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 events.
This CANNOT be easily measured
in the frequency domain, because you could have (as an example)
an 80 Hz signal coming from both the mains and the sub, and if
the sub is 12.5 msec late the two sources will "seem"
to be in phase but the sub really will be 360 degrees, or one
full wavelength late. It is the impulse smearing that this
affects, but people don't measure that because there is no simple
"hand held" phase or impulse meter as there is an SPL
meter. The REASON this meter does not and essentially cannot exist
is that in order to measure impulse response or phase response
you need a starting REFERENCE point, (in time) and in a speaker
system, since the signal has to travel through circuitry, amplifiers,
passive crossovers inside the speaker box and then hit the driver;
therefore the first reference point MUST be acoustic.
There ARE computer based impulse
response systems such as the TEF, ( very quick technical blurb
full story HERE
) but they are involved, require real instrumentation, are expensive,
have a seriously steep learning curve, and they are absolutely
not the kind of thing most 'consumers' -- or audiophles, can be
bothered with or have patience for.
So the overall view of adding
a sub is this: In essence you are designing and assembling
a new speaker system which is "2-way": the sub
is one way and everything "else" above it in frequency
is the 2nd way.
Simply connecting a sub to
existing mains speaker (or amp) terminals is the WORST POSSIBLE
WAY to do this. EVERYTHING scientific and acoustic about this
method is wrong, from the additive delay issues to the back EMF
of the mains affecting the LF signal. However there are plenty
of people who simply do not understand correctly integrated bass,
and they will be reasonably happy simply sticking another box
on to their system without regard to timing, phase and frequency
issues, and they will think it sounds "ok" or "good"
and for those people it doesn't really matter.
Indeed the only thing
that does matter is an individual's happiness with their system,
whether I or anyone else thinks it's right or wrong.
But I want you to know and
understand the truth, so to get purely technical...
There are a separate set of
issues for 2-channel stereo "audiophile" and Home Theater
systems, which we may call "Surround Sound", i.e., 5.1,
7.1 etc. systems. Later you may want to read my Surround Sound
page here: www.soundoctor.com/surround.htm
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 well 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 resonance of the wood STARTING in the subwoofer (that
means below 80 Hz, and you may wish to refer to my frequency-wavelength
chart here: www.soundoctor.com/freq.htm
) and the harmonics extending smoothly up through the various
drivers in the rest of the system.
Being a recording of actual
"wood", (and the strings!!!) these harmonics are mostly
even order. If we can correctly preserve the exact timing (and
therefore phase) relationships of the ratios of the harmonics
of these signals, we will preserve the imaging "in space"
of this instrument. If we do not do this, then the focus is lost.
One part of this assumption is that the instrument is correctly
recorded in the first place, ideally with a stereo pair
of microphones which therefore ARE picking up the 3-dimensional
phase and harmonic structure of the instrument in space.
You CANNOT have multiple low
frequency sources of differing phase relationships in a living
room-sized room. Let's examine the acoustic "spaces"
we might be dealing with. There are
3 useful separate sizes of acoustic spaces in life:
1) The inside
of a car, where you are essentially living inside the speaker
cabinet. (the pressure zone)
2) A large movie
theater, amphitheater, or outdoor space where there either are
no reflecting walls or the walls are so far away in
time that any reflections, partially because of the Haas effect
and frequency cancellation effects are essentially of no importance;
3) The inside
of a typical living room / home theater room. In this size room
you will ALWAYS have standing wave issues somewhere in the bass
passband from 20 -125 Hz. You CANNOT NOT have these issues
in a room this size unless you have a REAL acoustically
treated room with full size, perhaps 32' bass traps in the walls
and all the correct ratios of absorption vs diffusion especially
at low freqs. This does not mean a couple of pillows in the corners
or ineffectual 800 Hz absorbers on the side walls. IF you were
to have a room with REAL bass trapping then there would be no
bass standing waves because the LF signals hitting the walls would
be absorbed before they had a chance to bounce back. (what a concept!!!)
Rooms like this are a revelation, (not to mention extremely rare)
because for the first time you are actually able to hear the speaker,
and not the speaker "in" the room.
But back to "most rooms"...
If you have 2 LF sources of
differing phase relationships (that means timing relationships)
they will cancel. Period. And if they are "in phase",
but 1, 2, 3 or more full cycles (that means wavelengths) shifted,
(that means 360 or 720 or 1080 degrees out of phase) then the
overall frequency response will not seem bad but the impulse
response and clarity and focus will be smeared, and localization
and imaging will be lost. This is the main reason measuring in
the frequency domain especially in a home-sized room is such an
incredible waste of time. Your measurements "seem" pretty
flat and yet you don't like the end result - isn't as "clear"
as you think it should be, and it isn't as focused as you think
it should be. The issue is ONLY timing.
We can call the red
and green waves signals from 2 separate
"speakers", 2 separate subs, or a sub and a "mains"
speaker. Here are the diagrams that show this:
Fig 1. Obviously "in phase"
Fig 2. 90 degrees "out of phase"
(the red wave is lagging the green wave by 90 degrees)
Fig 3. 180 degrees out of phase (the net
result is complete cancellation)
Fig 4. ...an
example of group delay. This only shows one cycle of
many. It's entirely possible the signals are overlaid
so they look like they are "in phase"
but they are actually 360 degrees (one wavelength or cycle),
720 degrees (two wavelengths or cycles), or 1080 degrees
etc. shifted in time out of phase.
