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 / mastering engineer,
a Sci-Fi movie buff, or a professional musician, you have
your own rainbow of views, and objectives, and they each
have a "scale" whether they are emotional, financial,
audiophile, social, practical, experimental... therefore
you have to "fit" your purchase(s) and adjustments
into this range of ideas you have.
Well, be prepared
for a ride.
The ONLY correct way
to add a sub to system is to define everything ABOVE the
sub's [frequency] range as an entity; clearly define the
impulse, phase, and lastly frequency response of this
entity; and then make a new "2-way" system where
the sub is one way and everything above it is the 'other'
way. The parts must be combined correctly so that there
are no cancellations and no smearing of time-related musical
This CANNOT be easily
measured in the frequency domain, because you could have
(as an example) an 80 Hz signal coming from both the mains
and the sub, and if the sub is 12.5 msec late the two
sources will "seem" to be in phase but the sub
really will be 360 degrees, or one full wavelength
late. It is the impulse smearing that this affects,
but people don't measure that because there is no simple
"hand held" phase or impulse meter as there
is an SPL meter. The REASON this meter does not and essentially
cannot exist is that in order to measure impulse response
or phase response you need a starting REFERENCE point,
(in time) and in a speaker system, since the signal has
to travel through circuitry, amplifiers, passive crossovers
inside the speaker box and then hit the driver; therefore
the first reference point MUST be acoustic.
There ARE computer
based impulse response systems such as the TEF, ( very
quick technical blurb HERE;
full story HERE
) but they are involved, require real instrumentation,
are expensive, have a seriously steep learning curve,
and they are absolutely not the kind of thing most 'consumers'
-- or audiophles, can be bothered with or have patience
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.
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
Indeed the only
thing that does matter is an individual's happiness with
their system, whether I or anyone else thinks it's right
But I want you to
know and understand the truth, so to get purely technical...
There are a separate
set of issues for 2-channel stereo "audiophile"
and Home Theater systems, which we may call "Surround
Sound", i.e., 5.1, 7.1 etc. systems. Later you may
want to read my Surround Sound page here: www.soundoctor.com/surround.htm
NOT CONFUSE LFE with BASS. LFE is
a separate channel in a movie theater (called the
'boom' track in the industry) which is necessary because
there is not enough dynamic range (headroom, actually)
in the existing film optical sound tracks and their associated
playback hardware for additional "Low
Frequency Effects". In a movie theater, (as will
further be explained below) you CAN have multiple low
frequency sources because there are essentially no standing
waves of any consequence in that size room. This is also
a separate channel on a DVD or BluRay, as explained below.
With Home Theater,
there is relatively little coherent phase correlation
between the LFE channel (the true Low Freq Effects channel)
contained on the DVD / BluRay and all the frequencies
ABOVE 80; all that is really essential for the low frequency
part of movie enjoyment is the best coupling of the below
80 or 90 Hz effects to the room and indeed the listening
There is ALSO the
rest of the bass; all the below 80 or 90 Hz information
from all 5 channels that is stripped off and summed together
into mono and sent out the SUBWOOFER OUT connector on
every modern HT receiver / processor. That plus the LFE
channel (if it in fact exists on that particular DVD or
BluRay, and it may not) constitutes all the MANAGED
BASS. Therefore the MOST desirable scenario in a HT
situation is to best couple the sub(s) to the room FIRST,
and THEN timing and phase match the sub(s) to the rest
of the system. This way you will get the mostly sub Low
Freq "effects" coupled to your chair, AND the
correctly timed musical bass present typically in the
L and R channels.
In a living-room sized
room, the most desirable setup is to run all the
speakers as SMALL, and send the fully MANAGED BASS (NOT
'just' LFE) to the SUB(s). Another important criteria
is that you want the best "sound effects" from
movies and the best bass from 2-channel sources playing
back through the same speakers. There have been PLENTY
of complaints about "the movies are great but the
2-channel stuff sucks" or vice versa.
Audio is audio. Correct
audio correctly presented is what we are after whether
the source is AM, FM, a CD, an SACD, a DVD, a BluRay,
a simple analog computer out, simple or complex DACs,
esoteric music servers, streaming services, the analog
headphone jack out on a computer or iPhone... and so on.
Impulse response (NOT
frequency response) really is the holy grail of all of
audio. With more pristine 2-channel sound, (and when you
are playing music through your Home Theater system) as
we approach, want, or expect audiophile quality, the issue
is to get the IMPULSE RESPONSE through the crossover region
(and therefore both the phase response AND frequency response,
which is contained under the mathematical umbrella of
impulse response) as smooth as possible, so that IF we
were playing back a correctly recorded IMPULSE, for example
a well recorded kick drum, its fundamental (50-60 hz),
and its subharmonic, an octave lower (25-30 Hz) and its
mostly odd order harmonic structure (all the way up to
8 kHz and then some) are presented correctly by the time
they arrive at the acoustic summation point which is your
ears. This is the basis of "high fidelity".
We also have to assume
and this is a huge assumption that the manufacturers
of our "mains" speakers have ALREADY correctly
addressed the issues of both impulse response and frequency
response. So for the purposes of this discussion (my entire
book isn't ready yet) we will assume that whatever your
mains are, from a 2-way bookshelf to an 8 foot tall floorstander
monster, that within the desired passband of the mains,
the impulse response and frequency response are already
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
The net human result,
since you are standing or sitting in front of the drum,
is you feel and hear this positive pressure wave, and
your ears, body, intellect, social acuity, and previous
memories of such things all converge and you "hear"
this phenomena as a kick drum hit. You see it; you hear
it, you recognize it, and it fits your preconceived notions
about what a kick drum should sound like. In theory, this
sound is then picked up by a microphone. Positive [air]
pressure on the diaphragm of the mic produces a positive-going
(+) voltage at pin 2 (of the 3-pin connector); then this
goes into a mic preamp, the rest of the line amplification,
and at some point in the control room of the studio, out
to a monitor amp and then a loudspeaker. If all goes well,
we then stand in front of that speaker, and listening
to the monitor system, we are socially convinced there
is a drummer obliging us by playing a kick drum right
in front of our face. If the absolute polarity of that
impulse is "backwards" i.e. the polarity
of anything in the circuit is changed, such as the monitor
speaker is wired out of polarity then the absolute
polarity is not the same as the original and we can
hear that. This is one instance where this phenomena
is very easy to both set up as a test and easy to discern.
Clark Johnson has written a entire book about this called
The Wood Effect, available HERE.
