About This Page
All material in The Microphone Primer section is taken
from an enormous thread, started in April of 2001, at HomeRecording.com
in the section "Home Recording dot com BBS > Equipment Forums > Microphones".
It is a reply to the question: "How does diaphragm
size/polar pattern relate to mic applications?" The entire thread is located at:
https://homerecording.com/bbs/showthread.php?s=&threadid=27030 — view it for additions made after this compilation.
The thread started with Harvey Gerst deciding to spread the knowledge he had picked up during his 50-some years in the recording industry. Then the questions started coming in to him, and over the course of a couple of years the conversation turned into over 50 pages of text on the message board. I found working through the individual pages very slow, and that was the motivation to condense the information. At the time, Sept. of 2001, it was only 11 pages on the BBS.
The reason that the entire thread is just one big page is because that makes it searchable. Most of the discussions about "how great this thread is" and the "thanks" messages were removed, so what is left here is just the "meat" of the thread. Generally, redundancy wasn't removed and whole paragraphs were copied instead of just a significant line from a post; most of this is just a straight cut and paste of what was written. Some headings were added, and some posts were reorganized to keep topics together. During Q & A sections Harvey's, input usually either starts with an "A." for answer, or it looks like this (depending on which was easier for me to do at the time).
The primer was last updated with posts from 2/27/04 (page 27 of the thread).
Approximate Primer Outline
Sometimes the subjects overlap, making it difficult to decide when to start a new section. That's why this is just approximate.
- Getting Started
- Some history
- How a "dynamic" mic really works
- Some Mic Physics
- Mic Types
- More background
- Musical instrument radiation patterns
- Near-field placement vs. far-field placement
- Miking an acoustic guitar
- Miking Vocals
- Miking Other Instruments
- Horns, what mics and polar patterns to use
- Flutes, clarinets, and misc. woodwinds
- Concert and Celtic harps
- Misc. Percussion
- Digeredoos, harmonicas, and other odd shit
- Ok, let's mic an electric guitar
- Miking the guitar cabinet.
- Stereo Mic Techniques
- Drums, Here We Go.
- Pianos and Mics - No Simple Solutions
- Wrap up
- Microphone specs ("I don't think we're in Kansas anymore, Toto")
- Microphone Frequency Response - The Window
- Sensitivity - What's that all about?
- Maximum SPL - How Loud Can You Go?
- Polar Response - Turn, Turn, Turn
- Realistic Maximum Sound Pressure Levels for Dynamic Microphones
- Lots of Q & A
- Tube Mics and Preamps
- More Q & A
- How to make your own multi-pattern microphone
- Even More Q & A
- The List
- Final (?) Comments
- Lot of info at: www.dpamicrophones.com/
- Also see Ed's (Sonusman) thread over in "Mixing and Mastering" at https://homerecording.com/.
- Also see recordist.com/rap-faq/current section 10.
- There is a Word document of this thread made by Tekker at http://www.freewebs.com/tekker/harveygerstmicthreads.htm. I haven't checked it out.
- It's also i Word and pdf at http://www.kset.org/music/pub/
- Brad on Pro Sound Web gave Harvey a suggestion about a new mic pre thread. Harvey has taken the challenge and is inviting some real preamp heavyweights to take part. This new one looks to have a real big future and promise to turn into another one equal to the Big Mic Thread. I'm going to try to link here to the thread: Mic Pre Thread
The Microphone Primer
Intro - How It All Started
I won't bore anybody again with another "best mic for under $200" thread, but I'm curious to find out how you go about deciding which mic to use for which particular application... I understand that large diaphragm mics are often better for vocals, but what about other acoustic instruments? Most of the stuff I've read so far has pointed to small d. condensers for instrument applications, and yet most acoustic bassists I know seem to prefer LD mics in the studio. Is there a reason you would prefer a SD condenser over a LD for piano or double bass?
Also, I've done enough reading to understand (I think) what the different polar patterns are, but how do you make that choice when it comes time to record? Is there a FAQ on this subject that I could read up on, or some links to some other sources to study? Any/all opinions welcome.
Whoa, that's a lot to ask for. Are you sure you really wanna know? That could get pretty lengthy - almost book size. There's some information at http://www.phys.tue.nl/people/etimm...cordingFAQ.html in the FAQ, some good stuff on David Josephson's web site, as well as at Shure and other mic manufacturer's sites, but it would take a long, detailed post here to discuss everything you asked about. Be careful what you wish for....
Am I sure I wanna know? Umm, yes. Can I digest it all at once? Probably not...but I make about half of my living as a teacher (college music classes), so I'm okay with the idea of slow absorption mixed with a good deal of trial and error. I'll definitely check into the sources you mentioned (I'll get started this evening), and supplement with whatever I can find here and at any other audio sites I can dig up. If you know of any good books on the subject, I'd be happy to order those as well - I give my final exams this week, and then there's gonna be some free time to be had!
Am I sure you want to go into all of that detail to reply to a post on an internet forum from someone you've never heard of? Well, no....and I wouldn't blame you if you didn't want to deal with it. But I'd be glad to listen to anything you might have to say, since I've read most of your posts while researching microphones and you seems to make a good deal of sense - the kind that even a recording newbie like me can understand - almost every time you post. The idea of being able to pick the brain of a professional recording engineer on this subject is too good to pass up, so if you have the notion, do your worst!