Fig 5. Group
delay drawn another way. The GREEN
wave might be coming out of your "mains". The
RED wave is coming out
of your sub. Notice how at first they "look" as
if they are "in phase" but the red wave (from
the sub) is actually a full wavelength LATE.
How did the sub get to be
360 or more degrees late? It's the overall physics of how it's
built. The only correct way to implement a sub so the frequency
response and phase response can be controlled and
have it socially acceptable in a living room is to implement a
sealed box design, and that means EQ circuitry. Also most of the
better brands of subs, JL Audio included, use massive drivers
which have a relatively large X-max (that means cone movement).
The combination of the air pressure in the sealed box, and the
rest of the equalization circuitry necessary equal a mechanical
and electronic phenomena which equals an overall time or group
delay. Therefore IF the sub is already 8-10 msec late, AND it
is placed in the corners further away than the mains (just for
example) then relative to the mains it might be 12-16 msec late.
YOU CANNOT TAKE THIS DELAY AWAY.
If the sub has a VARIABLE
PHASE knob, (and not an incorrectly labelled "phase"
switch which is actually POLARITY), then as you turn the phase
knob "up" you are ADDING DELAY to the low freqencies
going through the sub.
You might enjoy referring
to my handy FREQUENCY-WAVELENGTH-PERIOD chart here: www.soundoctor.com/freq.htm
TYPES OF "MAIN" SPEAKERS
In addition to all the above,
there is the complex issue of the "main" speaker you
are coupling to. There are essentially 6 types of speakers that
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 acceptable integration.
A port is ALWAYS nothing more
than a cheap way to attempt to get free bass out of an enclosure
and /or driver that's too small. It's a holdover from the 1930's
when because of driver inefficiencies (especially when compared
to today's units) you had to do everything possible to increase
the useable output over the desired range of low frequencies.
At one level, all the guyz
want 9 foot speakers in the living room (read "man-cave").
All spouses, of whatever gender, want tiny 3" speaker cubes
that disappear, but expect 9-foot results from them. Since they
haven't repealed ohms law or any other laws of physics while we
were sleeping, the only way to get correct sound is to move a
correct amount of air.
Lets examine ported speakers.
We'll start with the worst case, the port in the front. At mid
bass frequencies, say 50-80 Hz, the LF driver moves IN the cabinet,
the air in the cabinet is elastic, and the port air moves out
of the cabinet. Because of the frequency at which the cone is
moving, by the time the cone moves out (forward) again, the port
air is now moving out, so in front of the cabinet the two air
pressure sources sum together and you get a fake bass "bump"
you go lower and lower in frequency, at some low frequency the
air pressure from the LF driver and the air pressure from the
port are exactly opposite each other, so they cancel, and there
is no more audio at that frequency: it disappears.
When the manufacturer of
a speaker cabinet defines the frequency response (i.e., 37 Hz
- 20kHz +/- 3dB) this is what is defined by the entire arrangement
of the port and the air in the cabinet and the driver. At some
low frequency the port air is exactly out of phase with the driver
air pressure and since they cancel, there is NO output from the
cabinet into the room. Therefore with a ported cabinet, the entire
sloppy concept is this juggling game between the response of the
drivers under air pressure, the passive crossover inside the box,
the port size and placement.
You must understand that ANY
driver goes down to 0 Hz, or DC. If you put a battery across a
speaker, the cone moves out and stays there. If you were to have
a DC coupled power amp feeding a speaker - ANY speaker, from a
1" dome tweeter to an 18" rock n roll stage bass driver
- and you put 4 Hz into it, it would simply move back and forth
at 4 Hz. Of course in order to actually "hear" the audio
it would have to be in the generally accepted passband of 20-20,000
Hz and the cone diameter would have to be enough to actually move
some air in the room. So it is the overall combination of the
driver size, the excursion, the box size, (therefore the air back
pressure) and many other factors that determines the overall response
of that "speaker" AS AN ENTITY.
That means IF you were to
simply put those same frequencies through the mains and the sub
(that means with no crossover, and this is the mistake that nearly
everyone makes) you would now have 3 sources of LF energy and
differing phase: the 'main' LF driver, the port, and the sub,
all fighting with each other in the time domain. A further corollary
is that since the air inside the [mains] cabinet is elastic, the
phase relationship of the port air to the driver air is also a
sliding one; that means it's "out of phase" and
smearing over a wider range of frequencies than you might
If the port is on the back,
again, a cheap attempt to use the back wave bouncing off a wall
to give 'additional' bass, you have the ADDITIONAL issue of the
transit time it takes for the back port pressure (already delayed
because of the elasticity) to leave the cabinet, travel back,
hit a wall, and bounce back around the front of the cabinet again;
therefore this LF wave MIGHT be "in phase" with the
front driver BUT BE 360 OR EVEN 720 DEGREES LATE; therefore it
sounds like the bass frequencies are ok in the frequency domain
but the IMPULSE RESPONSE is now muddied.