Imagine we are playing
back a well recorded cello: we have the fundamentals of
both the strings and the resonance of the wood STARTING
in the subwoofer (that means below 80 Hz, and you may
wish to refer to my frequency-wavelength chart here: www.soundoctor.com/freq.htm
) and the harmonics extending smoothly up through the
various drivers in the rest of the system.
Being a recording
of actual "wood", (and the strings!!!) these
harmonics are mostly even order. If we can correctly preserve
the exact timing (and therefore phase) relationships of
the ratios of the harmonics of these signals, we will
preserve the imaging "in space" of this instrument.
If we do not do this, then the focus is lost. One part
of this assumption is that the instrument is correctly
recorded in the first place, ideally with a stereo
pair of microphones which therefore ARE picking up the
3-dimensional phase and harmonic structure of the instrument
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:
inside of a car, where you are essentially living inside
the speaker cabinet. (the pressure zone)
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...
inside of a typical living room / home theater room. In
this size room you will ALWAYS have standing wave issues
somewhere in the bass passband from 20 -125 Hz. You CANNOT
NOT have these issues in a room this size unless
you have a REAL acoustically treated room with full size,
perhaps 32' bass traps in the walls and all the correct
ratios of absorption vs diffusion especially at low freqs.
This does not mean a couple of pillows in the corners
or ineffectual 800 Hz absorbers on the side walls. IF
you were to have a room with REAL bass trapping then there
would be no bass standing waves because the LF signals
hitting the walls would be absorbed before they had a
chance to bounce back. (what a concept!!!) Rooms like
this are a revelation, (not to mention extremely rare)
because for the first time you are actually able to hear
the speaker, and not the speaker "in" the room.
But back to "most
If you have 2 LF sources
of differing phase relationships (that means timing relationships)
they will cancel. Period. And if they are "in phase",
but 1, 2, 3 or more full cycles (that means wavelengths)
shifted, (that means 360 or 720 or 1080 degrees out of
phase) then the overall frequency response will not seem
bad but the impulse response and clarity and focus will
be smeared, and localization and imaging will be lost.
This is the main reason measuring in the frequency domain
especially in a home-sized room is such an incredible
waste of time. Your measurements "seem" pretty
flat and yet you don't like the end result - isn't as
"clear" as you think it should be, and it isn't
as focused as you think it should be. The issue is ONLY
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)
4. ...an example of group delay. This only
shows one cycle of many. It's entirely possible
the signals are overlaid so they look like
they are "in phase" but they are actually
360 degrees (one wavelength or cycle), 720 degrees
(two wavelengths or cycles), or 1080 degrees etc.
shifted in time out of phase.
5. Group delay drawn another way. The GREEN
wave might be coming out of your "mains".
The RED wave
is coming out of your sub. Notice how at first they
"look" as if they are "in phase"
but the red wave (from the sub) is actually a full
How did the sub get
to be 360 or more degrees late? It's the overall physics
of how it's built. The only correct way to implement a
sub so the frequency response and phase response can be
controlled and have it socially acceptable
in a living room is to implement a sealed box design,
and that means EQ circuitry. Also most of the better brands
of subs, JL Audio included, use massive drivers which
have a relatively large X-max (that means cone movement).
The combination of the air pressure in the sealed box,
and the rest of the equalization circuitry necessary equal
a mechanical and electronic phenomena which equals an
overall time or group delay. Therefore IF the sub is already
8-10 msec late, AND it is placed in the corners further
away than the mains (just for example) then relative to
the mains it might be 12-16 msec late. YOU CANNOT TAKE
THIS DELAY AWAY.
If the sub has a VARIABLE
PHASE knob, (and not an incorrectly labelled "phase"
switch which is actually POLARITY), then as you turn the
phase knob "up" you are ADDING DELAY to the
low freqencies going through the sub.
You might enjoy referring
to my handy FREQUENCY-WAVELENGTH-PERIOD chart here: www.soundoctor.com/freq.htm
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 in that room, and therein lie the problems in
A port is ALWAYS nothing
more than a cheap way to attempt to get free bass out
of an enclosure and /or driver that's too small. It's
a holdover from the 1930's when because of driver inefficiencies
(especially when compared to today's units) you had to
do everything possible to increase the useable output
over the desired range of low frequencies.
At one level, all
the guyz want 9 foot speakers in the living room (read
"man-cave"). All spouses, of whatever gender,
want tiny 3" speaker cubes that disappear, but expect
9-foot results from them. Since they haven't repealed
ohms law or any other laws of physics while we were sleeping,
the only way to get correct sound is to move a correct
amount of air.
Lets examine ported
speakers. We'll start with the worst case, the port in
the front. At mid bass frequencies, say 50-80 Hz, the
LF driver moves IN the cabinet, the air in the cabinet
is elastic, and the port air moves out of the cabinet.
Because of the frequency at which the cone is moving,
by the time the cone moves out (forward) again, the port
air is now moving out, so in front of the cabinet the
two air pressure sources sum together and you get a fake
bass "bump" or "boost".
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:
When the manufacturer
of a speaker cabinet defines the frequency response (i.e.,
37 Hz - 20kHz +/- 3dB) this is what is defined by the
entire arrangement of the port and the air in the cabinet
and the driver. At some low frequency the port air is
exactly out of phase with the driver air pressure and
since they cancel, there is NO output from the cabinet
into the room. Therefore with a ported cabinet, the entire
sloppy concept is this juggling game between the response
of the drivers under air pressure, the passive crossover
inside the box, the port size and placement.
You must understand
that ANY driver goes down to 0 Hz, or DC. If you put a
battery across a speaker, the cone moves out and stays
there. If you were to have a DC coupled power amp feeding
a speaker - ANY speaker, from a 1" dome tweeter to
an 18" rock n roll stage bass driver - and you put
4 Hz into it, it would simply move back and forth at 4
Hz. Of course in order to actually "hear" the
audio it would have to be in the generally accepted passband
of 20-20,000 Hz and the cone diameter would have to be
enough to actually move some air in the room. So it is
the overall combination of the driver size, the excursion,
the box size, (therefore the air back pressure) and many
other factors that determines the overall response of
that "speaker" AS AN ENTITY.
That means IF you
were to simply put those same frequencies through the
mains and the sub (that means with no crossover, and this
is the mistake that nearly everyone makes) you would now
have 3 sources of LF energy and differing phase: the 'main'
LF driver, the port, and the sub, all fighting with each
other in the time domain. A further corollary is that
since the air inside the [mains] cabinet is elastic, the
phase relationship of the port air to the driver air is
also a sliding one; that means it's "out of phase"
and smearing over a wider range of frequencies
than you might think.