I can promise you that it won't fall on deaf ears, and that I have no inclination to start flaming anyone who says something that I don't agree with/didn't want to hear. I'm here to learn, and anything on the topic you might have to offer would be more than I know right now, and as such would also be greatly appreciated.
Ok, you're on. At 64, maybe the best thing I can do with my life is to pass on what I've learned from great people that taught me when I was starting out. I think that's why Al Schmitt, George Massenburg, Ed Cherney, and some of the other really big guns spend so much time on the net. We all owe the guys that came before us a lot, and this is our way of paying them back. And that's the only thing I have in common with all those guys I just mentioned - we all kinda drank from the same well back in the 50s and 60s.
I've got a band in the studio right now that's taking up most of my time till Wednesday, but after that I'm pretty free. I'll start off with some basic concepts till we're all up to speed, and then I'll try to fill in some holes.
I'll try to cover as much ground as I can, to give everybody a good basic understanding of the different mic designs, advantages and disadvantages of each design, how mic polar patterns are created, advantages and disadvantages of each polar pattern, and finally where each type might be used, along with advantages and disadvantages of each usage. How's that for a course outline?
This is the frequency response curve of a Neumann TLM-103. Not very flat, is it? Does that mean it's a bad mic? Before we can answer that we hafta know how to read one of these curves and how to interpret it.
Go to: http://www.neumann.com/infopool/download.php?Datei=docu0002.PDF and download this pdf file. Don't worry if you don't understand the math. Just get out of it what you can.
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) untill 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.
How a "dynamic" mic really works
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.
Now ya gotta do all of this stuff JUST to get the response usable - never mind about the mic pattern yet!
Q. ...is it normal to have a zillion microphones?
A. Actually, no, it's not normal, but since I'm a bottom-feeder, it's cheaper for me to have a lot of low cost mics around for different colors than it is to have a lot of expensive mic pres - for different colors. Most of the mics I bought very cheap, at pawn shops, garage sales, ebay, newspaper ads, etc. I'm still paying on a few of the more expensive ones. At retail, it looks impressive, but I've never paid more than $50 for a Shure SM-57, way under $100 each for the Sennheiser 421s, etc.
My only "bought brand new" mics are my Coles, the Neumann TLM-103, the Oktavas, and the Marshalls - everything else was purchased used, for very cheap. And I actually traded some old stuff I had for the Neumann.
Q. 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 Ui 87? 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.
A. 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.
Some Mic Physics
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:
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)
Pressure gradient = condenser patterns such as figure 8, cardioid, hypercardioid etc.....
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".
A. By George, you've almost got it!! Forget the condensor/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 condensor mic.
A condensor 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 condensor mic to work - it has nothing to do with the patterns.
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!
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
Condensor 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 and an omni pattern results in a cardioid pattern).
Brüel & Kjær has been making measurement mics for about 50 years now. They're omnidirectional, and flat (within a few tenths of a dB) from about 5Hz to 40kHz, although the 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, 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.
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)
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 (!" and larger) sizes. Small diaphragm condensor 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 condensor 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 condensor 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 condensor omnis have the least, then cardioids, then hypercardioids, then figure 8, which has the most proximity effect).
You would use small diaphragm condensor omnis where you want the greatest accuracy or in high level situations where self-noise isn't a factor. Large diaphragm condensor mics are better used for quiet sources, or where you want a particular type of complimentary coloration.
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:
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.
Discussion, bold is by Harvey:
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".
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.
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.
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.
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.
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.
Wow, what a great thread this is turning out to be. I love to see the little lightbulbs turning on inside people's heads.
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.
1. Panning (Left to Right) - Very useful for separating instruments that occupy the same
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.
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 mx 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.
Miking 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 want 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, while you position a large diaphragm mic at the singers 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 - 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 shitty - you never know.
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.
Q. ...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-feild mic focusing on the freq's you wish to enhance in the mix? Provided, of course, that the whole mess goes mono. (?)
A. 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.
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: http://www.itrstudio.com/110xy.gif
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.
You use the angle to adjust the level of the center signal and the amount of left/right separation. And for any angle greater than 90º. you usually place the capsules one above the other, at their centerlines.
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.
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!). Harveys over the shoulder technique was close second (but we had problems with noise from the player's skirt).
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.
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, wind blasts are a concern and a pop stopper, foam wind screen, 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.
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.
The largest selling record of all time (75 million albums, Michael Jackson's "Thriller") used a Shure SM-7 for the main vocals instead of the artist's usual large condenser mic. The SM-7 is a pretty standard vocal mic in Nashville, and it sits very well next to a Neumann U47 or a Telefunken 251 for some voices. The Sennheiser 421 was designed primarily as a vocal mic when it first came out, and it's still a great mic for many vocals.
Use distance to adjust your low frequency response, and use angle to adjust your high frequency response.
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.
Miking 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 saxaphones.
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 !" 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 6 foot plywood sheet on the floor to help bounce a little more top end into the mics.
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.
Digeredoos, harmonicas, and other odd shit
A dig is like a single organ pipe - with weird ass 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 shit, I usually reach for an omni or small cardioid to "capture the moment".
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: http://www.itrstudio.com/speaker.gif
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 back side 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've 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 cancelling 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: http://www.itrstudio.com/guit_mic.gif
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. Pull back 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, or your kickdrum 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 are 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. Pull back 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, or your kickdrum 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 1/2 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.
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.
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.