Also, in the case of back
ported or (type 5) dipole speakers, since the path length from
the back of the speaker to the wall and bouncing back around to
the front of the speaker is a fixed physical entity, at some frequencies
you are adding and at some frequencies you are canceling: you
have simply made a physical/mechanical frequency comb filter that
you can't do anything about. Sound Lab's answer to this (for use
with their flat panel electrostatic speakers, which are dipoles)
is they sell you a "Sallie",
which is an absorber to absorb the entire back wave output of
the electrostatic panel. Since now there is no comb filtering;
all you are therefore hearing is the front signal.
A ported sub for home
use is even more wrong than ported mains. Now you would
be attempting to acoustically add together in the room at least
SIX low frequency sources with differing phase and frequency slope
conditions: the LF drivers in your two mains, their ports, the
sub driver, and its port. In addition, since it's a bandpass it
cannot go down low enough for serious Home Theater effects. (that
typically means a real 20Hz or close to it.)
some cases such as a bandpass sub used in a club or on
a modest-sized stage in your local pub, you are most concerned
with efficiency and not with getting frequency response "flat"
down to 20 Hz; therefore a correctly set up bandpass box that
might roll off at 35 to 45 Hz is quite sufficient and also very
efficient for the defined purpose. And again, as a point
of reference, "flat" response in the frequency domain
is FAR AND AWAY the LEAST important phenomena: impulse response
in the time domain is the most important, but it cannot be measured
with a handheld meter therefore almost everyone simply ignores
it. If you're interested in learning about the newest (and evolving)
pro sound system / stage methods of "steering"
bass, Dave Rat has some very cool videos here:
part 1 www.youtube.com/watch?v=VwLH7zP6Lwo
part 2 www.youtube.com/watch?v=B-3pURYOwfw
part 3 www.youtube.com/watch?v=aSZK9Altvm8
There's a nice article here:
But back to our Home / HI-FI
/ 2-channel / Audiophile / Surround Sound systems: There is ONLY
ONE truly correct way to "add a sub" to a system in
an controlled listening room situation: you must correctly cross
over the 2 sealed cabinets; and their timing must be correct.
ANY other method will lessen the focus and clarity and imaging
you have tried so hard to preserve.
I have many clients and customers
with extremely exotic high-end 2-channel systems that are all
chasing the holy grail of 3D holographic sound imaging, and until
they follow my distinct guidelines they are never completely satisfied
with the results.
THE BEST APPROACH
- IN ONE PARAGRAPH
A similar situation exists
with home theater setups where the customer THINKS that the front
speakers are "full range". Even so...
|The BEST overall
approach is to seal the ports, operate the 5 channels as "small",
crossover at 80 (or even a higher, like 90 Hz, but NEVER lower)
and correct the timing issues inherent in all modern subs
by setting (in the receiver or processor's setup menu) ALL
the distances THE SAME, and to a small number such as 7 feet;
then set the sub distance to 12 feet MORE (i.e. 19 feet) and
THEN use the variable phase control on the sub to fine tune
the relationship at the 80 Hz crossover point, at the listening
Some better speaker
companies that make "large" speakers (such as B&W)
are aware of this port issue and supply port plugs just for this
purpose. Kudos to them.
People who have fought with
their systems for weeks or years finally email and call me to
tell me that for the first time they are finally satisfied
in fact thrilled with the incredible integration
of their JL Audio, MK Sound, SVS sealed, or other fine subs.
All of this discussion (so
far) barely scratches the surface of the true complexity involved
in flawless integration, so let's continue.
The idea of setting exact
speaker distances is flawed from early mistakes made by both receiver
/ processor manufacturers and the somewhat misconstruing of the
acoustic and other technical differences between a large movie
theater and the home setup. I cover this in more detail on my
surround sound page, here:
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 orchestra.
There are precious few companies
who do pay attention to this; AIX
records is one. Chesky Records is another, here: www.chesky.com/content/binaural-series
But to think that any modern,
commercial pop recording mix has any true acoustic space (and
even uses real instruments!) is, for the most part, sadly mistaken.
(There will be MUCH more about
this in a long white paper to come in early 2019.)
Oh, and to touch upon "stereo
bass" for a moment... there almost is no such thing.
Going back to vinyl, every stereo vinyl record cut in the last
60 years has mono bass. It has to. If the bass were 180 degrees
out of phase L and R then there would be vertical modulation and
the stylus would jump out of the groove. Therefore most cutting
lathe electronics have a "compatalyzer" circuit, that
dumps frequencies below 160 hz into mono (typically a single-order
filter, therefore 6dB/octave). You MAY have out of phase
bass (i.e. "low frequencies") on a CD, but precious
few producers/engineers are savvy enough (or care enough to even
bother, since, typically, what's the point?) to make use of those
sort of tricks. There are some EDM dubstep dance trance psychedelia
eurotrash electronica club music releases where there are bass
tracks where there is stereo bass in the form of something like
24 Hz in one channel and 24.2 Hz in the other channel; therefore
you get an air pressure differential which travels around the
room. Cool! In the above example, the "traveling
pressure differential wave" would take 5 seconds to go back
and forth around the room. If you're a really bored or obsessive
techweenie you can have a lot of fun with this - we played with
this phenomena at Moog Synthesizer as far back as 1969. Expect
to either make your listeners nauseous or to watch their heads
rotate on their bodies not unlike the effect in the movie The
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!