If the port is on
the back, again, a cheap attempt to use the back wave
bouncing off a wall to give 'additional' bass, you have
the ADDITIONAL issue of the transit time it takes for
the back port pressure (already delayed because of the
elasticity) to leave the cabinet, travel back, hit a wall,
and bounce back around the front of the cabinet again;
therefore this LF wave MIGHT be "in phase" with
the front driver BUT BE 360 OR EVEN 720 DEGREES LATE;
therefore it sounds like the bass frequencies are ok in
the frequency domain but the IMPULSE RESPONSE is now muddied.
Also, in the case
of back ported or (type 5) dipole speakers, since the
path length from the back of the speaker to the wall and
bouncing back around to the front of the speaker is a
fixed physical entity, at some frequencies you are adding
and at some frequencies you are canceling: you have simply
made a physical/mechanical frequency comb filter that
you can't do anything about. Sound Lab's answer to this
(for use with their flat panel electrostatic speakers,
which are dipoles) is they sell you a "Sallie",
which is an absorber to absorb the entire back wave output
of the electrostatic panel. Since now there is no comb
filtering; all you are therefore hearing is the front
A ported sub for
home use is even more wrong than ported mains.
Now you would be attempting to acoustically add together
in the room at least SIX low frequency sources with differing
phase and frequency slope conditions: the LF drivers in
your two mains, their ports, the sub driver, and its port.
In addition, since it's a bandpass it cannot go down low
enough for serious Home Theater effects. (that typically
means a real 20Hz or close to it.)
some cases such as a bandpass sub used in a club
or on a modest-sized stage in your local pub, you are
most concerned with efficiency and not with getting frequency
response "flat" down to 20 Hz; therefore a correctly
set up bandpass box that might roll off at 35 to 45 Hz
is quite sufficient and also very efficient for the defined
purpose. And again, as a point of reference, "flat"
response in the frequency domain is FAR AND AWAY the LEAST
important phenomena: impulse response in the time domain
is the most important, but it cannot be measured with
a handheld meter therefore almost everyone simply ignores
it. If you're interested in learning about the newest
(and evolving) pro sound system / stage methods
of "steering" bass, Dave Rat has some very cool
part 1 www.youtube.com/watch?v=VwLH7zP6Lwo
part 2 www.youtube.com/watch?v=B-3pURYOwfw
part 3 www.youtube.com/watch?v=aSZK9Altvm8
There's a nice article
But back to our Home
/ HI-FI / 2-channel / Audiophile / Surround Sound systems:
There is ONLY ONE truly correct way to "add a sub"
to a system in an controlled listening room situation:
you must correctly cross over the 2 sealed cabinets; and
their timing must be correct. ANY other method will lessen
the focus and clarity and imaging you have tried so hard
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.
APPROACH - IN ONE PARAGRAPH
A similar situation
exists with home theater setups where the customer THINKS
that the front speakers are "full range". Even
BEST overall approach is to seal the ports, operate
the 5 channels as "small", crossover at
80 (or even a higher, like 90 Hz, but NEVER lower)
and correct the timing issues inherent in all modern
subs by setting (in the receiver or processor's
setup menu) ALL the distances THE SAME, and to a small
number such as 7 feet; then set the sub distance to
12 feet MORE (i.e. 19 feet) and THEN use the variable
phase control on the sub to fine tune the relationship
at the 80 Hz crossover point, at the listening
speaker companies that make "large" speakers
(such as B&W) are aware of this port issue and supply
port plugs just for this purpose. Kudos to them.
People who have fought
with their systems for weeks or years finally email and
call me to tell me that for the first time they are
finally satisfied in fact thrilled with
the incredible integration of their JL Audio, MK Sound,
SVS sealed, or other fine subs.
All of this discussion
(so far) barely scratches the surface of the true complexity
involved in flawless integration, so let's continue.
The idea of setting
exact speaker distances is flawed from early mistakes
made by both receiver / processor manufacturers and the
somewhat misconstruing of the acoustic and other technical
differences between a large movie theater and the home
setup. I cover this in more detail on my surround sound
THE RECORDING PROCESS
top of all the previous variables we have all the issues,
errors, and modern production values and practices inherent
in the recording process. It is simply laughable (and
pathetic) when I read the magazine articles where the
reviewer calls the "soundstage" of a rock recording
"palpable". Sorry, but every rock/pop recording
made in the last 50 years is composed of a series of panned
mono sources that have absolutely no "depth"
or "width". They are each separately sent to
an echo/reverb device, the delayed returns of which are
usually (but not always) panned somewhere in the left
to right soundstage 'width'. The combination of the 3
panned signals ("real", "echo return 1",
and "echo return 2") then present an auditory
fantasy (hallucination, actually) of a "soundstage".
The summation of all
the Left-Right panning placement and the summation of
all the reverb returns therefore fools you into thinking
there is a "soundstage". Sadly, precious few
recordings are made with any regard to true stereo or
binaural imaging sound in anything resembling a true form;
even better classical recordings of large orchestras have
morphed into combinations of stereo miking and "some"
local more-nearfield mono miking added to the mix to achieve
whatever the producer/engineer determine is a suitable
balance, perhaps between a soloist and the rest of the
There are precious
few companies who do pay attention to this; AIX
records is one. Chesky Records is another, here: www.chesky.com/content/binaural-series
But to think that
any modern, commercial pop recording mix has any true
acoustic space (and even uses real instruments!) is, for
the most part, sadly mistaken. (There
will be MUCH more about this in a long white paper to
come in early 2019.)
Oh, and to touch upon
"stereo bass" for a moment... there almost
is no such thing. Going back to vinyl, every stereo vinyl
record cut in the last 60 years has mono bass. It has
to. If the bass were 180 degrees out of phase L and R
then there would be vertical modulation and the stylus
would jump out of the groove. Therefore most cutting lathe
electronics have a "compatalyzer" circuit, that
dumps frequencies below 160 hz into mono (typically a
single-order filter, therefore 6dB/octave). You MAY
have out of phase bass (i.e. "low frequencies")
on a CD, but precious few producers/engineers are savvy
enough (or care enough to even bother, since, typically,
what's the point?) to make use of those sort of tricks.
There are some EDM dubstep dance trance psychedelia eurotrash
electronica club music releases where there are bass tracks
where there is stereo bass in the form of something like
24 Hz in one channel and 24.2 Hz in the other channel;
therefore you get an air pressure differential which
travels around the room. Cool! In the above
example, the "traveling pressure differential wave"
would take 5 seconds to go back and forth around the room.