I gave it a quick test and it sounds pretty good, micing my deluxe reverb with a Beta 52. I didn't put the amp up on a chair, but I did lean in back against the couch so it pointed upward at an angle. I stuck the mic right on the grill cloth pointed at the center of the speaker. Usually I use a 57 over at the edge of the speaker. It came out real good, especially on rhythm chords and on the bass strings (no surprise), but I played some solo notes in the high range and it sounds good on that too, even though it does not quite have the bite that I get from the 57. I guess it just shows there is more than one way to skin a cat. Perhaps this would be a good way to record a rhythm track, and then go back and use a 57 to record a lead guitar track?
Bingo, ya got it!! Except you might wanna record the rhythm track, move to an unused track and record a second rhythm track, pan them hard left and right, then switch to the 57 and record the solo straight down the middle. That gets the rhythm tracks really beefy, but out of the way of the solo and vocals.
Stereo Mic Techniques
there are 3 basic stereo techniques, and several variations within each technique:
These include X/Y, Blumlien Pair, M-S, and Phased Arrays
These include Jecklin Disk, ORTF, Binaural, Crossed X/Y, and some others that I'm forgetting right now.
Omnis (or cardioids), spaced far apart, with center fills sometimes.
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.
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.
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:
Mic University Stereo Techniques
But we'll discuss all of the techniques here. We've already covered "coincident" stereo mic techniques (M-S, X/Y, etc.), all of which 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.
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 an device which approximates the dimensions 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.
A-B (Wide-Spaced Stereo 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, orchestras, 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.
Ribbon mics are a wonderfully simple, yet elegant, solution to a number of mic design problems. Using a single strand of superlight 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.
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, barbers quartets might work better with near-coincidence mic techniques, while all three techniques might work well for larger groups.
You hafta ask yourself some question 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)?
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.
Drums and Piano
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):
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 knowledgable 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 drum head (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 center line, 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.
Recording drums - Things to think about
I have some time so here goes the start of this section:
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 shitty 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 hihat 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 hihat, 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.
Q. 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!
A. 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 torpedos". 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.
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?
Q. 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?
A. It's primarily to avoid phase anomalies.
Q. 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)?
A. I try to make sure that each mic follows the 3:1 rule, mentioned earlier in this thread. (I've been bitten on the ass many times for this in the past.)
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:
Q. 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?
A. That's part of it, but mostly, it's to keep the drums from bleeding into all the other open mics.
Q. That vocal mic picks up more snare than the overheads do, a result of Brian (my drumming brother) harmonizing using falsetto, which isn't too loud. Any thoughts on a way to get more of him and less of the drums on his vocal track?
A. A couple things that might help, use a boom stand instead of a gooseneck so that you can point the mic directly at his mouth and point the back (or null point) of the mic at the snare, use a mic with a tight cardiod pattern.
Usually, we'll use a headset mic to record a scratch vocal, and then replace it with a final vocal after everything else is done.
Q. 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 omis.
A. 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.
Pianos and Mics - No Simple Solutions
...I wrote part of an article for miking upright pianos for Recording magazine. ... It was for Electronic Musician, and 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 sightly 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 miked 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 condensor mics, but don't be afraid to try large condensors, 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.
Q. BTW any thoughts on using a couple of PZMs taped to the inside of the lid of a grand piano?
A. PZMs taped to the 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.
My question(s) is,I record in a small room and am concerned about having the two mics for the guitar within the proximity of the body,will I get to much boominess(is that a word?)
A pair of omni mics will reduce the boominess, or a figure 8 pattern mic turned sideways will work, with the mic's null pointed at the singer.
I was looking at either a pair of 012s or a single SM81, because it has a 3 position bass roll off,which I thought might help with this.
The MC012s with omni capsules would work well for the guitar if you move them in pretty close to capture just the guitar. but placement will be very critical.
How much difference do you think would be between stereo and mono for this type of recording?Most of it is going to be mixed with other instruments later,bass drums etc.
In that case, unless the stereo guitar is the most important element of the song, forget about recording it in stereo. Go for maximum separation.
Do you think an 81 is the best way to go for a single mic?
The 81 is a pretty damn good mic.
If I go with the 81 (or another mic you can recommend ) do you think Shure or other manufactures are consistent enough where I can purchase another one later and use them for a stereo pair?
The SM-81 is VERY consistent from unit to unit.
And if you or anyone else knows,is Octave the only one who sends their seconds to GC or could I run into problems with other mics bought there?
They don't get Oktava's "seconds", but they do get the uneven QC that Oktava is known for, and that makes getting an Oktava there a "crap shoot".
How is the earthworks stuff?
It's VERY good.
I'm looking at the SR71.
That's a nice hypercardioid mic.
How do they compare to the SM81?
Very well, but the -10dB pad and the 2 position roll-off filter on the SM-81 might give you a little more flexibility for recording guitar.
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.
I record in an unacousticly enhanced room (if you know what I mean)
You can hang some packing blankets on mic stands to create a smaller, deader space to record in.
What brand of figure 8 would you recommend?
I guess it would have to be a multi pattern?
The Studio Projects C3 and the Nady SCM1000 are both pretty good, low cost, multi-pattern mics.
Q. I blow alot of blues harp. I use Special 20s. I use a Green Bullet 520DX. When I play on stage, I run the GB through a small 30watt Marshall amp and mic it into the PA sys.