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
Referring to Fig 6, the subs
are also equidistant from your body. So the subs each couple to
the room however they do. The whole setup is essentially symmetrical.
Now here's another example.
Refer to Fig 7. We are still putting essentially the same signal
into both subs. There might be 3 ways to do this:
1) from the L and R of a stereo preamp (MUCH more on this later)
2) using a "Y" cord from the BASS MANAGED output from
a Home Theater receiver and
3) In the case of a JL Audio Fathom/Gotham series sub, it might
be a Master/Slave setup.
The point is, the sound leaves
both subs at exactly the same time. Notice in Fig 7 the R sub
is closer to your face. Perhaps the L sub is 11 feet away and
the R sub is 4 feet away. That's a 7msec time differential. So
you hear the leading edge of the bass wave from the Right sub,
then 7 msec later the leading edge of the L sub... then the note
dies away from the R sub and then 7 msec later the note dies away
from the L sub. What have you accomplished? Here comes the magic:
YOU HAVE FATTENED UP THE LOUDNESS ENVELOPE IN TIME ! This
is the magic that humans love. This is why someone says, "OMG,
two subs are SO much better than one!" So you have a combination
of the arrival time differential, and to a certain extent you
have the separate room coupling issues such that each sub is its
own entity coupling into the room with slightly differing standing
It's NOT that the two subs
are louder than one, since typically you would adjust both
to have the correct desired loudness leval AT the listening position.
A quick word about acoustic summing: In theory, 2 speakers in
the same room will sum so the result will be 6 dB louder. However
they will only do this if the phase relationship is the same.
Therefore because of the standing waves involved, if two subs
are right next to each other (or one is on top of the other) they
will essentially sum at about +5 dB. If they are apart (say, placed
next to each main) then they will essentially sum about 3 to 4
dB in the room.
So now we have 2 ways to view
the multiple sub issue: as a method of attempting to get better
coverage over a larger seating area of a multiple-seat Home Theater
room, or as a method of fattening up the bass presence
for one or two listeners in a sweet spot.
In the case of better subs,
that have variable phase adjustments, my suggestion in setups
like this is to use either method (1) or (2) above, and then adjust
the phase control knob on each sub for most accurate transitioning
at the crossover frequency. It's slightly tricky, but you will
keep the real phase relationship between each sub and the mains,
AND you will keep the arrival time differential that you "love".
Here's the magic trick if you have one
sub in the front and one in the "back" (as shown
in Fig 7, above.)
Turn off the back sub.
Align the front sub using the
out-of-phase nulling setup on my TEST CD page, here:
Now turn the MAINS off.
Flip the FRONT SUB's POLARITY SWITCH
to the opposite position from wherever it is.
Turn the BACK SUB on. Play a sine wave
at the crossover freq.
Null the back sub to the front sub AT
THE LISTENING POSITION by adjusting the back sub's phase
knob and level control.
When finished, put the FRONT SUB's POLARITY
SWITCH back to where it WAS.
Now the subs are both level matched
AND TIMED correctly AT THE LISTENING POSITION. If you accomplish
this correctly if you are playing a drum solo (as an example)
you should perceive the lower drum freqs (like from the
kick drum and floor toms) to be coming from in the front
of the room, as you would expect.
As a separate discussion I
should touch on this. In SOME instances, let's say in certain
Home Theater setups, you might have the option of "homogenizing"
the room, that is, making most of the seats sound "the same".
Depending on the number of seats, you might want to make ONE seat
as good as possible (your seat...) and for the rest, let the chips
fall where they may.
Here is why I actually suggest
this: Everyone wants something "different" from the
installation. The audiophile dude wants magic. The wife hates
audio. The mother-in-law likes movies but hates bass. (she is
going to move her chair around until she finds a bass null, believe
me) And the little kid wants to climb inside the subwoofer cabinet,
and the teenage son wants everything at 118 dB. If you decide
this scenario applies to you then you can use the room anomalies
(perhaps a bass null) to your advantage.
Perhaps the room is for just
two people; Then, typically, the chairs would not both
be centered; rather you would have to average out the sweet spot.
You decide, carefully.
Some people think that "bass
is non-directional". That is a mis-statement. The reality
is that as you go lower and lower it becomes less localizable
by your mechanism of hearing; above about 100 Hz you can start
to localize it and the precision of the localization depends on
the rest of the frequencies playing (or not); and the standing
waves in the room at the frequency you are trying to determine.
Feeding two subs with the same sine wave from a test oscillator
or test cd and adjusting the phase knobs separately will show
you just how directional it can be. It can be steered around the
room with surprising precision, and in my  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
Different frequency ranges
have VERY differing interactions with a room, AND with the speakers.