If you're a really bored or obsessive techweenie you can
have a lot of fun with this - we played with this phenomena
at Moog Synthesizer as far back as 1969. Expect to either
make your listeners nauseous or to watch their heads rotate
on their bodies not unlike the effect in the movie The
As far as PLAYING
BACK signals like this goes, as mentioned above, in a
large theater or outdoors you CAN have multiple bass sources
of differing phase because there are essentially no standing
waves, (and if there are, they are so delayed in time
that they are of no cancelling consequence in the audio
passband) and so your ears (and indeed your whole receptive
system) can process and differentiate and accept
all the phase issues. In a much smaller room like a living
room, it is more difficult but you might be able to pull
it off if your subs were more nearfield (the pressure
zone). Perhaps if you have a large room, with too much
low freq reverberation, you could put the sub(s) right
next to your listening chair and adjust their phase appropriately.
This would tighten the whole system up. If you invent
something new, let me know. Bass is fun!
USING TWO OR MORE SUBS
A symmetrical layout.
An asymmetrical layout.
So most people's reasons
for multiple subs in a room is "more even coverage".
Let's examine the instances of multiple subs and what
they do. One interesting issue with using multiple subs
concerns arrival times. Here's a hypothetical situation.
You are feeding the same signal to 2 subs. So this begs
the question what's your room like? Is it symmetrical?
L-shaped? A closed room? A Huge open space? Notice we're
back to acoustics.
Referring to Fig 6,
the subs are also equidistant from your body. So the subs
each couple to the room however they do. The whole setup
is essentially symmetrical.
Now here's another
example. Refer to Fig 7. We are still putting essentially
the same signal into both subs. There might be 3 ways
to do this:
1) from the L and R of a stereo preamp (MUCH more on this
2) using a "Y" cord from the BASS MANAGED output
from a Home Theater receiver and
3) In the case of a JL Audio Fathom/Gotham series sub,
it might be a Master/Slave setup.
The point is, the
sound leaves both subs at exactly the same time. Notice
in Fig 7 the R sub is closer to your face. Perhaps the
L sub is 11 feet away and the R sub is 4 feet away. That's
a 7msec time differential. So you hear the leading edge
of the bass wave from the Right sub, then 7 msec later
the leading edge of the L sub... then the note dies away
from the R sub and then 7 msec later the note dies away
from the L sub. What have you accomplished? Here comes
the magic: YOU HAVE FATTENED UP THE LOUDNESS ENVELOPE
IN TIME ! This is the magic that humans
love. This is why someone says, "OMG, two subs are
SO much better than one!" So you have a combination
of the arrival time differential, and to a certain extent
you have the separate room coupling issues such that each
sub is its own entity coupling into the room with slightly
differing standing waves.
It's NOT that the
two subs are louder than one, since typically you
would adjust both to have the correct desired loudness
leval AT the listening position. A quick word about acoustic
summing: In theory, 2 speakers in the same room will sum
so the result will be 6 dB louder. However they will only
do this if the phase relationship is the same. Therefore
because of the standing waves involved, if two subs are
right next to each other (or one is on top of the other)
they will essentially sum at about +5 dB. If they are
apart (say, placed next to each main) then they will essentially
sum about 3 to 4 dB in the room.
So now we have 2 ways
to view the multiple sub issue: as a method of attempting
to get better coverage over a larger seating area of a
multiple-seat Home Theater room, or as a method of fattening
up the bass presence for one or two listeners in
a sweet spot.
In the case of better
subs, that have variable phase adjustments, my suggestion
in setups like this is to use either method (1) or (2)
above, and then adjust the phase control knob on each
sub for most accurate transitioning at the crossover
frequency. It's slightly tricky, but you will keep the
real phase relationship between each sub and the mains,
AND you will keep the arrival time differential that you
Here's the magic trick if you
have one sub in the front and one in the "back"
(as shown in Fig 7, above.)
Turn off the back sub.
Align the front sub using
the out-of-phase nulling setup on my TEST CD page,
Now turn the MAINS off.
Flip the FRONT SUB's POLARITY
SWITCH to the opposite position from wherever it
Turn the BACK SUB on. Play a
sine wave at the crossover freq.
Null the back sub to the front
sub AT THE LISTENING POSITION by adjusting the back
sub's phase knob and level control.
When finished, put the FRONT
SUB's POLARITY SWITCH back to where it WAS.
Now the subs are both level
matched AND TIMED correctly AT THE LISTENING POSITION.
If you accomplish this correctly if you are playing
a drum solo (as an example) you should perceive
the lower drum freqs (like from the kick drum and
floor toms) to be coming from in the front of the
room, as you would expect.
As a separate discussion
I should touch on this. In SOME instances, let's say in
certain Home Theater setups, you might have the option
of "homogenizing" the room, that is, making
most of the seats sound "the same". Depending
on the number of seats, you might want to make ONE seat
as good as possible (your seat...) and for the rest, let
the chips fall where they may.
Here is why I actually
suggest this: Everyone wants something "different"
from the installation. The audiophile dude wants magic.
The wife hates audio. The mother-in-law likes movies but
hates bass. (she is going to move her chair around until
she finds a bass null, believe me) And the little kid
wants to climb inside the subwoofer cabinet, and the teenage
son wants everything at 118 dB. If you decide this scenario
applies to you then you can use the room anomalies (perhaps
a bass null) to your advantage.
Perhaps the room is
for just two people; Then, typically, the chairs would
not both be centered; rather you would have to
average out the sweet spot. You decide, carefully.
Some people think
that "bass is non-directional". That is a mis-statement.
The reality is that as you go lower and lower it becomes
less localizable by your mechanism of hearing;
above about 100 Hz you can start to localize it and the
precision of the localization depends on the rest of the
frequencies playing (or not); and the standing waves in
the room at the frequency you are trying to determine.
Feeding two subs with the same sine wave from a test oscillator
or test cd and adjusting the phase knobs separately will
show you just how directional it can be. It can be steered
around the room with surprising precision, and in my 
years of night club building we used to adjust the phase
steering of arrayed subs so that the bass was correct
on the dance floor where it belongs and much less off
the dance floor and in the corners of the club.
ROOM ACOUSTICS - AND THE
ONGOING PACK OF LIES
Understand that the
largest percentage of all audio issues is room acoustics.
You cannot put a great speaker in a marble shower stall
and expect it to sound good - it will sound like a speaker
in a marble shower stall. Room acoustics itself is a very
complex set of interactions of physics and perception.
Sadly, there are many
instances where manufacturers or individuals skew the
relevant terms and confuse people. For example, beware
of (and be aware of) the dangerous term "Room Tuning".
You CANNOT tune a room using an "equalizer".