A. ...sounds like a typical blues harp rig. I'd just put a mic in front of your amp and take that signal into the board. Possibly a short plate reverb and that's it. Hard to improve on your basic setup.
Microphone specs ("I don't think we're in Kansas anymore, Toto")
I know a lot of you read the specs of mics very carefully to help you make decisions before buying, but that could cause you some serious problems. Here's why:
Let's just look at two manufacturers that are highly acclaimed here; Marshall and Studio Projects. Keep in mind that I think very highly of both these lines, but their web pages show some serious inconsistencies. For example:
If you go to www.mxlmics.com and you want to find out about the 1000 series, you click on it and up comes a nice, very impressive specs page, complete with a frequency response curve that you can click to enlarge. When you move your pointer on top of the frequency response curve, it says it's about to show you "1000b.gif" (the big version of the picture). Now go to the 600 specs page (the 600, not the 603) and look at the frequency response curve. Hmm, looks very similar doesn't it? Let's slide our mouse pointer over it and see where that would take us. When you move your pointer on top of the frequency response curve, it says it's about to show you "1000b.gif" (the big version of the picture). Same damn curve. Is that accurate? Hell no, these two mics sound nothing alike. And that curve doesn't look anything like what either of the mics sound like.
Now let's wander over to the Sound projects page at www.PMIaudio.com. Let's click on the "Specifications" section at the top of the "Studio Projects" page and then let's click on the T3 specs. Curves look interesting, but not exactly what I was expecting, judging from the sound. Let's look at the polar patterns. The upper left polar pattern is the omni mode obviously, and it's pretty smooth. Let's really study that curve for a minute. Notice the little variations from perfectly round? Remember those.
Now go back and look at the omni polar pattern for the C3 mic. What's this. According to this these two mics are IDENTICAL in omni mode. Same little variations. So which should you believe? Well let's go look at the C1 for a minute since that's the mic most people are interested in initially. Hmmm, that doesn't look like a cardioid pattern polar response. Wait a minute, I know this curve. I've seen this before - just a minute ago. It's the same omni curve that's being used for the other two microphones.
So are Alan and Brent liars? No, they're not. But the specs sheets are wrong. If you're basing your decision on the specs, you're not getting the right information. I'm not picking on Brent or Alan - most microphone companies don't provide the specs you need to make an honest evaluation of the microphone's performance - and that includes all the big guys too. Speaker manufacturers are just as guilty. I just used Alan and Brent's pages to show you some pretty obvious screwups - some of the other manufacturers' screwups are less obvious ("downright sneaky" would be a better term).
In my next post, I'll get into how they create frequency response curves, different methods of measurements (steady tones, warble tones, white noise, pink noise, 1/3 octave, impulse, etc.), how they control the final results (pen speed, chart speed, smoothed, averaged, etc.), and what the curve REALLY means.
Stick around - it's gonna be fun. You're gonna hear about what manufacturers really DON'T want you to know.
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 side walls. 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 throw 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 judgment 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" look 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.
But you can trust a Neumann frequency response chart to be within 4 dB of what they say it is... Hmm. So one thing you're paying for in a Neumann is a mic that is what it claims to be on paper?
You can trust it to be within 2dB of the published curve - in each direction. It should never be more than 2 dB louder or softer than the published curve, once you match the levels at 1,000 Hz.
You know, I found an EQ preset in Cool Edit Pro that sounds good on my voice tracks - they call it 'Mackie High Band', but it matches the frequency response chart for a TLM 103. I wonder how many of Cool Edit's presets (or those of other editors) are nothing more than the published frequency response of famous mics. I think I'm going to do some tweaking with my Cool Edit EQ. LOL
Bingo, that's what some of the simpler mic modelers do.
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).
BTW, the first 5 mic graphs shown in the above example were all hand-created by me; no microphones were actually harmed or used during the making of these curves. The very bottom curve (showing off-axis responses) is a real curve of a real DPA mic.
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 200 hz for example, one mic reaches 10 dB and another mic reaches 18 dB.
Now suppose that apart from the 200 hz 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 200 hz 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.
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.
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.
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.
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.
Polar Response - Turn, Turn, Turn
I'm gonna use the sheet for the Shure SM-57 as the example, so download this and follow along:
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.
Just An "Old Fart" Recording Engineer
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?
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.
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.
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.
...where is the maximum SPL listed for the SM57? Do most manufacturers advertise this specification? If not, is there a way to figure it out?
You're right, Shure doesn't show max SPLs for the SM-57, 58, the Beta versions, or the SM-7. I'd guess at around 138 to 145dB max SPL for a good quality dynamic, but that's just a guess.
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? (text quoted below)
as asked in
Maximum SPL for dynamic mics (text quoted below)
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.
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.
Responses from Shure (referenced above) in the tables below:
|Can a dynamic microphone handle really loud sounds?|
|What is the maximum sound pressure level that a dynamic microphone can handle without distortion?|
Realistic Maximum Sound Pressure Levels for Dynamic Microphones
Microphone users often ask "What is the maximum sound pressure level that a dynamic microphone can handle without distortion?" Using the Shure SM58 as an example of a typical dynamic microphone, Shure Engineering performed experiments to answer this question. Like most technical matters, the answer is not simple.