At LOW freqs, 20 - 120 Hz or so, you have signals cooming from
a sub or subs, and the all-too-real issue of standing waves and
very distinct cancellations because of these standing waves.
At MEDIUM freqs, perhaps 125-2k or so, you have more issues with
first order reflection.
At HIGHER freqs, perhaps 2k - 15k, you might have flutter/ reverberation
Some people say they are going
to put a sub in the corner because of "room gain". Another
misnomer! There is no gain; there is no amplifier attached
to the room! What is taking place is the corner of a room has
the MOST EFFICIENT output coupling into the rest of the room at
the lowest frequencies because the 2 walls and the floor are acting
like 3 sides of a huge linear-sided horn at those large wavelengths.
So it's not that the corner has any gain; it's that
everywhere else in the room has apparent loss. The
middle of each wall has the most apparent loss, because the sound
leaves the driver, goes in all directions, reflects / folds (bounces)
back on itself and partially cancels out. If you put the sub in
the middle of a wall left and right and ALSO placed it in the
middle of the wall floor-to-ceiling you would get essentially
NO or very little bass in the room. Instead of thinking about
it as "room gain", think about it correctly as "room
loss". That will help to focus your thinking on where to
best put the sub(s) for the best coupling back to your chair.
So you have the 3 options
for sub placement:
1) where you think it
/ they should go
2) where your spouse
tells you to put them
3) where they ACTUALLY
For some very enlightening
articles about bass, room modes/nodes, standing waves, and room
coupling, see Art Noxon's articles HERE .
And for an in-depth listing of acousticians, acoustic materials,
design/build companies, and so on, see my links page on the Boston
Audio Society HERE.
So AFTER you have addressed
the issue of room acoustics to the best of your ability, and this
means you have decided if you have a 2-channel system, a Home
Theater system, (perhaps both, even perhaps separate!) what your
seating priorities might be, and the rest of your decor, you might
have decided to make the sub placement a priority. Or not. IF
YOU ARE ABLE, here is generally the best method: THE CRAWL-AROUND
TEST. While it might seem funny or silly the end result compared
to hours or days of computer analysis is usually spectacular.
And often better.
The methodology is outlined
on my test CD page, here: www.soundoctor.com/testcd.
The crawl-around test has nothing to do with the rest of your
system. What you are doing is coupling one or more subs back
to your listening position based on the physics of the room. AFTER
you have finished the test, you THEN match the subs with the rest
of your system in the frequency crossover mode, and in phase and
absolute timing mode.
If you DON'T couple the sub(s)
to your listening position or area as well as they might be, you
could be throwing away "a few" dB in coupling efficiency.
If you are "throwing away" 3dB PER SUB you might as
well not have bought the 2nd sub in the first place. Remember
3dB is twice the power, and 6dB is four times the power.
Most people who are NOT used
to audio tend to equate 10dB (10 times the power) as "twice
as loud", while engineers who are all too familiar with the
financial issues of trying to make something louder have learned
that 6dB is, in fact, twice (or half) the loudness, or Sound Pressure
Level. Actually there IS NO SUCH THING as "twice as loud".
Your brain and senses operate on a 20 log scale, and you should
learn how that equates to real life. It's fun. Similarly, there
IS NO SUCH THING as "50% louder". How do you learn this?
Get a good SPL meter or app for you phone (check out SPLnFFT)
and watch it all day, in a quiet room, in a loud room with audio
blasting, at a party, in your car, etc. Get a feel of you YOU
perceive the certain SPL's and how they relate to your spaces.
But back to reality: there
is a place in life for subs connected almost ANY way, even where
there's just another extra bass boom which impresses some people.
To someone who only has experienced a cheap table radio or a the
moral equivalent in any sort of surround system, ANY sub, even
one poorly set up will "seem" like a revelation!
Do not fall into the trap
of having a home theater receiver / processor with a "computer"
inside and your JL Audio sub with it's ARO inside (or other fine
sub) and think you are going to run these two computers and your
life is gwanna be great: you might be in for a rude awakening.
You will more than likely be like a person with 2 watches who
is never really sure exactly what time it is...
Your room is at least a 5
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 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 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
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.
(Not that that's slow!) It also takes a certain amount of time
for the signals to get through each piece of equipment, although
relative to other human events, this is quite fast: it might take
5-50 microseconds for the signal to go through a power amp, because
there are no mechanical devices in the way. Once we get a signal
into a mechanical device such as a speaker, whether it is passive
or active, we now have the sum total of all the electrical plus
mechanical phenomena to take into account. The typical group delay
through a modern, sealed box subwoofer, is perhaps 8 to 15 msec.
That's milliseconds, not microseconds.
In the digital world, delay
issues are often called latency. Specifically this refers
to some circuitry where the signal starts out as analog, goes
through an A:D converter (not an A/D converter as incorrectly
stated in much literature; it's all math and it's a ratio, not
a division... but I digress even further...) then gets processed
digitally in some fashion, then goes through a D:A converter,
and then we hear it as an analog signal. This is a HUGE issue
with modern recording studios and live venue "digital"
mixing boards and everyone is continually fighting against seemingly
impossible odds...sometimes there is so much latency when devices
are used in series with each other that the musicians hear themselves
as an echo and this makes it nearly impossible to play. The entire
premise of the "convenience" and "power" of
"digital" is sometimes negated by these latency issues
and the difficulties in "fixing" them.
