Harvey Thread

Harvey Gerst’s Mic Lessons
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Lesson 1
Posted by Harvey Gerst on 05-03-2001 09:23
Okay, let's start this with some interesting history as a prelude to the whole mic discussion. "Why" will become
pretty clear by the third or fourth paragraph:
In a way, the history of microphones and sound all started with Alexander Graham Bell, and Western Union. After
Bell won the lawsuit with Western Union over the invention of the telephone, his fledgling AT&T company
needed somebody to manufacture phones for them. Western Union had created a manufacturing division (Western
Electric) to make telegraph keys and telegraph equipment. Bell bought the Western Electric division and they had
the exclusive right to manufacture phones for Bell.
By 1910, Western Electric had the ambitious task of creating a coast to coast telephone hookup to tie in with the
opening of the Panama Canal, but the problem of amplifying a signal over long distances was still unsolved. In
1913, Dr. Harold Arnold (of Western Electric's research group) saw that Dr. Lee DeForest's "Audion vacuum tube"
was the possible solution, and they bought the rights to it and began work on a "high vacuum" tube.
This indeed solved their long distance problem, and led to another discovery - a "loud-speaking telephone". In
1916, they received a patent for what we now call a "loudspeaker". With the addition of the "high vacuum"
amplifying tube, and another little patent for a device called a "condenser mic", they were suddenly in the P.A.
business as well.
These inventions opened the door for radio, talking movies, and sound systems in general, and with their other
patent for a high quality "amplifier" in 1916, they pretty much defined the science of sound. It would be another 12
years (1928) until a young George Neumann would start his own mic company in Germany. That same year,
Western Electric received a patent for a "dynamic mic" design.
The designs Western Electric developed for movie speakers would eventually start companies like Altec and JBL
making horns and loudspeakers for Western Electric, and eventually those Western Electric designs became the
foundation for their own speaker lines.
Western Electric created their own Research and Development arm called "Bell Laboratories", which went on to
create the transistor and a host of audio related products. It was Western Electric and Bell Laboratories who we
must thank for the development and research into microphone design that we enjoy today.
Next, we'll look at some of the different types of microphone designs in terms of advantages and disadvantages.
How a "dynamic" mic really works will definitely surprise you (hint: it's NOT just a small speaker in reverse).
>
Lesson 2
Dynamic Mics
By far, the most popular mic on the market today is the dynamic cardioid mic, so that's as good a place as any to
start. "How does it work, what exactly is a cardioid, and how and where would you use it" will be our focus today.
Let's look inside one and see what we find:
Well, it has a cone (like a small speaker), a voice coil (like a small speaker), and it sits in a magnetic gap (like a
small speaker), so isn't it just a small speaker in reverse? Yes, and no. The operating principle is the same, but the
execution is very different. When's the last time you saw a 3/4" speaker that went down to 30 or 40 Hz? Here's
how it's done:
The system resonance is chosen for a mid band frequency. By itself, the capsule's response looks something like
this:
......./\
....../..\
...../....\
..../......\
.../........\
../..........\
./............\ - just one big resonant peak, with the response falling off rapidly on each side of the peak. Now you can
tame that peak by putting in a resonant chamber that's tuned to that peak, which will give you two smaller peaks on
either side, like this:
..../\..../\
.../..\../..\
../....\/....\ And if you add two more resonant chambers, tuned for each or those peaks, you wind up looking more
like this:
./\../\../\../\
/..\/..\/..\/..\ And if you make the chambers a little more broad band, the response starts to really flatten out:
._..._.._..._
/..\/..\/..\/..\ But remember, it's still a lot like a bunch of tuned coca cola bottles inside there.
Updated Graphic
Now ya gotta do all of this stuff JUST to get the response usable - never mind about the mic pattern yet!
A lot more to come!! Everybody still with me at this point? Any questions?
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Posted by Chris F on 05-05-2001 00:12
So, if I understand your first post, you explain that the frequency response of your basic garden variety dynamic
mic is not really a curve, but rather a series of mechanically engineered peaks, right? But we don't necessarily hear
it that way, because our ears/brains fill in the sonic spaces the same way our eyes/brains do when we look at a
newspaper photo that really consists of a bunch of dots rather than an actual picture. Is that pretty close?
Is the reason for that the size of the diaphragm? It would make sense that, in order to truly reproduce a sound in the
extreme low register, the diaphragm would need to be as large as the soundwave corresponding to the lowest note
on the recording, which would be both incredibly impractical and terribly funny....can you see Roger Daltrey
swinging one of those suckers around? So instead of that, the initial response peak is spread out so that it covers
more range more evenly.
Am I close, or did I get off track by assuming too much?

>
[QUOTE]Originally posted by Chris F
So, if I understand your first post, you explain that the frequency response of your basic garden variety
dynamic mic is not really a curve, but rather a series of mechanically engineered peaks, right? But we
don't necessarily hear it that way, because our ears/brains fill in the sonic spaces the same way our
eyes/brains do when we look at a newspaper photo that really consists of a bunch of dots rather than an
actual picture. Is that pretty close?
Yes and no. The broadband resonators and filters actually do smooth out those peaks pretty well, but you hafta
remember it's all done with mechanical tricks and if you hit it with enough energy in a susceptible frequency range,
it will resonate.
[QUOTE] Originally posted by Chris F

Is the reason for that the size of the diaphragm? It would make sense that, in order to truly reproduce a
sound in the extreme low register, the diaphragm would need to be as large as the soundwave
corresponding to the lowest note on the recording, which would be both incredibly impractical and terribly
funny....can you see Roger Daltrey swinging one of those suckers around? So instead of that, the initial
response peak is spread out so that it covers more range more evenly.
No, that's also a function of excursion and mic design. Small omnis for example, can get down to 1 Hz fairly
flat.

Am I close, or did I get off track by assuming too much?
A little too much assumption which I'll try to explain in the next installment.
>
[QUOTE] Originally posted by seriousturtle
Do you like the sound of the TLM-103 for vocals, which is what I assume you use it for? how much
different does it sound than the U87? keep preaching about the response of the mic, cuz I'm interested in
how an unflat response is considered good. since I hear crap about how you should always go for the flat
response.
Wow, difficult to answer, but I'll try.

I've used the 103 for vocals, acoustic guitar, mandolin, fiddle, and a host of other instruments. It's a very clear
sounding mic with a nice mid range warmth, and no shrillness. I'm not a big fan of the U87 but a lot of people love
them.
Finally, "flat" is hard to define. The TLM103 is flatter than a U87, but that's not the point. You want to go for the
most flattering mic for a particular instrument, but it must be a mic that doesn't add unpleasant coloration.
The flattest mics are/were made by B&K for test measurements and they're ruler flat (literally) usually from about
10 Hz to around 30 or 40 kHz. They are also pretty boring as mics for recording most music.
>
Lesson 3
Well, I haven't heard from David Satz as far as permission to reprint his post here, but I don't think he'll mind
(since he's a nice guy), so here it is:
RockyRoad wrote:
Could some kind person explain to me how the physics of these things work, and how sound from behind an omni
mic such as the KM183 can get around the metal side casing and into the mic.
Sorry for the dumb questions, but I'd like to know why things aren’t as they seem on the surface.
"Why are things not the way they seem?" is a question that I _so_ wish people would ask more often than they do.
Most folks seem to stop noticing that things aren't the way they seem, and start behaving as if that appearances are
all that matter. To me that's the essence of that form of spiritual death which we in this society call "adulthood." It's
why I believe that only children should be allowed to vote or own property--but failing that, there should be a law
(or better yet, a general agreement) that grown-ups ought to answer all honest questions honestly. Then maybe we
would not be such a culture of deception and self-deception, and people would retain their ability to notice things
that don't make sense.
The replies from Sean and Scott are spot on, but I'd like to try to help you visualize what these two types of
microphone are doing. Again, the relevant categories are "pressure transducer" (basically omnidirectional) and
"pressure gradient transducer" (basically figure-8, but by using dual diaphragms and other tricks, any other first-
order directional pattern can be synthesized including cardioid and super- or hypercardioid).
The model of a pressure transducer is a barometer. It measures air pressure in the space around it. The simplest,
grade-school science barometer is a sealed tin can with air in it. The lid of the can will flex in proportion to air
pressure changes in the room around it; you can attach a stick to the lid, and calibrate the stick's motions in terms
of whatever units of air pressure you want to use (inches of mercury or the standard metric unit, which is "bars").
The thing is, the can will get squeezed by increasing air pressure or it will expand in times of low air pressure,
regardless of which way you "aim" it. In fact the concept of "aiming" a barometer doesn't really exist because it's
integrating and responding to a phenomenon that is all around it. You just set it up in whatever physical orientation
is convenient for you, and it works.
You could think of the barometric pressure in a daily weather report as being the response of the barometer at
0.000011574 Hz if you want (one cycle per day). Essentially a barometer is a microphone with response down to
DC. And that is a real-world characteristic of pressure transducers: their low-frequency response can be extended
as far down as you like. Most pressure microphones have some small vent built in to prevent them from bursting
when transported by air, but they can very well be dead flat to below 1 Hz or 5 Hz, certainly to any audible
frequency.
OK. So the Pressure Transducer works precisely because only one side of the diaphragm (the lid of the can) is
exposed to the air pressure that is to be recorded; the air on the other side of the diaphragm is a constant mass, and
the diaphragm flexes in order to equalize the pressure on both its sides.
The other major category of transducer is Pressure-Gradient, which is a fancy way of saying that its diaphragm is
exposed to the sound field both on the front and the back, so it responds to the difference between the pressure that
exists on the front and the pressure on the back. If the pressure presented on both sides at a given moment is
identical, there is no net motion and no output. If the pressure on the front is greater than the pressure on the back,
the diaphragm will move toward its backplate (assuming a condenser microphone). If the opposite is true, the
diaphragm will move outwards, away from the backplate.
The thing is, if you just hang a microphone diaphragm out in space, it will be pushed around by wind or by air
currents of any kind (including if you just blow on it) but it won't pick up much in the audio frequency band
because it's a thin element and the pressure from sound waves will tend to be identical on both sides of the
diaphragm, at least until you get up to the high frequencies (which we'll talk about some other day), and when the
pressure is the same on both sides of the membrane there is no net movement and no output. But before I explain
why this type of arrangement picks up sound at all, let's observe that we've actually encountered something that is
true of pressure gradient microphones generally, which is that they are much more sensitive to wind, breath noise
and "popping" of consonants in vocal pickup than their omnidirectional counterparts are (when the omnis are
pressure transducers).
The trick which makes a pressure-gradient arrangement work for recording sound is that the sound reaching the
back of the membrane is delayed momentarily, by setting up a delay chamber in between the back vents of the
microphone and the back of the diaphragm. If you can make the pathway for sound even just a tiny fraction of an
inch longer before the sound reaches the rear of the diaphragm, then you will cause a phase shift between the
sound reaching the front and the sound reaching the back. That phase shift will be different at different
frequencies, of course, so there will really be only one frequency (plus its exact integer multiples) at which a
maximum of difference in pressure will result between the front and back of the diaphragm. At that frequency the
resulting microphone will have its highest sensitivity to sound. But if you arrange things so that this frequency
occurs somewhere other than at the very top or the very bottom of the audio range, you can do other tricks with
damping and filtering so as to flatten the overall response.
The thing is, this more complicated type of microphone is also sensitive to the direction from which sound is
arriving, because if sound is arriving from in front, it will strike the front of the diaphragm immediately, then when
it reaches the rear input ports it will pass through the acoustic delay chamber and eventually reach the back of the
diaphragm--so there will be a continually varying difference in the air pressure on the two sides of the diaphragm,
and that's what moves it and produces a signal. But if the sound is coming from behind the microphone, it will
reach the back inlets first, and pass through the delay chamber at the same rate of speed as the original wave is
traveling outside the microphone; by the time both waves reach the two sides of the diaphragm, they will be in
phase with one another and the result is no net motion of the diaphragm. (That's if the microphone is a
single-diaphragm cardioid.)
That should be enough to establish a basic viewpoint, I hope. (End of David Satz' post)
And we'll now head into "when and why to use what, and how" when we do the next posting on this.
>
Let me try again, and please enlighten me (again) if I'm wrong. So far, I understand that:
Pressure transducer = Omni-directional

Pressure gradient = Condenser patterns such as figure 8, cardioid, hypercardioid etc.....
If I'm understanding this correctly, condenser mics need to be powered with "phantom power" because the pressure
that they are receiving on the front of the diaphragm is balanced by pressure on the back of the diaphragm. The
difference in pressure - caused by a "phase delay" that can be measured in microseconds - between the front and
the back of the diaphragm is very small, and for this reason, the signal needs to be "amplified" by an electric
charge.
Further, the directional "pickup pattern" is determined by the design of the diaphragm capsule - more to the point,
the pattern is determined by the amount of "delay" engineered in to the back of the capsule. Because if an equal
amount of pressure reaches both the front and the back of the diaphragm at the same time, it won't move at all and
there will be no sound picked up from the direction that caused this to happen. But sounds coming from any
direction that causes the diaphragm to have more pressure on one side or the other will be picked up because
they're slightly "out of phase".
Am I getting any closer? I have some other questions but I'll hold them until I feel like I understand this issue
better
Sorry if I seem dense, I'm just trying to fully understand some of the basic concepts before you move on to the
REALLY DEEP stuff. Could you give a couple of examples of some common industry standard mics, which relate
to the pressure transducer vs. gradient schism?
Thanks for your patience. I'll get it eventually (I hope!). Believe it or not, I am trying.
Chris
>
By George, you've almost got it!! Forget the condenser/phantom power part and you've got it.
A pressure gradient mic depends on delays getting to the back of the diaphragm, whether it's a ribbon mic, a
dynamic moving coil mic, or a condenser mic.
A condenser mic works by the difference in voltage between the back plate and the diaphragm. The voltage can be
either a permanent pre-charged voltage (an electret), or a capsule that has 48 volts across one side or more (some
B&K mics use over 100 volts to charge the capsule). The phantom power is only one way to get a condenser mic
to work - it has nothing to do with the patterns.
Other than that, you've got it!!! The delay from sound hitting the backside of the diaphragm of any mic results in
the different patterns. If the backside of the diaphragm is sealed, it's strictly a pressure mic, and it's omni - always!
Very good!!!
>
You also asked for some examples of different patterns:
Pressure (omni) - all calibration mics, Earthworks.

Cardioid - Shure SM-57, Neumann TLM103
Hypercardioid - Beyer M201, AT25
Figure 8 (Bidirectional) RCA 44BX
Condenser mics which use two back to back diaphragms can simulate several patterns by electronically combining
the two diaphragms in different configurations (e.g., combining a figure 8 pattern and an omni pattern results in a
cardioid pattern).
>
[QUOTE]Originally posted by Harvey Gerst
By George, you've almost got it!! Forget the phantom/phantom power part and you've got it.
Consider it forgotten.

A pressure gradient mic depends on delays getting to the back of the diaphragm, whether it's a ribbon mic,
a dynamic moving coil mic, or a condenser mic.
Two things:
a) I don't think I understand the difference between "Dynamic" and "Condenser" mic designs other than the fact
that Condenser mics are powered and Dynamics are not. I have noticed that dynamics are often used for
applications where the mic is going to be exposed to really high SPL's - like on snare drums played by muscle-
bound rockers using tree trunks for sticks (ex - SM57), kick drums played by same (AT? 25 - the little stubby one),
and diva singers (that was meant as a non gender-specific term BTW) belting out their dramatic s*** with the mic
halfway down their throat during live shows (SM58). I think I must have been mistakenly confusing "Dynamic"
with what you describe as "Pressure Transducer". Until now, I always thought of condenser mics
as being much more of an "ambient" sounding mic.
b) I just figured out during your last post that the word "Condenser" is spelled with an "or" at the end instead of an
"er". DOH! Better late than never.

A condenser mic works by the difference in voltage between the back plate and the diaphragm. The voltage
can be either a permanent pre-charged voltage (an electret), or a capsule that has 48 volts across one side
or more (some B&K mics use over 100 volts to charge the capsule). The phantom power is only one way to
get a condenser mic to work - it has nothing to do with the patterns.
Gotcha.

Other than that, you've got it!!! The delay from sound hitting the back side of the diaphragm of any mic
results in the different patterns. If the back side of the diaphragm is sealed, it's strictly a pressure mic, and
it's omni - always!
I think I get it. But then, how does an omnidirectional condenser work? And what is a common type of pressure
mic that I can think of to try to visualize the difference between a pressure transducer and a pressure gradient mic?
If I'm understanding you so far, most of the mics used in the recording industry are all pressure gradient mics,
right?
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[QUOTE]Originally posted by Kelly Holdridge
I'm especially interested in how B&K made a microphone with flat response. I'd imagine it's a mix of
various patterns and electrical tricks... Also, how much did it cost and was it a big deal when they
announced it?
Brüel & Kjær has been making measurement mics for about 50 years now. They're omnidirectional, and flat
(within a fer tenths of a dB) from about 5Hz to 40kHz, although they have some models that are flat down to 1Hz
and other models that are flat to about 140kHz. Every mic manufacturers uses B&K to see what their own mics are
doing; the test is very simple:
You point the B&K mic and the mic you wanna test at any sound source and record the two response curves. You
subtract the B&K results from the mic under test, and any differences from the B&K response - well, that's the
your mic's response curve.
They made two basic types of omni test mics - one for pressure field (on axis) and one for diffuse field (90 degrees
off axis). Their DPA web site (DPA is their name for their studio type mics) contains a whole bunch of good,
objective info about the differences between small and large diaphragm mics, but it's pretty techie oriented.
(Bottom line: Small Diaphragm mics - have higher noise, flatter response, and greater dynamic range.
Large Diaphragm mics - have higher output, lower noise, and less dynamic range and frequency response.)
Dick Rosmini in California was a big champion of using B&K test mics for recording and we usta have long
arguments about it, since I found them kinda boring.
>
Lesson 4
Let's review what we've learned so far.
Okay, no pop quiz today, but let's review some of the stuff we've gone thru so far. For most applications, the 3
basic mic designs are:
1. Condenser mics
2. Dynamic (moving coil)
3. Ribbon mics (a special class of dynamic mics)
The basic Polar patterns are:

1. Omni-directional (pressure)
2. Uni-directional (cardioid)
3. Bi-directional (figure 8)
4. Hyper-cardioid
True omni-directional mics have a sealed back chamber and only allow sound to hit the front of the diaphragm.
The other polar patterns are created by using "pressure gradient" techniques to delay and let some of the sound hit
the back of the diaphragm.
Condenser mics can be made in small (1/2" or smaller), medium (5/8" to 7/8"), and large diaphragm (1" and larger)
sizes. Small diaphragm condenser mics have these advantages:
1. Flatter, extended frequency response
2. Higher SPL levels
3. Better off-axis response
4. Greater accuracy
They have these disadvantages:
1. Lower output levels
2. Higher self-noise
Large diaphragm condenser mics have these advantages:

1. Higher output levels
2. Lower self-noise
They have these disadvantages:

1. Poorer limited frequency response
2. Lower SPL levels
3. Uneven off-axis response
4. Less accuracy
However, some of the resonances in a large diaphragm condenser mic can be very pleasing and musical, and can
often compliment the voice and some instruments very well.
"Pure" pressure mics do not have proximity effect (bass buildup as you get closer) – All pressure gradient mics DO
have proximity effect (dual diaphragm condenser omnis have the least, then cardioids, then hypercardioids, then
figure 8, which has the most proximity effect).
You would use small diaphragm condenser omnis where you want the greatest accuracy or in high level situations
where self-noise isn't a factor. Large diaphragm condenser mics are better used for quiet sources, or where you
want a particular type of complimentary coloration.
We'll get into choice of mic, patterns and placements in the next installment.
>
Lesson 5
A few more things to think about
OK, before we get into what mic to use for what purpose, and where to place it, here are a few more things you
hafta be aware of. One is called "musical instrument radiation patterns" and the other is "near-field placement vs.
far-field placement".
The most common question here is "how do I mic an acoustic guitar?", followed by vocal mic techniques. Let's
look at the first question because it's more complex than it appears and it's actually made up of two parts.
Musical instrument radiation patterns
Guitars, violins, stringed instruments, in fact, all instruments radiate notes differently at different frequencies!!
Read that again: Guitars, violins, stringed instruments, in fact, all instruments radiate notes differently at different
frequencies!!
What does that mean exactly? It means that different parts of the instrument's body are used to produce different
notes! Just pointing a mic at a guitar is no guarantee that you'll get what you want. Unless you understand how
guitars generate sound, the best you can hope for is to somehow get lucky. Here are two links that show how the
guitar top changes with each note:
Chladni Guitar Top Radiation Patterns:

http://www.phys.unsw.edu.au/music/guitar/guitarchladni_engl.html
Radiation Patterns:
http://www.astro.cf.ac.uk/groups/acoucomp/MAGResMeth.html#fotos3
As you can see, different notes come from different places on a guitar, which brings us to the next section:
Near-field placement vs. far-field placement
Ok, so what the hell does that mean? Well, let's do a thought experiment to illustrate this concept:
Think of a tall column of speakers - about 6 feet tall, with woofers on the bottom, midrange speakers in the middle,
and tweeters at the top. Now imagine that you walk right up to it and put your ear about 4" away from the system;
what will you hear?
If you answered that it depends on whether your ear is near the tweeters, mids, or woofers, you're absolutely right.
So where would you hafta stand to hear the whole system evenly balanced? At least 6 feet away is the correct
answer - and that 6' away point is the boundary between the "near-field" and the "far-field" in this example. Any
closer than 6 feet and you don't hear the whole sound, because you're in the "near field".
Now let's look at a typical acoustic guitar. The body is about 2 feet across. Put a mic any closer than 2 feet and
you're in the "near field" of the guitar, and those two links I posted show you that you will be hearing uneven
sound, depending on the note being played.
So, The First Rule To Remember Is: "The near field distance is defined as being equal to the length of the longest
part of the vibrating section of the instrument."
The Second Rule To Remember Is: "Inside the near field of an instrument, the sound will change drastically with
different mic placements".
We'll get into mic choices, polar patterns, and mic placements in our next installment, but this "radiation pattern"
and "near-field" vs. "far-field" stuff is really important to remember when you're trying to get a good instrumental
sound.
>
[QUOTE] Originally posted by Harvey Gerst
If you answered that it depends on whether your ear is near the tweeters, mids, or woofers, you're
absolutely right. So where would you hafta stand to hear the whole system evenly balanced? At least 6 feet
away is the correct answer - and that 6' away point is the boundry between the "near-field" and the "far-
field" in this example. Any closer than 6 feet and you don't hear the whole sound, because you're in the
"near field".
This makes perfect sense, but it also brings up a couple of points for later if I understand you right:
a) The person who is playing the instrument is in the near field, and if they're an accomplished musician it means
that they've been practicing for years and years in the near field, which would mean that there's at least a decent
chance that the person creating the music doesn't really know what the instrument really sounds like to someone
else when they're playing it. And a microphone is the proverbial someone else in this situation. This might explain
why many acoustic musicians become so confused/disconcerted when asked to wear phones in the studio…
Because they are accustomed to hearing only from a certain place in the near field and reacting to that, and all of a
sudden someone has moved their "ears" to another location by making them wear headphones. Either that, or I'm
making this stuff up as a rationalization for why I hate recording with phones on… It makes me feel as if I'm
suddenly not playing the same instrument anymore.
b) If you get further away than 6 feet, you might be hearing "the whole sound", but you're also hearing more than
that, because the further you move away from the sound source, the more your listening environment (i.e. - room
or hall) is coloring your sound with reflections of some sort. So this is why the speakers mounted on either side of
the desk in most studios are called "near field monitors", right?

Now let's look at a typical acoustic guitar. The body is about 2 feet across. Put a mic any closer than 2 feet
and you're in the "near field" of the guitar, and those two links I posted show you that you will be hearing
uneven sound, depending on the note being played.
So, the first rule to remember is: "The near field distance is defined as being equal to the length of the
longest part of the vibrating section of the instrument."
A couple of questions related to (a) above:
For stringed instruments of the fretless variety, is the sound of the string vibrating on the fingerboard (i.e. - the
"growl" of the low notes on a Double Bass) also part of this "near field distance", or do you only count the length
of the body itself? I'm only asking because this would change the length of "near field" somewhat.
For a grand piano, do you count the length of the longest (bass) portion of the soundboard? And how do you
decide whether to mic the top or bottom of it?

The second rule to remember is: "Inside the near field of an instrument, the sound will change drastically
with different mic placements".
Gotcha. This is making more and more sense. Thanks.