As a point of reference, 140 dB SPL is the accepted threshold of pain for the human ear. The maximum sound pressure level (max SPL) from a human voice as measured by Shure is 135 dB SPL at 1 inch from the mouth. A kick drum played very loudly may exceed 140 dB SPL, but has never been measured by Shure above 150 dB SPL. The loudest orchestral instrument, a trumpet, can theoretically produce a MAX SPL of 155 dB SPL at 1 inch, but only in its upper register. Note that the distribution of energy (sound pressure) in speech, music, and noise is dependent on the frequency. For example, the human voice does not produce much energy below 100 Hz and its frequency of MAX SPL would be higher than 100 Hz. Exactly how much higher depends on the individual voice.
Unlike a condenser microphone which has internal electronics that may overload, a dynamic microphone distorts when its diaphragm hits a physical barrier, like the magnetic pole piece, and can move no further. The excursion of the diaphragm is frequency dependent and the excursion is greatest at the resonant frequency of the diaphragm. Therefore, the MAX SPL of a dynamic microphone like the SM58 is frequency dependent. This means that low frequencies will produce distortion at a lower SPL than higher frequencies.
For the SM58, the frequency range to first exhibit distortion is centered around 100 Hz, close to the resonant frequency of the microphone's diaphragm. At 100 Hz, the measured MAX SPL is 150 dB SPL and the electrical output of the microphone is 0 dB V or 1.0 volts. Note this is a line level signal, not a mic level signal.
In the 1 kHz range, the SM58 measured MAX SPL is about 160 dB SPL due to the change in microphone sensitivity at the higher frequencies. The electrical output of the microphone at 160 dB SPL is +10 dBV or 3.2 volts.
In the 10 kHz range, 180 dB SPL is the MAX SPL of the SM58. However, this is a calculated measurement as Shure Engineering had no means to create such enormous and dangerous SPL. For comparison, NASA reports that a space shuttle launch measures 180 dB SPL and higher at 10 meters.
In the 20 kHz range, the MAX SPL is calculated to be around 190, due to the response falloff of the SM58. But now the point of absurdity has been reached because at 194 dB SPL the sound pressure varies from twice normal atmospheric pressure (at the wave peak) to a total vacuum (at the wave trough). Plus the sound source must be moving at the speed of sound just to generate a wave of this intensity.
In summary, a well-designed dynamic microphone of professional quality will never reach its distortion point in "normal" conditions. If one does encounter distortion when using a professional dynamic microphone for an extremely loud source, it is most likely that the electrical output of the microphone is clipping the input of the microphone preamplifier. [Remember that at 150 dB SPL, the SM58 will provide a line level output!] To solve this problem, an in-line attenuator ("pad") must be placed before the preamplifier input, or the microphone must be moved farther from the sound source. In general, the sound pressure level will decrease 6 dB for each doubling of the distance.
|Maximum SPL for dynamic mics|
|I would like to know the maximum input sound pressure levels for the following mics I own. SM57, SM58, BG 3.0 and BG 3.1. I am having trouble recording an acceptable for crash cymbals closely miked. This info could help me select the correct mike's.|
In general, condenser mics are used for cymbals due to the better transients response of the condensor element. Consider the SM81; the SM94; the BG4.1
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?
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.
I'm still trying to figure the best way to record guitar and vocals at the same time.
I was thinking,watching groups performing on T.V. they always use a dynamic or black electret for vocals and a small condenser for acoustic. The sound they get is equal to or better than the studio sound.
Do you think this is a possible way to go in a home studio? It seems like the plus for a dynamic or black 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?
It's a little tricky to explain without the design theory, but lemme see if this'll help a little bit:
Cardioids have a heartshaped 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 condensor 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.
Lots of Q & A
There is a lot about recording instruments and solo singers, but what about recording a 4 part male a capella 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.
Harvey and you other guys - I just started a new thread in the Recording Techniques forum, that I'd be really grateful if you'd take a look at. It's about micing a bass amp. Here it is:
I was wondering if anyone 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.
Have you tried a search at RAP?
It is a very large pro audio group.
Every engineer who has used the 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.
...but Duran Duran are one of my biggest influences. Would you happen to know what vocal mic Simon LeBon was using during the "Rio" era?
comment: Colin Thurston was the producer for the first 2 Duran Duran records, along with virtually every other huge english act around the late 70's early 80's.
Based on that pre-digital effects era, you are going to be dealing with high end analog equipment and engineers who knew how to use a room/studio to get a good sound and probably alot of those "bowie" tricks...
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):
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 dynamic)
2 D112 (cardiod dynamic "the egg")
1 D19C (cardiod dynamic)
8 D190E (cardiod dynamic)
2 D202ES (cardiod dynamic)
1 D224ES (cardiod dynamic)
2 D25 (cardiod dynamic)
2 M160 (hyper cardiod double ribbon)
2 M201 (hyper cardiod dynamic)
2 M260N (hyper cardiod ribbon)
4 Blue Bottle (multi-capsule tube mic system)
4 B4 Capsule (perspex sphere pressure omni)
11 4038 (classic BBC ribbon)
2 4003 (omni condenser)
2 4007 (omni condenser)
2 4011 (cardiod condenser)
2 PZM (pressure zone effect)
1 RE15 (cardiod dynamic)
3 RE20 (cardiod dynamic)
1 PL80 (cardiod dynamic)
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 (miniature omni fet 80 series)
9 KM84 (miniature cardiod fet 80 series)
4 KM86 (miniature selectable fet 80 series)
1 KM184 (miniature 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 diaphragm cardiod tube)
5 TLM170 (large diaphragm fet 100 series)
1 DC21 ()
1 RV ("the bullet")
2 CU41 (cardiod condenser)
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)
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)
10 SM57 (classic instrument cardiod dynamic)
1 SM58 (classic vocal cardiod dynamic)
1 SM98 (miniature super cardiod condenser)
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.