This is also an issue inside
Home Theater receivers/processors, where the purely digital HDMI
signal is stripped apart and reconverted back to analog. Collectively,
this mess is partially responsible for instances where the picture
and sound are "out of sync" in modern equipment. Since
you CAN'T get rid of the delay, the only answer is to delay something
else so it all "matches up" in the end. In the analog
world it still takes time for a signal to go through a circuit,
and although the phenomena should probably be called transit
time, group delay is what has stuck; a holdover from the early
telephony days, when the concern was the delay of the audio frequencies,
not the DC control or bell ringing signals (all carried on the
same lines), and the term meant a "group" of frequencies
we were concerned about.
Let's start with a 2-channel
(stereo) setup and look at this block diagram:
|Fig 8. The same
signal applied to both the main power amp and the sub are
delayed going through the sub.
As shown, the delay of the sub would be 1 wavelength at 80
Hz, or 12.5 msec.
Fig 8. shows
THE INCORRECT METHOD many people use when connecting a
sub. It pains me to even have to use this diagram. NO crossover
is shown. The full range signal goes through the power amp and
into the mains; and the full range signal goes into the sub, where
the sub's own LOW PASS / HIGH CUT filter is engaged.
Here's the clincher: since
the sub is always at least 8, 9, 10, 11 msec late, the phase
relationship CAN NEVER be correct. It can be corrected
in one of 2 ways only: you can use some electronic means to ADD
the same amount of delay to the top (mains); or you can move the
sub(s) closer to your body the correct number of msec. You CANNOT
match the phase of the sub to the mains because you CANNOT use
the phase control on any sub to remove delay; you can ONLY ADD
Crossovers are always a slippery
issue. Many 'audiophile' dealers don't necessarily sell them because
(go ahead: squirm) they don't really understand them, and
they require a lot of handholding therefore they can't
make any money on them... and most speaker manufacturers won't
admit or suggest that their speakers need a sub because they don't
(or may not) make a sub; therefore they port their speakers in
an attempt to get extra "free" bass and therefore the
coupling and delay timing issue is made ever so much more complicated.
Many customers that I talk to simply buy a sub (or two) parallel
("Y") the output of their preamp into the main amp and
the sub, and are then unhappy with the results. They think
that because their speakers go down to 38 Hz that
they ONLY want to use the sub between 20 and 40 Hz... it simply
doesn't work like that, because of the incorrect port, and
the fact that the sub is simply not matched to the mains. The
results are muddy, indistinct bass, and users who incorrectly
attempt this setup often wrongly blame the sub.
One brief word about all the
terms being bandied about: yes, a LOW CUT and a HIGH
PASS are the same thing. It is MOST USEFUL
to use the terminology so it fits the use of the situation. In
one example, we have a filter in a recording studio Microphone
Preamp. Of course WE KNOW THE AUDIO GOES "THROUGH" the
thing; what we want to know is what we are doing - what "change"
we are going to hear when we click the switch! We are CUTTING
THE LOWS. In this instance the correct terminology is LOW CUT
FILTER. In the case of "filtering" a signal that's going
to our mains, yes, of course we are "letting the highs through"
and we are also "blocking the lows". So the typical
useage for this would be "HIGH PASS" filter. Technically
and mathematically, either is correct. But it's always a good
idea to use the term which will yield the least confusion, especially
where people are concerned who don't necessarily have audio as
a first language. Manufacturers, pay attention...
Be aware that there is very
annoying current marketing/sales term where some manufacturers
say there is a "crossover" in the sub and it is only
a low pass filter. THERE IS NO HIGH FREQ OUT to go back
to your amp, so it is a lie, plain and simple. There ARE, however,
subs with real crossovers in them. For example the JL Audio E
subs HAVE a real crossover in them, with HF OUTPUTS which then
go back to your power amp. The Fathom series does not; it has
a low pass filter.
audiophiles don't want to introduce yet another active "thing"
in their precious signal path, not realizing that adding the crossover
is very much the lesser of two evils.
Actually adding a crossover
is really a WIN-WIN situation:
WIN # 1) Since
you are now NOT putting in 20 Hz - 80 Hz into the
mains you are not using up the available LF cone movement with
bass, so the LF cone in your mains is able to play its
higher freqs (up to IT'S crossover point) much more cleanly.
You get an apparent 6dB or more dynamic range. You
can play your system LOUDER, and also with less compression distortion
in the LF driver when you're having that Saturday night dance
party and you're playing urban bass technopop at 110+ dB. Really.
WIN # 2) Since
you are not putting bass into that same driver you are not Doppler
modulating everything between 80 and 600, or whatever the next
crossover point is. This means cleaner mids. By far.
WIN #3) You are
not sucking current out of your main power amp at low frequencies,
so there is more current reserve to play those highs louder...