We'll get into mic choices, polar patterns, and mic placements in our next installment, but this "radiation
pattern" and "near-field" vs. "far-field" stuff is really important to remember when you're trying to get a
good instrumental sound.
So any time you mic in the near field, you're really getting an incomplete sound, and if you use only one mic, or
two or more mics placed more that about 5" apart, you're recording an "artificial" or "manufactured" sound since
your ears could never pick up that sound in a natural acoustic setting. How would the "polar pattern" of a set of
human ears be described using general microphone terminology? Stereo omnidirectional?
Thanks again for your time...this stuff is not only fascinating, but also addictive!
>
Posted by c7sus on 05-14-2001 09
What I wanna know is why the field of the instrument is the width (length?) of the body and not the length of the
string...which as you play changes constantly....
Now I may be getting way ahead of myself...but if the field is 2 feet out from the front of my guitar, wouldn't I
want to use the tightest pattern possible to capture just the sound of the instrument and not an omni that is gonna
pick up the room… Then, since I'm 2 feet from my source, won't I start running into S/N problems....
Maybe time for a little experiment....
>
Posted by The Axis on 05-14-2001 10:52
Near Field...
C7sus:
There is no sharp dividing line between the "near" and "far" fields. The differences just gradually fade into
obscurity. That is why it is somewhat irrelevant what the length of the string is. The entire guitar body is always
vibrating, including neck, headstock, strings, and body top, sides, and back…
My experience on acoustic guitar is that omnis sound more natural because they are more similar to the human ear,
which is *closer* to omni than cardioid. They are also generally flatter (more accurate) in frequency response. If
you move your head around in front of someone playing guitar, the tone does change somewhat, but it is not the
same dramatic differences as when you move a cardioid mike around slightly (wild variations in tone).
Even well inside the "near field" (4 - 8 inches) an omni sounds much more natural than a cardioid. Then the S/N
ratio problem is solved because the guitar is so much *relatively* louder than any other noise in the room. Having
said that though, I keep my mikes about 14-18" out, because it does allow the sound to come together a little better.
An experiment is a very good thing on this phenomenon. Many people (myself included) have been virtually
brainwashed into thinking that cardioid is the only way to go. A brief experiment with a good omni will blow your
mind!
Rick
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Posted by Chris Shaeffer on 05-14-2001 15:56

Another consideration after looking at the resonance maps...
I would imagine that different instruments resonate differently- obviously- and we can easily here the different
timbres of different acoustic instruments. Different guitars can even sound vastly different.
Question is do similar instruments project the same frequencies from the same areas? For example, do ALL guitars
tend to sound bassy in the same area?
Seems like this would be handy stuff to know when deciding where to start with near-field miking. I can also see
that the different mike patterns will interact with different areas of sound projecting from the instrument...
I'm imagining a figure-8 pattern mike placed right between a guitar and a reflective wall that are 4 feet apart with
the guitar facing the wall.
This is really neat stuff to think about! I'm looking forward to applying it (and seeing how well/poorly those ideas
work!)
Take care,
Chris Shaeffer
>
Posed by Harvey Gerst on 05-14-2001 17:54
Chris, you sure do ask a lotta questions!
Originally posted by Harvey Gerst

If you answered that it depends on whether your ear is near the tweeters, mids, or woofers, you're
absolutely right. So where would you hafta stand to hear the whole system evenly balanced? At
least 6 feet away is the correct answer - and that 6' away point is the boundry between the "near-
field" and the "far-field" in this example. Any closer than 6 feet and you don't hear the whole sound,
because you're in the "near field".
This makes perfect sense, but it also brings up a couple of points for later if I'm understanding you right:
a) The person who is playing the instrument is in the near field, and if they're an accomplished musician it
means that they've been practicing for years and years in the near field, which would mean that there's at
least a decent chance that the person creating the music doesn't really know what the instrument really
sounds like to someone else when they're playing it. And a microphone is the proverbial someone else in
this situation. This might explain why many acoustic musicians become so confused/disconcerted when
asked to wear phones in the studio.... because they are accustomed to hearing only from a certain place in
the near field and reacting to that, and all of a sudden someone has moved their "ears" to another location
by making them wear headphones. Either that, or I'm making this sh*t up as a rationalization for why I
hate recording with phones on.... it makes me feel as if I'm suddenly not playing the same instrument
anymore.
That's probably a part of it, and it also explains why a lot of guitarists like my over the shoulder technique of guitar
miking.
[QOUTE] Originally posted by Chris F

b) If you get further away than 6 feet, you might be hearing "the whole sound", but you're also hearing
more than that, because the further you move away from the sound source, the more your listening
environment (i.e. - room or hall) is coloring your sound with reflections of some sort. So this is why the
speakers mounted on either side of the desk in most studios are called "near field monitors", right?
Yes, as you move further back, more of the room comes into play. That's also when you start changing polar
patterns to compensate, or to use more of the room sound (but that's all covered in the next installment). With
regards to monitors, yes they are in "your" nearfield so you hear them before you hear any room reflections.
[QOUTE] Originally posted by Chris F

Now let's look at a typical acoustic guitar. The body is about 2 feet across. Put a mic any closer than
2 feet and you're in the "near field" of the guitar, and those two links I posted show you that you
will be hearing uneven sound, depending on the note being played.
So, the first rule to remember is: "The near field distance is defined as being equal to the length of
the longest part of the vibrating section of the instrument."
A couple of questions related to (a) above:

For stringed instruments of the fretless variety, is the sound of the string vibrating on the fingerboard (i.e. -
the "growl" of the low notes on a Double Bass) also part of this "near field distance", or do you only count
the length of the body itself? I'm only asking because this would change the length of "near field"
somewhat.
Since the body accounts for the bulk of the instrument's radiating energy, I tend to just consider the widest part of
the body itself, unless you're close miking.
For a grand piano, do you count the length of the longest (bass) portion of the soundboard? And how do
you decide whether to mic the top or bottom of it?
Piano is one of the hardest instruments to record and I planned to go into that in greater detail later. For now, I
consider the widest dimension as the length, and I mic from out front if it's a solo instrument, or in close if it's to sit
in a mix.
The second rule to remember is: "Inside the near field of an instrument, the sound will change
drastically with different mic placements".
Gotcha. This is making more and more sense. Thanks.
We'll get into mic choices, polar patterns, and mic placements in our next installment, but this
"radiation pattern" and "near-field" vs. "far-field" stuff is really important to remember when you're
trying to get a good instrumental sound.
So any time you mic in the near field, you're really getting an incomplete sound, and if you use only one
mic, or two or more mics placed more that about 5" apart, you're recording an "artificial" or
"manufactured" sound since your ears could never pick up that sound in a natural acoustic setting. How
would the "polar pattern" of a set of human ears be described using general microphone terminology?
Stereo omnidirectional?
Yes, but that "artificial" or "manufactured" sound is not always bad, if it works better in the mix.
Ears are basically pressure transducers with increased directionality at higher frequencies.
Thanks again for your time...this stuff is not only fascinating, but also addictive.
It's a fascinating subject, and the more you learn, the more you realize how much more there is to go.
>
[QUOTE] Originally posted by c7sus
What I wanna know is why the field of the instrument is the width (length?) of the body and not the length
of the string…. which as you play changes constantly....
Some people do use the whole string length to calculate the near field distance. I don't since the radiating area of
the neck isn't very big once you're out to the body distance. It adds a bit, just not that much, IMO
Now I may be getting way ahead of myself.... but if the field is 2 feet out from the front of my guitar,
wouldn't I want to use the tightest pattern possible to capture just the sound of the instrument and not an
omni that is gonna pick up the room.... then, since I'm 2 feet from my source, won't I start running into S/N
problems....
Maybe time for a little experiment....

Ahhh, the light is coming on. Yes, when you're out of the nearfield, you start thinking about other mic polar
patterns. There's where you might wanna switch to larger diaphragm mics (remember condenser diaphragm size
determines output level, all else being equal). But you give up some accuracy when you trade off to a larger size.
>
[QUOTE] Originally posted by Chris Shaeffer
Another consideration after looking at the resonance maps...
I would imagine that different instruments resonate differently- obviously- and we can easily here the
different timbres of different acoustic instruments. Different guitars can even sound vastly different.
Question is do similar instruments project the same frequencies from the same areas? For example, do
ALL guitars tend to sound bassy in the same area?
No they don't all radiate exactly the same, and that's what makes it so frustrating and individual - everything must
be decided on a case-by-case basis. But there will be some similarities and that can help. Like the bass boom
coming from the sound hole.
Seems like this would be handy stuff to know when deciding where to start with nearfield miking. I can also
see that the different mike patterns will interact with different areas of sound projecting from the
instrument...
I'm imagining a figure-8 pattern mike placed right between a guitar and a reflective wall that are 4 feet
apart with the guitar facing the wall.
This is really neat stuff to think about! I'm looking forward to applying it. (and seeing how well/poorly
those ideas work!)
You hafta be very careful with figure 8 patterns, since it's impossible to predict easily whether the reflections from
a hard surface will help or cancel some other critical frequencies. But we'll get into that in the next installment.
>
Lesson 6
Fortress Of Doom
Ok, one more quick review and we're off into placement and mic choice land:
1. Small mics generally tend to be more accurate than large mics. Large mics are generally more flattering than
small mics.
2. Omni mics generally have the greatest accuracy but the smaller most accurate omnis have a higher self-noise
level, but they can handle higher SPLs as well.
3. Pressure gradient mics (cardioid, hyper-cardioid, and figure 8) use delayed sound coming into the backside of
the diaphragm to create their patterns.
Any mic can be used to record any sound

Think about that for a minute. A mic doesn't care what it's recording, but some mics are better suited for certain
things. If you don't have the one "perfect" mic for the job, any mic will work, with some trade-offs in sound
quality. Understanding what those "trade-offs" are is the hard part.
How does this part fit into the song?
Does it need to be miced in stereo or will mono work? Is it gonna be an upfront part or does it hafta blend in? Is
there another instrument operating in the same frequency range that might cause a conflict? The right choice of
mics can help in all these situations, but you hafta make sure BEFORE you set up any mics just exactly what it is
that you hope to accomplish. Here is how I think about where a part fits in:
Notice there are 3 important elements to this box:
1. Panning (Left to Right) - Very useful for separating instruments that occupy the same frequency range.
2. Level (Front to Back) - In combination with reverb, this creates the illusion of near and far, and can also
separate instruments in the same range.
3. Frequency (Lo to Hi) - the most overlooked aspect of getting a good blend when you're first starting out.
Most people solo a track and then work to get a killer sound (bass, guitar, whatever), then move on to the
next track. Wrong way to think. Instead, think about the song; which instrument should cover the bottom
octave, electric bass or kick drum? If it's the kick, roll off some of the bottom on the electric bass and listen
to make sure the two instruments aren't fighting for the same space.
Is the vocal important? Put it in the center, right up front. Are the guitars conflicting with the vocal? Move them
out of the way with the pan control.
But what if it's just a solo guitar or piano track? Ahhh, there's where you need to decide if a stereo recording would
be best. If it's an accompaniment to a vocal, a stereo-recorded guitar or piano can sound very nice contrasted with a
mono vocal.
Sorry this has turned into a rambling diatribe, but these are things that people tend to overlook in their haste to
record stuff. But it's exactly this stuff that determines what mic, polar pattern, and placement you should be using -
before you even plug in the first mic. We'll cover exactly that part next - I promise.
>
That last post was very elucidating. The one part I'm not sure I understand is the "near-far" or "depth"
part - is this aspect actually created by micing certain layers closer in the "near field" than others, or is
this some kind of illusion produced by mechanical means?
Near-far "depth" is achieved by using a combination of close and distant miking techniques AND the judicious use
of reverb to place instruments at different distances in the mix.
For example, you may wanna record the vocals at "point blank" range and add just a touch of a cathedral reverb, so
that the vocal is still up close, but you get a sense of a larger room. Strings might require that you mic from further
away, roll off a little bit of the top end and add more reverb to simulate how they would sound if they were in a
large room.
You control depth of field by careful use of mic placement, final mix level, and reverb when planning the mix. Just
remember that heavy reverb tends to blur the sound of the instrument, so don't overdo. I know of one engineer who
worked for two weeks just on the fine-tuning of the EQ on the reverb for the snare.
>
Posted by Chris Shaeffer on 05-16-2001 23:48
1. Small mics generally tend to be more accurate than large mics. Large mics are generally more flattering
than small mics.
Now that I see WHY this is true (smaller diaphrams move more quickly in response to the details of a sound) it
gets me thinking.
1) "Accurate" doesn't always mean "Sounds better" and "Flattering" doesn't mean "Accurate." I know that seems
like a 'duh!' statement but it seems important.
2) I wonder what it is about human hearing that makes certain kinds of "less accurate" sound better to us. It
reminds me of the difference between consumer stereo speakers (flattering) and monitors (accurate.)
3) I also notice that there doesn't seem to be any measurement that quantifies the accuracy or speed of
responsiveness of a mike- frequency response isn't really the same thing. Also, no measurement that I know of for
that elusive term "color." Is there more to the "color" of a mike (or amp, or, speaker, or...) than the frequency
response?
and finally....
4) 1-3 above seem like clear proof to me that choosing a mike and placement that sounds the "best" for a given
source is going to involve knowledge of mike and sound source characteristics (which we seem to have covered
*really* well), some ideas of where to start with placing those mikes, creativity, perseverance, and luck. I suppose
that experience can make up for the lack of luck.
It makes me want to take REALLY good notes about how I get the sounds I like- as well as the sounds I don't like.
This seems like an art and science that deserves a lot of attention. I'm beginning to see every mike/sound
source/placement experiment as time well spent even if it fails. Good stuff to have in one's head.
Thanks, everyone. I do believe we are creating a really valuable bundle of information here. And thanks, Harvey,
for fueling this fire so well. (How's your back, by the way? I'm sure I speak for all in that I hope you're feeling
better.)
take care,
Chris Shaeffer
>
[QUOTE] Originally posted by Chris Shaeffer
Now that I see WHY this is true (smaller diaphragms move more quickly in response to the details of a
sound) it gets me thinking.
1) "Accurate" doesn't always mean "Sounds better" and "Flattering" doesn't mean "Accurate." I know that
seems like a 'duh!' statement but it seems important.
You've got it!! That's also why no one mic can do it all.
2) I wonder what it is about human hearing that makes certain kinds of "less accurate" sound better to us.
It reminds me of the difference between consumer stereo speakers (flattering) and monitors (accurate.)
Don't think of it as "less accurate"; think of it as "more flattering".
3) I also notice that there doesn't seem to be any measurement that quantifies the accuracy or speed of
responsiveness of a mike- frequency response isn't really the same thing. Also, no measurement that I know
of for that elusive term "color." Is there more to the "color" of a mike (or amp, or, speaker, or...) than the
frequency response?
Actually, there ARE measurements that will measure the accuracy, but it's a lot more technical, and doesn't really
help this discussion. But in answer to that question, yes, there's more to it than just frequency response. Remember
the pictures of the guitar top vibrating at different places with different frequencies? Well, mic capsules do that
too. Capsule tensioning, damping, thickness, mass, stiffness, and excursion all affect the sound of the mic.
and finally....
4) 1-3 above seem like clear proof to me that choosing a mike and placement that sounds the "best" for a
given source is going to involve knowledge of mike and sound source characteristics (which we seem to
have covered *really* well), some ideas of where to start with placing those mikes, creativity,
perseverance, and luck. I suppose that experience can make up for the lack of luck.
Yup, and that's exactly what the next section of this overly long diatribe is going to get into. Some of you (like
Chris) are beginning to see why all the stuff we covered early on in this thread is going to be more important than
you first realized.
It makes me want to take REALLY good notes about how I get the sounds I like-as well as the sounds I
don't like. This seems like an art and science that deserves a lot of attention. I'm beginning to see every
mike/sound source/placement experiment as time well spent even if it fails. Good stuff to have it one's head.
It's 50% art, 50% science, and 50% luck.
Thanks, everyone. I do believe we are creating a really valuable bundle of information here. And thanks,
Harvey, for fueling this fire so well. (How's your back, by the way? I'm sure I speak for all in that I hope
you're feeling better.)
3 visits to the chiropractor, and my right leg is still killing me (and he's closed today). As soon as the pain goes
down a little, I'll try to write this next section about mic choices, patterns, and placements for as many instruments
as I can think of.
Some great insights, Chris (and from the rest of you as well). Just hang in there gang, we're almost thru. This last
part will be a series of multiple posts, since we'll be covering so many techniques and instruments (in as much
detail as possible).
>
[QUOTE] Originally posted by manchild
Harvey,
Is it true that mic frequency plays apart of the source that you are recording? And that you are much better
off using a mic that is as close to the frequency of the source you are trying to record? ie. piano, singing
voice, drums, guitar, flute, etc, etc.....
Yes, and no!! The mic's frequency characteristics are of course a vital part of deciding which
mic to use where, but it's often a choice of complimenting instead of capturing. In other words,
sometimes you use a mic to flatter and enhance the sound, not because it has a similar range
or is the most accurate choice.
>
Lesson 7
Ok, here we go,
Micing an acoustic guitar
Many acoustic guitars today have built in pickups, and it's gonna hafta be your choice whether you add that to the
mix or not - that's a whole 'nother subject. Before you reach for a mic, you hafta decide a few things:
Is it a solo guitar, strictly as a backdrop for vocal, or is it one part of a group mix (where there'll be other
instruments like drums and bass and electric guitars going on)? Does it need to be recorded in stereo or is mono
ok? Is it gonna wind up being in your face, or buried in the mix?
Solo Guitar or Background For Vocal?
If it's a great sounding guitar, and you have a good room, you want to use the best mics you have and record in
stereo. You can use omnis, or a pair of good cardioids in an X/Y configuration (capsules almost touching, angle of
about 110 degrees between the two mics) and about two feet out from the instrument.
A dynamic or condenser mic will work fine as long as the mic has a fairly smooth response. Smaller condenser
mics are usually more accurate, but if it's not a killer instrument, don't be afraid to try large diaphragm mics to get
a more flattering sound. The mics should be pretty closely matched otherwise the stereo image can shift as you
play different notes.
If the sound ain't working for you, that's the time to move in closer and see if you can find spots nearer the guitar
that produce a better tone (even if it's just for that song). Try to get as close as possible to the final sound you want
BEFORE you reach for EQ and/or effects.
After you get the tone damn near perfect from placement and selection, then do a little touchup with the EQ to nail
it. (If you hafta boost or cut more than 4 dB in any frequency range, you either haven't got the placement right yet,
or it's a really crappy guitar.)
Acoustic Guitar as a Rock Track With a Band
You need a tone that's gonna cut thru the other instruments and if there's gonna be drums, bass, electric guitars
going at the same time, record the guitar on the thin side (some bass cut and treble boost). Make it brighter than
you normally like it, and don't worry about how it sounds soloed - it's how it sounds when it's all mixed that will
count. I usually mic in close (about 6 to 8"), from slightly below, looking up directly at the bridge. Roll off the bass
below 100 Hz, and boost around 2 to 4 kHz (move the frequency around to where it sounds bright, but not shrill).
The Singer/Songwriter Syndrome
The singer also wants to play guitar at the same time, and you want some decent separations between the vocals
and the guitar. One trick is to use a X/Y stereo pair of small cardioids down low, aimed at the guitar, hile you
position a large diaphragm mic at the singer’s forehead, tilted just slightly forward, toward his/her nose.
Some Points To Ponder
These techniques should work for banjo, mandolin, 12 string, uke, and other small stringed instruments. But
sometimes they don't always work as planned. If you're not hearing the sound you want, try moving the mic
around, even to the point of miking the side of the instrument instead of the front. Violins, cellos, and upright
basses are a whole special category, which will be discussed later.
A good trick is to stick your finger in one ear and move around till you find a spot that sounds good, then put the
mic there for starters. Remember that each guitar is different, each mic is different, each room is different, and
sometimes just going up or down a 1/2 step will change everything. Starting from the outside edge of the
"nearfield" is a great starting point.
Some Mics To Try First:
Dynamics: Shure SM57, Sennheiser 421, and Beyer M201
Ribbons: Beyers, RCA, any ribbon mic.
Small Condensers: Oktava MC012, Marshall 603S, AT 4041, Neumann KM184, any small
cardioid or omni condenser mic.
Large Condensers: These mics add a great deal of color to the sound, so "try" anything you
happen to own. It may work great or not - you never know.
End of the first part - more to come!

>
Posted by KaBudokan on 05-17-2001 22:00
Harvey, what kind of placement strategies (or starting points) would you suggest for mixing
large and small diaphragm condenser mics for micing acoustic guitar?
I have an NT2 that I've been using to mic my acoustic with mixed results. I picked up one of
the Marshall MXL1000's from e-bay the other day ($50, I figured even a poor grad student like
me couldn't go wrong), and I am wondering what may be the best approach if I am going for a
stereo mix using those two mics.
I'll experiment, of course, just wondering what good starting points may be. I'm thinking, try to
find a solid sound I am happy with using the MXL, and then compliment with the NT2...
(Knowing the NT2 is a bit harsh on guitar, I'll probably try to find a spot, which mellows that
shrillness.)
Thanks.
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[QUOTE] Originally posted by KaBudokan
Harvey, what kind of placement strategies (or starting points) would you suggest for mixing large and
small diaphragm condenser mics for micing acoustic guitar?
Start around 2 feet out with the two mics set up in an X/Y array and see if you luck out. If not, try the Marshall
1000 by itself over your shoulder, figure out what's missing and then try to get that from NT2. Take the ball off the
1000 and put it in a closet somewhere.
I have an NT2 that I've been using to mic my acoustic with mixed results. I picked up one of the Marshall
MXL1000's from ebay the other day ($50, I figured even a poor grad student like me couldn't go wrong),
and I am wondering what may be the best approach if I am going for a stereo mix using those two mics.
I think the 1000 should be the main sound for your guitar with the NT2 as kind of a fill-in mic.
I'll experiment, of course, just wondering what good starting points may be. I'm thinking try to find a solid
sound I am happy with using the MXL, and then compliment with the NT2... (Knowing the NT2 is a bit
harsh on guitar, I'll probably try to find a spot, which mellows that shrillness.)
Read my whole description of guitar miking techniques a few posts above this one - I tried to get pretty detailed
>
Posted by KaBudokan on 05-17-2001 22:31

Take the ball off the 1000 and put it in a closet somewhere.
That's my first plan as soon as it arrives.
I think the 1000 should be the main sound for your guitar with the NT2 as kind of a fill-in mic.
That is also what I was thinking...
I like the idea of the "Over The Shoulder Technique". I am hoping between the two mic's I will be able to get a
good, interesting sound. I can always pick up another Marshall to do stereo micing with too.
I'll let you know what I think of the combination when I get to try it out.
>
The mic placement ideas for acoustic guitar make a lot of sense. I guess the part I didn't realize was the "wild-card"
factor of the large diaphragm mics. I understand that they "color" the sound, but from what you seem to be saying,
there's no way to predict in just what way they will color it....which is kind of a bummer. I notice when people are
using them for voice that there is a certain "depth" added to the spoken sound that the smaller condensers don't
seem to add, but I have no idea how this would translate to instruments.
I'm happy to wait for the installment on grand piano, but even with my own NEWBIE mic placement I notice that
the mxl 603s gets a much better sound than anything I've used before (including AT Pro 37). The "xy" thing is
going to require a special mounting device, and it probably means I need to order at least 1 or 2 new mics for the
other instruments since I want to do 90% live recording.
What about acoustic bass? The biggest problem I have noticed with trying to record this instrument is that there is
one note (the open "D") string which seems to record way hotter than the rest of the range of the instrument, so that
when the rest of the range is in balance and sounds great, that D comes in and bottoms the whole track out. How
should I deal with this? Bass roll of on the board while recording? Moving the mics further away? This may be an
EQ question rather than a mic question...
I know a lot depends on the instrument, the player, and the mic, BUT....when recording acoustic bass, do you often
use a large diaphragm mic? If so, have you ever used the V67G for this purpose? Over at my bass site, there is a lot
of heated discussion about the issue, but it's coming from players - most of whom (like myself) don't know much
about recording – rather than engineers. I'd love to hear your input, even though I realize that there is no definitive
answer for all situations.
>
Posted by dobro on 05-18-2001 08:58
"The Singer/Songwriter Syndrome
The singer also wants to play guitar at the same time, and you want some decent separations between the
vocals and the guitar. One trick is to use a X/Y stereo pair of small cardioids down low..."
How low? Below the guitar and pointing up? Under the guitar basically? This is the first time I've heard about this.
How far out? Doesn't it have to be close in to maximize the guitar and reduce the vocal? I'm gonna find out
eventually, but you can help the process along if you like.
>
The mic placement ideas for acoustic guitar make a lot of sense. I guess the part I didn't realize was the
"wild-card" factor of the large diaphragm mics. I understand that they "color" the sound, but from what
you seem to be saying, there's no way to predict in just what way they will color it...which is kind of a
bummer. I notice when people are using them for voice that there is a certain "depth" added to the spoken
sound that the smaller condensers don't seem to add, but I have no idea how this would translate to
instruments.
It's a major problem, Chris, especially when talking about this new crop of low cost, condenser microphones from
Russia and China. The Quality Control from many of the distributors leaves a lot to be desired. Some capsules
have a very ragged and peaky top end, while others either have a diminished bottom end or they're bloated in close.
The depth you hear on vocals is usually from the "proximity effect" when singing in the microphone's near field,
boosting the bass at around 250 to 400 Hz to produce a fullness that's usually very desirable.
I'm happy to wait for the installment on grand piano, but even with my own NEWBIE mic placement I
notice that the mxl 603s gets a much better sound than anything I've used before (including AT Pro 37).
The "xy" thing is going to require a special mounting device, and it probably means I need to order at least
1 or 2 new mics for the other instruments since I want to do 90% live recording.
Yes, the 603S mics are an amazing buy right now for x/y use and many other applications. Acoustic bass, cello,
flute, and fiddle are other uses for the 603S, which I'll try to cover in further installments.
What about acoustic bass? The biggest problem I have noticed with trying to record this instrument is that
there is one note (the open "D") string which seems to record way hotter than the rest of the range of the
instrument, so that when the rest of the range is in balance and sounds great, that D comes in and bottoms
the whole track out. How should I deal with this? Bass roll of on the board while recording? Moving the
mics further away? This may be an EQ question rather than a mic question...
There are several mic positions that can lessen that "hyped D" effect, like miking on the bass side of the bridge,
and even with a mic stuffed part way inside the "f" hole. You can also use a parametric equalizer to lessen the
effect. I'll get into all that in another posting later in the series.
I know a lot depends on the instrument, the player, and the mic, BUT...when recording acoustic bass, do
you often use a large diaphragm mic? If so, have you ever used the V67G for this purpose? Over at my
bass site, there is a lot of heated discussion about the issue, but it's coming from players - most of whom
(like myself) don't know much about recording - rather than engineers. I'd love to hear your input, even
though I realize that there is no definitive answer for all situations.
There are times when a large diaphragm mic can sound very good on upright bass, but it's usually from the middle
of the nearfield (about 2 feet) to the far edge of the nearfield (about 42")where it sounds best. Any closer and the
proximity effect will produce a boomy upper bass tone that destroys any definition and detail in the tone.
>
[QUOTE] Originally posted by dobro
You know, there must be 49 issues that are gonna get addressed here, and this is gonna turn into the
mother of all microphone forum threads, but just before it tops a thousand, I want to get my own particular
question in.
"The Singer/Songwriter Syndrome
The singer also wants to play guitar at the same time, and you want some decent separations
between the vocals and the guitar. One trick is to use a X/Y stereo pair of small cardioids down
low..."
How low? Below the guitar and pointing up? Under the guitar basically? This is the first time I've heard
about this. How far out? Doesn't it have to be close in to maximize the guitar and reduce the vocal? I'm
gonna find out eventually, but you can help the process along if you like.
As usual, it will depend on the guitar, but right near the bottom of the lower bout is a good place to start. Getting in
close WILL reduce the vocal, but, as you now know, that can create other problems. Start at about 18" away with
the mics slightly below and aiming JUST SLIGHTLY up toward the body. The trick is to get the vocal mic up high
and close to the singer without using a windscreen.
>
I have read about guys using music stands to isolate the instrument mics from the vocal mic in these
situations. But I wonder if the added baffle of the music stand doesn't start causing it's own problems in
this.
That can sometimes create more problems that it will fix. In essence, you're creating a mechanical comb filter.
Adding some padding can often minimize the combing effect, but you can also position the mics close to the stand
and create a Blumlien pair, which will give you very good separation. You can build a Blumlien stand by using a
piece of plastic about 12" x 12" and gluing a mouse pad on each side.
>
[QUOTE] Originally posted by rjbutchko
Harvey, sorry to add to the barrage of questions, but...
...as far as acoustic guitar is concerned, as long as that's the example we're working with, would it be a
good idea to mic the guitar from a couple of feet away, as you suggest, and add a near-field mic focusing
on the freq's you wish to enhance in the mix? Provided, of course, that the whole mess goes mono. (?)
As long as you observe the 3:1 Rule (the Second Mic must be at least 3 Times Further Away than the First Mic), it
might work well. It depends on moving that mic in the near field till you find the perfect spot.
>
Harvey, One more quick question and I'll shut up for a while (I promise!)
You mentioned the "Hyped D" effect regarding acoustic bass in your last response. Did you do that only
because I mentioned it first, or is the open D a problem on many acoustic basses?
Most basses, upright and electric usually have one, sometimes two, predominant strings. The old Fenders were
very notable in that the bottom two strings sounded completely different from the two high strings. With upright
basses, it's also a function of the resonant chamber, the porting, and the woods used. It's usually the two upper
strings that create the most problems for recording.
I'm asking because both my carved bass and my plywood do this when I record, though neither does it at
the fingered unison, octave, or double octave. My carved bass used to pop out a pretty ugly open "G" as
well, but I tempered it by putting in some velvet padding at the nut on that string. Maybe I should try this
for the D string as well, even though the "hyped" effect isn't all that noticeable to the human ear when
heard live. Hmmm....
You're in the near field when you're playing, so it may not be accurate, plus the G and D may be really booming
off the upper bout.
BTW, I know we should all make out our checks payable to you, but where do we
send them?
All checks should be made payable to:

The Harvey Gerst Rest Home For The Permanently Senile
8676 Travis Road
Sanger, TX 76266
Attn: Drug Procurement Dept.
>
Posted by dobro on 05-19-2001 22:55
Okay, I've been trying miking the gitbox from down under (no, I haven't booked a studio in Melbourne), and I'm
getting good results. Close works better than 18-24" out, probably because of the room, I think.
I tried the 'forehead level, point it at your nose' approach for the vocal mic too, and that's good too, but because of
the room I'm in, I actually get a slightly better sound by having the mic at nose level and pointing it at my
forehead! LOL
>
Lesson 8
An interesting, easy experiment:
Put your hand out about one hand's width from your face, even with your mouth, and try to blow straight ahead.
Feel where the air blast is actually hitting your hand. Surprise!!!!
If you're like most people, the blast will actually hit the bottom two fingers of your hand. We all tend to blow
slightly downward, so putting a vocal mic at nose height (or higher) actually misses the bulk of the air blast that
causes popping on words that have a "p", "f", "b", or "v" in them.
Tell the singer NOT to aim into the mic - just have them sing straight ahead, or set up another mic in the standard
vocal mic position (the "dummy mic" trick), and have them sing into that one. Or put the lyrics in the standard
vocal mic position, so that they hafta keep facing straight ahead to read the words.
X/Y miking setup
For some of you that may not know what "X/Y miking" is, here's a diagram of two cardioids set up for X/Y
miking:
Notice the capsules are almost touching and the angle between them is 110 degrees. This can also work with two
omni mics, but with lower stereo separation.
More on all this stuff later…
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I guess what I don't understand yet about the 110 degree concept is the difference between creating the
stereo image with the mic placement as opposed to using the pan left/right on the mixer....this is a concept
which had not even occurred to me until now. I guess that's why it says "newbie" under my name....
You still pan the mics hard left and right, but the placement is critical for maintaining the accuracy of the stereo
image. If done correctly, there are no phasing problems what so ever, and you get a perfect stereo spread. That's
the goal of every two-mic stereo imaging system and they all have different advantages and disadvantages,
depending on the source.
>
Posted by Harvey Gerst On 05-22-2001 14:55

[QUOTE] Originally posted by Wil Davis
I always thought a Blumlein pair was a pair of figure-of-eights mounted coincident at 90 deg to each other,
and at 45 deg to the stereo-plane. The plastic device you describe sounds more like a Jecklin Disk used in
"Baffled Stereo", or am I missing something ? (I usually do, so it'll be no surprise... :> )
Wil you didn't miss anything. I meant the Jecklin Disk. That'll teach me to type when I'm in pain.
The 3 coincidence techniques are:
X/Y (which does used 2 cardioid or omni pattern mics mounted coincident to each other at classically 90º, and at
45º to the stereo-plane),
The Jecklin Disc (which is actually what I was describing),
And the Blumlein pair (which does used 2 Figure 8 mics, mounted coincident to each other at 90º, and at 45º to the
stereo-plane). I'll get into the pictures with circles and arrows a little later.
Thanks for catching that and keeping me honest.