You are tuning THE SOUND SYSTEM with the equalizer - the
room is still the same. REAL room tuning means anything
from sticking pillows in the corners to rebuilding the
room (perhaps correctly) from scratch, incorporating a
set of acoustic devices and parameters which sometimes
seem nebulous but get a desired result. Because of this
nebulosity of all the acoustics terminology (not
to mention the international differences in measuring
techniques, terminology, and 'scales', which are substantial)
it is often difficult for an end user (and many audio
professionals, for that matter) to be able to mentally
visualize just what a room without standing waves will
sound like, or a room which is so rolled off that the
high frequencies seem to "fall to the floor".
To make matters even
worse, a term like "soundproofing" is essentially
a meaningless audio non sequitur; you would have to define
how many dB, and at what frequencies... and what is the
ambient noise level of the areas of interest to begin
with? And do you mean sounds coming IN or sounds LEAVING
an area? And so on.
So real room tuning
is one entire entity, and then once the room is deemed
to be as useable as it's going to get, then we
enter the realm of SYSTEM TUNING. The big trick of course
is getting the correct balance of all of these ducks in
a row, so you have an end result you like.
ranges have VERY differing interactions with a room, AND
with the speakers.
At LOW freqs, 20 - 120 Hz or so, you have signals cooming
from a sub or subs, and the all-too-real issue of standing
waves and very distinct cancellations because of these
At MEDIUM freqs, perhaps 125-2k or so, you have more issues
with first order reflection.
At HIGHER freqs, perhaps 2k - 15k, you might have flutter/
Some people say they
are going to put a sub in the corner because of "room
gain". Another misnomer! There is no gain;
there is no amplifier attached to the room! What
is taking place is the corner of a room has the MOST EFFICIENT
output coupling into the rest of the room at the lowest
frequencies because the 2 walls and the floor are acting
like 3 sides of a huge linear-sided horn at those large
wavelengths. So it's not that the corner has any gain;
it's that everywhere else in the room has apparent
loss. The middle of each wall has the most apparent
loss, because the sound leaves the driver, goes in all
directions, reflects / folds (bounces) back on itself
and partially cancels out. If you put the sub in the middle
of a wall left and right and ALSO placed it in the middle
of the wall floor-to-ceiling you would get essentially
NO or very little bass in the room. Instead of thinking
about it as "room gain", think about it correctly
as "room loss". That will help to focus your
thinking on where to best put the sub(s) for the best
coupling back to your chair.
So you have the 3
options for sub placement:
1) where you
think it / they should go
2) where your
spouse tells you to put them
3) where they
For some very enlightening
articles about bass, room modes/nodes, standing waves,
and room coupling, see Art Noxon's articles HERE .
And for an in-depth listing of acousticians, acoustic
materials, design/build companies, and so on, see my links
page on the Boston Audio Society HERE.
So AFTER you have
addressed the issue of room acoustics to the best of your
ability, and this means you have decided if you have a
2-channel system, a Home Theater system, (perhaps both,
even perhaps separate!) what your seating priorities might
be, and the rest of your decor, you might have decided
to make the sub placement a priority. Or not. IF YOU ARE
ABLE, here is generally the best method: THE CRAWL-AROUND
TEST. While it might seem funny or silly the end result
compared to hours or days of computer analysis is usually
spectacular. And often better.
The methodology is
outlined on my test CD page, here: www.soundoctor.com/testcd.
The crawl-around test has nothing to do with the rest
of your system. What you are doing is coupling one
or more subs back to your listening position based on
the physics of the room. AFTER you have finished the test,
you THEN match the subs with the rest of your system in
the frequency crossover mode, and in phase and absolute
If you DON'T couple
the sub(s) to your listening position or area as well
as they might be, you could be throwing away "a few"
dB in coupling efficiency. If you are "throwing away"
3dB PER SUB you might as well not have bought the 2nd
sub in the first place. Remember 3dB is twice the power,
and 6dB is four times the power.
Most people who are
NOT used to audio tend to equate 10dB (10 times the power)
as "twice as loud", while engineers who are
all too familiar with the financial issues of trying to
make something louder have learned that 6dB is, in fact,
twice (or half) the loudness, or Sound Pressure Level.
Actually there IS NO SUCH THING as "twice as loud".
Your brain and senses operate on a 20 log scale, and you
should learn how that equates to real life. It's fun.
Similarly, there IS NO SUCH THING as "50% louder".
How do you learn this? Get a good SPL meter or app for
you phone (check out SPLnFFT) and watch it all
day, in a quiet room, in a loud room with audio blasting,
at a party, in your car, etc. Get a feel of you YOU perceive
the certain SPL's and how they relate to your spaces.
But back to reality:
there is a place in life for subs connected almost ANY
way, even where there's just another extra bass boom which
impresses some people. To someone who only has experienced
a cheap table radio or a the moral equivalent in any sort
of surround system, ANY sub, even one poorly set up will
"seem" like a revelation!
Do not fall into the
trap of having a home theater receiver / processor with
a "computer" inside and your JL Audio sub with
it's ARO inside (or other fine sub) and think you are
going to run these two computers and your life is gwanna
be great: you might be in for a rude awakening. You will
more than likely be like a person with 2 watches who is
never really sure exactly what time it is...
Your room is at least
a 5 dimensional system:
Height x Width x Depth; and Frequency and Time,
which includes reflections and their subsequent cancellations.
Until there is a real
holographic computer / Lidar correlation / deconvolving
system which really can sample the room in a 3 dimensional
grid (for example in 36 or 48 places) the best we can
do right now it to attempt to approximate the net results
in a room at a few (1, 2, 3, or 4) places. In SOME setups
like this the results can be great. But here is where
it sometimes falls apart: If the room is so bad that you
really "need" a setup computer in the first
place, it can't necessarily determine what is real,
what is reflection, what is standing waves, and so on,
and it simply won't work as you expect. Imagine
trying to adjust a sound system in the aforementioned
marble shower stall. You cannot fix or change the room
reverberation or standing waves no matter what you do
with a computer. Someday there will probably be computers
powerful enough to do subtractive room decorrelation,
and they will probably work by scanning the room with
laser interferometers first, then build a 5 dimensional
graphic of the room, (by then probably in n-dimensional
space, but I digress) then correlate all the standing
waves at all frequencies, calculate all the Rt60 times
at all frequencies, then adjust the output of all the
amps to decorrelate all this... (hear that, Darpa?) but
don't hold your breath. It will initially probably be
very, very sloppy.
My suggestion is to
follow the necessary steps separately AND MANUALLY, and
in the correct order; learn the equipment,
and then experiment with ONE "computer" at a
time (I would suggest the JL Audio ARO / DARO first) and
determine if it helps you. If not, try something else.