WIN # 4) Since
the cones aren't moving as far at the low freqs the driver itself
is not generating as much back EMF therefore the damping factor
and all of its issues are greatly negated. And you don't need
to run silver plated cold water pipes to your mains as speaker
wires because there is less current draw by the speakers.
WIN # 5) Freqs
below 80 are now NOT causing transient intermodulation distortion
with the higher freqs (and vice versa) in your power amp.
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
|Fig 9. Here's
the Marchand XM46SB PASSIVE Filter.
Here's how it's connected to a typical 2-channel
|Fig 10. The passive
filter used as "half" the crossover
So you roll off the mains
at some frequency, such as 80 or 90 Hz, 24 dB/octave; (you have
to purchase the frequency you want, since it is custom made, and
I HIGHLY SUGGEST 90 Hz, 24 dB/octave) and you set the low
pass filter in the sub the same way. If you want (for some reason)
to only use passive capacitors and inductors in your system, this
is one answer. Overall I do not necessarily recommend this though.
More modern solutions are FAR better. I only want to show the
option. Please be aware that your precious audio signal has gone
through MANY THOUSANDS of opamps from the microphone through
a myriad of 'stuff' in and out of computers or tape recorders
or both, and then THAT signal has gone through further opamps
in the mastering process, and so on. Using an active filter with
a few more opamps is not going to destroy your 'precious' audio.
To use an active filter (if
it has 2 or more sections we can now call it a CROSSOVER), there
are many choices some of which are each explained below.
There are the Bryston
crossovers, very handsome, built like a tank, and with a terrific
warranty... except the ordering options are quite complicated
and many people wind up getting the 10b 'standard' (which does
NOT have a 24 dB/octave setting) when the better choice would
be the 10b SUB, or LR. Then the crossover winds up on Audiogon
or Audiomart, because the user is frustrated. If you order the
SUB version or the third LR version then you must order separate
plug-in parts for different frequencies. Be very careful reading
their very complex user manual.
Many versions of the Marchand
(solid state, tube, balanced, unbalanced, 1 way, 2 way, rotary
knob, precision stepped attenuator...) are available.
|Fig 11. One variation
of a Marchand crossover (XM9) showing stepped volume control
Another choice for simpler
experimentation and budgetary concerns is the dbx 223 series,
|Fig 12. The dbx
223 xs crossover
Note the dbx has separate
models for use with XLR or Phone/RCA connectors. BOTH models are
balanced (but may be wired unbalanced) - only the connectors
are different. If you are intending to use UNbalanced RCA's then
you must get these RCA to 1/4" TS (Tip/Sleeve) adapters:
(you will need SIX).
|Fig 13. RCA to
1/4" Tip/Sleeve adapter
Here is the dirt simple front panel setup
for the dbx units:
|Fig 14. Dirt
simple dbx XO setup for 90 Hz
Putting the active, 2-way crossover in your
system is done like this:
|Fig 15. Showing
a typical active crossover in a "typical" system
Since ALL the filtering is
done IN the crossover, you turn OFF the [low pass] filter in the
sub. For fine level matching adjustments you typically have HIGH
and LOW output knobs on the crossover to play with.
There's also, at the higher
end, the Pass
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.
|Fig 16. The JL
Audio CR1 Crossover. (Click on any of the pix for a larger
Here are some of its UNIQUE
1) In the crossover mode, you can use either/and/or
RCA, Balanced XLR, or balanced or unbalanced 1/4" (Tip/Sleeve
Tip/Ring/Sleeve) input or output connectors at any time.
In the [hard] bypass mode (see the color
diagram above) of course the RCA input connectors connect directly
to the RCA output connectors, and the XLR input connectors connect
directly to the XLR output connectors.
2) The frequency controls are SEPARATE -
that is, you can set them the same, or overlap or underlap to
accomodate ANY preferences you might have.
3) The Bypass/On switch will give you the
PERFECT A/B comparison: in the "on"/operate mode the
sub is crossed over (as it should be); and so are the mains.
In the bypass mode, the mains are operating full range and there
is no sub, so you can FULLY and IMMEDIATELY appreciate exactly
the benefits of what the crossover (and sub) is doing in your
room. NO OTHER DEVICE is capable of this - and it is this A/B
demo feature that blows everyone's mind.
4) The bypass switch may also be used for
Home Theater bypass.
5) Separate HF and LF Damping controls give
you subtle and desirable control. This becomes the final "I
gotta add salt and pepper to the chef's creation" buttons,
because everyone really wants one more knob to turn! Or not.
6) The "balance" control enables
you to have less or more sub or mains for late night listening,
or to assist with mixes which could use a bit of "help".
It has a zero reference detent in the middle to easily return
exactly to normal.
7) MUTE buttons enable you to discern anything
- such as muting the mains to hear if the subs are vibrating
something in your room, etc. The separate L and R mains (satellite)
and sub mute switches also greatly assist in setup using my
unique TEST CD, here: www.soundoctor.com/testcd
8) yes, you can have separated "stereo"
subs if you want.
9) Note for you pristine analog fanatics:
There is NO "digital audio" in the CR1. The audio
path is completely analog.
...and there's much more! Here's the User
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!