>
Posted by c7sus on 05-22-2001 19:29
Ok, so I gave it a try.....
I got from the diagram that the mic bodies are 110 degrees to each other, so that puts the capsules at 70 degrees
face to face.... right? The mics are an NTV and an NT2. I'm recording a 1985 Martin D-45 with light strings that
are fairly worn.....
Anyway, I started at 2 feet with the faders set at unity and trim up about 3-4 dB for the NTV and up about 9-10 dB
for the NT2. Bummer that the NTV is so much hotter...... all the EQ is set flat.
Anyway, the first pass I finally made was about 18" out with the mics above the sound hole angled down toward it.
I got a fairly "strong" signal through but a lot of harshness, which I am assuming is from all the gain on the Mackie
pres.....
So I'm a bit disappointed right now..... so now what? The guitar sounds pretty "honky" too in the mids...... is it the
mics or the placement or a combination of both? Hell, maybe that's just what the guitar sounds like.... I've got a
VC6Q too, which allows a lot more gain without harshness..... but alas only one of those......
I'm thinking the next thing is close miking with the hotter mic behind the bridge and the NT2 at the 14th fret.....
Any suggestions?
>
Posted by h kuhn on 05-22-2001 20:25
The mics are an NTV and an NT2….
It is more common to use 2 identical mics for xy. the difference in frequency response between 2 different mics
may cause phasing effects. BTW the nicest sound we got from an acoustic was with two mc012 about 70cm apart
and 1m from the source, with omni capsules (this, of course, is NOT coincident and has got nothing to do with
xy!). Harvey’s over the shoulder technique was close second (but we had problems with noise from the player's
shirt). The aim of the recording was to achieve a "being right there" feeling, lots of room while mono compatibility
was no issue.
>
Posted by Wil Davis on 05-22-2001 23:12
I notice that nobody has mentioned that much under-used (in my opinion) coincident set up of MS (Mid-Side). I
never cease to be amazed at the results possible from what is essentially a very simple technique. The main
advantages of using this method being (1) the two channels can be recorded and the stereo mixed at a later time,
and changed as seen fit, and (2) the mono-compatibility of MS is just about perfect. Anyone else use it?
>
[QUOTE] Originally posted by Elle
Harvey,
What would you use to record piano? (Upright)
With an upright piano, there are no rules, but in general, I would use a couple of small condenser mics, but where
is the big question, rather than what. Sound radiates from an upright piano from a lot of different directions.
Opening the top and aiming a mic into the piano's innards usually results in very unpleasant jangle.
I'd first pull the piano away from any nearby walls, and try a mic pointed at the backside of the piano (the sounding
board), listen and decide what else was missing. Maybe a mic on the front somewhere as well, checking for phase,
and trying to balance the two mics to get the full range sound without the jangle.
Every upright piano is different and you hafta experiment with different mic locations. Getting the mics further
back in a good room will give you a more balanced sound, but room noise then becomes a factor.
>
Lesson 9
Harvey Gerst on 05-27-2001 11:57
Vocals - Why Are They So Hard To Get Right?
The two most asked questions are what's a good mic for acoustic guitar and what's a good mic for vocals. Most
people actually want one mic for both, but if you've been following this whole thread, you know that the mic
requirements for acoustic guitar are different than the mic requirements for vocals. So, what's so special about
vocals and vocal mics?
There are three types of mics that are usually used for vocals:
1. Large condenser mics

2. Dynamic (moving coil) mics
3. Ribbon mics (a special class of dynamic mic design)
There are two types of patterns usually used for vocals:
1. Cardioid (most typical)

2. Bi-directional (Figure 8)
All of the above mics and patterns have "proximity effect" in common (more upper bass boost as you get closer to
the mic).
With these mics, you can adjust the distance and the angle between the singer and the mic to get a wide variety of
tonal effects till you find the right balance for a particular singer and song. Off axis response will often vary
dramatically with large condenser and dynamic mics, and when coupled with the "proximity effect", you have a
wide range of tones to choose from.
The general working range for most LD condenser and ribbon mics is anywhere from 6 to 18" away. Dynamic
mics are usually best under 6" away. But there is no hard and fast rule there. For intimate softer ballads, you may
want the singer to "eat the mic", recording them from 2" away, or even closer. Up close, windblasts are a concern
and a pop stopper, foam windscreen, or even both may be required.
Remember that "proximity effect" starts in the upper bass (around 400 Hz), and this is exactly the start of the
human vocal range. It can add richness to a thinner voice, but as with most things, it can be overdone. You adjust
“Proximity Effect” by adjusting the distance between the mic and the singer - Closer for more, further back for
less.
Use different mic angles to adjust the “High Frequency Response” - Straight on for maximum highs, off axis for
less highs.
As mentioned earlier, most singers’ breath blasts are aimed slightly downward, so try to get the mic above that
blast when possible. I try to mic from about nose or forehead high, aimed slightly down towards the mouth, but if a
person is more comfortable with a stage mic at mouth level, don't be afraid to give it a try.
Some condenser mics tend to have some bright high end peaks which may help a singer that doesn't have a lot of
high frequency content in their voice, but it's all too easy to just end up with an overly bright vocal. You usually
look for a mic with a smooth top end (like a ribbon), or a mic with a gentle high frequency rise.
With mics like the AKG C3000, some of the Rode mics, or the lower end LD ATs, watch for peaky high-end
response that may result in an overly bright vocal track that high end EQ can't fix later.
Try to choose the mic that doesn't require any EQ when recording, if possible. That's where the right sound begins.
Use compression sparingly when doing the tracking - you can always add more later.
I try to avoid committing to any effects while tracking, so that I have more options available during mixing. You
can't turn the vocal reverb down later if you record with it during tracking.
Chris Fitzgerald has been doing some experimenting with upright bass mic placements and he might want to share
some of the things he's found so far.
After that, we'll discuss miking drums, then grand pianos, then horns, and exotic instruments.
>
[QUOTE] Originally posted by JerryD
I understood your last posts a lot more than the X/Y technique stuff. I do plan to go back and re-read
everything.
Newbie questions:
#1 Would you ever mic vocals using two mics? To get a fuller sound? Is there another interesting technique
for this?
If you mean would I mic a vocal in stereo, the answer is no, never. With most vocalists moving around even
slightly, the phasing problems would be enormous. There are two times when I would use two or more mics on
vocals:
1. If the singer has a very wide dynamic range (from a whisper to shouting in the same song), I might use two mics
recording to two tracks; one mic set so that it doesn't clip on the loud parts, and the second mic set up to pick up
just the soft parts and to hell with clipping on the loud parts, then compile them in the mix, bringing up the soft mic
in the mix during the soft sections, and killing it during the loud parts.
2. An interesting technique that David Bowie used is to set up a second mic about 15 feet away and gate it so that it
only comes on during louder parts, and then a third mic set up around 30 feet away and gated to only come on
during the very loudest part of the vocals. This gives you two natural delays of 15ms and 30 ms, yet keeps the
main vocals very up front. Very cool trick.

#2 What music in the public domain do you feel is well produced? ie miking, tones, ect...
"Public Domain" has a very specific meaning in the music business. It means music that is free of copyright
protection and may be recorded by anyone without having to pay royalties (like old folk songs). I assume you
meant what music is out right now that I think is well produced etc.
Anything produced by George Massenburg or Al Schmitt will always be well recorded and an inspiration for me to
try and come close, even though I know I won't. There are some artists I don't even particularly like, but their
recordings are beautifully recorded and well produced.
>
[QUOTE] Originally posted by h kuhn
What is the advantage/disadvantage of using a figure 8 mic over a cardioid for single voice? Would you set
up the figure of 8 differently (absorbing surface at the back?)?
I was primarily thinking of Figure 8 ribbon mics when I wrote that but some of it holds true for all figure 8
patterns. Figure 8 patterns have the most proximity effect possible, which can really enhance some vocals. Ribbons
have a smooth, silky sound to them, which compliments a great number of voices. Figure 8 patterns have the
smoothest off-axis response of all gradient polar patterns.
Yes, I use some form of absorption for the backside when using figure 8 mics for vocals.
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[QUOTE] Originally posted by chessparov
Hi again Harvey! A more appropriate question from me as relates to this thread would be how your
experience has been using large diaphragm dynamic microphones for vocals, and how to use them more
effectively for us home recordists. I notice a lot of people recommend using EQ to cut the midrange
frequencies even after proper mike placement (2000 to 3000 hertz) about 2 or 3 decibels. Does that seem
typical to you also?
No, it doesn't "seem typical" to me. If anything, I'll sometimes boost those frequencies slightly, then cut the same
frequencies in other tracks to give the vocals more separation in a mix.
Keep in mind that it also depends on mic choice, since some mics do have an artificial boost (or peakyness) in the
2000 to 8000 Hz region. High, narrow high frequency peaks in microphone response drive me crazy, since they
can't be tamed with general EQ without dulling down the whole top end, and it takes forever to track them down
using surgical parametric EQs.
As I mentioned many times in this thread, mic selection and placement are everything. Use Distance to adjust your
Low Frequency Response, and use Angle to adjust your High Frequency Response.
By the way, the record that sold 75 million copies and used a Shure SM-7 as the main vocal mic was….
…Michael Jackson's "Thriller".
>
[QUOTE] Originally posted by Dolemite
Hey Harvey,
Don't you have that actual SM-7 used on Thriller? I think I remember reading that on R.A.P.
When I first got it, I thought it was the SM-7 that Bruce Swedien used on Michael, but I'm not sure any more. This
is one of Bruce Swedien's SM-7 mics that he had around the time "Thriller" was cut. It still has his name on it in
blue Dymo labeler tape that he puts on all his mics. When I bought it, I didn't know it originally belonged to Bruce
Swedien - I bought it because it was a great mic.
The SM-7 stays on a stand in the studio 24 hours a day, wired up and ready to go. It's one of 5 mics that are ready
to be used at all times (the other 4 are the Audix TR-40 omni, the Neumann TLM-103, the Marshall MXL-V67G,
and a Shure SM-57).
The Audix TR-40 omni is used for misc. acoustic guitars, flute, violin, and various percussion instruments.
The Shure 57 is ready for adding electric guitar solos and overdubs.
The other 3 (the SM-7, the TLM-103, and the V67G) are first grabs for vocals, with the LOMO/MC012, and the
ribbon mics standing by for other flavors. I'll also try these mics for horns, if a Sennheiser 421 doesn't give me
what I want.
>
Hey Harvey, I'm very new microphones and recording (6 months).
What is different about a valve microphone? How do these compare to a FET type mic?
Is the rant about Rode mics, NTK and NT1000, mostly hype?
Well, lemme see if I can dispel some myths here. There are many reasons why a tube is used in some many
condenser mics and is preferred to transistors (like FETs); some reasons make sense, while others are just hype and
nonsense:
A condenser mic capsule requires seeing a high impedance source to keep the signal from being loaded down,
usually in the multi-megohm to gigohm range. These can get pretty expensive. A tube, on the other hand, loves to
see that kind of load, so it's a good match up.
Tubes, when used in a class A configuration, can be made to clip softly and symmetrically, producing a more
musical tone, rather than the sharp clipping characteristic of transistors.
Also, the heat (from the tube inside the mic housing) can serve as a built-in heater to drive humidity and moisture
out of the case. All of the above traits are very good things. But there are some bad things as well.
You have to supple 12V DC heater and 250V plate voltages for the tube inside the mic, which require a separate
power supply and a special cable running between the power supply and the mic. Tubes also tend to be a tad noisy.
Different tubes can have different sonic properties which are either desirable, or not, depending on the
manufacturer, or even the conditions on a particular manufacturing date.
All mics are a combination of hype and reality, mixed up and served by the manufacturer or the distributor, served
up to fit their marketing plan. If you've been following this thread, you know there is no one microphone that will
work best for everything. Large diaphragm vocal mics are the prime example - a mic might be perfect for one
voice and sound like crap with another voice.
As far as tubes vs. FET mic designs go, a good tube mic will beat the s*** out of a bad FET mic, but a good FET
mic will beat the s*** out of a bad tube design. Tubes tend to smooth out the sound a little bit, hiding some of the
things that might be harsh in other designs.
As far as the Rode mics are concerned, I haven't heard those two models. My main concern with the Rode mics is
consistency of sound from unit to unit.
>
Lesson 10
Nearing the end.
Ok, here's some tips on miking some other instruments:
Horns
What mics and polar patterns to use. Most horns put out a lot of energy so close miking is not a good idea. About
18" away is a good starting place. Start at the bell level and work your way up to pick up more breathtones and
output from the keys, in the case of saxophones.
Generally, you can use a large diaphragm dynamic, a ribbon, or a large diaphragm condenser mic with pretty good
results. The Sennheiser 421 is a great choice for a dynamic, almost any kind of ribbon, or a 1" condenser mic.
Cardioid is usually the pattern of choice.
Choirs, pipe organs, and large stuff:

Small diaphragm spaced condenser omnis or cardioids generally work best, pulled way back to capture some of the
room as well. I usually split the sound source into 3rds and put a mic at the 1/3 and 2/3 points. (If the source is 60
feet wide, I'd start with mics 20 feet in and 20 feet apart and adjust from there.) How far away depends on how
much reverb would sound good. I'd start with about 1/3 of the way back from the source, then move in closer or
out further, depending on how much reverb I heard in the phones.
Flutes, clarinets, and misc. woodwinds:

I like small omnis, but cardioids may also work fine too. Try miking from below the instrument to minimize breath
noises, but this really requires a lot of trial and error experimenting to get right.
Concert and Celtic harps:

Very difficult to get a good sound in a less than perfect room, but generally, a pair of X/Y small cardioids about 4
to 6 feet away, placed as high as the instrument is tall, and aimed slightly downward. Use your quietest mics, since
these things don't put out a lot of sound and pedal noises can often be a problem. A couple of hints: Try putting the
harpist (harpy?) in the corner of the room, facing out, to pick up a little more fullness without using EQ. Also, try
putting the harp on a 6 foot by 6foot plywood sheet on the floor to help bounce a little more top end into the mics.
Misc. Percussion:
For tambourines, cowbells, etc., a small omni is usually best, placed about 2 feet above the instrument. Watch your
levels, and don't exceed -10dB, since most of the energy won't be shown on your meters. For misc. drums, try a
large diaphragm dynamic mic, or a small cardioid condenser, and try miking close to the top, and even try miking
from below the drum.
Didgeridoos, harmonicas, and other odd stuff:
A dig is like a single organ pipe - with weird mouth stuff attached. Use a small condenser cardioid slightly off-axis
from the end of the thing and adjust the height to pick up more mouth action. Harmonica players usually have their
own mic, but if not, try your cheapest dynamic into a small guitar amp with a little bit of distortion. For clean
harmonica, try an omni, about 1 foot above the player and 6" in front, aimed straight down. For other odd stuff, I
usually reach for an omni or small cardioid to "capture the moment".
OK, what have I left out?
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Lesson 11
Ok, let's mic an electric guitar.
This is gonna be another very long post, so hang in there. I'll try and keep all the techie stuff to a minimum, but
there are some concepts that are kinda hard to explain without getting a little technical, so ask questions if you
don't understand something - it's probably just due to my poor explanation. Before we get into the miking part, we
hafta talk about how speakers radiate sound, so here comes the first drawing:
Figure 1. Imagine a speaker suspended in space - the sound comes off the front of the cone, AND off the back of
the cone, more or less equally. The problem with this kind of setup is that the low notes coming off the backside of
the speaker cancel the low notes coming off the front of the speaker. Their wavelength is much bigger than the
diameter of the speaker and they just go around the frame easily.
Figure 2. Now imagine we've mounted the speaker in the exact center of a huge board 40 feet wide by 40 feet tall.
The speaker is still radiating in all directions, but unless a low note is at least 20 feet long, it ain't gonna get around
the edge of that board easily. Since we eliminated the possibility of cancellations, the bass comes way up when
your standing in front of the speaker, compared to the speaker that was just hanging there on a string. As far as
we're concerned, it's now radiating into a hemisphere.
Figure 3. Now put the speaker down low on the board and imagine a floor has been added. What happens? The
bass notes double in volume since they're now radiating into one half of a hemisphere. If you put the speaker at the
junction of the floor and two walls (a corner), the bass would double again, since all the bass is now radiating into
a quarter of a hemisphere. But what does this hafta do with miking an electric guitar? You're about to find out right
now.
Figure 4. If we fold the board (shown in Figure 2.) into an open-backed box, we can still prevent a lot of the bass
from wrapping around and canceling out. Starting to get it? Bingo, you're basically looking at a side view of most
open backed guitar cabinets, like a Fender Twin. The box prevents some of the low notes coming off the back of
the speaker cone from getting around to the front and interfering with the notes coming off the front of the speaker.
This arrangement works ok till you get down to around 90 - 120 Hz, and below, right at the bottom end range of a
guitar. So how do we get a little more bottom end?
Figure 5. Make the box a little bigger and seal it completely. Now the back notes can't interfere. Recognize the
design? A Marshall cabinet? Right!!!
Figure 6. As long as we've come this far, I threw this in. You take the sealed box, cut a hole in it, and then you can
tune the air in the cabinet to create a "blowing across a Coke bottle" effect, to add some bottom where the speaker
starts to give out.
Keep some of this in mind when I start this next section:
Miking the guitar cabinet.
Guitar amps come in many different configurations, but I'm gonna focus on miking the three most popular speaker
designs:
Open back cabinet, single speaker.

Open back cabinet, dual speakers
Closed back cabinet, with 4 speakers.
Here comes another one of those damn drawings:
Figure 1. The two-12" open back speaker combo is one of the most popular units of all time. There are 4 basic mic
positions, with several variations:
1. Stick a mic right into the speaker, aimed at the center of the cone. Maximum high end, and least outside
noise.
2. Stick a mic right into the speaker, aimed at the edge of the cone. Less high end, and a little more bass.
3. Pullback a bit (12 to 24") and aim a mic right between the speakers. More realistic, but increased chance of
phasing problems and more susceptible to room noise.
4. Use any of the first 3 methods and add a mic aimed at the back of the speaker. Try the phase switch and
choose the position that sounds best to you.
What mic to use?
Try the Shure SM-57, your kick drum mic, or any good dynamic for positions 1 and 2. Positions 3 and 4 might use
a ribbon or condenser mic to get a little fatter sound. I usually start with position 1 (one mic, pointed into the center
of the cone), but I may add something like an AKG D122 on the outside edge of the other speaker to emphasize the
bottom end a little.
Figure 2. The single speaker open back speaker cabinet is another popular design. The same 4 basic mic positions
are used:
1. Stick a mic right into the speaker, aimed at the center of the cone. Maximum high end, and least outside
noise.
2. Stick a mic right into the speaker, aimed at the edge of the cone. Less high end, and a little more bass.
3. Pullback a bit (12 to 24") and aim a mic at the speaker. More realistic, but increased chance of phasing
problems and more susceptible to room noise.
4. Use any of the first 3 methods and add a mic aimed at the back of the speaker. Try the phase switch and
choose the position that sounds best to you.
5. Repeat all 4 mic techniques, but put the amp on a bar stool or chair. Why? Go back to the very first
drawing and look at Figure 3. By raising the amp, it now feeds into a hemisphere instead of a half
hemisphere, lowering the bottom end a little. Pull the amp away from a wall for less bass, in closer to the
wall for more bass. See how the first drawing is starting to fit in?
What mic to use?
Try the Shure SM-57, your kick drum mic, or any good dynamic for positions 1 or 2. Positions 3 and 4 might use a
ribbon or condenser mic to get a little fatter sound. I usually start with position 1 (one mic, pointed into the center
of the cone), but I may slide it till it's at the outside edge of the speaker to emphasize the bottom end a little.
Figure 3. Is simply there to use up some space. I just thought it looked better with 6 drawings instead of 5.
Figure 4. Is a standard 4x12 Marshall cabinet. You would use mic positions 1 and 2 for adjusting the high end
relative to the bottom end (and remember, you're getting that ½ hemisphere bass boost from the floor). To lower
some of the bottom end, move the mics to positions 3 and 4 (or try a 57 at position 3 AND a D112 at position 2,
then blend them to one track, or record them wide apart to two tracks).
Figure 5. Marshall cabinet with distant miking. Try a ribbon mic, or a big condenser mic to get a fuller sound.
Adjust the mic anywhere from about 2 to 10 feet away. If needed, also use one of the mic techniques in Figure 4.
Figure 6. Actually this one is for any cabinet. Scenario: The guitar player isn't happy with any of the mic setups
you've tried so far. Have the guitar player play with the controls till he's happy with the sound. Tell him to freeze,
right there. Put a mic close to his ear, pointed at the center of the cabinet, and go back and listen. Either an omni,
small cardioid (dynamic or condenser), or a large cardioid will usually do fine. The mic is now hearing "exactly"
what the guitar player heard in the room when he said he liked the sound. That should end any conflict.
Hey, we're nearing the end of this whole mess - just a few more things to clean up, and then we're done!!!
>
Question #1;
Why wouldn't you put the dual speaker with the open back end into a chair? Wouldn't it have the same
problems as the other cabinet configurations?
Good call!! Yes, moving the speaker off the floor can help an overly heavy bottom end and get the speaker sound
closer to what the guitarist is hearing, once he adjusts the tone to compensate for the movement. And it's an
alternative position for any kind of speaker cabinet, not just a single speaker setup.
Question #2;
Why wouldn't you leave the single speaker config on the floor so it would have more bass? I guess my
question is why are you trying to reduce the bass when the single speaker would have less bass and need to
be increased?
Most players don't realize how much boost they're getting from the floor. In many cases, there's simply too much
bottom end and it interferes with everything from the kick all the way to the main vocals.
Question #3;
Could you give a brief explanation of phasing problems or point me to a article that clears this problem up
for me?
Pretty simple really. When miking a multiple speaker cabinet from a distance (rather than right at the cone), there
is a chance that you might get some comb filtering caused by phase differences between the different path lengths
from the mic to each of the multiple speakers.
On the good side, this is one of the most benign distortions there is. The old Bozak and Bose speakers used
multiple drivers and nobody really ever complained about them. There are probably some old AES papers on
multiple driver phase cancellations somewhere, but I don't know of any on the net.
PAGE 9
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Lesson 12
I'm gonna break this into sections and then post the basics first, then talk about how and when to use them.
Coincident stereo mic techniques.

In the late 1920's and early 1930's, Alan Blumlein (in Britain) and RCA were both working on recording
techniques, using only a small number of recording channels for reproduction over a pair of loudspeakers.
The technique developed by Alan Blumlein consisted of a pair of microphones with figure 8 patterns, mounted
close together,with the front lobe of one mic pointing 45 degrees to the left, and the front lobe of the other pointing
45 degrees to the right (Figure 2.). Although it provided excellent stereo imagining, sounds coming from the rear
are also picked up and when reproduced over a pair of loudspeakers, these sounds were also mixed into the
speakers. This results in a sound which is too reveberant for many people.
"Purists" who liked the simplicity and accuracy of the Blumlein technique modified it in order to remove this
problem. By replacing the figure 8 microphones with cardioids and changing the angle between them to include
the desired soundstage, it is possible to use the cardioid mic's lack of rear response to reduce the rear reveberant
sounds. This results in a much more acceptable, if less accurate sound image. Typically, the angle between the
mics should not be more than about 135 degrees, or less than 90. This technique is the popular "X/Y" stereo
recording system. (Figure 1.)
Mid-Side (M-S) techniques use a figure 8 mic, sideways to the sound source, and a cardioid mic facing the source
(Figure 3.). By inverting the signal from the real lobe of the figure 8 mic and using a matrix network, it is possible
to adjust the width of the sound stage to almost any size.
>
Q1: In the X/Y pair I'm assuming that the mics have to be identical. Correct?
Yes, it helps keep the stereo image from wandering. (That means if one mic has a peak at a certain frequency,
every time that note comes along, the mic will hear it louder and play back as if the sound is coming from one
side.)
Q2: In the X/Y pair where and how far away would the sound source be?
Depends on the X/Y angle you use, but in general, no further back than the mics pointed at the outside edges of
whatever you're recording. At 90°, the mics would form the apex of an triangle, with one mic pointed toward the
left edge of the group you're recording and the other mic etc., etc.. As you move the mics in closer to the source,
you would widen the angle accordingly. it's not a hard and fast rule. For example, I usually aim the mics about 1/4
of the way in from the outside of the source, and then move in and out till I get the sound I want.
Q3: You lost me on the Matrix network. What is that?

