ADDING A SUB
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 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...
you have to "fit" your purchase(s) and adjustments
into this range of ideas you have.
Well, be prepared for
The ONLY correct way to
add a sub to system is to define everything ABOVE the sub's
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 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 can not 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;
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' can
be bothered with.
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 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, 6.1, 7.1 etc. systems. Later you may want to read
my Surround Sound page here.
With Home Theater, there
is relatively little correlation between the LFE channel (the
true Low Freq Effects channel) contained on the DVD 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 Hz effects to the room and indeed the listening
position. There is ALSO the rest of the bass; all the below
80 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. 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 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.
But let's 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 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 handled.
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 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
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
400 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
If you have 2 LF sources
of differing phase relationships and/or 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 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)
...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
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.
You might enjoy referring
to my handy FREQUENCY-WAVELENGTH-PERIOD chart here: www.soundoctor.com/freq.htm
TYPES OF "MAIN"
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:
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
Each of these speaker
types couples somewhat differently to the room, and certainly
to a sub.
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 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 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".
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.
This defines the -3dB "cutoff" point of the cabinet
in question. When the manufacturer of a speaker cabinet defines
the frequency response (i.e., 37 Hz - 20kHz +/- 4dB) this
is what is defined by the entire arrangement of the port and
the air in the cabinet and the driver.
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. 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
some cases such as a bandpass sub used in a club or
on stage, you are most concerned with efficiency and directionality
and not with getting frequency response "flat" down
to 20 Hz; therefore a correctly set up bandpass box that might
roll off at 34 to 40 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 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
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 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
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 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 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. 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 subs.
All of this discussion
barely scratches the surface of the true complexity involved
in flawless integration, so let's continue.
THE RECORDING PROCESS
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 the magazine articles where the
"soundstage" of a rock recording is "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
(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 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 every cutting lathe on the planet has 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 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 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 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. 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
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 subs
were more nearfield (the pressure zone). If you invent something
new, let me know. Bass is fun!
USING TWO OR MORE SUBS
|Fig 6. A
|Fig 7. An
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
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
2) using a "Y" cord from the BASS MANAGED output
from a Home Theater receiver and
3) In the case of the JL Audio sub, it might be a Master/Slave
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 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
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".
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 with surprising precision, and
in my years of night club building we used to adjust the steering
of arrayed subs 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 of all audio issues is room acoustics. You cannot
put a great speaker in a marble shower stall and expect it
to sound good. 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 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'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 items
in a row, so you have an end result you like.
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 coupling at the lowest frequencies
because the 2 walls and the floor are acting like 3 sides
of a 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, folds back on itself and 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 NO bass in the room.
So you have the 3 options
for sub placement:
1) Wherever you
can, or wherever your spouse tells you to put it
2) Where it's mathematically
correct to couple to the most applicable part of the room
3) By doing the
crawl-around test and matching up the sub 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 .
And for an in-depth listing of acousticians, acoustic materials,
design/build companies, and so on, see my links list 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.
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.
But back to reality: there
is a place in life for subs connected almost ANY way, 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
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 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...
Until there is a real
holographic computer 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 4
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.
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 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 does NOT correct issues in the time domain. It only attempts
to correct one frequency anomaly and flatten that out if it
finds one. You should make every attempt to correct the timing
FIRST. And if you have two JL Audio subs (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 sub ONLY coupling into the room.
Some people incorrectly
use a Y cords 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 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 level in the room and you 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.
ELF TRIM and BOUNDARY SETTINGS
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 paragraph
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 twice the power!
have THX and other 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 !
ABOUT GROUP DELAY AND IMPULSE
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. 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,
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
"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 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 or 32 Hz or
27 Hz that they ONLY want to use the sub between 20 and 32
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 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...
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
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
WIN # 5) Freqs
below 80 are now NOT causing transient intermodulation distortion
with the higher freqs (and vice versa) in your power amp.
So let's start with the
simplest method: a passive, "filter" that blocks
below 80 Hz from going to your "mains":
|Fig 9. Here's
the Marchand XM46SB PASSIVE Filter.
Here's how it's connected to a typical
|Fig 9. The
passive filter used as "half" the crossover
So you roll off the mains
at 80 Hz, 24 dB/octave; and you set the filter in the sub
the same way.
To use an active filter
(if it has 2 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:
|Fig 9. The
active crossover installed.
Since ALL the filtering
is done IN the crossover, you turn OFF the filter in the sub.
For fine level matching adjustments you typically have a HIGH
and LOW knob on the crossover to play with.
|Fig 10. One
variation of a Marchand crossover showing volume control
So all of this seems easy
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?
FIXING THE GROUP DELAY
We CANNOT change (or fix)
the inherent 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 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
is ALWAYS the magic frequency. 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
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,
or the Mcintosh version, here.
There is also the BSS Studio processor here.
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.
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 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 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 correct
the timing issue your results will be everything you hoped
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.
ONE LAST BIT OF RELIEF
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
about using 2 subs. Notice that humans actually LIKE the 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!