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):
, and there is also integration with the Roon player, here:
The DEQX models are here
The DATASAT www.datasatdigital.com
DIRAC is here: www.dirac.com
...and the MiniDSP collection
of products, which include Dirac www.minidsp.com
There's SONARWORKS www.sonarworks.com
and TRINNOV www.trinnov.com
The AudioVero products include
ACOURATE and The ACOURATE CONVOLVER, here :www.audiovero.de/en/
The JUICE Audiolens is here
I have posted many more links
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
|Fig 17. The Bryston
B135 Integrated Amp with PRE OUT - PWR AMP IN connectivity
Then there's the issue of
all the other Integrated Amps which do NOT have the abovementioned
loop, but almost all of them DO have a "Tape Loop" -
a holdover from the days when people connected cassette decks...
So the issue there is the TAPE OUTPUT is at a full, fixed level,
taken off at an earlier stage, before the volume control. So you
CANNOT connect a regular crossover in that loop because the low
freq outs of the crossover will be at a fixed level (full), while
the high freqs only will be adjustable. So for the over a thousand
(!) people who contacted me to ask about this, there is really
only ONE WAY to accomplish the nearly impossible: The Marchand
XM9 or XM44 crossovers, which have SEPARATE LOW and HIGH frequency
level controls. And for wonderful convenience, these are available
with precision stepped attenuators with repeatable click positions.
This is what the front panel
with the separate click stepped level controls looks like.
|Fig 18. One variation
of a Marchand crossover (XM9) showing stepped volume control
This is how you would connect
it - you MUST use a crossover with separate high out and
low out level controls. The issue is if you try to use a crossover
with "regular" non-stepped controls, (that means a plain
old potentiometer) you will be frustrated trying to always match
and fine tune the levels.
|Fig 19. Connecting
a Marchand crossover in the TAPE LOOP
Now you set the maximum level
you want in the room by putting the four crossover levels all
the way up (L high, L low, R high, R low) and adjusting the main
volume control on the amp. Then you use the 4 controls on the
crossover to attenuate to the volume in the room to what
you want. This the ONLY way to accomplish this if you have a tape
loop only and can't insert a crossover between the premp out and
power amp in.
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 room.
OPTION 2) We
must introduce an equivalent delay TO THE TOP (mains) to match
the inherent delay in the sub; then we can super fine-tune the
match by using the phase knob on the sub.
Some notes about phase knobs:
If you have a (toggle) SWITCH on a sub labelled "phase"
that is wrong. It is not phase; it is POLARITY. Phase is ANY NUMBER
of degrees shifted, from 1 degree to 360 degrees to 3600 degrees
and so on. Polarity is either 0 degrees or 180 degrees, period.
(see Fig.1 and Fig.3 above) If you have a phase KNOB on
a sub, the circuit is usually designed to only ADD DELAY. You
cannot take away the inherent delay in the entire electro-mechanical
physics of the sub, but you can ADD further electrical delay.
Some subs are calibrated in electrical degrees of waveform at
80 hz, because 80 hz was the original suggested crossover freq
for Home Theater/Surround Sound systems. Therefore IF the knob
says 180 degrees it is actually adding 6.25 msec of delay to the
sub signal; this is the equivalent of moving the sub 7 feet FURTHER
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.
(see a longer list above) All of these are audiophile grade devices.
That means that UNLIKE all the "digital" speaker gadgets
intended for use in nightclubs and rock n roll systems, these
do NOT operate at 44kHz, (or even 48kHz) and you will NOT be disappointed
with what the "processing" has done to your precious
highs. Many of the so-called "professional" units are
perfectly suitable for a noisy bar or a rock touring PA system
but you might be very disappointed if it is your intent to use
them in a critical audiophile listening/monitoring situation.
That means beware of $99 - $299 processors. But even if you DO
get a very inexpensive processor, say on ebay or Craigslist, etc.,
the learning experience is well worth it, if you have the patience.
In the instance of Home Theater
processors, there is an easy method. We can take advantage of
the somewhat flawed concept of "speaker distance settings"
to perfectly fix the sub timing issues. Simply set ALL the top
speakers ( L C R Ls Rs) to 7 feet where they belong, and
set the sub distance to 18-19-20 feet. Now, because all consumer
equipment operates backwards (!!!) you are introducing 10-12 msec
delay TO ALL THE TOP SPEAKERS. Now you can fine tune the phase
control on the sub to add a bit more delay to the sub to perfectly
match the mains and the results should be spectacular. My test
CD and the two different procedures to accomplish this are all
carefully explained here: www.soundoctor.com/testcd.
Once you correctly place the
sub(s) in your room so they correctly couple to your desired area,
cross over the mains to the sub correctly, and fix the timing
issue your results will be everything you hoped for.
IN SUMMATION (pun
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 above
about using 2 subs. Notice that humans actually LIKE the (slight)
fattening up of the bass loudness envelope in time. Therefore
even IF your sub is 12 msec late, and you are one wavelength off,
as long as you get that delayed wavelength to line up with the
bass coming out of your mains, your frequency response
will be pretty good and you won't have any awful objections, again,
assuming you get as much else right as possible.
Enjoy your audio journey!
And let me know your results!