Ok, this M-S stuff works by additive/subtractive matrixing. True figure 8 mics have the best off axis response of
any pressure gradient design and a perfect null in the middle (at exactly right angles to the front of the mic. When
you turn a figure 8 mic sideways to the sound, none of the sound from dead center is heard. Only from the left side,
or the right side (but one side is out of phase, cuz it's coming off the back of the mic).
You add a cardioid pointed at the center of the music, and not only does it fill in the hole, but it is combined with
the figure 8 to create sum and difference combinations (usually thru a matrixing box) to let you control the
absolute width of the stereo image. You can dial in anything from a perfect mono signal to wide stereo, all with
perfect phase coherency.
A more complete source for how it does this is available from my friend Wes Dooley at
http://www.wesdooley.com. He makes M-S matrix boxes and he has a complete article on how M-S stereo works
on his web site. Wes is one of the leading authorities on M-S Stereo miking. (I can't afford any of his damn matrix
boxes, but it's good reading to understand the principles).
Q4: On the X/Y pair it looks like if you go over 90 degrees on your angle you would start missing the
center of your stereo image due to the limitation of the mic pickup pattern. Why would you ever want to go
over 90 degrees?
To increase the apparent width of the sound stage. But, as you widen the angle between the mics, you move in
closer to the source, and you move in closer to the center, more than you do to the ends of the source, so the center
level increases to offset the loss from the wider angle. In simpler terms, it all works out.
P A G E 10
http://homerecording.com/bbs//showthread.php?
s=a029271b38004871d985d32ca6e4512a&threadid=27030&perpage=25&pagenumber=10
Lesson 13
OK, let's finish up stereo mic techniques and move on.
I'm not gonna go thru all the pictures with circles and arrows this, since most of these techniques are listed in detail
with pictures at the DPA Microphone University site at:
http://www.dpamicrophones.com/index.htm
But we'll discuss all of the techniques here.
Coincident Stereo Mic Techniques:

Involve stereo from phase differences between the sound sources. The mic diaphragms are spaces close together,
and the angle between the mics is used to control the phase differences. (M-S, X/Y, etc.)
Near Coincidence Stereo Techniques:

These use time and level differences between the two mics to achieve the stereo effect. The mic diaphragms are
spaced at different distances or something is put in between them to approximate the way the human ear hears
stereo. The near coincidence methods include:
ORTF (France): A pair of cardioids at 90° (pointed away from each other) spaced 17 cm apart. Looks like this: \ /
(except the actual angle is 90°).
DIN (Dutch): A pair of cardioids at 90° (pointed away from each other) spaced 20 cm apart. Looks like this: \ /
(except the actual angle is 90°).
NOS (Netherlands): A pair of cardioids at 90° (pointed away from each other) spaced 30 cm apart. Looks like
this: \ / (except the actual angle is 90°).
Jecklin: A pair of omnis spaced 36 cm apart, with a disk between them, like this:
.....||
|....||....|
.....||
Binaural: A pair of omnis, usually mounted in a device, which approximates the dimensions of a human head.
All of these near coincidence techniques depend on time and level differences to get the stereo image. Which you
would use depends on the source of the sound you want to capture, the room and equipment you are using, and the
accuracy of the stereo image desired.
Wide-Spaced Stereo (A-B) Techniques:

A-B techniques all use time and/or level differences to get the stereo effect. For large groups, a third (center) mic is
often used. The standard A-B miking uses a pair of omnis spaced about 40 to 60 cm apart, facing the sound source.
Cardioid can often be used instead, and a third mic in the center can be added to fill in any hole the wide spacing
causes.
Stereo Wrap-Up Thoughts:

The effect of all these different stereo miking techniques will vary, depending on whether playback is for speakers
or for phones, and the width of the source being recorded.
Generally, you'll get more accurate stereo effect using the coincidence or near coincidence methods, especially for
smaller single instruments or small groups, like a string quartet or bluegrass band.
A-B wide spacing usually works better for loudspeakers and wide sound sources, like choirs, orhestras, and pipe
organs. But M-S, Blumlien, Jecklin, and binaural can also work well on large sources.
All these stereo techniques are time-tested tools that work well on a lot of different things. As the person doing the
recording, it's important that you understand how they work.
The choice of which to use for a given project is up to you, but now, you at least know all of the techniques that are
normally used on a lot of records that you've heard.
Still to come: Drum miking, piano miking, percussion miking, and how to read (and understand) mic
specifications.
>
Posted by tubedude on 08-04-2001 04:17
Ribbon Mics
Harvey,
A big part of my interest last few months or so has been in ribbon mics. How are they better/worse, etc. I hear they
make great room mics, overheads, and get a huge dirty guitar sound. Maybe it’s just the Royers. Been wanting to
check out the Beyer ribbons and find some comparisons against the Royers.
Peace,
Paul
>
Paul,
Ribbon mics are a wonderfully simple, yet elegant, solution to a number of mic design problems. Using a single
strand of super light corrugated aluminum ribbon, they eliminate a large part of the resonant, transient, and axial
difficulties of other microphone designs. They have the disadvantage of being fragile, having very low output,
susceptible to hum fields and wind blasts, and the older ribbon mic designs were heavy as hell.
I own four ribbon mics, and I'd sell almost everything else in the studio BEFORE I'd part with them.
>
Lesson 14
A few more thoughts about stereo miking.
First, I want to add a few pointers about stereo miking that might help clarify some terms for people that are new to
recording.
Coincident Mic Techniques:

That's a fancy term for any stereo mic technique where the two matched capsules are as close together as possible
and only the phase differences are used to get the stereo image. The advantage to these techniques is that they
collapse to mono very well, sound good on speakers, and amazing on headphones.
Near-Coincident Mic Techniques:

Another fancy term for any stereo mic technique where two matched cardioid (or omni for some techniques)
capsules are wider apart, pointed parallel (or away from each other) and use level differences and time delays to
get the stereo image. The different spacings determine the stereo image spread. The advantage to these techniques
is that they sound excellent on speakers, and very good on headphones.
Wide Spacing Mic Techniques:

A term for any stereo mic technique where the two or more similar cardioid or omni capsules are wide apart, aimed
parallel to each other (or angled in) and use level differences and time delays to get the stereo image. The
advantage to this technique is that the sound is amazing on speakers, but not very good on headphones.
So which technique should you use for your recording? It depends on what you're recording, and what part it plays
in the final recording. If it's an acoustic guitar part for a full rock band, forget about stereo and just use one mic,
usually. For a singer/songwriter, try M-S or coincidence recording.
If the recording might get a lot of FM radio play, coincidence recording will give the best mono signal when you
get into weak signal areas. String quartets, bluegrass groups, barbershop quartets might work better with near-
coincidence mic techniques, while all three techniques might work well for larger groups.
You hafta ask yourself some questions when choosing a stereo mic technique… How important is mono? How
much spread do I want? How important is one element in the stereo field (like a soloist in a choir)?
Like everything else in recording, experience comes with time, experimentation, and failures. We learn from our
mistakes. And sometimes, our mistakes save our asses. Record, listen, then analyze:
What's missing? Why?

What's too loud? Why?
What's too boomy? Why?
What's too muffled? Why?
If you constantly hafta reach for eq to correct things, it usually means that the mic is:
1. In the wrong place, or

2. The wrong mic for the job.
Any more questions about stereo recording techniques, or does this about cover the subject?
>
Any more questions about stereo recording techniques, or does this about cover the subject.
Yes! Why is it that in classical recordings the mics are very often set up above the orchestra/ensemble, while the
audience normally sits below? wouldn't it be more logical to put them where the listener’s ears is? The other
question: how important is mono compatibility nowadays? I mean, how many % of the population (even in 3rd
world countries) uses a mono playback device? I haven't seen one since many years. Thank you
Harald
>
Many orchestras DO put two mics in the listener's position, but it often captures too much of the hall's
reverberation, so they use spot mics to cover different sections of the orchestra, and then they blend the various
sections together during mixdown from a multi-track recorder.
Mono capability is still important in two separate areas - Television, and FM radio. There are still a lot of TVs out
there with only one mono capability. And when you get into a lower signal strength area, your FM radio
automatically switches to "mono receive" mode.
Now, let's say you recorded your rhythm guitar track on the left channel, and then ran the same track thru a short
delay to the right channel - to make the guitar sound bigger (a common home studio technique). It'll sound fine in
stereo, but the guitar will DISSAPPEAR COMPLETELY when summed to mono.
>
Many orchestras DO put two mics in the listener's position, but it often captures too much of the hall's
reverberation, so they use spot mics to cover different sections of the orchestra, and then they blend the
various sections together during mixdown from a multi-track recorder.
But have you ever seen a Decca tree at listener’s position? As far as I know they are usually hung about 2m above
the conductor’s head, aren't they?

Now, let's say you recorded your rhythm guitar track on the left channel, and then ran the same track thru
a short delay to the right channel - to make the guitar sound bigger (a common home studio technique).
It'll sound fine in stereo, but the guitar will DISSAPPEAR COMPLETELY when summed to mono.
OK, but would you avoid to use a spaced pair of omnis to record an orchestra if you knew that the concert will be
transmitted on TV? Or maybe I should ask: Can you achieve satisfactory mono-compatibility (what a word) with
that kind of setup? Or would you prefer a coincident technique, even if the soundstage doesn't translate that well?
>
[QUOTE] Originally posted by h kuhn
But have you ever seen a Decca tree at listener’s position? As far as I know, they are usually hung about
2m above the conductor's head, aren't they?
Yes, and that's mainly because the listener's position would have too much of the reverberant field present. You
must also account for the fact that the end listener will probably be listening to the final product in their own
reverberant field (i.e., their living room). The conductor's position would (or should) be the most balanced spot in
the hall, since he controls the level of each section of the orchestra.
OK, but would you avoid to use a spaced pair of omnis to record an orchestra if you knew that the concert
will be transmitted on TV? Or maybe I should ask: Can you achieve satisfactory mono-compatibility (what
a word) with that kind of setup? Or would you prefer a coincident technique, even if the soundstage doesn't
translate that well?
When you hafta get it right quickly, you go the safest possible route, which means a coincident pair, if
you're restricted to using two mics and you need a recording in a hurry without a sound check beforehand. M-S
would allow you to adjust the soundstage in post-production, and it would be mono compatible.
The "Making Of 'The Producers' Soundtrack" is going to be on PBS tonight and I'm looking forward to
finding out how they recorded it. But, to answer your question, yes, a pair of spaced omnis would
probably work fine, but most commercial orchestral recordings are usually made with each section
miced.
>

Yes, and that's mainly because the listener's position would have too much of the reverberant field present.
You must also account for the fact that the end listener will probably be listening to the final product in
their own reverberant field (i.e., their living room). The conductor's position would (or should) be the most
balanced spot in the hall, since he controls the level of each section of the orchestra.
Harvey,
I think I found a different answer (which coincides very well with what you said about radiation patterns): I
recorded a violin player last week. The venue was a fairly big theatre, I decided to try xy from about 2m high and
3m away from the player. The accompanying piano (a 6ft steinway, sigh..) had the lid half closed and was directly
behind the violinist. The violin sounded great from above and only so and so from aprox 1m height and the same
distance, while the steinway sounded better. I guess that the violin is simply radiating upwards/side wards, very
similar to the cymbals of a drum set. And much of the orchestral sound comes from the strings, while they seem to
be the weakest part concerning SPL, so it makes sense to accentuate them a bit. Are these very stupid ideas or am I
on the right track?
Harald
>
Harald, you're exactly on the right track. Now that you have a good idea about radiation patterns, some different
positions open up to you and you begin thinking "outside the box" - a fancy term for avoiding most people's
preconceived notions of how something should be done, based on limited experience, or just reading "how-to"
books.
>
Lesson 15
Drums, Here We Go….
This is gonna be a pretty big addition, so I'll break it up into several sections. Obviously, recording drums depends
on a lot of different elements; the actual drums used, the drummer, the room, the style of music, the mics available,
mic placement, number of tracks available, stereo or mono, and how important the drums are to the particular
song. Let's look at each of the above elements in a little more detail (although I'm gonna go into a "lot of detail"
about kick and snare right now):
The Drums
Bad drums will never sound great. The drums hafta be in good shape, tuned correctly, and properly set up. If they
sound bad in the room, they'll probably sound bad on tape. A good engineer sometimes has to be a good instrument
tech. I've had to tune drums many times, intonate guitars and basses, rewire pickups, etc. Just because the drummer
knows how to play drums is no guarantee that he/she can tune them.
Every drum has a natural resonance. You can hear the note by lightly tapping on the side of the drum. That's
usually what you tune the top head to, with the bottom head tuned a little lower. There's a range of about 2 or so
notes each way from that natural resonant frequency that will work fine, but you need to stay in that range to get
the power out of the drum. Drums are usually tuned in fourths, starting with the high tom. If you're not
knowledgeable about drum tuning, it would be well worth it to have a good drummer come in one time and show
you how to tune drums.
I'll get into drum tuning in another post if anybody's interested in that, but right now, just make sure the drums are
tuned correctly, and they sound good in the room. We usually use Ambassador coated heads for our drums and
they record very well. We avoid the oil-filled heads (too dead-sounding), and we stick with the single ply, coated
top heads for everything, with clear heads on the bottom.
The Kick Drum

The kick, along with the snare and electric bass, is usually the backbone of the song - these instruments provide the
"groove" and "drive" of most rock music, and they require the greatest attention. For rock, especially metal, the
kick also provides another element - the beater "click", needed to hear the speed or complexity of the bass drum
patterns.
Most rock drummers have a hole cut in the front head (the head facing the audience), but few drummers
understand the hole's function. Most do it for looks, because "all the other drummers do it".
The hole is for mic access to the back drumhead (the head being hit by the foot pedal), to let the mic get close
enough to pick up more of the beater "click". The hole should be 4 to 6" in diameter, and located above the
centerline, to make it easy to get the mic (mounted on a short stand and boom arm) inside the drum.
I usually have the drummer loosen and turn the head till the hole is in the upper right quadrant, and I'll bring the
mic in, angled toward the floor tom, about 3 to 4" away from where the beater hits the head. Angling the mic
towards the floor tom reduces the amount of snare bleed, which will help later on if I need to gate the kick drum.
For drums without the access port, I'll also try miking the kick from the pedal side of the drum. If I need a really
"huge" sounding kick, I'll construct a tunnel from a packing blanket off the ported head and add a large condenser
or ribbon mic about 3 to 4 feet away (inside the "tunnel"), just to pick up the low end. I've even made a "tunnel" by
removing the front head entirely and placing a second kick drum in front of the first (removing the back head from
the second kick, and miking the second kick at the hole.
I avoid gating or compressing the kick during the recording stage, but I might do it during the mix. I usually add a
few dB of boost between 2 and 4 kHz to emphasize the beater click. I'll crank the boost all the way up, and then
sweep till I find the desired click sound, then back off on the boost. For tape-based systems, this should be standard
procedure, since boosting top end later on will also add hiss.
I'll scoop out a big hole down low, using a parametric, anywhere from 250 to about 800 Hz, eliminating the
"boom" frequencies. I don't usually add any low bottom boost during recording, since it's easy to add later during
mixdown.
The Mics For Kick Drums

For rock drums, the mic choices are usually: AKG D-112 (older recordings use the AKG D-12E), ElectroVoice
RE-20, AudioTechnica ATM-25, Sennheiser 421, Shure Beta 52, and the new Sennheiser 602. These are all
dynamic mics, either cardioid or hypercardioid, and pretty large diaphragms. The Shure SM-57 is also used for
kick, and will work ok, but not usually as well as the mics listed above.
Mics for use in a kick tunnel or for distant kick miking are usually either large diaphragm condensers (the
Neumann U-47fet is the most popular choice), or ribbon mics like the Royer 121, the RCA 44BX, or the Coles
4038 - all high dollar mics. Some good low-cost choices would be the Marshall V67G and the Studio Projects C1.
That should do it for a while. I'll continue the rest of this a little later.
>
Lesson 16
Recording drums - Things to think about:
Drums: One instrument?
There are two schools of thought on this. Since each drum and cymbal basically produces just one note each, it
may be thought of as simply one large instrument.
You can mic a drum set with just one mic, but it's tricky. You pretty much move the mic around till you find the
right balance between the snare, toms, kick, cymbals, and high hats. That's usually about 6 to 8 feet away, and
about 6 to 8 feet up in the air.
BUT that means you're also picking up a lot of the room, and if you have a bad room, the drums won't sound all
that great. So how do you get around that? It's actually the same problem as miking an acoustic guitar or a grand
piano. Move into the instrument's near field (get closer), but when you do that, all bets are off.
In the near field, you have to use more mics, watch for phase problems, and realize you're gonna hear resonances a
lot louder than normal. To keep phasing problems down, try just two mics (above the drummer's head, aimed at
each end of the drum kit), then see what needs more oomph. You may need to add a snare mic, or a mic on the
kick drum, but at least you won't be fighting the room.
If you have a lot of available tracks, you don't even need to commit to a particular drum balance right now - just
put a close mic on every tom, the snare, kick, even the high hats, and worry about balancing everything out at the
mixdown. (We'll talk about phasing problems that can occur in a little while.)
The snare and kick - the heart of the drum set. Ok, so it has two hearts. In actual fact, the snare is the heart of the
set with the kick a close second. Everything revolves around the snare. When you set your overhead mics to pick
up the cymbals, use a tape measure so that each mic is exactly the same distance from the center of the snare head.
Miking the snare up close - decisions, decisions.

I use a hypercardioid mic (the Beyer M201) to mic snares, since it has some nulls at 130° off axis, so I can angle it
to reduce hi-hat bleed into the snare mic channel, but even the Shure SM57 kills for snare if placed right.
One of the first places I try is the spot between the high tom and the hi-hat, aimed at the center of the snare, about
1" above and 1" inside the snare rim. If that doesn't sound good, I then check the actual sound of the snare to make
sure it's tuned right and not creating a lot of problems. Sometimes, a little butterfly of duct tape on the snare head,
right in front of the mic, will reduce ringing and spurious resonances enough to get a usable sound.
If that doesn't work, I'll check different mic placements, even to the point of pulling the mic back a few inches and
moving it up and down the height of the shell, looking for a good balance (yes, actually pointing the mic at the
shell, not the snare head).
Finding the right place for the snare mic can actually take upwards of an hour, but it's well worth the time spent.
Once I have the snare sound about 80% nailed, I'll go to the eq and do any trimming that's needed, roll off some
bottom, add a little mid crack, or some high end.
I usually add some short plate reverb to the sound of the snare, even if it's just in the headphones for now. I don't
get "super anal" about the final sound, since I know it'll hafta change a little bit when I have all the other
instruments mixed in.
Then I move on to the kick, which I'll cover in the next post on drums.
>
Some of you may not be familiar with the duct tape "Butterfly" used on drums. I've (hopefully) attached a picture
that explains it a little better. It use a piece of duct tape shaped a little like an upside down butterfly to use for
damping out any nasty head resonances.
It works better than a flat piece of duct tape, since the footprint can be smaller and it adds more mass in a given
space, so that it damps just the problem spot without affecting the surrounding area. It can be as little as 1" wide
and 3/4" high.
Any questions?
>
Posted by mixsit on 09-24-2001 23:02
Harvey, when you referred to the two overhead mics being 'aimed at each end of the kit', did you mean front and
rear? In other references to a three mic technique, the primary mics seem to be directed as much front/back as
left/right. If that's the case, would this call for more of a mono/center blend mix on the drums, rather than paned? It
could be I misunderstood altogether!
>
[QUOTE] Originally posted by mixsit
Harvey, when you referred to the two overhead mics being 'aimed at each end of the kit', did you mean
front and rear? In other references to a 'three mic technique, the primary mics seem to be directed as much
front/back as left/right. If that's the case, would this call for more of a mono/center blend mix on the drums,
rather than paned? It could be I misunderstood altogether!
Actually, I meant left to right. Think of a plastic dome over the snare, with the center of the dome being the center
of the snare head. (Or think of it as a force field around the drum kit, with the center of the snare head being
Voyager, or the Enterprise).
Think of the overhead mics as "photon torpedoes". You want to place the two overhead mics so that they are at the
same distance from the snare (like touching the surface of the dome). Now without moving the front of the two
overhead mics, you angle the mics outward, so that they are aimed more toward the outside edges of the kit.
I usually aim one of the overheads at a spot between the high tom, snare, crash, and high hat, and the other mic is
aimed at a spot that sees the floor tom and the front edge of the ride cymbal.
But the business end of each overhead mic is equidistant from the snare.
Does that make it clearer?
P A G E 11
s=f2f812f5370cca8b19b0acd1e5dabae7&postid=230962#post230962
Lesson 17
A few more thoughts to wrap up drum miking
Before we move on the last section (understanding mic specs), I want to finish up this section with some tips,
tricks, and general thoughts about drum miking:
You don't always need "stereo drums". Unless there are a lot of tom rolls (or the drums play a huge part in the
song), sometimes mono drums will work better to preserve the mood of the song.
If you DO mix the drums in stereo, you don't always need to pan everything wide. Closing up the stereo width will
help bring the set together. Most drum sets aren't as wide as the distance between your home stereo speakers.
GENERALLY SPEAKING, you use large diaphragm dynamic mics on the drums, and you use small diaphragm
condenser mics to bring out the detail of the cymbals.
A large condenser room mic can sometimes help bring the whole drum set into focus - if you have a decent room.
Works best at 6 to 20 feet away.
The better the room, the less mics you need.
You "close mic" drums for two reasons - either the room sucks, or you want more options later, during mixdown.
Avoid gating and compression during tracking. Use it during mixdown if you need to.
Avoid recording the drums with reverb or effects. Add those later if possible. If not, add a little plate reverb to the
snare during recording and just a touch to the rest of the drums. Record the kick dry.
If you have enough tracks AND you're close miking, record each drum to a separate track. If you're track limited,
record the kick to it's own track, the snare to it's own track, and the rest of the set to a stereo track.
Use EQ to get into the ballpark of what you envision as the final drum sound, but don't add big amounts of bass
boost at this point - you can add that later, during mixdown. The snare will be the hardest to get right, but getting
all the toms to sound even will also be a big challenge.
You can get a bigger deeper kick sound by building a tunnel and adding a second mic about 4' to 6' out from the
kick, in the tunnel.
You can also put a second kick drum in front of the first kick drum, and mic that - with or without heads.
You can run the whole drum mix thru a pair of speakers and mic the speakers and mix that in with the original mix
to fatten the drums.
You can lay a speaker on top of the snare, mic the underside, and feed the snare track to that speaker to add body,
more snare rattle, or change the sound of the snare.
You can run the drum mix thru two compressors to really fatten the drum mix - here's how:
Set the first compressor for maximum compression (20:1 or greater), set the threshold for about -8 dB, and set the
attack and release pretty slow (a little past halfway - you'll need to experiment to find the right settings). That
knocks down the really loud parts without touching the faster initial peaks.
Take the output of the first compressor and feed it into a second compressor. Set the second compressor for
maximum compression (20:1 or greater), set the threshold for about -3 dB, and set the attack and release to their
fastest settings. That trims the fast stuff and what you get back is a huge sounding drum kit. The output of the
second compressor is your final drum sound.
Hopefully, this covers all the drum stuff, so the next section (the last section ?) will cover understanding mic specs,
separating the truth from the hype and BS, and how to really read a mic frequency response curve (how they're
created, and what they REALLY mean).
Did I miss anything?

>
[QUOTE] Originally posted by Henrik
Ditto that! Harvey, it's really very generous of you to take your time writing all this. I can't wait to try out
the double compressor-trick on drums!
However, I have two questions:

1. When you emphasize the importance of placing each overhead mic at exactly the same distance from the
snare drum, is this to avoid phase anomalies, or is it because you think it's important to have the snare in
the center of the stereo image?
It's primarily to avoid phase anomalies.
2. If you mic a drum (or anything) with two mics at different distances, as you
suggested for the kick drum in your previous post - what steps do you take to
avoid phase cancellation (if any)?
I try to make sure that each mic follows the 3:1 rule, mentioned earlier in this thread. (I've been bitten many times
for this in the past.)
>
[OUOTE] Originally posted by Henrik
Harvey, you mentioned some time ago that you would explain why many studios use small drum booths
instead of nice sounding rooms. So why is that? They trust their digital reverbs and want the drums as dry
as possible when tracking?
That's part of it, but mostly, it's to keep the drums from bleeding into all the other open mics.
>
Posted by Kelly Holdridge on 09-27-2001 03:31
Ok, I've been going off of John Sayers' approach to drum micing, which includes putting the snare in the middle of
the stereo overheads by offsetting the axis of the stereo spread. Here's a picture of what I understand John to be
talking about:
(Told you I'd post it, lol)
Ok. What I'm wondering is why worry about phase cancellation with precise measurements if you can get the
diaphragms close enough together in an X-Y configure? Is it because of the middle tom (or whatever sits below
this off-centered axis) being louder than anything else? Is there a distinct problem with this setup? (The snare and
hi-hat both have the same off-axis position to the 57's)
Oh, and that's an RE-235 hanging from the ceiling, our only omni.
>
[QUOTE] Originally posted by ausrock
Harvey, are you intending to start a thread on drum tuning in the future, as threatened? I think it would be
valuable in it's own right as even a lot of drummers don't really know how to tune their kits correctly.
You're right about some drummers not understanding how to tune their drums. I was going to get into a whole
section of how to tune drums, but then, I found this web site that covers it pretty well:
http://www.drumweb.com/profsound.shtml
>
Posted by gnarled on 09-27-2001 17:23
[QUOTE] Originally posted by Kelly Holdridge
Ok. What I'm wondering is why worry about phase cancellation with precise measurements if you can get
the diaphragms close enough together in an X-Y configure? Is it because of the middle tom (or whatever
sits below this off-centered axis) being louder than anything else? Is there a distinct problem with this
setup? (the snare and hi-hat both have the same off-axis position to the 57's)
Well, I think with the John Sayers approach he's talking more about spaced pairs, not coincident pairs. With spaced
pairs (referring to your picture) you might put one overhead over the drummer's left shoulder and the other one in
front of the kit on the side with the crash and high rack tom. That way, both mics will get the snare at
approximately the same level so it will be centered when panned and (if you measure the distance to the snare) the
snare will be in phase.
>
Posted by mixsit on 09-28-2001 10:05
Thanks Harvey. Sorry for the delay, been out of town...
So, if the mics are close together above the drummers head, you've got sort of a narrow XY stereo set up, except
the mics point out instead of across? I definitely like the idea getting a little behind the kit. I used to do XY above
the kit, but switched to each side near the drummers shoulders when I went to Earth omnis. It got me closer to the
drums (skins), more isolation from the other instruments and farther from the cymbals. Almost never see the need
to close mic the toms. The hat being a little too loud and left seems to be the main disadvantage (but that's usually
because he's playing it too loud). In keeping with the mics being equal-distance from the snare, the left mic gets
pulled back a bit, which also helps.
>
I prefer using wide spaced omnis (Audix TR-40, MXL603S, or Oktava MC012s w/Omni capsules), or cardioids
(Shure SM-81 or Oktava MC012s w/Cardioid capsules) for overheads. It gives me a feeling of greater control over
the stereo image, but it's all about whatever works for the music.
One last note about drums before we move onto the last section of this whole mess:
I usually stand in the drum room and listen to the drummer and watch him for a while BEFORE I start picking or
placing mics. I watch where he hits each piece of the set and try to establish a feeling of where the mics are gonna
be out of the way, AND it helps me decide on which mics to use.
If he's heavy on the high hats, I'll position the overhead to pick up more crash, or switch him to a 12" set of high
hats to soften them. Heavy on the snare? Pull the snare mic back a little (and maybe angle the kick mic away from
the snare side a little more). Light on the snare? I'll move in closer. Lots of tom rolls? I'll double check the sound
of the tom mics against the overheads for possible phasing problems. Lot's of light cymbal work? I might move the
overheads in a little closer to pick up the delicacy and shimmer.
But really listening and understanding how the drummer approaches the drum set is very important to getting the
right sound.
Ready for the next part?