The only way you can determine if something works is to
make one change at a time. Remember, the JL Audio ARO
/ DARO does NOT correct issues in the time domain. It
only attempts to frequency anomalies and smooth that out.
You should make every attempt to correct the overall timing
FIRST. And if you have two JL Audio subs (or more) and
have followed the rest of the procedures, then I suggest
NEVER running them in master/slave, because the ARO /
DARO results from each placed in the room will be different.
They can only be the same if the room is TRULY flawlessly
symmetrical or if the subs are RIGHT NEXT to each other.
Even if one sub is on tho of the other sub it will be
different because the top sub is now coupling modally
different into the room.
Be aware that on the
JL Audio series of Fathom and Gotham subs, the V1 ARO
uses one band of determined EQ freq and attenuation and
the V2 DARO uses 18 bands of 1/6 octave EQ and attenuation.
Some people incorrectly
use a "Y cord" to feed both inputs of a sub.
This is or should be completely unnecessary; all it does
is the same thing as turning
up the level on the sub (or the send level from the receiver/processor)
+6dB. And if you happen to have TWO subs you should actually
wind up turning each one down 3dB, so you wind up with
the correct resultant level in the room and you will have
gained 3 dB of HEADROOM in each sub. If you were
to leave each volume at its reference level you might
find that it's easier to turn DOWN the SUBWOOFER LEVEL
in the setup menu of your Home Theater receiver/processor.
ELF TRIM and BOUNDARY
On the JL audio subs,
the ELF trim is an equalizer operating in the 25 Hz region
which can compensate for the [apparent] bass buildup
if you are placing the sub in the corner. (See the paragraphs
on room acoustics, above) Typically IF you placed the
sub in the corner you might want to turn the control down.
If for some reason you place the sub at the middle of
a wall or in another less than desirable position, you
can add 3dB. Remember 3dB is using twice the power!
have THX and other proprietary settings for "boundary"
effects, and these are similar to the ELF trim on the
JL Audio subs.
A further discussion
includes crossovers, whether passive, active, tube, solid
state, analog, digital, balanced or unbalanced; and proper
methodology of both measuring and correcting the inherent
group delays in modern equipment to fine tune the impulse
response. We're getting to that !
ABOUT GROUP DELAY AND
So now let's examine
the aforementioned group delay. It takes time for a signal
to go through a circuit. Inasmuch as everyone thinks electricity
travels at the speed of light, that's not quite true.
Electrons going through a wire, which we can call a transmission
line are slowed down by a certain amount. For some
types of cables this is called the velocity factor, and
it's typically 66% of the speed of light. (Not that that's
slow!) It also takes a certain amount of time for the
signals to get through each piece of equipment, although
relative to other human events, this is quite fast: it
might take 5-50 microseconds for the signal to go through
a power amp, because there are no mechanical devices in
the way. Once we get a signal into a mechanical device
such as a speaker, whether it is passive or active, we
now have the sum total of all the electrical plus mechanical
phenomena to take into account. The typical group delay
through a modern, sealed box subwoofer, is perhaps 8 to
15 msec. That's milliseconds, not microseconds.
In the digital world,
delay issues are often called latency. Specifically
this refers to some circuitry where the signal starts
out as analog, goes through an A:D converter (not an A/D
converter as incorrectly stated in much literature; it's
all math and it's a ratio, not a division... but I digress
even further...) then gets processed digitally in some
fashion, then goes through a D:A converter, and then we
hear it as an analog signal. This is a HUGE issue with
modern recording studios and live venue "digital"
mixing boards and everyone is continually fighting against
seemingly impossible odds...sometimes there is so much
latency when devices are used in series with each other
that the musicians hear themselves as an echo and this
makes it nearly impossible to play. The entire premise
of the "convenience" and "power" of
"digital" is sometimes negated by these latency
issues and the difficulties in "fixing" them.
This is also an issue
inside Home Theater receivers/processors, where the purely
digital HDMI signal is stripped apart and reconverted
back to analog. Collectively, this mess is partially responsible
for instances where the picture and sound are "out
of sync" in modern equipment. Since you CAN'T get
rid of the delay, the only answer is to delay something
else so it all "matches up" in the end.
In the analog world it still takes time for a signal to
go through a circuit, and although the phenomena should
probably be called transit time, group delay
is what has stuck; a holdover from the early telephony
days, when the concern was the delay of the audio frequencies,
not the DC control or bell ringing signals (all carried
on the same lines), and the term meant a "group"
of frequencies we were concerned about.
Let's start with a
2-channel (stereo) setup and look at this block diagram:
The same signal applied to both the main power amp
and the sub are delayed going through the sub.
As shown, the delay of the sub would be 1 wavelength
at 80 Hz, or 12.5 msec.
Fig 8. shows
THE INCORRECT METHOD many people use when connecting
a sub. It pains me to even have to use this diagram. NO
crossover is shown. The full range signal goes through
the power amp and into the mains; and the full range signal
goes into the sub, where the sub's own LOW PASS / HIGH
CUT filter is engaged.
Here's the clincher:
since the sub is always at least 8, 9, 10, 11 msec late,
the phase relationship CAN NEVER be correct.
It can be corrected in one of 2 ways only: you can use
some electronic means to ADD the same amount of delay
to the top (mains); or you can move the sub(s) closer
to your body the correct number of msec. You CANNOT match
the phase of the sub to the mains because you CANNOT use
the phase control on any sub to remove delay; you can
ONLY ADD DELAY.
Crossovers are always
a slippery issue. Many 'audiophile' dealers don't necessarily
sell them because (go ahead: squirm) they don't really
understand them, and they require a lot of handholding
therefore they can't make any money on them... and most
speaker manufacturers won't admit or suggest that their
speakers need a sub because they don't (or may not) make
a sub; therefore they port their speakers in an attempt
to get extra "free" bass and therefore the coupling
and delay timing issue is made ever so much more complicated.
Many customers that I talk to simply buy a sub (or two)
parallel ("Y") the output of their preamp into
the main amp and the sub, and are then unhappy
with the results. They think that because their
speakers go down to 38 Hz that they ONLY want
to use the sub between 20 and 40 Hz... it simply doesn't
work like that, because of the incorrect port, and
the fact that the sub is simply not matched to the mains.
The results are muddy, indistinct bass, and users who
incorrectly attempt this setup often wrongly blame the
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,
Be aware that there
is very annoying current marketing/sales term where some
manufacturers say there is a "crossover" in
the sub and it is only a low pass filter. THERE
IS NO HIGH FREQ OUT to go back to your amp, so it is a
lie, plain and simple. There ARE, however, subs with real
crossovers in them. For example the JL Audio E subs HAVE
a real crossover in them, with HF OUTPUTS which then go
back to your power amp. The Fathom series does not; it
has a low pass filter.