1) Really brite, cheap plate reverb.
2) Lots and lots of chorus.
3) A little more chorus.
Done. No need to ditch that 57 . . . no need to take up a summer internship with Sir George, feather-dusting his massive mic collection.
...suppose you have a pair of subcardioids like the 603:s. 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 wonder 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.
... Do smaller drums record better than larger dia. ones. Part of this is due to alway 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.
I have a small studio doing the light contemporary Folk sort of stuff. I would like to purchase a kit for the studio and have been having trouble pulling the trigger partially due to this one question. Drum kits are a hard one to "try before you buy" kind of purchase. At least for recording.
I figure having a studio kit would make getting reproducible results easier.
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!!
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!
I'm in the business of getting a stereo pair of mics for drum overheads etc. The Oktava M012 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.
I'm also considering the ADK A51sc - small diaphragm condensator. Apparently it's voiced like a KM84. Tried them? Heard of anyone who has? Any good?
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 pict 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.
I haven't heard the ADK mic.
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.
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 coincidenct spacings to see if that gives you a
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.
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 flat out bitch to record. It's also a lot easier to record a strummed J-45 than it is to record a fingerpicked 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
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 the 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
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.
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 8 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.
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 condensors with selectable patterns that can produce a decent figure 8?
Also, I understand your favourite 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? 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)?
Gimme another day to get over this damn flu and think about it for a little bit more. Yes, you can use almost any figure 8 mic as the "side" mic, but there are some other considerations. Lemme see if I can make this easier than Wes Dooley's explanation, (which I have a hard time following). I'll try to post a complete answer by Sunday.
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?
Q. 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?
A. 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.
Q. I'm still wondering - what are your reasons for preferring spaced omnis to M-S while recording drums?
A. 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.
Q. 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.
A. 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.
...OK, I can see where my response might have been confusing so let's see if I can clear things up a bit.
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 drumset...
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 coincendence" 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.
Does all of this make a little more sense to you now?
...some time later...
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.
Originally posted by h kuhn
this link might be interesting, complementing some of the information given by Harvey (especially the drawings are nice):
(Comment) All very nice, but the English (if it is indeed English) is reminiscent of the instructions from Roland, and I think the MS diagram is wrong... (but it's good to keep this one afloat... )
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
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 convertor, which also works as a gain stage, limiter, and as an even order distortion generator, when pushed hard. One lessor 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 48 volt 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.
Q. how would the drawing differ for a class A/B or class B circuit?
Class AB and B divide the signal up into two section: 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.
Q. Originally posted by Harvey-
"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?
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 absorption 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.
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.
Unfortunately,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.
More Q & A
I have three questions:
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, 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 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 matchup 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 ambiance.
Member replies in this section, not Harvey's.
Q. 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?
A. The most silent mic I came across until now, was the Rode NT1000, with a specified self-noise of 6dB SPL (A-weighted).
A. 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.
Q. ... I am about ready to record is a bass oriented (some call it "lead bass") rock album ala Stuart Hamm and John Paul Jones ... and the usual direct recording for bass is not going to cut it. It has become painfully obvious that I need to mic a cabinate 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.
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.
Q. ... What I don't see in here is how to mic a Leslie...
A. (from Member) 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.
A. (Harvey) For those of you not familiar with a Leslie, it uses a rotating horn for the 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.
Q. 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.
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 condensor 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.
A. 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?
A. 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 questions 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 an 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.
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 microphones, 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 should 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've gone from cardioid to a narrow cardioid, a hyper-cardioid, a super-cardioid, and finally to a figure 8 pattern.
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 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?
Q. Can you substitute using 2 cardios instead of purchasing a multipattern 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 multipattern 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.
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 multipatterm mic under $1,000 (actually, the suggested list price was $399).
The idea of using dual cardioids was from Neumann's original patents in 1936. I happened to be reading some old AES articles in 1987 and I thought to myself, "Hmmmm, this should also work with just a couple of these new small electret cardioid capsules." and the idea was born.
Even More Q & A
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.
(Member comment)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.
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 dia. being less bothered with room reflections than large. Is this so??
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.
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 rehearsal 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 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.
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.
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.
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.
... I just wondered what your opinion was on the concept of mic modeling. Do you think this is a good way to go (given that I can't afford these expensive mics)? Do you think that mic modeling as a concept is worthwhile, am I just being silly? I don't know enough about mics to know how successfully modeling 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.
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) can not
be completely eliminated by eqing the response (frequency domain fixes).
Differences in off axis response can not be taken into account
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.
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 diaphragm, 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.
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?
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 serviceable 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.
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 ambiance 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).
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."...
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?
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.
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 diaphragm 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.
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 center line thru the choir. Each mic is positioned on the center line 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 it's shock mount so that it's 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.
(Member comment)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 otherwised missed.
(Member comment)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
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.
original quote: "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.
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.
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 itr, 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!!
"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"?
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.
It was covered pretty well in these two threads:
I do have some unanswered questions regarding acoustic guitar recording which I
did not see addressed sufficiently to overcome these problems.