OK, if we're done with drums, we can move on to understanding mic specs (and I promise you that there will be a
lot of surprises there). Polar patterns are NOT what they "appear to be", "fudge factors" are major parts of most
spec sheets, and microphone frequency response curves are often meaningless, misleading, or worthless, either by
accident or by intent.
Stick around - it's about to get real interesting, real quick.
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HomeRec, did I miss pianos? I'll get to it before we finish. I know I wrote part of an article for miking upright
pianos for Electronic Musician....
....Here's what I wrote:
Pianos and Mics - No Simple Solutions
Just as with most acoustic stringed instruments, the bulk of the sound is produced by the sounding board to which
the strings are attached. In guitars and violins, it's the top of the instrument; in pianos, it's the sounding board. You
don't mic the picks, the bows, or the hammers - they produce very little sound.
There are several considerations when placing mics for piano recording. Foremost, will the instrument be recorded
by itself, or with other instruments playing at the same time? Those two situations require different mic techniques.
Is it a grand piano or an upright piano? Each requires different mic techniques. Finally, where will the recording
take place? That may also require different mic techniques.
If the purpose of the recording is accuracy, and you're micing a solo concert grand piano, then you'll need some
good, small diaphragm condensor mics, placed some distance from the piano, usually around 6 to 8 feet away. You
can use an x-y setup for cardioids, or a wider spaced ORTF setup with omnis or cardioids.
The piano lid is used to direct some of the sound towards the mics. IF the piano is part of a group of instruments,
you can get better isolation by micing the underside of the instrument, using a slightly wide spacing with omnis or
cardioids. Mics placed inside the top of the instrument can also be used, but it's harder to achieve a good balance or
isolation since the piano lid will also reflect sounds from the other instruments into the mics.
Large diaphragm mics can also be used, but the response changes as the sound enters from different angles and the
larger mics add coloration (which can sometimes add an unexpected richness to the sound).
Upright pianos should be miced from the back of the instrument, but try to avoid having the soundboard too close
to a wall. The distance from the wall will create a standing wave, which will interfere with the sound. If the piano
has to be near a wall, angle the piano so that it doesn't sit parallel to the wall. Be especially attentive to a ringing
sound when micing upright pianos.
This ringing is caused by resonances within the piano, and usually can be solved or reduced by moving the mics
around till you find a dead area, free of the ringing. Just as with a concert grand, close micing is not advised, but
since an upright piano is usually part of a group, it's not possible to mic from a distance and still have isolation.
To sum it up, first choices for recording a piano would be small omni or cardioid condenser mics, but don't be
afraid to try large condensers, ribbons, or dynamic mics (if that's all you have). Mic from a distance if possible.
Second choice would be under the piano, and finally, from the top of the piano, but watch out for ringing and
reflections from that position.
>
[QUOTE] Originally posted by Wil Davis
Thanks again for a great thread, Harvey. BTW any thoughts on using a couple of PZMs taped to the inside
of the lid of a grand piano?
PZMs taped to the underside of the lid wouldn't exactly be my first choice for miking a grand piano. It might be ok
if the piano was being used in a rock setting, but even then, I would probably try some mics under the piano first.
>
[QOUTE] Originally posted by muzeman
Do you think I can use just one for mono on an acoustic with vocals on another mic, or do you think I
would do better with a large diaphragm figure 8 on the acoustic?
The figure-8 pattern will almost completely eliminate the voice from the guitar track. Other than recording them
separately to begin with (which would be the best way), a figure-8 pattern on the guitar would be best.
>
Lesson 18
Microphone Frequency Response - The Window
Most microphone manufacturers quote frequency response numbers somewhere on their spec page, and it's usually
something like "20 - 20 kHz" (or "30 - 15 kHz"), but what does that really mean, and how does it relate to what
you hear?
For that you'll need to know how to read a frequency response curve, add in what they "don't tell you", and
understand the amount of deviation possible between identical units. But before we can do that, you need to know
how microphones are measured.
Even though computer measurements have replaced a lot of the mechanical measurement systems, companies (like
B&K) still provide precision microphone test equipment, consisting of a frequency sweep oscillator, synched to a
chart recorder, and a ruler flat test microphone.
Basically, you feed the oscillator signal into something that will generate the sound, hook up the mic you want to
test, and the calibrated mic, then sweep the entire audible frequency range while you chart the "difference"
between the calibrated flat mic and the mic you're testing. The resulting chart is the frequency response of that one
microphone.
Calibration mics usually come in two flavors: direct measurement mics (on-axis), and diffused field mics (usually
90° off-axis). Direct measurement mics are used in anechoic chambers where there is no sound bouncing around so
the mic can be designed to be absolutely flat on-axis (i.e. pointed straight at the sound source). As you aim the mic
away from the sound source, the high end response of the microphone drops off dramatically.
Diffused field microphones are used in normal type rooms where pointing the mic directly at the speaker will pick
up unwanted reflections. When making measurements with diffuse field mics, they're usually pointed 90° off-axis
(towards the ceiling, the floor, or one of the sidewalls). Diffuse field microphones are flat 90° off-axis, but they
have a large rising frequency response on-axis.
So we now measure our mic, using one of the two methods described above and we look at the chart that was
produced, but that only tells us about that one mic. . Here's the mic curve for "our mic":
We'll need to run a batch of the same mics to see how much they'll vary from this one mic we just tested. To make
it easy to compare the frequency response, we'll adjust the level so that each mic is set to the same level at 1,000
Hz (although we'll keep track of how much the level needed to be adjusted for each mic). Let's say we test 50 mics.
We lay out the 50 charts and we also have a blank piece of chart paper in front of us.
We find the lowest frequency (20 Hz) on each chart, and look for the highest signal level (loudest), and the lowest
signal level softest) we measured at 20Hz. We put two marks (shown in red)on our blank piece of chart paper at 20
Hz. We do the same thing at each line, peak or dip on the chart, until we have an upper and lower row of dots that
represent the maximum and minimum range of frequency responses from this batch of mics. Here's the curve for
"our mic" and the variations we found in testing 50 mics:
We then connect all the upper red dots, and we connect all the lower red dots (with the final curves shown in grey):
We can then draw a line (the blue curve) exactly centered between the upper and lower dots and that's our "typical
response curve" that we submit to the marketing department. Understand, at this point, the curve could look fairly
flat, but individual mics can vary by 5dB or more from the "average curve", and still be considered "normal".
(Remember we also adjusted the output level for a constant 1,000 Hz signal from each mic? That will thow off the
results even more and be critical when it comes to finding a matched pair).
Well, our sample (in black) isn't too far off the average (in blue), but we might find some mics in that batch that are
better in the bottom end. How tight to hold the "deviation from average" window is a judgement call by the
company and then carried out by the quality control department. At companies like Neumann, they use a 4dB
window, which means that all mics must fit within a +/- 2dB window (4 dB overall) of their published curve. B&K
test mics may use a window as small as +/- 1/10th of a dB variation from their published curves.
But our "average curve" may still look "too jagged" for public consumption" from the marketing department's
point of view, so the curve can be "smoothed" by averaging some of the jagged peaks, or slowing down the pen
speed on the chart recorder (so it doesn't move as fast up and down and makes the curve look smoother by simply
ignoring all the little jagged short bursts). These are usually marketing decisions, so that "our curve" looks similar
to "other companies' curves":
And there we have the final "respectable" frequency response curve that is published in the advertising literature.
Now, here's another "gotcha" for most pressure gradient mics: the frequency response will change, depending on
the distance from the sound source, or the angle to the mic. Some manufacturers will actually show the "proximity
effect" on the frequency response chart, showing how the bass is boosted as you get closer. Some will also show
the frequency response at different angles (usually 0°, 30°, 60°, 90°, and 180°), like this:
When you look at a number like "Frequency Response: 20 - 20k", look at the published curve to see what the
"usable response" really is, and remember that the curve you see is "averaged and smoothed. Unless the deviation
is shown (either as a gray area or a line above and below the curve, or a number like +/- 3 dB), you really don't
know what your mic is really doing. That's why it's so hard for the average person to tell what a mic might sound
like, judging from the frequency response curve, or just reading the specs.
Any questions so far?
>
Three more things I forgot to point out.
1. There is no mic (in any batch you test) that will match the advertised blue curve.
2. If you happen to get a worst case mic that has the horrible peak at 200Hz and at 7,000Hz, it might sound very
bloated and screechy if you have a singer with a lot of energy in those ranges, or it could sound "full and detailed"
if the singer doesn't have a lot of energy in those areas.
3. Even though the response takes a nose dive after 10kHz, and starts to rolloff below 100Hz, it is still capable of
responding to energy from 20Hz - 20kHz, and the manufacturer can advertise it as such (and not bother to publish
a curve).
>
Yeah, I have a question, or rather a clarification so I'm sure I understand this properly:
Assume you're comparing some mics of the same model, as in your second jpeg, with the red dots. At
200Hz for example, one mic reaches 10 dB and another mic reaches 18 dB.
Now suppose that apart from the 200hz deviation, these two mics have similar response curves (a very
theoretical assumption no doubt). Would you then be able to make a recording with the mic with the 10 dB
response, and on your EQ boost the recording 200hz 8 dB, and as a result have the sound you would have
obtained if you had recorded with the other mic?
Assuming the peak was the only difference and you could match the Q of the peak (the Q is what determines the
shape of the frequency boost or cut) with a parametric equalizer, would that eliminate it? Not exactly.
See, that's a resonant peak, which means something in the design is resonating at that frequency. A parametric EQ
won't make the peak go away entirely because there's always gonna be some resonant energy hanging in there after
the note (that excites it) stops.
If so, then I can understand that's a helluva difference! I mean boosting any recording 8 dB really alters it
A LOT. Are there any real mics out there that show these big differences between the individual mics?
Even Neumann's guaranteed +/- 2dB could really make a noticeable difference (if you were unlucky
enough to get hold of two mics on each end of the spectrum).
Take a look at the curves from Beyer, Sennheiser, and a few other major companies; +/- 2dB is VERY good as far
as tolerances go. To be fair, most of the big names do hold good tolerances but it can get hairy in the high
frequency end of the spectrum on the lower priced models.
Can we trust manufacturers that sell what they claim are matched pairs of their mics?
Usually, yes. The manufacturers who sell matched sets can usually use their test curves to find two mics that are
similar in response and sensitivity. (Kinda like going thru those 50 curves we ran and finding the two curves that
are very similar: that's our "matched pair" of mics.) Not necessarily the "best two" mics, but the "closest two" mics.
>
[QUOTE] Originally posted by dobro
A comment, a question.
So, in the end, forget the published graphs and trust your ears.
Well, kind of. You CAN trust some of the curves, once you know how to interpret them, and if you can find the
manufacturer's stated tolerance (usually buried in the fine print, or a dB range printed right on the graph itself).
When you see a wide tolerance number, ask yourself "Why do they need a large tolerance if their quality control is
capable of much smaller numbers?"
However, the Neumann published graphs are pretty trustworthy, from what you say. Any other companies
you know about Harvey that put out more or less reliable frequency response charts, 'smoothing'
notwithstanding?
Shure is pretty accurate, and so are many of the good quality mics (like Schoeps and Earthworks and DPA and
B.L.U.E., just to name a few). I'm sure Soundelux and Brauner and Soundfield are also pretty honest. All the
Oktavas I bought from the sound room had curves with them that matched very well with what I heard.
>
Lesson 19
Sensitivity - What's that all about?
Sensitivity is the measurement that tells you how hard your preamp is going to have to work to get the signal up to
a useful level. It's found by feeding a specific sound level into the microphone and measuring the output level of
the mic.
The older standard was µbars (where 1 µbar equaled a 74dB SPL). The new standard is Pascals (where 1Pa equals
a 94dB SPL). If the measurement is shown in µbars, simply add 20 dB to the output level to convert it to Pascals.
Here are some typical microphone output levels:
1.1 mV/1Pa = -59dB (very low output - requires almost 60dB of gain to hit 0 on the meters - typical ribbon mic
output)
1.2 mV/1Pa = -57dB
2 mV/1Pa = -54dB (typical dynamic mic output)
2.3 mV/1Pa = -53dB
5.6 mV/1Pa = -45dB
10 mV/1Pa = -40dB (typical condenser mic output)
20 mV/1Pa = -34dB
25 mV/1Pa = -32dB (very hot condenser mic output)
If your preamp gets noisy at high gain, avoid using mics with a big negative dB number. All that -dB number is
showing is how much preamp gain you're going to need to bring the signal up to a useful level.
Finally, you may see a number thrown into the sensitivity measurement that says "+/- 1.5dB" or "+/- 2dB" - that's
how much variation in output is allowed by the manufacturer between units of the same model of mic. "+/- 1.5dB"
means that one mic may have 3 dB more output (or 3 dB less output) than another mic of the same exact model.
>
[QUOTE] Originally posted by chessparov
Harvey, what do you consider a good SPL rating for vocal mics, and have you had any singers "blow out"
any?
A mic in the 128 to 135 dB max SPL level should be more than adequate for most singers. I've never had anyone
blow out one of my vocal mics.
>
Lesson 20
Maximum SPL - How Loud Can You Go?
Since chessparov just brought it up, let's discuss "Maximum SPL" and what that specification means.
"Maximum SPL" is the maximum Sound Pressure Level a microphone can take, at a specified level of permissible
distortion.
The problem with this spec is that some microphone manufacturers don't tell you what the distortion level is, or
whether it's the capsule or the mic preamp (inside the mic body) that's distorting, or they calculate the level at a
distortion level that's different from other manufacturers. If the distortion isn't mentioned, figure it's either 1/2% to
1% (tolerable), or 5% (starting to get gross).
If they show only one distortion vs. SPL figure, it's easy to convert that number to distortion figures other
manufacturers use, to help make a fair comparison. Here's how:
For a round diaphragm mic, distortion will usually double for every 6dB increase in SPL. So, if someone shows a
max SPL of 1/2% @ 128dB, it's gonna be around 1% @ 134dB, 2% @ 140dB, and around 5% @ about 148dB.
You can control too much output level by two methods: by placing the artist further back from the mic (which will
also help reduce wild variations in level due to movement), or by using compression to smooth out level
inconsistencies, but with the liability of a possible increase in room noise.
As you move the person back, the inconsistencies from note to note smooth out, but you pay the price of added
room noise.
Usually, you'll want to strike a balance by moving the artist back just a little (to control levels, but not so much that
you pick up a lot of room sound). For condenser mics, I like somewhere between 1 and 2" away (for ballads and
very soft singers), and 6 to 12" back (for the "belters"). I might also add some compression if their levels get really
wild (anywhere from 2:1 to 10:1 ratios, but only triggered on the VERY LOUDEST peaks).
The final section (on polar patterns) is coming up next, and that should wrap this whole thread up.
I'm sure they'll be some questions, and I'll try to answer them all, if I can. Thanks to everybody for hanging around
this long.
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Lesson 21
Polar Response - Turn, Turn, Turn
Ok, I noticed my last post was #299 in this thread, so I might as well finish this whole thread off at post #300, so
here's the last section, on polar response. I'm gonna use the sheet for the Shure SM-57 as the example, so download
this and follow along:
http://www.shure.com/pdf/userguides/guides_wiredmics/sm57_en.pdf
Notice the polar response looks very smooth and you can almost visualize what it would sound like if you moved
30° off to the side of the mic, but what "you see" isn't exactly what "you get". Look at the frequency response
curve to see why. Notice that the response of the mic is down a couple of dB at 125 Hz and it's got a 7 dB peak at
around 5kHz or so?
Now look at the polar response curve and find those 2 frequencies. Notice at 0°, they show up on the 0dB line? If
you were on axis, the 5kHz signal would actually be 7 dB louder than a 1kHz signal, but they're shown as identical
levels at 0°. In other words, they've been "normalized" to be the same level at 0° as all the other frequencies. In
reality, that 5kHZ line should be 7dB louder on the graph, all the way around.
The only thing the polar pattern shows is the general pattern of the mic at different angles; it does not represent
reality with regards to the actual signal levels you may get off axis. For that, you have to use the frequency
response curve and extrapolate (i.e., "guess") the actual off axis response from there.
And that's the "last secret" of this whole thread. We started off discussing "diaphragm size", and this last post
covers the polar pattern part of the question, although we discussed different polar patterns and their use earlier.
I hope you enjoyed this as much as I did, and I'll try to answer all questions, and clarify anything I didn't explain as
well as I should of. It's a big subject and I didn't go into it as fully as I maybe should have, but tried to make it as
useful as I could to the broadest spectrum of posters here. If it was too simple and basic for some of you, I
apologize, but just look at it as a refresher course. For the rest, I hope it's given you some insights into how to
improve your own recordings. Whatever - I've enjoyed the hell out of this.
Sincerely,
Harvey Gerst,
Just An "Old Fart" Recording Engineer 

>
Posted by chesterfield on 10-15-2001 05:03
Harvey......... with regard to close-micing loud sources (guitar amp, snare, trumpet), isn't high SPL the most
important factor? I just bought a Shure SM57 because it's universally acclaimed as THE mic for those applications.
Is it because this mic has high SPL?
It's been said before, but thanks again for all your time and effort putting this info together. Now I've got to go
back and find the link to print out the main text so I can save it and refer to it often.
>
Actually, it's the fact that the frequency response of the mic compliments the sound of the snare and electric guitar
that makes it so desirable. It adds a nice high frequency boost at the top end and has a natural rolloff at the bottom
end, just damn near perfect for rock snare and electric guitar.
>
Posted by Henrik on 10-15-2001 06:55
Hmm...I used to think SPL meant that louder sounds might break the mic (so a low SPL figure meant something
like "guarantee is void if you stick this into a kick drum").
One thing I wonder: Is it always desirable to have as high SPL as possible? Or are there situations where it could
be an advantage to use a mic with a lower SPL?
>
SPLs way louder than the maximum SPL usually means increased distortion. Some mic designs (like the
Sennheiser MD421) can handle 150dB without any problems - so can a lot of small omnis.
In general, large diaphragm condenser mics CAN handle quite a bit, but the internal preamp often overloads first,
so a pad is used to lower the capsule output by 10 to 20dB typically. Kick drum is a special situation, since the
main concern is the draft of air the kick head causes. That blast of air is what can kill the mic, not the high SPL
level.
Some mic designs (older ribbon mics for example) CAN also be damaged by high SPLs or drafts from fans,
speaker ports, doors closing, or even shutting the lid too fast on the carrying case.
So why use a ribbon mic (that typically has a low SPL rating)? Ahhh, the sound is wonderful. Many ribbon mics
use ribbons that are only 7/10ths of a micron thick (a typical human hair is about 20 microns thick by comparison).
The ribbon in a ribbon mic has essentially minimal mass and responds beautifully to a lot of signals, such as voice,
strings, horns, etc.. The resonance can be set very low (20Hz on an RCA 44BX), and that single piece of
corrugated aluminum ribbon has almost no other resonances, so it's very flat and smooth throughout its entire
response range.
An RCA 44BX or 77DX on a Marshall 4x12 cab, about 2 feet away, is a sound to die for. Lush, rich, beefy,
gorgeous, you pick the adjectives. As the frequency goes up, the SPL rating of most ribbons increase dramatically -
about 6dB more power handling per octave. Many pros will use a ribbon mic or large condenser mic for kick, by
placing the mic away from the kick (4 to 6 feet), in a long packing blanket tunnel in front of the kick.
>
[QUOTE] Originally posted by cyork
Are there frequency bands that tend to distort first? Are various mics sensitive to distort over specific
frequency bands depending on their internal resonances? Proximity effect comes to mind as an obvious
low frequency problem, but is it always in the low and low-mid that the mic will hit its max SPL first?
Well, the diaphragm has to move the furthest at low frequencies, so that's usually where the problem starts. For the
same output level, a mic diaphragm has to move twice as far for each lower octave.
>
Posted by HomeRec on 10-15-2001 14:28
Speaking of physics equations...
Apparently, this is Shure's answer to "What is the maximum SPL rating for an SM57?"
Can a dynamic microphone handle really loud sounds?

http://shure.custhelp.com/cgi-bin/shure.cfg/php/enduser/std_adp.php?p_refno=000217-
000007&p_created=950774400
as asked in
Maximum SPL for dynamic mics
http://shure.custhelp.com/cgi-bin/shure.cfg/php/enduser/std_adp.php?
p_sid=ZfTZkKZf&p_lva=&p_refno=010926-000008&p_created=1001509489&
p_sp=cF9ncmlkc29ydD0mcF9yb3dfY250PTExMyZwX3NlYXJjaF90ZXh0PXNtNTcgc3BsIHJhdGluZyZwX3Nl
YXJjaF90eXBlPTM<br>
mcF9wcm9kX2x2bDE9fmFueX4mcF9wcm9kX2x2bDI9fmFueX4mcF9jYXRfbHZsMT1_YW55fiZwX3NvcnRf
Ynk9ZGZsdCZwX3BhZ2U<br>9Mg**&p_li=
Any thoughts, Harvey?

>
Posted by vox on 10-15-2001 14:37
Harvey, the MXL 603 specs claim 137db for 0.5%THD, I know that you have a pair and that you have stated that
some of their printed specs may not be completely accurate, do you think this SPL rating is correct? would you feel
comfortable using the 603 on a snare or cranked up amp for example?
>
HomeRec,
I think my answer is pretty consistent with Shure's answer, and I think my numbers may even
be a little more realistic, since I believe they're talking about what kind of SPLs would cause
physical damage to the mic (as opposed to moderate distortion which I was discussing). We
both talked about how the low end is the most damaging.
vox,
That 603 spec sounds about right. For 1% distortion, that would translate to about 142dB. I
wouldn't be worried in the least about using it on a snare or a loud guitar cabinet.
>
Posted by HomeRec on 10-16-2001 12:11
So the maximum SPL rating for an SM57 would vary according to the kind of sound being recorded, instead of
there being a single SPL rating for all applications?
>
Well, actually, kinda, sorta, yeah. It depends on the mic design, the resonance of the diaphragm, and the day of the
month, but in general, max SPLs are often measured at either 1 kHz, or around 250 Hz. Ribbon mics, like the
Coles 4038, even specify the maximum permitted SPL at different frequencies.
Most mics can handle most signals, but ribbon mics and a lot of pressure gradient condenser mics don't like singers
in close, singing F, P, B, V, W, and T, or anything that produces an air blast, which can bottom out the diaphragm
or stretch the ribbon.
>
Posted by muzeman on 10-27-2001 21:06
Hi Harvey,
Thanks again for your help and all the great info in this thread, I'm still trying to figure the best way to record
guitar and vocals at the same time. I hope you don't mind me squeezing in here to ask you about this again.
I was thinking, watching groups performing on T.V. they always use a dynamic or back electret for vocals and a
small condenser for acoustic. The sound they get is equal to or better than the studio sound. (Stained was on
unplugged the other nite, they were fantastic!)
Do you think this is a possible way to go in a home studio?
It seems like the plus for a dynamic or back electric on vocals would be elimination of room and background noise,
and the advantage of good close proximity effect for untrained voices, one of which I have the pleasure of owning.
The big disadvantage being, you don't get the sound of a large diaphragm. Am I on the right track with this?
I'm also a little confused about the different cardioid patterns, can you clarify the difference between cardioid,
hypercardioid, and supercardioid. Are there advantages and disadvantages to the different designs?
I’ve been looking at the Shure beta 87a and c ,one is cardioid and the other is supercardioid, both back electret I
believe. Also the sm7, which is a dynamic. I'm not sure which would be the best for my situation. Any advice or
other mic options would be appreciated.
Thanks again
Pete
>
Muzeman,
It's a little tricky to explain without the design theory, but lemme see if this'll help a little bit:
Cardioids have a heart-shaped polar pattern at most frequencies, but they tend to be more omnidirectional at low
frequencies.
Hypercardioids are less wide compared to cardioids, but still have some omni characteristics at lower frequencies.
Supercardioids are similar to hypercardioids at high frequencies, but they act more even at low frequencies by
creating a deeper rejection point at around 125° off axis.
So what the hell does all this mean when it comes to choosing the right mic, based on polar patterns?
If you're getting a mic for recording just your voice, it's easier to use a smooth cardioid mic that will be fairly flat
and natural and you don't worry too much about picking up bleed from other instruments due to the wide pattern of
most cardioid mics.
If you're playing guitar at the same time, you want to try and keep the sound of the guitar out of the vocal mic, so
you need a tighter mic pattern (like a hypercardioid) and you try to put the guitar in the null of the pattern so that it
doesn't get heard by the vocal mic. And it holds true for the guitar as well; you might use a second hypercardioid
on the guitar to keep the vocal out of the guitar mic. But hypercardioids aren't perfect, especially at lower
frequencies.
That's where the supercardioid comes in; it's got a solid null point at 125° at just about all frequencies.
So why not just use hypercardioids and supercardioids for everything? Part of the problem is that hypercardioids
and supercardioids don't always have the best frequency response, so you pay a price in performance for that
deeper rejection. And they have more proximity effect (which is not always a good thing).
For stage and live work, the rejection in a hypercardioid and supercardioid mic is a blessing, especially when
working with on-stage monitors, but it's not as important in the studio, where accuracy counts more.
Does it make a little more sense now?