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
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+
WIN # 2) Since
you are not putting bass into that same driver you are
not Doppler modulating everything between 80 and 600,
or whatever the next crossover point is. This means cleaner
mids. By far.
WIN #3) You
are not sucking current out of your main power amp at
low frequencies, so there is more current reserve to play
those highs louder...
WIN # 4) Since
the cones aren't moving as far at the low freqs the driver
itself is not generating as much back EMF therefore the
damping factor and all of its issues are greatly negated.
And you don't need to run silver plated cold water pipes
to your mains as speaker wires because there is less current
draw by the speakers.
WIN # 5) Freqs
below 80 are now NOT causing transient intermodulation
distortion with the higher freqs (and vice versa) in
your power amp. Cleaner still.
So let's start with
the simplest method: a passive "filter" that
blocks below 80 Hz from going to your "mains",
and PASSES the highs to your mains:
Here's the Marchand XM46SB PASSIVE Filter.
Here's how it's connected to a typical
The passive filter used as "half" the crossover
So you roll off the
mains at some frequency, such as 80 or 90 Hz, 24 dB/octave;
(you have to purchase the frequency you want, since it
is custom made, and I HIGHLY SUGGEST 90 Hz, 24
dB/octave) and you set the low pass filter in the sub
the same way. If you want (for some reason) to only use
passive capacitors and inductors in your system, this
is one answer. Overall I do not necessarily recommend
this though. More modern solutions are FAR better.
I only want to show the option. Please be aware that your
precious audio signal has gone through MANY THOUSANDS
of opamps from the microphone through a myriad of 'stuff'
in and out of computers or tape recorders or both, and
then THAT signal has gone through further opamps in the
mastering process, and so on. Using an active filter with
a few more opamps is not going to destroy your 'precious'
To use an active filter
(if it has 2 or more sections we can now call it a CROSSOVER),
there are many choices some of which are each explained
There are the Bryston
crossovers, very handsome, built like a tank, and with
a terrific warranty... except the ordering options are
quite complicated and many people wind up getting the
10b 'standard' (which does NOT have a 24 dB/octave setting)
when the better choice would be the 10b SUB, or LR. Then
the crossover winds up on Audiogon or Audiomart, because
the user is frustrated. If you order the SUB version or
the third LR version then you must order separate plug-in
parts for different frequencies. Be very careful reading
their very complex user manual.
Many versions of the
(solid state, tube, balanced, unbalanced, 1 way, 2 way,
rotary knob, precision stepped attenuator...) are available.
One variation of a Marchand crossover (XM9) showing
stepped volume control knobs.
Another choice for
simpler experimentation and budgetary concerns is the
dbx 223 series, here: https://dbxpro.com/en-US/product_families/crossovers
The dbx 223 xs crossover
Note the dbx has separate
models for use with XLR or Phone/RCA connectors. BOTH
models are balanced (but may be wired unbalanced)
- only the connectors are different. If you are intending
to use UNbalanced RCA's then you must get these RCA to
1/4" TS (Tip/Sleeve) adapters: (you will need SIX).
RCA to 1/4" Tip/Sleeve adapter
Here is the dirt simple front panel
setup for the dbx units:
Dirt simple dbx XO setup for 90 Hz
Putting the active, 2-way crossover
in your system is done like this:
Showing a typical active crossover in a "typical"
Since ALL the filtering
is done IN the crossover, you turn OFF the [low pass]
filter in the sub. For fine level matching adjustments
you typically have HIGH and LOW output knobs on the crossover
to play with.
There's also, at the
higher end, the Pass
Here is the very
best cut-to-the-chase analog answer: The JL Audio
CR1 Crossover. It is VERY comprehensive and the cleanest
device there is. It was my concept/idea while at JL Audio
and it took the amazing engineering department and I nearly
4 years to finalize the design, development, and production.
The JL Audio CR1 Crossover. (Click on any of the pix
for a larger page)
Here are some of its
1) In the crossover mode, you can
use either/and/or RCA,
Balanced XLR, or balanced or unbalanced 1/4" (Tip/Sleeve
Tip/Ring/Sleeve) input or output connectors at any
In the [hard] bypass mode (see the
color diagram above) of course the RCA input connectors
connect directly to the RCA output connectors, and the
XLR input connectors connect directly to the XLR output
2) The frequency controls are SEPARATE
- that is, you can set them the same, or overlap or
underlap to accomodate ANY preferences you might have.
3) The Bypass/On switch will give
you the PERFECT A/B comparison: in the "on"/operate
mode the sub is crossed over (as it should be); and
so are the mains. In the bypass mode, the mains are
operating full range and there is no sub, so you can
FULLY and IMMEDIATELY appreciate exactly the benefits
of what the crossover (and sub) is doing in your room.
NO OTHER DEVICE is capable of this - and it is this
A/B demo feature that blows everyone's mind.
4) The bypass switch may also be
used for Home Theater bypass.
5) Separate HF and LF Damping controls
give you subtle and desirable control. This becomes
the final "I gotta add salt and pepper to the chef's
creation" buttons, because everyone really wants
one more knob to turn! Or not.
6) The "balance" control
enables you to have less or more sub or mains for late
night listening, or to assist with mixes which could
use a bit of "help". It has a zero reference
detent in the middle to easily return exactly to normal.
7) MUTE buttons enable you to discern
anything - such as muting the mains to hear if the subs
are vibrating something in your room, etc. The separate
L and R mains (satellite) and sub mute switches also
greatly assist in setup using my unique TEST CD, here:
8) yes, you can have separated "stereo"
subs if you want.
9) Note for you pristine analog
fanatics: There is NO "digital audio" in the
CR1. The audio path is completely analog.
...and there's much more! Here's
What I have determined
is that sometimes, a customer might be reluctant to purchase
such a device as the JL CR1. But here's the easy path.
First get something very simple (and inexpensive) such
as the dbx. Experiment with it for a bit! Once you learn
the benefits of correctly applying a crossover to your
system, you can sell the dbx in a heartbeat and get the
CR1 you really want and deserve!
CROSSOVERS TO COMPUTERIZED ADJUSTMENTS
Some people WANT to
get more detailed and be VERY involved with complex and
comprehensive setups, and want to turn into an engineer.
