simply flips the polarity. The phase knob adds delay.
If you have a phase knob on a subwoofer, the circuit is usually designed to only add delay. You cannot take away the inherent intrinsic delay in the entire electro-mechanical physics of the subwoofer, but you can add further electrical delay. Some subwoofers (including the JL Audio subwoofers) are calibrated in electrical degrees of waveform at 80 Hertz, because 80 Hertz is almost always the magic frequency. Therefore if the knob says 180 degrees it is actually adding 6.25 millisecond of delay to the subwoofer signal; this is the equivalent of moving the subwoofer 7 feet further away.
So how do we add delay to the “top”? We would have to introduce a real (digital) processor to do that. The options are a devices like the DEQX, the Lyngdorf, or the Mcintosh version, the MEN220. There are also devices like the BSS Studio processor. All of these are of audiophile grade devices. That means that unlike all the “digital” speaker gadgets intended for use in nightclubs and rock n roll systems, these typically do not operate at a 44 kilohertz, (or 48 kilohertz) sampling rate and you will not be disappointed with what the “processing” has done to your precious highs. While some of these enchanting devices exist, and they may be an endless source of fun and tweaking, the JL Audio crossover is designed to cut through much of the tedious setup pains and to make both necessary adjustments easy and experimental tweak adjustments also easy.
Many 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 inexpensive processors which purport to fix everything in the “digital” domain. Most of those do operate at a slower sampling rate.
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 subwoofer timing issues. Simply set all the top speaker distances in the unit ( L C R Ls Rs) to 7 feet where they belong, and set the subwoofer distance to 19 feet. Now, because most consumer equipment operates backwards (!) you are introducing about 12 millisecond delay to all the top speakers.
Now you can fine tune the phase control on the subwoofer to add a bit more delay to the subwoofer to perfectly match the main speakers and the results should be spectacular. My test CD and the two different procedures to accomplish this are all carefully explained here.
Once you correctly place the subwoofer(s) in your room so they couple to your desired area, cross over the main speakers to the subwoofer(s) correctly, and correct the inherent timing issue your results will be everything you hoped for.
In Summation (pun intended…)
I am therefore not suggesting that everyone force themselves to be so fanatic an audiophile, or to necessarily get this crazy when setting up a subwoofer (Ok, yes I am…). 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, or even better, absolute perfection? Yes. Is it going to adjust itself? Sorry, no. That is why you are an audiophile, right?
One Last Bit Of Relief
Even if you cannot get the timing of your subwoofer(s) to match your main speakers as closely as it can be done, there is a saving grace: re-read the paragraph above about using two subwoofers. Notice that humans actually like the fattening up of the bass loudness envelope in time. Therefore even if your subwoofer is 12 millisecond 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 main speakers, your frequency response will be flat (throughout the crossover region, where it matters) and you will not have any awful objections, again, assuming you get as much else right as possible.
Enjoy your audio journey!