BTW, it's a "back electret", not a "black electret". That means the condenser element is pre-polarized (carries a
permanent charge) by putting the charge on the back plate rather than the diaphragm. The difference in voltage
between the diaphragm and the back plate is how a condenser mic works.
>
Posted by gershwin on 11-10-2001 18:57
There is a lot about recording instruments and solo singers, but what about recording a 4-part male a cappella
group? We have 13 guys, 4 basses, 3 baritones, 4 Tenor 2's, and 2 Tenor 1's. At any one time, we have 11 guys
singing backgrounds, 1 soloist, and one Vocal Percussionist.
What would you suggest as far as recording this group? Would it be best to record the entire group together,
further out, with an X-Y? Should we add to that mics for each section? Or should we record each section
separately?
>
Wow, I'd love to record you guys. Without hearing you or seeing your staging, that's a difficult question to answer.
If I just had two mics, I'd probably start with X/Y or an NOS pair and adjust the spacing till the middle sounded
right to me. If I were to add 2 mics, it would be for the soloist and the vocal percussionist next. Finally (assuming 8
tracks), four more mics to spot each section.
Two mics are all you need usually, but you really hafta get the angle and distance just right to get the full stereo
spread, and the right balance of natural reverb. I'd get a "gofer" to move the mics around while I listened to the
sound. looking for the "sweet spot" where it all comes together.
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Posted by muzeman on 11-11-2001 18:07

I was wondering if Harvey or anyone else has used the Neumann KMS105, I have a chance to get a good deal on
one. Thinking about trying it on vocals, with another mic on acoustic to record both at the same time.
>
Every engineer that has used the Neumann KMS105 has raved about it. These comments are from people I know
who’s ears I trust. They say it's the finest stage mic they've ever heard.
>
Posted by Eurythmic on 11-26-2001 19:48
I realize that there aren't a lot of people left who still like this band, but Duran Duran is one of my biggest
influences. Would you happen to know what vocal mic Simon LeBon was using during the "Rio" era? I'm thinking
it must be the same microphone across the whole album, because that effect is there on every song. Although I'm
not enthralled with the sound on that album as a whole, I really like the way the vocals sound. I wouldn't want to
sound like that all the time, but if I could figure it out, it would be a great new color to add to my "arsenal" (which
is currently sadly lacking in variety).
>
Duran Duran's "Rio" was done at Sir George Martin's Air Studios, London. You could go to http://airstudios.com
and see if someone there has the tracking session sheets. If that doesn't work, here's a list of the mics they
have (and it's a safe bet they probably used one of these):
AKG
1 CK5 (capsule for 451)
2 C12VR (reissue of classic valve)
9 C414 (large diaphragm condenser)
3 C460 (condenser)
8 D451 (cardiod condenser)
2 D12 (cardiod dymanic)
2 D112 (cardiod dymanic "the egg")
1 D19C (cardiod dynamic)
8 D190E (cardiod dynamic)
2 D202ES (cardiod dynamic)
1 D224ES (cardiod dynamic)
2 D25 (cardiod dynamic)
Beyer
2 M160 (hyper cardiod double ribbon)
2 M201 (hyper cardiod dynamic)
2 M260N (hyper cardiod ribbon)
Blue
4 Blue Bottle (multi-capsule tube mic system)
4 B4 Capsule (perspex sphere pressure omni)
Coles
11 4038 (classic BBC ribbon)
B+K
2 4003 (omni condenser)
2 4007 (omni condenser)
2 4011 (cardiod condenser)
Crown
2 PZM (pressure zone effect)
EV
1 RE15 (cardiod dynamic)
3 RE20 (cardiod dynamic)
1 PL80 (cardiod dynamic)
Manley
2 Valve
Nakamichi
1 DM1000
Neumann
6 U47 fet (large diaphragm)
3 M49 (classic large diaphragm valve)
3 M50 (classic spherical omni valve)
3 M150 (spherical omni M50 reissue)
4 TLM50 (spherical omni)
5 U67 (classic large diaphragm valve)
2 KM83 (minature omni fet 80 series)
9 KM84 (minature cardiod fet 80 series)
4 KM86 (minature selectable fet 80 series)
1 KM184 (minature cardiod fet 180 series)
14 U87 (large diaphragm selectable pattern)
6 U87Ai (large diaphragm selectable pattern)
2 U89 (large diaphragm 5 polar patterns)
6 TLM103 (large diaphragm cardiod)
5 M147 (large diaghragm cardiod tube)
5 TLM170 (large diaphragm fet 100 series)
Pearl
1 DC21 ()
Reslo
1 RV ("the bullet")
Sanken
2 CU41 (cardiod condenser)
Schoeps
12 CMC5 (condensor microphone body)
6 CMC6 (condensor microphone body)
3 MK2H (omni capsule)
3 MK2S (omni capsule)
7 MK4 (cardiod capsule)
9 MK21 (wide cardiod capsule)
2 MK21H (wide cardiod capsule)
3 MK41 (hyper-cardiod capsule)
1 Schoeps Stereo (Dual Capsule Microphone)
Sennheiser
14 421 (cardiod dynamic tom mic)
3 441 (similar to 421)
2 MKH20 (omni condenser)
6 MKH40 (cardiod condenser)
2 MKH50 (hyper-cardiod condenser)
4 MKH800 (extended response switchable condenser)
Shure
10 SM57 (classic instrument cardiod dynamic)
1 SM58 (classic vocal cardiod dynamic)
1 SM98 (minature super cardiod condenser)
Sony
3 C8000 (modern valve)
If I had to bet money on which mic he used (without any more information), I'd bet Simon used one of their 5
Neumann U-67s.
>
Posted by Henrik on 11-27-2001 13:53
I was wondering, suppose you have a pair of subcardioids like the 603:s (which I don't since Marshall lacks
distribution up here in the cold North). Anyway. Suppose you'd like their polar pattern to be a bit more narrow, for
instance if you want to x/y mic with them, or if you are trying record a guitar while the guitar player is singing into
another mic at the same time. Could it then be a good idea to tape some foam rubber directly onto the mic, forming
a small tunnel around the diaphragm, but of course leaving the front opening uncovered?
I've seen some mics that come with an optional pipe, about a foot long, which you can attach to the mic in order to
make it very directional. Also the AKG c1000 comes with a small plastic cover that you can put over the
diaphragm to change the polar pattern from cardioid to hyper. I was just thinking that this idea may be applicable
to other mics. What do you think?
>
If you block the side vents of a cardioid, you get....
.... An omni mic. Ever wondered why a singer gets feedback when he/she cups the mic? Now you know.
Adding mechanical stuff can work, but it's really hit and miss without proper equipment to see what you're doing
to the response. Chances are you'll get some resonant peaks that are more directional.
>
Posted by Folkcafe on 11-28-2001 19:11
I know this is going to be off the subject but the section on drums prompted this question that I've never been able
to get a satisfactory answer to. Do smaller drums record better than larger ones. Part of this is due to always having
drums in the studio that are of the more traditional (larger size). I continue to hear and read this but would like
some insight to this issue.
>
Don, if the drums are tuned correctly, you don't need a big set, even for heavy metal. Our most popular kit here at
the studio consists of:
A 22"x16" Tama Rockstar DX Kick drum

A 10"x10" Tama Rockstar DX High Tom
A 12"x11" Tama Rockstar DX Mid Tom
A 13"x12" Tama Rockstar DX Low Tom
And a 16"x16" Tama Rockstar DX Floor Tom
The 13"x 12" doubles as a floor tom (replacing the 16"x16") or as the third rack tom, depending on the group.
When tuned and miked correctly, the sound is huge!!

>
Posted by nezpierce on 11-29-2001 06:46
I would totally agree with Harvey on the tuning part. New heads for the drums (which can get costly if you have to
change them every time you record) help immensely if you are going for a real tone-y sound. If you just like the
drums to be pitched "thuds", you can get away with older (less than 6 months) or thicker heads (ala remo pinstripes
etc).
I find that smaller thinner cymbals tend to work better as well. for example, I once did cymbal overdubs for a
record that had a lot of triggered drum sounds and I used a 12 UFIP splash as a crash cymbal. It sounded fantastic!
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Posted by Henrik on 12-11-2001 21:33:
I'm in the business of getting a stereo pair of mics for drum overheads etc. The Oktava MC012 is a bit out of my
price range, but I'll of course go there if I must. But in your big test of Marshall mics (another near classic Harvey
thread) you mentioned that the MXL603's sound almost identical to the Oktavas. One difference however seems to
be the polar pattern, the 603's are wide cardioid ("near omni" I think you said). When stereo micing, I prefer the
x/y setup, since a lot of my stuff goes on the Internet, and mono compatibility is necessary for lightweight files.
Bla bla bla, anyway - do you think the 603's wide cardioid pattern makes them unsuitable for x/y micing?
Plus, do you think it's OK to buy a pair without having heard them, or should they be matched? This would present
a problem for me, since I will have to buy them online.
>
Posted by Harvey Gerst on 12-12-2001 00:31:
I would have no problems buying two 603Ss and not having them matched, of course I would like them exactly
matched if I could get them that way. I would also pick up a pair of those dirt-cheap Behringer reference mics and
try those as well, wide spaced.
The 603S should work fine as an X/Y pair. They're wide, but still very cardioid.
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[QUOTE] Originally posted by Al Sim
Harvey, would a pair of Behringer ECM8000s be good for X/Y recording an acoustic guitar that
has a lot of bass response? Or would Marshall MXL 603s be better? Any other budget suggestions?
The guitar is a wonderful Gibson from the 40's and I'm doing a terrible job with a Rode NT1 and an
A-T MB4000C. Thanks, you are truly the man.
I don't think you'll get enough stereo separation with two ECM8000s in X/Y, but hey, try it. Try NOS or one of the
other near coincident spacings to see if that gives you a better image.
The use of omnis will certainly help tame the bottom end on an old Gibson (is it an old Gibson J45?). You might
also try playing around with different string gauges on the low E and A strings to control levels.
>
Posted by Al Sim on 01-09-2002 17:17
Yes, I believe it is a J45. I know it's a something-45. I assume you've dealt with this instrument before. Man, you
are good.
The guitar belongs to the singer-songwriter I work with. I don't think I'll have much luck asking him to change
string gauges, so that'll have to be a last resort. It's a truly great-sounding guitar. I haven't bought the ECM8000s
yet. I'm trying to figure out what to get on a budget of $200 for recording this particular instrument. A pair of
MXL603s and an ECM8000? Any other mics you'd suggest?
Have you ever encountered an EV 676 and is it good for anything?

>
The J-45 can be a very difficult instrument to record. What the audience hears is NOT what the player hears. They
can sound downright "tubby" from out in front of the guitar. I'd probably try the ECM8000s first, doing the "over
the shoulder" routine, and then dink around with the placement from there.
This is the one guitar where you might need to add some EQ, if nothing else works (some "bass" to "low mid" cut,
anywhere from 80Hz to 250 Hz, depending on the guitar and the song).
For the right song, there's nothing that comes close to a J-45, but they can be a pain to record. It's also a lot easier
to record a strummed J-45 than it is to record a finger picked J-45.
I've had pretty good luck using a Shure SM-81, but that puts it out of your price range.
I'm not familiar with the EV, but most EVs always have "some" use in a studio. Does EV have a historical
database on line? You might start there, or ask them via email if they have any information about the 676.
>
Posted by Al Sim on 01-09-2002 20:04:
I will order an ECM8000 for over-the-shoulder. There are songs that I would like to try X/Y on, if not on the J45,
then on one of our other guitars. Would a pair of MXL603s be good for that, or is there another mic in that price
range that you would recommend? And do I have any hope of getting good results doing X/Y on the J45, or should
I not waste time trying? That should be it for questions from me, at least until the mics arrive. Meanwhile, I'm off
to the EV website to see what I can learn about the 676.
>
Yes, a pair of 603S's in X/Y should sound fine, as long as you're not too close. As far as if it's worth trying, yes - at
least you'll learn something about the technique, even if it doesn't work for this particular application (and it might
work perfectly for this).
Please keep in mind that even though stereo is great, it's not always the best technique for maximum emotional
impact. Just as black and white is sometimes better for conveying a mood (instead of color) in movies, don't be
afraid to think in terms of mono instead of stereo when it might be more appropriate.
>
I'm thinking of getting some figure 8 pattern mic, and my question is: Will all figure 8 mics give an equal
frequency response on both sides of the mic, or can this differ?
No, it can differ greatly depending on how the figure 8 pattern is achieved. Ribbon mics that are designed strictly
as figure 8 mics have the most perfect figure 8 pattern AND the flattest off-axis response.
If so, are mics with selectable polar pattern more likely to have different frequency responses in figure
position, compared to a mic that can only do figure 8 (like a ribbon)?
It depends on how well each diaphragm is matched and balanced.
I also wonder, for M/S-recording, is it necessary that the two mics have to be very similar (like in X/Y-
recording), or can they differ some? Could you, for example, use a C3 for side and an NTK for mid?
Yes, you could use that combination.
Any recommendations on good figure 8 mics
RCA 44BX, Royer, Coles 4038, Beyer 130?, etc…

>
Harvey,
Thanks for this. But I wonder - you are only mentioning ribbon mics when I asked for some examples of decent
figure 8 mics. Do you mean to say there are no condensers with selectable patterns that can produce a decent
figure-8?
Also, I understand your favorite method of overhead drum micing is a spaced pair of omnis. Does this mean you're
not too worried about the mono compatibility, or do you have a trick up your sleeve in order to make the spaced
pair work flawlessly in mono? (Hopefully some of the music you are recording gets aired in radio and TV). Why
don't you use the MS technique, which I understand is foolproof for mono?
Just came to think of another thing I never really grasped about the MS-technique (even after having read Wes
Dooley's web site. I must be dim. I'm a bass player, OK?):
If I understand it correctly the sound of the mid mic is subtracted to the sound of the side mic on one channel, and
added to the sound of the side mic on the other channel. But what exactly does it mean to subtract a sound from
another (as opposed to adding one)?
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Lesson 22
Breaking down the M/S technique in simple terms.
Okay, I'm gonna try and do this without vectors, math, "sum and difference" signal theory, and no pictures - except
for the pictures we'll draw in our mind. Let's start by imagining a figure-8 ribbon mic and how it works, since the
concept is really simple.
A figure 8 ribbon mic basically consists of a piece of thin metal ribbon, hanging in a large magnetic field. The
ribbon is fasten at the top and bottom, and a small wire is connected to each end of the ribbon. When there is a
sound, it causes the ribbon to move inside the magnetic field; a small voltage is generated, and that's the signal that
comes down the cable.
The ribbon is free to move forward and backward, but not side to side. If a sound comes in from the front, it puts
out a positive signal. If a sound comes in from the back, it puts out a negative signal. If a sound comes in from the
side, it hits both the front and back of the ribbon equally, and no signal comes out.
Okay, now let's put this mic into our mixing board, set all the balance controls straight up, and listen to it.
If we put a singer in front of the mic and another singer in back of the mic, it'll pick up both singers pretty equally,
because the front and back of the figure 8 ribbon mic are both open. A singer singing into the side of the mic won't
be heard at all.
Okay, is everybody cool on all this stuff so far? Cuz here's where we get a little tricky.
Let's imagine 3 trumpet players spread out across a stage. The player on the left is A, in the center is B, and at the
right is C.
At the same time, we're gonna turn our mic sideways, so that the front of the mic is facing the trumpet player (A)
standing at the left side of the stage, the back of the mic is facing the trumpet player (C) standing at the right side
of the stage, and the side of the mic is facing the trumpet player (B), standing in the middle of the stage. When we
listen to the recording, we can hear A and C clearly, and we won't hear B hardly at all, as expected.
OK, here's where the "sleight of hand" magic trick comes in.
Let's run the mic into a Y cable, except one of the ends of the Y cable is wired backwards. This means we'll have a
normal positive signal going to one channel, AND a mirror image negative signal going to the other channel.
If you bring up the level of either channel by itself, you'll get a good sound. But if you leave one channel up and
bring the other channel up slowly, the ENTIRE signal will eventually disappear when the levels are exactly equal.
The plus and minus signals exactly cancel out. But here's where it gets wild.
Stick a cardioid mic on top of the figure 8 and point it towards the center of the stage, and bring up it's level on a
third mixer channel, again with the pan control centered. We pretty much hear the middle trumpet now and we
hear the two trumpets on both sides of the stage, but softer than the middle one. And the ribbon mic is still silent,
so why even bother with it?
Here's the magic trick:
Move the pan control on the first ribbon mic's channel all the way to the left. Move the pan control on the
backwards wired channel all the way to the right. Whoa, full, glorious stereo. Kill the middle mono mic and the
entire signal disappears again. Bring back the mono center mic and we have full stereo again. What the hell is
going on - what's happening?
Think of each trumpet as putting out a little puff of positive air. As the trumpet note from A hits the ribbon the mic
faithfully records it to the track that corresponds to that ribbon mic's channel. At the same time it records a mirror
image negative to the next ribbon track (since it's wired backwards). It also records a positive puff to the cardioid
in the center).
When you add that center cardioid mic, it combines with the ribbon signal from the left, and moves the apparent
sound all the way to the left. The trumpet on the right gets the reversed channel's attention and it combines with the
center mic to produce a very right hand sounding signal. The center trumpet is only heard by the cardioid so it only
comes back thru the center.
By adjusting the level of the side channels relative to the center mic, you can get any stereo spread you want, from
full stereo to perfect mono.
So can you use condenser mics that have a figure eight pattern, or three cardioids, each pointing in a different
direction? Yes, as long as you can get the two side channels to null completely, or damn close to it. One mic or
side has to face left and the other side/mic has to face right, and the two channels hafta cancel out almost
completely. How you do that is up to you. The cardioid mic faces forward and completes the stereo information
gathering.
That's how M/S works. I use/like ribbons for the side mic cuz they just have one moving part, no phantom power
to worry about, and if the mixer channel has a polarity reverse switch, a simple Y cord will work. And the signal is
automatically identical to start with.
Does this about cover it?

>
Posted by Chris Shaeffer on 01-28-2002 03:52:
Hot Damn!
OK, I may not have it exactly right, but it sure is fun! Here's what I did:
GT 5sm in figure 8 into Art MP Channel 1

1/4" out to Track 1 in Cubase
XLR out to Art MP Channel 2 with phase reversed
1/4" out to Track 2 in Cubase
C1 in the center into Track 3 in Cubase
Set levels on the MP so that they are identical (which was quite a game), set levels equally into the computer...test
in mono- yup the sides pretty much disappear.
Play some Tom Petty-ish stuff from the center position. (Feel like an idiot...) Sounded like doo doo in the cans.
Back in Cubase, panned 1 & 2 hard left and right. Left 3 dead center. Left all levels where they were- roughly
equal. Pressed play...
WHOA!!!!! That's some W___I____D____E stereo! And loud as heck. OK, hit the mono switch to see what it
does....
Drops about 12dB. Is it supposed to do that?
Hmmm.... Re-read Harvey's post several times.
I fiddled with the levels and found that if I keep the side channels lower (about -15dB) I can still get a nice full
stereo image that isn't too affected when I sum to mono (and doesn't sound too much like my...ah..."studio" room).
I'm not entirely sure what's going on, or even whether I have it right or wrong...
When I mute the center and have the sides panned out I get a really yucky, quiet, thin stereo. I was expecting it to
cancel out almost completely. If I then hit the mono switch it *almost* disappears- there is only a tiny bit left-
probably the inconsistencies in the two signal chains. The meters say it peaks at -22dB but I can hardly hear it.
When I bring the center back the stereo effect is much more noticeable than the tinny sides by themselves so I
think its working the way it should, but I don't really have perfect mono compatibility because of my half-arsed
signal chaining rig.
I wonder if:
- The figure 8 pattern on the GT isn't tight enough.

- My funky signal chain is causing this volume change in mono.
- It would work better recording more than one instrument.
- I somehow got it perfect and this is normal. (Not likely....)
Yehaw! Another fun evening courtesy of Harvey Gerst. Someday I might even understand some of what I'm doing.
>
What are the pros and cons of using the technique you describe above compared with using a matrix box or a
preamp that can function as one (such as Joemeek VC7)? If you do the latter, can you control the amount of stereo
spread after having recorded it, or are you stuck with the relative levels of the M and S mics?
If you connect a mic to a Y-cable, will it perform as well as it would with just one cable?
Also, what are your opinions on the different ribbon mics mention? Any recommendations?
>
The matrix box means not having to make up special cables or worry about matching stuff. It just makes life a little
easier.
Any good ribbon or figure 8 mic should do a pretty good job. The Coles 4038 is a very good choice and many
people like the AKG C414 for M/S. The stereo spread is determined by the level of the M mic relative to the S
signal.
>
Harvey,
I'm still wondering - what are your reasons for preferring spaced omnis to M-S while recording drums?
>
Mainly because:
1. One of our drum rooms is too frigging small to really get a great natural drum sound in.
2. Most of the stuff I do is rock and metal, and spaced omnis or spaced cardioid miking gives me a few more
options later during mixdown.
3. It makes the drummer feel more professional when I run 10 channels of drums for a 4-piece rock group. That
leaves a channel for the vocal, a channel for bass, and 12 channels for the guitar, which works out about right these
days.
>
A friend of mine just visited a studio where they were recording drums - using no less than 32 mics! Now what's
THAT all about!? Well, maybe they were trying to make the drummer feel important.
>
Some engineers are very anal about miking drums. They mic each head (top AND bottom), each cymbal, and then
throw some room mics in as well. Most engineers aren't usually that persnickity.
PAGE 18
Posted by wes480 on 04-12-2002 01:21:

Maybe this has been covered - but I haven't seen it.
On the one hand, the 3:1 rule is recommended...
Each mic should be 3 times as far away from each other as the first one is to the source.
So why does XY micing work?
If the mics are so close. Seems like if say, one was 2 feet from the guitar...according to the 3:1 rule, the next mic
would have to be like 6 feet away from that mic.
With XY...they are right next to each other - I don't get it
>
The 3:1 rule is really for when you're recording mono instruments, although it has some bearing in stereo (like
drum overheads).
XY places the mic capsules so close together that they essentially pick up exactly the same signal from an
instrument directly in front of them, without phase problems. As the instruments spread out from the center, they
are picked up by one mic more than the other. Because they are so close together, the phase problems are
minimized.
You can also use two mics, if they are exactly equal distance from the source, with out phase problems. That's the
basis behind ORTF, NOS, and several "near coincidence" stereo-miking techniques.
>
Harvey and others,
I just wanted to say thanks for the tip on Behringer ECM8000's. I couldn't get them in Sweden for some reason, so
I had to order them from Germany. They cost about the same as you pay in the US.
And boy, was it worth the effort! As of yet I have only tried recording drums with them. I placed the mics in front
of the kit instead of hanging above it. That way, they picked up more of the resonant heads of the drums, and
generally gave a more true image of the drums, which I liked a lot. First I placed them about two feet above the
floor. Lots of resonant heads, but the drummer preferred the sound we got at cymbal height, so I folded.
You can hear the results at http://polyester.just.nu - The song called “Medicine”.
>
OK, I can see where my response might have been confusing so let's see if I can clear things up a bit.
[QUOTE] Originally posted by wes480
So - why does XY miking sound better than just using 1 mic?
X/Y is a recording technique for getting a good stereo image of a wide sound source, using just two microphones.
If they are basically picking up the same source.... just so you can easily pan? Is the only goal with that?
Makes sense I guess....
Using multiple mics at different distances from a mono source is a technique for getting different tonal colors and
interesting time delays that can add a distinctive character to the sound that isn't possible with just one mic.
As far as 3:1 - "mono instruments"? Obviously that doesn't mean an instrument you are recording with just
1 mic.... because then 3:1 wouldn't come into play, so... I am not sure what a "mono instrument" is.
An electric guitar where the sound comes out of one speaker would be a good example of a "mono source". Any
instrument that isn't going to have a stereo image in the final mix would be a mono source, even if you use several
mics on the instrument. When you do use several mics on one instrument that will be blended together into one
sound, you have to watch for phasing problems and that's where the 3:1 rule comes into play.
In terms of equidistant from the source.... Is that like you said always have your overheads the exact same
distant from the snare? On a drum set...
It's a good practice to follow, but sometimes you ignore the snare if you're trying to get more cymbal sounds and
you use the overhead mics spread very wide. You have to listen for possible phasing problems, but if you mic the
cymbals pretty close, it's not a big problem.
I haven't tried it yet... but it seems like that would really hurt the stereo image. You just need to bring one
mic in closer than the other?
Since the snare is probably the main focus in a drum set, you always make sure that your multiple mic setup on
drums isn't causing problems with the basic tone of the snare from mic phasing problems due to poor mic
placement.
X/Y, ORTF, NOS, "near coincidence" miking are just techniques for getting a good stereo image of a wide source,
using just two mics.
Whenever you go to more than just those two microphones (e.g., close miking a set of drums with 3 or more mics),
you can create phasing problems caused by multiple mics picking up the same source from small differences in
distance, which creates phasing problems.
PAGE 19

[QUOTE] Originally posted by wes480
So in terms of acoustic instruments...the 3:1 rule isn't a big deal.
Yeah, it is a big deal if the mics aren't the same distance from the instrument; then the 3:1 rule comes into play.
And, if you are recording an acoustic instrument with only 2 mics.... Then you really don't have to worry
about phasing issues most of the time?
If the mics are in X/Y (or one of the standard stereo miking configurations), no, you don't hafta worry about
phasing. If the mics are at different distances from the acoustic instrument, yes, you hafta worry about phasing.
And XY miking is essentially just like using an AT "stereo mic" or something?
Yeah, kinda like that.

>
Posted by wes480 on 04-13-2002 06:06:
"Yeah, it is a big deal if the mics aren't the same distance from the instrument; then the 3:1 rule comes into
play."
Ah...so, for stereo miking like that, you either do 1:1 or 3:1...and you are set.
>
Posted by h kuhn on 05-01-2002 02:08:
This link might be interesting, complementing some of the information given by Harvey (especially the drawings
are nice):
http://members.aol.com/mihartkopf/lexicon.htm
>
Lesson 23
Clearing up some misconceptions about tube mics and preamps.
OK, here we go about tubes. As far as most tube circuits in mics and preamps are concerned, they're usually a
single tube operating as a "Class A" device. Here's a picture of how a Class A circuit works:
See the signal coming in, at the bottom of the drawing? It goes into the tube (which is represented by that "S"
shaped curve), hits the transfer function and can come out amplified (as in the right side of the drawing), or simply
come out equally, depending on the circuit designer's intent.
It's important to realize that the output of a condenser capsule is very low, while the capsule's impedance is VERY,
VERY high. Very high impedance sources don't travel well over long distances, so it's important to convert that
high impedance to a lower impedance as soon as possible.
You have two choices: a Field Effect Transistor (commonly called an FET), or a Tube (some tubes love seeing
very high impedance sources). If you use an FET, it won't do much in the way of giving you any more signal, so
you'll need to add some more transistors to boost the signal a little bit. And most transistor circuits tend to distort
very easily (and in a nasty way) if pushed too hard.
Tubes, on the other hand, tend to simply round off the signal if they approach the top and bottom of the "S" shaped
"Transfer Curve", resulting in more musical distortion components, i.e., more 2nd and 4th harmonic overtones,
which are musically correct, creating a "warmer, fatter tone. That's what makes tube distortion so desirable in
guitar amplifiers.
Some tube preamp designs add more distortion by using a very small plate voltage to effectively shrink the length
of that straight-line part of the "Transfer Function" so that the tube saturates quicker and distorts faster. To me, it
sounds a little fake and un-natural, but a lot of people seem to like it.
So the main advantages to using tubes in mics are: Natural impedance converter, which also works as a gain
stage, limiter, and as an even order distortion generator, when pushed hard. One lesser-known aspect of using a
tube inside the body of a microphone is that the heat from the tube helps drive out any moisture in the capsule
when used in humid environments.
Since the tube must have heater and plate voltages supplied from an outboard power supply, it also makes sense to
generate the 48volt phantom power voltage from the power supply as well. This brings up another possibility when
using dual membrane capsules for multiple polar patterns: Continuously variable remote polar pattern selection
from the power supply.
Remember earlier in this thread, we discussed how the different pressure gradient polar patterns are created by
mixing the sound from two polar patterns; omni and figure 8? We can take that one step further since a dual
membrane condenser mic is made into an omni by having both capsules charged. Flip the polarity of the back
capsule's signal and you have a figure 8.
As you continuously adjust the level and polarity of the back capsule, the mic will slowly change the polar patterns
starting with Omni, passing thru Wide Cardioid, Sub Cardioid, Hyper-Cardioid, and Super-Cardioid on it's way to
Figure 8.
If you are using one of these continuously variable polar pattern mics, it can be used to remotely change the tone
and the amount of proximity effect of the mic as well. As you move from Omni to Figure 8, the proximity effect
goes from almost none to maximum.
Many engineers will use the pattern selector switch as a tone control and ignore the different polar pattern choices
for a particular singer, since the mic is used in a pretty absorbent situation; an iso booth, or a very dead room for
example.
Often, the decision to use a tube mic is mistakenly made to increase distortion, resulting in what some people
describe as "tube warmth". In most modern mic designs, tubes are used for the performance reasons (listed above),
not to add distortion, but to eliminate the often unpleasant distortions caused by poorly designed transistor mic
circuits, which can often be described as harsh, edgy, brittle, etc.
One last point about LD mic design: a 1" wavelength corresponds to a frequency around 5 to 7 kHz. Ever wonder
why all 1" capsules have a peak right around that frequency range? Now you know. Explaining what to do about it
is a whole 'nother subject, which we'll get into at another time.
I hope some of this has proved helpful to at least somebody out there.
>
Posted by h kuhn on 05-16-2002 20:36:
Harvey,
How would the drawing differ for a class A/B or class B circuit?
>
Class AB and B divide the signal up into two sections: The positive half, and the negative half. A separate tube
drives each half of the signal and they're recombined in the output stage. Unless the circuit is designed very
carefully, right where each side shuts off (as it hits the zero point) can create a slight lag, causing what's called
"crossover" or "notch" distortion. It's a very ugly sound.
Class AB tries to prevent this by having each side operate as Class A at very low levels (i.e., both sides passing the
whole signal), gradually switching to class B as the signal gets louder and louder.
Class A is usually used in low-level signal amplification (preamps, mics, etc.), whereas speaker amplifiers
generally use Class AB, or other classes of amplification.
>
Posted by muzeman on 05-17-2002 02:24:
"If you are using one of these continuously variable polar pattern mics, it can be used to remotely change
the tone and the amount of proximity effect of the mic as well. As you move from omni to figure 8, the
proximity effect goes from almost none to maximum”
When you said change the tone and proximity effect, does this mean you can change the tonal character of the mic
to match the singers voice as well as the room?
When looking for a tube mic, are there certain electronic components to look for, other than class A circuitry, and
others to avoid?
>
Yes, by using the polar pattern selector to combine the two diaphragms in different ways, the frequency response
will change quite a bit. If the room is not contributing to the sound (by using an iso booth, or close-in absorbtion
baffles), you can use the pattern selector as a kind of weird tone control, since it alters several characteristics at the
same time (like frequency response, angular response, and proximity effect). It gives you a lot more color
variations to choose from.
>
[QUOTE] Originally posted by muzeman
Do you think there's a great advantage/disadvantage between class A verses A/B,B designs in tube mics
and pres?
Pretty much most single tube devices that amplify have to be Class A, but there's good Class A design and bad
Class A design. The other classes come into play when you need a lot of power and you want to split the work
between two or more output devices, like a power amplifier.