Not everyone does. But in case you do, you can do everything
yourself with a computer, test microphones, and products
like the ones below. You'll never want to come out of
your room, your spouse (ex-spouse, by now) or various
buddies will have to throw in cold pizza, warm coke, and
an occasional piece of raw meat into your room, but as
a now devoted for life audiophile engineer, you WILL be
able to control the world! Onward!
There's REW (Room
EQ Wizard): www.roomeqwizard.com
, and there is also integration with the Roon player,
The DEQX models are
DIRAC is here: www.dirac.com
...and the MiniDSP
collection of products, which include Dirac www.minidsp.com
and TRINNOV www.trinnov.com
The AudioVero products
include ACOURATE and The ACOURATE CONVOLVER, here :www.audiovero.de/en/
The JUICE Audiolens
is here www.juicehifi.com
I have posted many
more links HERE: www.bostonaudiosociety.org/links3.htm#SOFTWARE
ISSUES TO CONSIDER: INTEGRATED AMPS & TAPE LOOPS
There are two other
annoying problems that have to do with so-called "Integrated
Amps". MOST of them sadly do not have a method
for connecting a crossover between the PREAMP OUT and
the POWER AMP IN. The terrific Bryston B135 (and also
their B60r) DO HAVE the availability to correctly
do this. A few (very few) other brands do have this feature.
The Bryston B135 Integrated Amp with PRE OUT - PWR
AMP IN connectivity
Then there's the issue
of all the other Integrated Amps which do NOT have the
abovementioned loop, but almost all of them DO have a
"Tape Loop" - a holdover from the days when
people connected cassette decks... So the issue there
is the TAPE OUTPUT is at a full, fixed level, taken off
at an earlier stage, before the volume control. So you
CANNOT connect a regular crossover in that loop because
the low freq outs of the crossover will be at a fixed
level (full), while the high freqs only will be adjustable.
So for the over a thousand (!) people who contacted me
to ask about this, there is really only ONE WAY to accomplish
the nearly impossible: The Marchand XM9 or XM44 crossovers,
which have SEPARATE LOW and HIGH frequency level controls.
And for wonderful convenience, these are available with
precision stepped attenuators with repeatable click positions.
This is what the front
panel with the separate click stepped level controls looks
One variation of a Marchand crossover (XM9) showing
stepped volume control knobs.
This is how you would
connect it - you MUST use a crossover with separate
high out and low out level controls. The issue is if you
try to use a crossover with "regular" non-stepped
controls, (that means a plain old potentiometer) you will
be frustrated trying to always match and fine tune the
Connecting a Marchand crossover in the TAPE LOOP
Now you set the maximum
level you want in the room by putting the four crossover
levels all the way up (L high, L low, R high, R low) and
adjusting the main volume control on the amp. Then you
use the 4 controls on the crossover to attenuate
to the volume in the room to what you want. This the ONLY
way to accomplish this if you have a tape loop only and
can't insert a crossover between the premp out and power
FIXING THE GROUP DELAY
So all of this crossover
setup so far seems moderately easy (you just... plug it
in...) and yet with ANY of the passive or active crossovers
we have not YET addressed the critical issue of the group
delay in the sub. SO even though we have made everything
lovely in the frequency domain, the INHERENT delay in
the sub is still there. What are our options?
We CANNOT change (or
fix) the inherent / intrinsic group delay in modern subs.
That leaves us with two choices IF WE ARE INTENDING
TO BE FANATIC !
OPTION 1) We
can move the sub closer to our body about 9-10 feet or
so, and then use the phase control on the sub itself to
fine tune the match. This is not necessarily as crazy
as it sounds. We do this successfully in studios all the
time. Of course this might not work in your particular
OPTION 2) We
must introduce an equivalent delay TO THE TOP (mains)
to match the inherent delay in the sub; then we can super
fine-tune the match by using the phase knob on the sub.
Some notes about phase
knobs: If you have a (toggle) SWITCH on a sub labelled
"phase" that is wrong. It is not phase; it is
POLARITY. Phase is ANY NUMBER of degrees shifted, from
1 degree to 360 degrees to 3600 degrees and so on. Polarity
is either 0 degrees or 180 degrees, period. (see Fig.1
and Fig.3 above) If you have a phase KNOB on a sub,
the circuit is usually designed to only ADD DELAY. You
cannot take away the inherent delay in the entire electro-mechanical
physics of the sub, but you can ADD further electrical
delay. Some subs are calibrated in electrical degrees
of waveform at 80 hz, because 80 hz was the original suggested
crossover freq for Home Theater/Surround Sound systems.
Therefore IF the knob says 180 degrees it is actually
adding 6.25 msec of delay to the sub signal; this is the
equivalent of moving the sub 7 feet FURTHER AWAY.
So how do we add delay
to the "top"? We would have to introduce a real
processor to do that. The options are a device like the
or the Mcintosh version, here.
(see a longer list above) All of these are audiophile
grade devices. That means that UNLIKE all the "digital"
speaker gadgets intended for use in nightclubs and rock
n roll systems, these do NOT operate at 44kHz, (or even
48kHz) and you will NOT be disappointed with what the
"processing" has done to your precious highs.
Many of the so-called "professional" units are
perfectly suitable for a noisy bar or a rock touring PA
system but you might be very disappointed if it is your
intent to use them in a critical audiophile listening/monitoring
situation. That means beware of $99 - $299 processors.
But even if you DO get a very inexpensive processor, say
on ebay or Craigslist, etc., the learning experience is
well worth it, if you have the patience.
In the instance of
Home Theater processors, there is an easy method. We can
take advantage of the somewhat flawed concept of "speaker
distance settings" to perfectly fix the sub timing
issues. Simply set ALL the top speakers ( L C R Ls Rs)
to 7 feet where they belong, and set the sub distance
to 18-19-20 feet. Now, because all consumer equipment
operates backwards (!!!) you are introducing 10-12 msec
delay TO ALL THE TOP SPEAKERS. Now you can fine tune the
phase control on the sub to add a bit more delay to the
sub to perfectly match the mains and the results
should be spectacular. My test CD and the two different
procedures to accomplish this are all carefully explained
Once you correctly
place the sub(s) in your room so they correctly couple
to your desired area, cross over the mains to the sub
correctly, and fix the timing issue your results will
be everything you hoped for.
IN SUMMATION (pun
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,
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
about using 2 subs. Notice that humans actually LIKE the
(slight) fattening up of the bass loudness envelope in
time. Therefore even IF your sub is 12 msec late, and
you are one wavelength off, as long as you get that delayed
wavelength to line up with the bass coming out of your
mains, your frequency response will be pretty good
and you won't have any awful objections, again, assuming
you get as much else right as possible.
Enjoy your audio journey!
And let me know your results!