1) what are some approaches to limit the fret,string and other transient noises made by the player when recording an acoustic guitar. BY using your techniques I have achieved much better tone than I ever have,. the noise from string buzz/whine, and fret slap as well as right hand noises against the body or pick/string seem to remain at an objectionable level. What tricks do you have up your sleeve for this?
Distance is your friend as far as lowering all the extra sounds. The other trick is something you aren't gonna like: practice. You learn how to lift your fingers off the strings and where to place them to eliminate fret buzz and reduce string squeeks.
2) As you mentioned, Every acoustic guitar has certain resonant frequencies which seem to dominante the recorded sound and can even mask the beauty of the instrument or ruin the desired tone as well as drowning out what is being played.
I know you mentioned moving/experimenting with the mic positions to minimize this but could you elaborate on any other tricks or positioning strategies to help with this. The recorded sound which I am getting seems exaggerated many fold to what I hear from that same instrument live.
Again, distance is your friend and, yes, mic placement is critical.
You are right that practice would really help to overcome a lot of that noise . Unfortunately I'm not one of those gifted players who can achieve a constant benefit from practice...
I had a long talk a while back
about this very subject with Rick Ruskin, one of the finest finger pickers on
the planet, and he said basically the same thing. It takes practice to remove
the squeeks and buzzes - there's really no other way to get rid of
Tone is way more elusive. Sounds come from all over the face of the guitar, so it's really hard to find the "best position" as a formula. The "over the shoulder" technique gets the sound closer to what the guitar player normally hears, but it may not be the best placement to fit nicely in a particular track.
Moving the mics out a little bit gives you a more "accurate" picture of the "whole guitar", but sometimes, that's not desirable.
Bottom line: "It depends."
I'm pretty new to the
whole recording thing, and I've got to say, having read this entire thread last
night, I feel like I'm starting out on the right foot. I just won an auction for
an AT 3035, which I've seen some of you say nice things about. And i'm about to
pick up a pair of 603Ss to go along with the one 57 I already had.
After reading everything, though, I've got one concern that wasn't really mentioned in the thread: how helpful will this knowledge of mics and mic placement be without using preamps? This is another thing I've gotten mixed responses about in the past, and I'm not sure if the preamps in the crappy little behringer mixer I'm going to buy will be enough.
I'm really not able to spend money on standalone pres at the moment, so I'm just wondering if I'll be able to get any quality recordings without them.
Thanks for everything.
The Behringer mixer or your microphones won't be a problem in getting good quality recordings. The music, the talent, mic placements, the room, and "how" a person uses the equipment they have, are usually the biggest problems in getting a good recording.
In your honest opinion, how important is the room you record in in the overall scheme of things?
All I can ever give you is my honest opinion. The answer is: It depends. It
depends on what you're recording, and how you're recording it. For example, with
close miking (within a few inches of the sound source), the room usually doesn't
But, if you're recording an operatic singer, a flute, harp, cello, acoustic guitar, or a violin (where you need a little distance between the mic and the source to capture all the dynamics or the full instrument), then the room comes into play. As the mic to source distance increases, the room becomes VERY important.
If you're limited by the room size (i.e., a very small room), your only choices are to make the room as dead as possible, or to at least make one part of the room as dead as possible, with baffles, blankets, pillows, whatever. You can then add some electronic reverberation to simulate a better room.
I am curious about the effects of age on microphones. Do different types of mics decay in quality over time? If so, in what way and which mics are most vulnerable? Can you offer any suggestions for people who may be shopping around for used mics? What to look for and what to avoid?
Yes, all microphones decay in some way over time.
Electret condenser microphones lose their charge, although the newer electret materials tend to hold their charge longer than the older materials. Because of the lower costs to produce an electret mic, often the tensioning, manufacturing techniques, and long term stability suffer, resulting in everything from poor initial performance to very peaky response at higher frequencies.
Condensor mic diaphragms in general suffer from stretching, arcing, and dirt buildup, due to electrostatic attraction. Older condensor mics (which used PVC instead of mylar as the base material) suffer from hardening and cracking of the PVC base material.
Ribbons sag over time, stretch, and can rub when they do, decreasing the output greatly.
Dynamic microphones can come loose at the outer surround due to continuous flexing, or deform so that the coil rubs against the pole piece, causing buzzes and shorts.
Q1: It seems the possibilities are pretty much open-ended. But are there a couple of things you should never do with a microphone?
A1: Avoid subjecting any microphone to large blasts of air, and extremely humid conditions.
Q2: What are your thoughts and comments on Decca Tree type arrangements - or other similar minimal microphone arrangements. Say for example miking a drum-set with three microphones?
A2: The Decca Tree was a unique solution to the problem of stereo imaging, using the unusual directional properties of the Neumann M50 microphone. It basically consisted of 3 mics (all M50s), with the L/R mics about 6 feet apart, and a center mic about 4 feet in front of the l/r pair. It maintained great stereo imaging, since the omni M50s became almost hypercardioid at high frequencies, and the center M50 kept the stereo image from wandering too far.
You can achieve similar results today, using a "wide cardioid" set of 3 mics, and even 3 cardioids (and 3 figure 8s) have been used with Decca Trees, with varying good results.
The Decca Tree was primarily designed to record symphony orchestras; it wouldn't be a good choice for miking a set of drums. It needs distance to work properly. The normal placement was 3 to 4 feet behind the conductor and about 10 feet above the conductor's head.