Unfourtanatly,most of us around here could never afford a class A tube mic or pre.
Prices are dropping on these things, but again, there's good Class A design and bad Class A design.

Would it be better to stick with a non tube mic or pre that's class A, than go for an AB/B tube design?
There's no tube mic I know of that uses a Class A/B or B circuit. The best mics and pres will be the best sounding
units that have well designed circuits and don't worry about the class - that's the designer's job - to figure out how
to make the device sound good.
When people make these things, there are only two or three possible choices; Make them almost one at a time,
using the best components possible, and sell them for enough money to make a decent profit. This is the way they
build Manley, Brauner, Milinia Media, or when you send something to Stephen Paul for modifications.
Or you can setup a factory, hire knowledgeable people, use good components, build them very well, and sell a lot
of them at a lower price.
The last way to lower your cost and lower the price is to have somebody else build something in a place that has
very cheap labor and is set up to crank out a ton of products. In order to do that, you need to install a large amount
of QC to insure the product does what it's supposed to and that the outside supplier hasn't screwed up somewhere.
P A G E 20
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[QUOTE] Originally posted by Grotius
1. I noticed that Harvey recommended a (matched pair?) of small diaphragm omni or cardioid
condensers for recording classical grand piano, which is my main interest. Noise and dynamic
range are obviously important to me. My main question is about the relative importance of various
specifications, especially self-noise and SPL. Is low self-noise more important than high maximum
SPL? (FYI, I have a new 7-foot Bluthner in my living room, and my tastes lean toward "quieter"
classical music like Chopin and Debussy, though I do enjoy playing the occasional Beethoven and
Rachmaninoff.)
Dynamic range isn't as much of a problem as self-noise when it comes to recording classical piano.
2. I'm also curious whether Harvey (or anyone else) has a preference among specific microphones
for this application. I've heard various recommendations, including the KM184, KM183, TLM103,
Rode NT2, AT 4033, AT822, ATM25, ATM87R, Oktava MC012s, and MXL 2003. Or, the DPA
4011, if one can afford it (which I can't). Also, in the low-self-noise department, does the TLM103
take the crown?
The ATM25 should be removed from your list. The Neumanns are pretty standard for miking piano, but
they have a tendency to be a little bright. A good pair of Soundroom MC012s or even the Marshall MXL
603S might work for you. All of the mics you list are pretty good for piano, but finding the best match-up
for your piano and playing style will take some work.
3. Finally, I gather that a single stereo pair is enough? Too many phase problems if one tries 4
mikes, e.g. to close-mike and distance-mike at the same time?
A stereo pair will usually do the job, although adding an extra mic for the room is often a nice way to get
real ambience.
>
Posted by Speeddemon on 06-10-2002 20:59:
Also, in the low-self-noise department, does the TLM103 take the crown?
The most silent mic I came across until now, was the Røde NT1000, with a specified self-noise of 6dB SPL (A-
weighted). That's pretty damn silent if you ask me!
>
Posted by Chris F on 06-11-2002 03:58:
2. I'm also curious whether Harvey (or anyone else) has a preference among specific microphones for this
application. I've heard various recommendations, including the KM184, KM183, TLM103, Rode NT2, AT
4033, AT822, ATM25, ATM87R, Oktava MC012s, and MXL 2003. Or, er, the DPA 4011, if one can afford
it (which I can't). Also, in the low-self-noise department, does the TLM103 take the crown?
The best sound I've ever gotten on my grand (1937 Baldwin 6'3") came from an Earthworks omni, but that was
borrowed and I can't afford the price tag. I get good results with a pair of MXL 603's in the x/y configuration,
although you'll have to do a lot of testing to figure out what kind of sound you're after. You'll get a very powerful
sound if you go close to the strings, but you'll also hear the mechanism of the action and the "whoosh" of the
dampers every time you pedal when you do this. Each placement scheme gives a slightly different color, and only
your ears can tell you what kind of sound suits your taste.
Two things to be careful of: A bright piano (or an old one with old hammers) brings all of the imperfections of the
instrument to the forefront of the recording, and you can hear some wolf-tones (overtones, etc.) on the playback
that you never even knew existed. Also, if your action hasn't been regulated and your strings leveled, you might be
in for some strange surprises as the mics are able to pick up a level of detail that your ears don't usually catch from
further away.
>
Posted by Grotius on 06-11-2002 17:54:
Harvey: Many thanks for the helpful feedback and suggestions.
Speeddemon: Wow, that's quite a self-noise spec on that Rode. The TLM103's spec is 7dB SPL (A-weighted), I
think. Can one hear the difference between that level of self-noise, and, say, 24dB SPL (A-weighted)? Between
35dB (my cheap mic's spec) and 24?
ChrisF: Thanks for your input. My piano is brand new, so the hammers are new too; it doesn't sound too bright
(yet). Mostly, it needs its first post-delivery tuning. When recording (with a cheap Sony mic), what bothers me
most is my playing! The rubato that sounds so wonderful to me while I'm playing sounds schmaltzy on tape. Some
pieces sound rushed; some sound too slow. My little microphone picks up every little technical deficiency in my
fingers. Recording is a great practice tool: There's nowhere to hide.
>
Posted by H2oskiphil on 08-07-2002 03:55:
NEWBIES TAKE NOTE!!!!!!!!
Gotta bump this up...As a newbie, speaking to other newbies...
YOU MUST READ THIS THREAD BEFORE POSTING!!!!!!
It will answer 85% of the questions you have. Print it, read it, use it as a reference source. Put it in a 3 ring binder
by your DAW or mixer. Tell your friends to read it. Tell the band to read it. When you're done reading it, read it
again! Take my word for it-it's worth the time invested.
You can't buy a book like this one...
P A G E 21
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(OFF Topic!!)
Posted by gibsonez on 08-11-2002 12:13:
In my heavy electric guitar recording, I get a lot of hum or build-up on certain frequencies between 100Hz and
180Hz. I'm not sure if this is the same as "hyped D" I read about before or if it's caused by AC hum or something
else. I'm using a Marshall amp and cab recording with a SM57 onto a Yamaha AW4416. If there is a way for me to
get rid of these problems that you may know of, please let me know.
Oh, yeah. And what is "hyped D"?

>
Posted by Axis on 08-11-2002 14:27:
What kind of guitar are you using? It may be the pickups, single coil pickups tend to hum. You may have bad
circuitry in the guitar(s)
Also it could be the cables if you have cheap cables they will make noise.
There are pedals I think there called hum eliminators or something like that that are supposed to reduce hum.
>
I use an Ibanez AX120 with a Gibson 500T in the bridge and the original AX1 pickup in the neck. The instrument
and mic cables are not the best, but I don't think they're horrible. That could be it. Maybe the inst and mic cables
are too close to the power cables. Could that be it? A hum eliminator or whatever would probably be a good idea,
though.
>
Posted by Axis on 08-11-2002 16:01:
When I get to close to my amp w/ any of my guitars the hum increases in volume, also at certain standing positions
it does the same. What kind of cables do you use? I use horizon and they seem to work good enough for me.
I doubt now that it’s your guitars. How many pedals do you have in your signal path if any? The more you have
the more the noise increases.
>
To be honest, I'm not sure what brand. They are cables made for a music store I used to work at. I have Horizon
cables too and I think the demo cable is just as good. That could be my "weak link". I think the problem lies
elsewhere though, like in my setup. I was just wondering if my problem was something typically encountered in
similar situations.
(Back On Topic!!)
Posted by Bass Master "K" on 08-19-2002 17:32:
I have a question for Harvey that believe it or not has not been addressed (at least as far as I can tell and I've read
through this several times)....
One of my projects that I am about ready to record is a bass oriented (some call it "lead bass") rock album ala
Stuart Hamm and John Paul Jones (at least his last solo album, not the Zeppelin years) and the usual direct
recording for bass is not going to cut it. It has become painfully obvious that I need to mic a cabinet to get the
high-end sound I am looking for. Should I treat it the same as a regular guitar cab?
I have a Harke 4x10 cab and a Fender Spectrum 1x18/2x10. I am planning on using the Harke because the Fender
is a little boomy for the sound I am looking for. My mikes: 2 Shure SM58, 3 Shure SM 57, 1 AT25, and 2 SP C-1.
I figured I'd use two 57's up on the cones and maybe a C-1 further back.... If the C-1 doesn't add anything then I'd
dump it. There are no restrictions in the number of tracks available. Does this sound right? Any advice is
appreciated.
Thanks goes out to:

1) Harvey, for sharing all of this knowledge with us.
2) Harvey again because he has saved me so much time and tears of frustration.
3) Tekker, for compiling it all into one document. I wish it could be put on page one because it would have saved
me about 5 hours of copy and pasting until I came upon his thread listing his link
4) Harvey one last time, for taking the time to do all of this question and answer stuff on top of just posting the
basic info. I can't tell you how much I appreciate this thread.
>
Sorry, I missed your question first time around. You might try the AT25 on one of the Hartke speakers, either of
the bottom two speakers for bass, or the top two for a little more high-end. We've been using the D112 for miking
bass cabinets lately, and it sounds pretty good when combined with a DI signal from our outboard preamps, like
the Great River MP-2NV or the Milinia Media SST-1.
>
Posted by Bass Master "K" on 08-23-2002 14:20:
Thank you very much Harvey. After I typed it out and re-read my thread I was wondering about that AT25... I
remember the guys at the local sound store said it had great bass response. I wasn't sure if those 57's were going to
have enough low-end response, but since this project has lots of finger tapping and I am looking for real crisp high
end I thought I might give them a try. I'll try the AT25....
>
Posted by dafduc on 10-11-2002 13:24:
Miking Leslies?
I killed a good 4 hours reading this thread yesterday! What a great thing you have done for us, Mr. Gerst!
What I don't see in here is how to mic a Leslie. My church has a great little L-100, hooked to a beefed-up Leslie
147. For recording organ parts, I'll usually just use a B-4 in my studio, but I've got a couple of gospel-style pieces
that need the Mavis-n-Lucky treatment, which means I've got to mic the Leslie, and cover the bass as well as the
manuals.
Mics at my disposal:
4 Shure AXS-4s (similar to the BE-4)
1 MK-319
1 SM-57
3 SM-58s
1 Green Bullet
I'm also considering buying:

2 MC012s or
2 Marshall 603s and/or
2 Behringer 8000s,
But am open to other suggestions.
One last limitation:

If I use my VS-840 for the recording, I'm essentially stuck with a stereo mix - Despite its 4 inputs, the 840's zip
disk is so slow that it can't write the 3rd or 4th track unless I seriously drop the resolution.
That's a WHOLE lot of detail for a VERY specific task, so maybe the questions for this thread need to be: How do
I mic a Leslie if I'm kicking bass? What if I'm not? Do the mics I use depend on where the organ needs to stand in
the mix? What if it's a different instrument (RIP, Danny G.!) through the Leslie? And does the answer change if I
use a 122 or a MotionSound instead of a 147?
Thanks again for a great thread. BOOMP!
Jay
>
Posted by ChuckU on 10-15-2002 20:47:
Jay,
A friend of mine went into the studio a few weeks ago to lay down some B3 tracks. The engineer/studio owner
stuck a 57 right on the slot that lines up with the horn. The engineer is also a keyboard player and that's what works
for him.
>
Posted by dafduc on 10-16-2002 07:27:
So, if I really wanted that stereo doppler thing, could I X-Y a pair of 57s? I don't think I've seen X-Y and SM-57 in
the same sentence yet...
>
Posted by Bigus Dickus on 10-25-2002 11:14:
Harvey,
I don't know if you still check out this thread from time to time, but I thought 'why not take a shot at a few
questions.' At the least, perhaps some other knowledgeable folks will have some advice.
You touched on grand pianos several times, but never really got that in depth on them. To give you a bit of
background, I want to make recordings of my grand in my home. It's in a fairly small and dead room, and this is
the only instrument I intend on recording in the foreseeable future (of course, like any project or hobby I get
involved with, it will probably become an addiction). I've read this thread entirely, and I also read and absorbed the
"Microphones: Method of Operation and Type Examples" by Gore and Peus that you recommended.
If I have indeed understood things, then I should be looking for a small diaphragm condenser or dynamic
microphone for accuracy instead of coloration (that larger diaphragms give), and a fairly low self-noise to catch
intimate passages adequately. I also gather that I should be looking for an omni-directional or wide cardioid
pattern. And, I should probably look for a pair (matched if possible) for stereo recording.
I'll stop here and ask a question: There was a list of possible microphones given a few pages back, and you gave
some general comments on those (and I've made a note of that list). In my case, I don't have the luxury of having
multiple microphones to play with looking for the sound I want, or a convenient way of purchasing on trial
different mics to audition. Instead, I'll have to make the best-informed decision I can, and hope that the mic pair I
purchase is satisfactory. Could you give any suggestions for a specific mic? If you only had one shot at recording a
grand (not in a concert hall, mind you), and could only choose one mic model and hope for the best, what mic
would you choose?
Now, about placement. As I said the room is fairly small and dead (and could easily be made 'more dead' if that
would indeed be helpful). From reading your wonderful advice, it would seem that an X/Y pair (or perhaps even
near-coincident) a couple of feet to the side of a high-stick lid would be a good place to start with placement... If
the room was nice. Mine is not. Perhaps bringing the mics closer (under the open lid) to record in their near field
would help to reduce the effects of room modes and odd wall reflections.
Which brings me to my second line of questions. I've heard several people speak of good results with wide spaced
micing under the lid of a piano. Wouldn't that cause phasing problems? I'm not really interested in mono
compatibility (since this will be primarily for friends and family), but I want to avoid as many complications as
possible. Would an X/Y setup under the lid have trouble with a balanced frequency response, especially
considering that some strings might be in the near-field while more distant ones would not? It would seem for that
omnis might be better suited than cardioid...
I suppose the bottom line question is: If you were attempting to record a grand in such a room, where would you
start? I know that mic placement plays a role in mic choice as well, so that makes it that much more difficult for
me. I think you understand what I'm after... The crucial part is choosing a mic pair that is workable, and I can play
with placement ad infinitum (but starting suggestions are nice).
Also, I'd like to add my thanks to that of many others over the past year and a half. I can't imagine how much time
it would have taken me to assemble the collection of information contained in this one thread by scouring the web.
For that, I am forever grateful.
>
dafduc,
Sorry, I was at the AES show when you posted and your question slipped in under the radar.
For those of you not familiar with a Leslie, it uses a rotating horn for the high end. There's only one rotating horn
on top (the other end of the arm is a counter balance). There's also a rotating baffle for the low-end speaker.
Probably the widest "swirl" would be a couple of mics placed 90° apart, one pointing towards the louvers on one
side, and the other pointed at the louvers on an adjacent side, both mics about a foot away from the cabinet. Some
people also use a 421 or similar dynamic to mic the bottom speaker and mix it on with the horn tracks.
But like anything else, it depends on how the Leslie is supposed to fit in with the other tracks. If it's not the
featured instrument, use a 57 on one side, aimed at the louvers and be done with it.
If you're recording Booker T., mic everything you can think of, even putting a mic on the keyboard to capture
keyclicks and any singing along – You don't hafta use them in the final mix.
P A G E 22 http://www.homerecording.com/bbs/showthread.php?
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[QUOTE] Originally posted by Bigus Dickus
You touched on grand pianos several times, but never really got that in depth on them. To give you a bit of
background, I want to make recordings of my grand in my home. It's in a fairly small and dead room, and
this is the only instrument I intend on recording in the foreseeable future.
If I have indeed understood things, then I should be looking for a small diaphragm condenser or dynamic
microphone for accuracy instead of coloration (that larger diaphragms give), and a fairly low self-noise to
catch intimate passages adequately. I also gather that I should be looking for an omni-directional or wide
cardioid pattern. And, I should probably look for a pair (matched if possible) for stereo recording.
Yes, those mic patterns are usually the best choices for people in your situation (i.e., small, non-treated room,
home recording, small budget, and not a lot of previous mic placement/acoustic treatment knowledge). That wasn't
meant as a putdown, BTW. You have the advantage of time, compared to studio recordings, where the clock
becomes very important.
Stereo piano recordings are usually the most natural sounding, as far as recording techniques, but it's also possible
to do an artificial stereo recording that duplicates the emotional impact of a stereo recording, without using
recognized stereo techniques (like recording from underneath the piano for the low notes, and from above for the
treble).
I'll stop here and ask a question: There was a list of possible microphones given a few pages back, and you
gave some general comments on those (and I've made a note of that list). In my case, I don't have the
luxury of having multiple microphones to play with looking for the sound I want, or a convenient way of
purchasing on trial different mics to audition. Instead, I'll have to make the best-informed decision I can,
and hope that the mic pair I purchase is satisfactory. Could you give any suggestions for a specific mic? If
you only had one shot at recording a grand (not in a concert hall, mind you), and could only choose one
mic model and hope for the best, what mic would you choose?
I can't really answer that without hearing the actual room and the actual piano. My first concern would be the
actual acoustics of the room, and then I would be concerned about the evenness of the tone of the piano.
Am I hearing any peaks or dips in the sound as you play while I walk around the room? Do I need to put up some
packing blankets or move furniture around? Is the piano thin or full in different ranges? Is it bright or mellow?
As far as mics, I'd be more concerned about acoustics and placement than I would be about the choice of
microphones. Most 1/2" condenser mics are pretty similar, so whichever mics were fairly flat, and low noise,
would be the ones I'd bring. I'd also bring a pair of the Behringer ECM8000s, just on the chance they MIGHT
work well, since $70 for a pair is kinda a no-brainer.
Now, about placement. As I said the room is fairly small and dead (and could easily be made 'more dead' if
that would indeed be helpful). From reading your wonderful advice, it would seem that an X/Y pair (or
perhaps even near-coincident) a couple of feet to the side of a high-stick lid would be a good place to start
with placement... If the room was nice. Mine is not. Perhaps bringing the mics closer (under the open lid)
to record in their near field would help to reduce the effects of room modes and odd wall reflections.
Yes, no, maybe. You have the luxury of time. Try everything you can think of. As I said, my first concern would
be the effects of room modes and odd wall reflections, and I would try to eliminate them (or reduce them) as much
as possible. Once those are tamed, then it would be time for experimenting with mic placements.
Which brings me to my second line of questions. I've heard several people speak of good results with wide
spaced micing under the lid of a piano. Wouldn't that cause phasing problems? I'm not really interested in
mono compatibility (since this will be primarily for friends and family), but I want to avoid as many
complications as possible. Would an X/Y setup under the lid have trouble with a balanced frequency
response, especially considering that some strings might be in the near-field while more distant ones would
not? It would seem for that omnis might be better suited than cardioid...
Yes, no, maybe - see paragraph above. First, fix the room, then understand the instrument (and the way it radiates
sound), then choose the best location to capture the desired sound.
Usually, my motto is, "The worse the room, the closer the mics!"
I suppose the bottom line question is: If you were attempting to record a grand in such a room, where
would you start? I know that mic placement plays a role in mic choice as well, so that makes it that much
more difficult for me. I think you understand what I'm after... The crucial part is choosing a mic pair that is
workable, and I can play with placement ad infinitum (but starting suggestions are nice).
All the above suggestions would be the way I would approach recording a strange piano in a strange location. But
if it's a paying gig, I would bring almost every damn mic I had that I think just might sound good. That's not a
viable option for you. What's your budget for mics? Is the piano bright or dark? What kinda music? What registers
will the bulk of the playing be in, and how does the piano sound in those registers? How much is the room altering
the sound? Without a better handle on those questions, I can't help much more on your situation.
>
[QUOTE] Originally posted by dafduc
Thank you, Harvey!
So the 421 would be about a foot away, too?
Yeah, about that, and pretty low to the floor.
The great thing about Leslies is the sound. The bad thing about Leslies is the mechanical noise when you get too
close.
>
Posted by Bigus Dickus on 10-25-2002 15:53:
Wow, I didn't expect a reply so fast (nor do I expect to be so lucky again)! Though you may have thought your
answers were vague, they've already helped tremendously in just confirming that I wasn't completely in the dark
anymore. Feeling more confident that I do actually understand a bit of what you have taught, I will be less timid in
trying things.
To fill in a few of the holes:

What's your budget for mics?
I'd be willing to spend a couple of hundred on mics, maybe ~$300 maximum.
Is the piano bright or dark? What kinda music? What registers will the bulk of the playing be in, and how
does the piano sound in those registers?
There is a Sibelius piece that covers practically the entire range, within a couple of notes of either extreme. I also
want to record a Listz etude that covers most of the range as well. The piano is a Baldwin artisan, and has a
wonderful sound from about 400Hz down. It gets a bit metallic sounding around 2000Hz, but I'm suspicious that it
may be the room as much as the instrument. From ~5000Hz up it's pretty smooth again. As far as musical style, the
Liszt etude is both beautiful and very dynamic, as is the Sibelius piece. Both have some very soft, and some rather
powerful sections. I'm naturally a heavy player as well, and tend to lean towards that style. However, I also want to
include a nice Chopin prelude or two.
How much is the room altering the sound?
That's difficult for me to say, because I've never heard the instrument in any other setting. My guess is that most if
not all of the mid/high range thinness and metallic sound is due to the room. I'm doing some research on what kind
of damping might help.
Without a better handle on those questions, I can't help much more on your situation.
You've already helped more than you could imagine. I was lost just three days ago... now I have an idea of where I
am headed. I will scour the mic forum a bit more, and see what condenser mics people are pleased with that have
low noise and fairly flat freq. response. The most important thing is knowing that the mic purchase isn't going to
make or break the recording (sure, I'm sure there is a perfect mic for me, but it sounds like there are many that will
give satisfactory results). And as you said, I have all the time in the world to play with placement, and you've given
valuable insight on what to try as starting points. If it takes a year to find the "sweet spot" then I'll just be that much
better rehearsed when the serious recording comes. More than that, I now will understand when I move a mic
around why the sound changes like it does. Nothing wrong with trial and error, as long as it isn't completely in the
dark... IMO.
>
Lesson 24
How to make your own multi-pattern microphone
While this thread is near the top, I thought this might be a nice addition, in that it will help many people understand
exactly how multi-pattern condenser microphones work.
We are going to make a multipattern mic. You can either actually try this, or just follow the thought process.
You will need two similar sounding cardioid microphnes, the closer the match, the better. If they are side address
(like a B1 or MXL990), even better, but a pair pf SM-57s will work fine.
Position them so that one capsule faces forward, while the other capsule is above the first and faces backward
(exactly 180°). With side address mics, you can use two stands and position one of the mics upside down over the
first mic, with the top of the screens almost touching. Just remember one of the mics shold be facing forward, and
one facing backwards.
The forward facing cardioid mic represents the front diaphragm in a multi-pattern mic. The rear facing cardioid
mic represents the back diaphragm in a multi-pattern mic.
Bring each mic into a separate channel on the board and set the pan controls on each channel to straight up, with no
EQ, so that both channels are identical. Bring up the level of just the forward facing mic - Keep the rear-facing mic
turned off. That's exactly what happens when you choose the cardioid pattern in a multi-pattern mic; they shut off
the rear diaphragm, just as you have the rear cardioid shut off.
Now start to bring up the rear facing cardioid slowly while you listen to the sound change. Think about what is
happening. A cardioid picks up whatever is in front of it, and then tapers off as you approach the rear of the mic.
But now, the second mic is picking up the sound from the rear, and then tapers off as you approach the front.
What's happening is that the cardioid pattern is starting to balloon in back as you bring up the rear mic level. When
you get to the point where the levels are equal, congratulations, you've just made an omni mic out of two cardioids.
As you moved the slider for the rear mic up, you went from Cardioid to Wide Cardioid to Omni.
OK, bring the rear mic's slider all the way down, and hit the polarity switch for the rear mic. You're now back to
cardioid (only the front facing mic is heard).
Since the rear mic is now reversed polarity, it's gonna act in a subtractive way, taking away some of the sound
from the front facing cardioid mic. OK, lets start raising the level of the rear-facing mic. Remember, with it off,
we're actually starting at the cardioid pattern again (front facing mic only is on). Let's start bringing up the rear-
facing mic again and try to imagine what is happening to the polar patterns.
The first place any change appears is in the rear (dead center), and at the right angle points. A small lobe starts to
appear behind the cardioid pattern and the sides of the cardioid pattern shrink in size. Why? Because sounds from
the side are hitting both diaphragms and cancelling out a little.
As you bring the level up, the back lobe directly to the rear starts to get bigger and bigger, while the front pattern
begins to shrink in width, due to the cancellations mentioned above. Imagine the patterns are like a balloon that
you squeeze in the middle. At first the sides kinda shrink till you finally wind up with what looks like the number
"8" - two smaller perfect circles. At that point, any sounds coming in from the exact right or left side should cancel
out since both mics are hearing it equally, but reversed.
Starting with the cardioid pattern, by raising the slider on the rear microphone (with the polarity reversed), you
went from Cardioid, to a Narrow Cardioid, a Hyper-Cardioid, a Super-Cardioid, and finally a Figure-8.
So, is there a little guy inside a multi-pattern mic working a little mixer? Yeah, kinda, except they do it with the
polarizing voltage on the rear diaphragm, raising, lowering, and the electronic equivalent of reversing it.
To get cardioid, they just use the front diaphragm. For omni, they turn on the back diaphragm full and add it to the
front diaphragm. For figure 8, the reverse the polarity of the back diaphragm and add it to the front pattern.
Every other pattern variation is made by just raising and lowering the level of the back diaphragm with the polarity
normal (for all patterns between Cardioid and Omni), and reversing the back diaphragm's polarity (for all patterns
between Cardioid and Figure 8).
So, do multi-pattern mics make a little more sense now?
>
Posted by Son of Mixerman on 11-12-2002 10:49:
Yes, very clear. I never thought of it that way before. Next question, can you substitute using 2 cardioids instead of
purchasing a multi-pattern mic in a pinch? A little more work, maybe a bit more variability? Thoughts?
>
Yes, actually, you can substitute a couple of cardioids, but the spacing won't get you as accurate patterns as two
diaphragms located on the same center line and only a few millimeters apart. You'll also get some phase problems
up high (the exact frequencies will depend on the diaphragm size and the distance apart). Some people will hear a
slight problem, most people won't. Smaller cardioids work better for this.
That was actually the idea I came up with which eventually resulted in the CAD E-200; use a pair of 1/2"
cardioids, back to back, and combine them like a standard multi-pattern condenser mics. In the original design, I
used a pair of matched YASU cardioid capsules, but Equitek decided to go with the Primo cardioid capsules
instead, which I always thought was a mistake.
The two diaphragms in a dual-diaphragm multi-pattern mic also interact, since there are holes in the backplate,
which are used to create the delay network (and that creates the cardioid pattern).
The whole idea of how to use both diaphragms in a condenser mic to create different patterns is really interesting.
>
Posted by BasPer on 11-12-2002 12:29:
I almost dare not post in this "the mighty thread", but multiple cardioids are not entirely new to me. Check below:
http://www.lineaudio.se/qm12man.htm
>
BasPer,
I didn't say it was a new concept. The design that eventually became the CAD E-200 I started working on in 1987,
as I recall, and I first showed it at the 1987 or '88 AES show in New York. It was called "The Mic" and it created a
great deal of interest as the first multi-patterm mic under $1,000 (actually, the suggested list price was $399).
And this is hardly a "mighty thread"; just a very big thread that I hope new recordists will find interesting and
informative.
>
BTW, I wound up using a Behringer ECM8000 to record an upright bass in a bluegrass group and the results were
wonderful.
No big setup either, just pointed the ECM8000 straight up towards the ceiling about 3 feet off the ground,
positioned about a foot away from the bass, so that the mic came up to the neck/body joint, just to the right of the
bridge. I had positioned it there since he was the leader of the group and it was intended to be a talkback mic so we
could communicate.
I had planned to take his bass direct at a later session, but it turned out so well for bass, that I just used it in the
final track.
Ya just never know.

>
Posted by Chris F on 11-12-2002 15:20:
That's VERY true. I've used the ECM8000 a time or two for the Double Bass as a "supporting solo mic" which
only gets faded in during upper register passages. Due to the "thick" color it gets on my bass, it's too much for a
whole track, but it works beautifully to fatten up the high stuff. For Bluegrass, the position you describe would be
great because it would pick up both the pitch of the notes AND the percussive slap that's needed to drive the
rhythm section. If I ever do a 'grass session, I'll try that.
>
P A G E 23
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Posted by Jeeper on 01-10-2003 01:34:

Maybe you answered already
I hope I've not missed this somewhere if so tell me and I'll sift back through until I find it.
Problem job for me I've yet to cure and there probably is not one but I'm asking how to minimize. From time to
time I'm asked to record a choir:
1. An overly reflective space.
2. Not enough real room to back up sufficiently either with mics.
3. The reflections and wide angle for stereo pair (not enough room be back up pair in space) have been a killer.
I've tried spaced pair large diaphragm and small, XY large, and ORTF and it's variations with large and I'm still not
happy with sound. Some methods have been better some worse. I do not have figure 8 in locker to try MS but
please say that would be your choice if it is.
Somewhere I've seen reference to small diameter mics being less bothered with room reflections than large. Is this
so??
What would your experienced first try at this job??
After all experience is the best teacher but mine has not taught me enough yet.
>
If you're recording a choir without an audience, try packing blankets set up in a "V" around the mics to try and
tame some of the side and rear reflections.
If you hafta mic in close with an X/Y pair, open the angle between them a bit wider than 90°, try as wide as 120°.
I might also try some wide spaced omnis in close, too.
It's a tough call without actually being in the room and hearing the problems.
Small diaphragm mics usually work best on wide sound sources because of better off axis response.
>
Harvey thanks for your speedy reply.
The choir I'm working with does not do well at all without an audience. That may be different but I've done sound
checks at final rehersal several times and they sound bad but when performance time comes they sound like
seasoned pros. My tapes back this up also so it's not just my thought at the time.
Your point about the small dia. may be well worth trying with xy. I'll probably put that up next with them. At the
same time I've got plenty tracks to use so I'll probably try large dia spaced pair mainly because I've not tried that in
this location and some of my better tapes was spaced pair with smalls.
I'm going to have to try to borrow or rent some omni's to try that idea. I have never had what I thought was a use
for an omni before but I still value your experienced opinion. I have been considering some multi pattern mics to
try ms and Blumlin (not sure if I spelled his name right) and that would cover the omni thing also. Again maybe I
need to go back and reread the rest of this thread about omnis.
BTW your mic closet is probably proof that it's selecting the proper tool for the job because your site lists almost
none of the got to have expensive mics. I hope my ear develops 1/3 that good for selection of mics.
>
Didn't ask this above
I do not follow your choice of the omni. Why use an omni in a reflective space that you are attempting to tame???
Or is that not your intention. Makes me wonder if maybe your intention is to leave the reflections in as space's
character and then go for your real first choice on any group.
Thanks I'm listening and probably are more people.

>
My suggestion about using omnis was based on three assumptions:
1. It was going to be for a "choir only" recording, no audience, so you had some mic placement options, but none
that were far enough away.
2. That you could set up packing blankets on either side of the mics to reduce side and rear reflections. A 4' to 6'
"V" around each mic would certainly provide some control of unwanted reflections.
3. Wide spaced omnis will often work when no other choices will. Omnis will also give you a little more space in
the recordings, even when used up close.
>
Thanks Harvey, I now I'll have something else to try when I'm able to get a pair of omnis available.
Your ideas are appreciated but it may be several weeks before I'm back on that location. Maybe they will not be
offended by me burning up a bunch of space in front and trying several things next round. I have way more tracks
than inputs and if the track does not work archive for latter comparisons. It will not be the first time with many
notes on track sheets.
>
Posted by MISTERQCUE on 01-23-2003 18:11:
Hello Harve Master Gee, is it at all possible to increase the diaphragm size of the mic thru DIY means and change
the coloring of the mic via store bought electronics?
In other words, I have a cheap Radio Whack cardioid Mic; can I alter the specifications of mic using current
technology and components found in upper grade mics. I have been doing some experimentation with this and a
COBY omni and was wondering if it can be upgraded and is it worth it?
>
Posted by cottonmouth71 on 02-09-2003 12:52:
In reading the thread I noticed you didn't have much to say at that time about Rode's NTK or NT1000. Have you
considered giving them another try? With your knowledge and experience I am interested in your opinion of these
microphones. I find nothing but praise. Hype or good marketing?
>
I tried to avoid being too microphone specific, since tastes and techniques often change over time.
The simple answer is that I've never heard any of the Rode mics, except for the NT-1 (and it was one of the very
early units). I thought it was excessively bright at the time.
I haven't heard any of the new models, so I can't really comment about them.
>
Posted by Buck62 on 02-13-2003 06:59:
Hey Harvey....
Any chance you might get your hands on an NTK for a trial run and review?
There's a whole lot of us NTK owners who would be interested in your "highly regarded" opinions on it.
>
I got a nice letter from the US Rode distributor about a year ago, saying that he hoped I wouldn't judge the whole
line of Rode mics based on my one experience with an early Rode NT-1 and he encouraged me to try some of the
other models.
I replied that I hadn't heard any of the newer models and I would be glad to give them a listen. I also told him that
I'm very careful to point out in any posting I make about Rodes mics, that I've only listened to one early Rode NT-
1 and that I haven't heard any of the newer models.
I think I offered to give them a listen if he wanted to ship me some test units, but that one post is the only
communication I've had with anyone from Rode. He never answered my return post, and that's where it was left at.
>
Posted by moley on 02-25-2003 12:56:
Sorry if you've covered this ground already and I missed it but...
Harvey, I have a Roland VS2480 DAW, and I recently bought an AKG C3000B (which I have since seen you
describe as sounding like "singing into a cardboard box" ) to use with it.
As you may or may not know, the VS2480 has a mic modelling algorithm - the idea being that you can use a lower
end mic to simulate a range of higher end mics. However, obviously, to do this, the algorithm has to know the
characteristics of the low end mic - so there is only a few specific mics you can use for this. And one of the lower
end mics that you can use to do the modelling is the AKG C3000B (this, combined with the fact that it was on sale
was partly why I went for this mic).
As well as using the C3000B to simulate higher end mics, you can also bring it to flat line (i.e. flat response curve).
I haven't had a chance to try out this properly yet (and even if I had, not having had any experience with the
particular mics it models, I wouldn't be able to say how accurate it is).
I just wondered what your opinion was on the concept of mic modelling. Do you think this is a good way to go
(given that I can't afford these expensive mocs)? Do you think that mic modelling as a concept is worthwhile, am I
just being silly? I don't know enough about mics to know how successfully modelling of this sort can be done - e.g.
whether it's possible to make a C3000B sound like it has a flat response curve etc.
Any ideas?
>
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I'm not sure that I have a good answer for your question. Mic modeling can only go just so far, even in theory, let
alone in practice. It can compensate for "some" of the problems in a microphone, but things like resonances (which
are time domain problems) cannot be completely eliminated by EQing the response (frequency domain fixes).
Differences in off axis response cannot be taken into account either.
But in many cases, it can help (or at least, it couldn't hurt) when you're dealing with less than ideal mics anyway as
your source.
If the mic you're using as a source is less than ideal, it's possible to use mic modeling to improve at least the on-
axis response and wind up with a usable end product. At least, in theory anyway.
Maybe I should explain that whole "resonance/time domain vs. eq/frequency domain" bit, since it has broader
implications than just microphone modeling.
Resonances are buildups of certain frequencies caused by sympathetic vibrations at those frequencies. Blowing
across a Coke bottle is the classic example. The air in the bottle combined with the bottle opening creates a
Helmholtz resonator which produces a specific pitch when excited.
Any thing which has a suspended mass (that's somewhat free to move) has a natural resonant point. The air column
in a wind instrument has a natural resonant frequency, but trumpets, trombones, flutes, etc. get around this by
adding devices that let the players adjust the length of the air column, thereby changing notes. Pipe organs simply
have a separate air column for each desired note - each pipe has its own note.
Even rooms resonate, and you'll often hear people complaining about their room causing problems at different
frequencies. But the important point to remember is that it takes some time to get something to resonate, and it
takes time for the resonance to die away. It's a buildup/die down problem - over time. Resonances don't start the
instant a specific frequency is played and it doesn't completely stop when the exciting force is removed. It builds
up and dies down slowly.
Just lowering the level of a specific note won't do anything to solve the time delay problem, it just reduces the level
at which the resonant peak is heard, but the time smear is still there, and very short bass notes will suffer, since
they are reduced as well, and may not have been long enough duration to even excite the resonance, so you're
screwing with some notes that didn't need any help, but the eq doesn't know that.
That's why it's not a good idea to try to tune a control room by using an equalizer. You fix problems in the
frequency domain, but it does nothing to really stop time domain problems, and it only "helps" at one specific
point in the room. Move a foot in any direction and you may have made the problems far worse.
It's the same problem with microphones - they have time domain resonance problems that simple eq can't fix.
But if using the mic modelers makes the sound "better" to your ears, go for it. It may not be the perfect solution,
but if it works, it's a good solution.
>
Posted by moley on 02-27-2003 17:34:
Thanks Harvey!
I understand what you're saying about time domain vs. frequency domain.
I don't know whether the mic modeling algorithm in the VS2480 is simply using EQ, or whether there's any more
sophisticated processing in there to adjust the characteristics of the microphone.
I'll mainly be using it for vocals, anyway, so it'll be on-axis, and relatively close up. I'll also be recording trumpet,
and possibly some sax, as well some percussion (individual drums etc. not a full kit) - and maybe some acoustic &
electric guitars. How the mic (with or without modeling) will fare recording guitars, remains to be seen.
>
Posted by Tomcat on 02-28-2003 14:58:
Hi Harvey, got a question for you,
If you take an SM57 and wrap a piece of tape (or anything) around the plastic piece that rotates blocking off the
time delay back access to the diagphram, will that turn the SM57 into an omni mic? If so, will it be any good as an
omni?
I got to wondering because I want to record an organ in a church and I have two SM57s and don't yet have enough
money saved up to get two SP B3s so I was wondering if it would be worth the time and effort to try using the
SM57s. The reason I want to use omnis is because there are antiphonal speakers at the front of the church which
make a great deal of difference in the sound than when they are turned off and using the SM57s as made, being
fairly directional, they wouldn't pick up much of the antiphonal speakers. Thanks a lot.
>
Yes, it will work, kinda. It might also create some other resonances that aren't pleasant. Can you give it a test run
with the covered SM57s a few days before your session?
What about using a pair of $35 Behringer ECM8000?

>
I'm the church organist and go up and practice at night, so I've got all the time I need. It isn't a critical project so it
can take as long as I want. I intend to eventually get a pair of SP B3s but I got to wondering about the SM57s as
something to maybe try just for fun and learning in the meantime but didn't want to waste my time if it was a
totally absurd thing to do. Thanks for your reply.
>
Is this an electronic organ or a pipe organ? SM-57s won't make you sound like E. Power Biggs, but they should
give you a very servicable recording, if they're placed right.
>
It's a current top of the line two manual Rodgers Trillium 807 electronic that sounds amazingly like a real pipe
organ but without the reeds going bad all the time and without the temperature pitch changes. There are two sets of
three speakers about 15 feet apart at the front railing of the choir loft and then two, one on each side, antiphonal
speakers at the front of the church about 70 to 80 feet away from the rear speakers.
I have thought about setting the mics up at the front of the church pointed to the rear in order to pick up both sets
of speakers but am certainly open to suggestions for best placement. Thanks again.
>
Since I'm basically lazy, my first question would be, "Does the organ have a stereo line out, and what does that
sound like?". You might wanna just record that and add some room ambience mics to fill it out.
If you want to go the all miked routine, I'd get another organist in there, and while he/she is playing, walk around
the room, looking for "sweet spots" to place your mics. For starters move in close to the main speakers and then
back away till you hear a nice balance between the direct sound and the reverberation in the room. That may be
anywhere from 1/3 of the room away to 2/3 of the room.
Too much reverb will produce a muddy, indistinct recording, while too little reverb will create a dry, clinical
sound. Look for a good balance between those two extremes. Go for a little less reverb than you think sounds right
(a little reverb goes a very long way).
>
The organ is fully midied and has a little black box called a PR300 which allows me to start recording, set the
pistons I want, then record whatever I play and store it on floppy. I can then play it back on looped and go
downstairs and listen to it anywhere in the church, over and over as long as I want.
I was really amazed when they finished the installation and I played back the demo disk I got and walked around
the entire church because there was no dead spot! It sounds very even all over the church from back to front and
left to right; there isn't even very much volume difference right under the rear speakers from what there is at the
front (if the antiphonals are on). The Rodgers folks did a very, very good installation. The guy who did most of the
installation was George Kirkwood, who worked for Rodgers for over 30 years and retired as head designer and
chief of the design department. He had been the organ caretaker for Virgil Fox and went on tour with him to make
sure the organ was always in top notch condition in addition to having designed "black beauty", the organ Fox
played. George is a year older than I and was "encouraged" to retire at 65, ie, he was given a bonus and sent out the
door!
George told me that those particular speakers have an ellipsoid sound pattern aligned with their long axis so four of
the back speakers are vertically oriented as two pair one on top of the other and the other two come off the middle
of the pair and are horizontal, ie there are two vertical and one horizontal on each side of the choir loft. That way
they cover the church space more evenly.
The organ does have a line out, but I don't particularly care for the sound as it's pretty sterile. The ring time in the
church is about 2 seconds so it has a good ambience and doesn't get very muddy.
>
Is George still available to possibly advise you on mic placements? I'm sure he would have some excellent
suggestions, since he's far more knowledgable than I am in this area.
My only other suggestion is to put on the demo disk and walk around the church, till you find the best spot to put
up your mics.
It sounds like a really fun project.

>
Posted by audiofreak on 03-03-2003 13:03:
Omni vs. True Pressure Transducer
Harvey, in your posts in this big thread you have mentioned a number of times the term "true omni" -- meaning a
true pressure transducer in the context of your posts. It seems that in practice the two terms omni and pressure
transducer are in fact not interchangeable -- hence the qualifier "true". I'd appreciate your time in explaining it a
bit.
1. Does it follow that there are other mics which claim to be omni's but actually are NOT true pressure
transducers? Then what type of mic are they actually -- a special type of pressure gradient transducers modified to
achieve an omni pattern (just guessing)?
I'll call them "non-pt" omni in this post -- meaning non-pressure transducer -- for want of a better term.
2. What form do these "non-pt omni's" take, e.g. dual diaphragm switchable pattern mics? In fact, Schoeps' web
site implies that dual diaphragm omni's are not true pressure transducers -
..."Unlike dual-diaphragm capsules, in the omnidirectional setting it is a true pressure transducer with flat response
down to the lowest frequencies."...
http://www.schoeps.de/E/mk-ccm5.html
3. Suppose I am shopping for a true pressure transducer, and I come across a mic that claims to be an omni. How
can I tell whether it is a "true" pressure transducer or not, short of asking the manufacturer? Can I tell it just by the
look of it or by the functions/parts that the mic possesses/does not possess?
4. What exactly are the differences in characteristics between a "true" omni and a "non-pt" omni? In other words,
what are the characteristics possessed by a true omni but not a "non-pt" omni, e.g.
- Directivity at high frequency?
- Relatively high self noise?
- Higher accuracy?
- Flatter on-axis response?
- No proximity effect?
- Ability to be used in the far-field?
Many thanks.
>
You've answered most of the question already. Pure pressure transducers are not "multi-pattern, dual diaphragm
mics". A dual diaphragm multi-pattern mic can mimic a true omni's non-directional characteristics, but not at all
frequencies, and without the omni's inherent flat response.
True omnis have some directivity at high frequency, but unless they are designed for diffuse field measurements,
they'll be dead flat on axis.
In the case of omni capsules, the relatively high self-noise is usually due to the smaller size of the diaphragm.
Omnis are known for their higher accuracy and flatter on-axis response, without proximity effect.
Because they respond to pressure only, omnis have the ability to be used in the far-field, without loss in frequency
response.
>
Posted by ambi on 03-18-2003 04:17:
So the Marshal MXL 603s are cardiod SD condensors. Correct? And what would be an omni Condensor? Omni's
are SD condensors correct? What would the Behringer ECM8000's be?
Also how would you do a stereo recording with Two large diaphram condensors? Like of the ice cream cone style
LD's. XY seems like it would be difficult because of the shape and size of the mics.
>
Sorry if I didn't make that very clear. Yes, the MXL 603 is a small diaphragm cardioid mic and the Behringer
ECM8000 is a small diaphragm omni mic. But omnis don't hafta be small diaphragm mics in order to work. The
DPA 4041 is a large diaphragm omni mic.
With large diaphragm, side address mics, the typical x/y set up would be one mic positioned upside down above an
identical mic, with their tops as close as possible.
>
Posted by ambi on 03-18-2003 08:28:
THATS WHAT I THOUGHT!!
Wow that's cool, I just put together the most logical way to do it in my mind and I came up with that. Nice to know
I got one thing right!
>
Posted by Kelsey on 03-18-2003 18:32:
I am trying to build a good mic cabinet for home recording that will grow with me. I am considering the FMR RNP
and RNC to use as the front end for laptop-based recording. The music is primarily blues-based material, and the
primary mic requirements are for vocals (mostly male, but some female as well), electric guitar, and acoustic
guitar, with occasional fiddle, cello, and percussion.
Right now I am thinking about buying a Shure SM57 and a Sennheiser MD421-II, but I'm torn between several
options beyond those two mics. I am considering a Shure SM7B, a BLUE Baby Bottle, and an AT4047/4040/4033
mic to cover vocals, but I'm not sure where that leaves me for acoustic instruments. My recording environment has
no sound treatment now, so I am concerned about how well a LD condenser will work for me at this stage. Would
the SM7B be a better choice for vocals, and does it work well with the RNP/RNC? What mic would be best for
acoustic instruments under these circumstances? Is an SD condenser as sensitive to ambient noise as an LD
condenser?
>
Posted by ScienceOne on 03-18-2003 20:01:
I don't understand the description of a LD x/y pattern. Could you describe it just a little better for me? Would the
capsules essentially be parallel to each other, just in different horizontal planes, or would they be vertical, one atop
the other, in a type of "v" shape? Maybe I'm just way off?
>
Imagine two identical LD condenser mics, like a pair of V67s, or a pair of SP B1s or C1s, since everybody should
be familiar with those mics. You're going to record a choir, using those two mics arranged in an x/y setup, with
each mic pointing about 45° away from a centerline thru the choir. Each mic is positioned on the centerline thusly:
The right side mic is on a straight stand in the center about 10 feet out from the choir. The XLR cord is hanging
down from the bottom normally, and the mic is rotated so that it's aimed at the right edge of the choir.
The second mic is mounted upside down on a boom mic stand, so that the XLR connected is aimed at the ceiling.
This 2nd mic is placed ABOVE the first mic, so that the tops of each mic are nearly touching. The 2nd mic is then
rotated in its shock mount so that its capsule is pointed at the left side of the choir.
Is that any clearer?

>
Sort of like a "half Blumlein.”
>
Exactly, but with a pair of cardioids rather than figure 8s.
>
I don't claim to be an expert or even advanced at stereo work, as matter of fact one choir is running me up the wall
with a reflective space (but that's another post). I just thought I'd give you a link though for some simple pictures of
stereo mic setups. First try this one http://www.dpamicrophones.com/eng_pub/. Then click on Microphone
University and select through different stereo pair setups. Also try Coincident and near Coincident for a search
terms that gets some sites otherwise missed.
Please don't get flamed if above link is under you but it's one of the more linked stereo pair locations on the net.
Good luck and if you’re in a place where you can and have the tracks available more mics and locations will not be
a waste. You may find that the extras are the ones you will use, I've done just that several times when I did not
foresee a problem with a setup.
>
Granted several of these links are deep but glance at them anyway many times reading over my head some good
info soaks in anyway.
Josephson Mike Technique
Hope this is helpful in some way.

>
Two excellent sources, jeepers. I mentioned David's site in the very first post I did in this thread, and I believe I
mentioned the DPA site in the section on stereo miking techniques. But it's good to repeat the links - both are
wonderful sources of great, accurate information.
>
Posted by hixmix on 04-19-2003 06:51:
"How does diaphragm size/polar pattern relate to mic applications?"
I've been thinking about this for a long, long time, but I still don't have a clue.
>
PAGE 25
s=03721567be1f1612322299dd5648ee0b&threadid=27030&perpage=25&pagenumber=25

OK, let's see if I can sum it up in just a few words:
"Use small diaphragm mics with wide polar patterns at a distance to record large sources. Use large diaphragm
mics with narrower polar patterns in closer to record smaller sources."
"Small diaphragm mics have more accurate off-axis response. Large diaphragm mics have more interesting off-
axis response. Which is most important is up to you."
Ignore these rules if your final results sound better.

>
Posted by hixmix on 04-19-2003 07:57:
Thanks, Harvey!
>
Posted by Sen on 04-19-2003 08:02:

The right side mic is on a straight stand in the center about 10 feet out from the choir. The XLR cord is
hanging down from the bottom normally, and the mic is rotated so that it's aimed at the right edge of the
choir.
The second mic is mounted upside down on a boom mic stand, so that the XLR connected is aimed at the
ceiling. This 2nd mic is placed ABOVE the first mic, so that the tops of each mic are nearly touching. The
2nd mic is then rotated in it's shock mount so that it's capsule is pointed at the left side of the choir.
Harvey, do you ever position your overheads like this? Why?

>
No, I don't use coincident xy setups for drum overheads, simply because I have far greater control over what the
mics pick up when I use wide spaced omnis or cardioids and careful positioning.
Plus, it requires a very even-handed drummer, which we don't get in very often. I would consider it for recording
jazz drummers, if they were very good, and I wanted a more minimal setup for a particular sound.
>
Posted by Sen on 04-19-2003 08:38:
Cool, thanks Harvey... So wide spaced is the way to go....
Thanks sir
>
Posted by jmorris on 04-20-2003 04:41:
Harvey, I'm lazy I guess. I dig out the mic I think will work the best, stick it on what I'm trying to record.........and
just listen. Move the mic, turn it, back it up a bit, EQ it. I’m not very tech… But I like what you are doing. Maybe
a little simpler for us guys that are the hit and miss type!! Thanks! Jim
>
"Hit and miss" is a wonderful luxury, but it takes time, and some people just don't have that kind of time, or they're
on the clock. If you're already to the point where you can "dig out the mic I think will work the best", then maybe
you're a little more advanced than the people this thread is aiming to help.
For many people, learning by trial and error is a wonderful way to understand the do's and don'ts of mic choice and
placements, but understanding the principles behind what the mic was designed for and how the mic actually picks
up the sound can often save valuable setup time.
It also helps answer some questions, like "why does my cardioid vocal mic feedback when I cup my hands around
it"?
>
Posted by carlosguardia on 04-24-2003 02:07:
I have a Question that I've looked to be answered here but don't see an answer for yet. I'm going to record a violin
player and would love to know what you think would be the best way to pick up a single violin. The violin is one
instrument that still has to be discussed in this thread. Thank you in advance,
>
It was covered pretty well in these two threads:
http://homerecording.com/bbs//showthread.php?s=&threadid=26563
http://homerecording.com/bbs//showthread.php?s=&threadid=42024
>
The END???
Posted by Harvey “The Man” Gerst on 12-08-2001 18:13:
“I'm very flattered you found some things of interest in the thread. The big problem has always
been that the information is out there, but nobody took the time to show how it all relates to each
other. The biggest thanks should go to Chris F. who took the time to formulate a question that
went to the heart of the matter; namely, how does all this stuff relate to each other?
Some of my posts are sketchy at best, or incomplete, but they at least give people a starting point
to learn more about one particular aspect of something. Overall, I'm very proud of this thread. It
was something that's been needed for a long time, and I'm glad I was able to contribute in a
small way to paying back an industry that has sustained me for most of my life, and rewarded me
with so many good friends and associates.”
Awards:
Awarded by MISTERQCUE on 01-10-2003 18:23:
 This thread awarded the coveted,
"MOST PROLIFIC M.V.P THREAD EVER POSTED" 2002-2003!!
100 out of 100 votes for HALL OF MICROPHONE FAME entrance!

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Harvey Gerst’s Mic Lessons

Am I close, or did I get off track by assuming too much?

[QUOTE] Originally posted by Chris F

[QUOTE] Originally posted by Chris F

Wow, difficult to answer, but I'll try.

Pressure transducer = Omni-directional

Pressure (omni) - all calibration mics, Earthworks.

[QUOTE]Originally posted by Harvey Gerst

[QUOTE]Originally posted by Harvey Gerst

[QUOTE]Originally posted by Harvey Gerst

The basic Polar patterns are:

Large diaphragm condenser mics have these advantages:

They have these disadvantages:

Musical instrument radiation patterns

Chladni Guitar Top Radiation Patterns:

Near-field placement vs. far-field placement

[QUOTE] Originally posted by Harvey Gerst

A couple of questions related to (a) above:

[QUOTE] Originally posted by Harvey Gerst

Gotcha. This is making more and more sense. Thanks.

[QUOTE] Originally posted by Harvey Gerst

Posted by Chris Shaeffer on 05-14-2001 15:56

[QUOTE] Originally posted by Chris F

Originally posted by Harvey Gerst

[QOUTE] Originally posted by Chris F

[QOUTE] Originally posted by Chris F

[QUOTE] Originally posted by Harvey Gerst

A couple of questions related to (a) above:

Gotcha. This is making more and more sense. Thanks.

Maybe time for a little experiment....

Any mic can be used to record any sound

How does this part fit into the song?

Notice there are 3 important elements to this box:

Don't think of it as "less accurate"; think of it as "more flattering".

It's 50% art, 50% science, and 50% luck.

Micing an acoustic guitar

Solo Guitar or Background For Vocal?

Acoustic Guitar as a Rock Track With a Band

The Singer/Songwriter Syndrome

Some Points To Ponder

Some Mics To Try First:

Dynamics: Shure SM57, Sennheiser 421, and Beyer M201

Ribbons: Beyers, RCA, any ribbon mic.

End of the first part - more to come!

Posted by Harvey Gerst on 05-17-2001 22:25

[QUOTE] Originally posted by Harvey Gerst

That's my first plan as soon as it arrives.

That is also what I was thinking...

All checks should be made payable to:

X/Y miking setup

More on all this stuff later…

Posted by Harvey Gerst on 05-22-2001 09:10

Posted by Harvey Gerst On 05-22-2001 14:55

The 3 coincidence techniques are:

The Jecklin Disc (which is actually what I was describing),

Thanks for catching that and keeping me honest.

1. Large condenser mics

There are two types of patterns usually used for vocals:

1. Cardioid (most typical)

[QUOTE] Originally posted by JerryD