Do you know of any mics that have crap transient response that I could use for close micing drums (for the purpose of recording to digital)?
You really want mics that have good transient response for close miking drums.
For kick, there's the ATM25, the D112, the 421, the Beta 52, the ATPro25, and a host of others that work well. The Sennheiser 504 and 604 work well for toms as does the Shure 57. The 421 works great for most floor toms. The 57 and Beyer M201 work well for snare.
For higher budgets, the U87 is often used for overheads and the U47fet for kick, with a 414 for snare. Other overheads range from the Coles 4038s and the new AEA R84s down to the Behringer ECM8000s.
It really depends on the type of music you're recording, the drummer and the kit, and the sound of the room.
Most engineers will make microphone choices based on two possible scenarios: Do they want absolute accuracy, or do they want to emphasis certain elements of the instrument being miked? The choice of mics determines which goal is achieved.
I had the idea of using mics with crap response suggested to me by a friend as a way of getting around the massive transient spike coming off the drum, so that the signal I would be getting would be mainly the tone rather than the crack of the drum - the idea was to us the overheads to capture the transients, which would theoretically "spread out" the energy from the transients.
Remember this is for digital recording, I'd definitely agree that recording to tape I would use mics that would capture the transient in detail, but this is just something different that I'd like to try.
A compressor (set to a fast attack and release) will eliminate the transient, if that's what you want.
A "blurry" mic would simply produce a lot of boom that won't reproduce the detail of the percussion. You can also roll off the high end of the mic, since transient response is a function of rise time and high end response.
Regarding the Decca Tree, or any 3-microphone setup, I was hoping to learn of your own personal experiences.
Harvey, how do you like to mic things?
That is, have you done some cool, weird (to us) microphone setups that have yielded something beautiful or unexpected? ...
That kind of thing. Got any stories? As in practical applications?
As far as the personal stories,
they aren't appropriate for this forum. I don't have any set mic techniques to
speak of. I try to listen first to anything I'm going to record, and then figure
out what mics and placements might be best to capture, or enhance the music.
I'm also concerned with staying out of the musicians' way, so they're not as aware of the recording process, letting them concentrate on establishing a groove and communicating with each other. Every engineer has his/her own style of doing things. What I do will depend on what the musicians need to feel comfortable.
I haven't worked with large sources in a long time. By large sources, I mean big bands, orchestras, choirs, and pipe organs, but I would still approach those tasks the same way. What do I need and (where do I need it) to capture the excitement of this music? Can I do it without disturbing the "flow" of the music? Does the music call for intimacy, or a sense of space?
My only rule is "Honor The Music". Kinda like the medical professions' rule, "First, do no harm".
On the studio wall, I do have a list of stuff to think about (and for the band to think about), but they're not rules exactly, just a few things to think about while recording.
Care to share the list?
Only the words in all caps appear on the list:
The song will tell you exactly what it wants - if you're open enough to listen.
WHERE'S THE "HOOK"?
That's the part of the song you mostly remember, either a catchy phrase, or melody, or both. It can even be an unusual instrument. Most hit songs have a "hook". All time great lyric hook? Probably Steppenwolf's "Born To Be Wild". All time great melody hook? Hendrix "Purple Haze".
GET TO THE "HOOK" IN UNDER ONE MINUTE
If you want to be a star, don't waste your time setting up a long, complicated intro to a song. Get to the heart of the song quickly. Consider the examples listed above. When you're doing your stage show, then you can do the long version. A record executive will give you about 12 seconds of his time. If you spend two minutes just getting into the song, you haven't got a chance.
IS IT "RADIO-FRIENDLY"?
Is it the kind of music you're likely to hear on the radio? If a radio station won't touch it, chances are a record exec won't either.
IS IT TOO LONG?
Leave the long version for the stage show. Tell your story in 3 to 3½ minutes. (This isn't a hard and fast rule, but if you're gonna take 6 minutes to say what you want to say, it better be important stuff that people want to hear.)
DOES IT MAKE SENSE?
Is your lyric really tight? Are you just throwing in lines to stretch the song? You've got 3 to 3½ minutes to tell your story - make every word count.
IS IT SOUP YET?
That's from an old Campbells soup commercial, about a kid standing near the kitchen window, wanting to know if the soup is ready. Some groups keep adding more and more layers, till the arrangement is so full, the original song gets lost. Learn to know when a song is finished, and stop there.
FEELING IS MORE IMPORTANT THAN TECHNIQUE
Nobody thinks Smashing Pumpkins are the best musicians on the planet and they'll never win a Grammy for "Instrumental of the Year". Unless you're Dream Theater, go for feeling.
CAN YOU DO IT BETTER, OR JUST DIFFERENT?
Guitar players, and some singers, are funny sometimes. If they improvise, they want to lay down 20 tracks and choose the best parts. That's OK if you have unlimited time and money, but most of the time, any good take will work fine.
A "MISTAKE" IS OBVIOUS TO EVERYBODY
During a session, somebody will sometimes hit a note they didn't mean to hit. Is it a mistake? Yes, no, maybe, or maybe not. If the song is in E minor and the guitar player hits an E major, it's probably a mistake, but if the bass player hits a B instead of an E, it may not be a "mistake" - it may make the song better.
If you have a small group (Bass, Guitar, Drums, and Vocals), you don't need 6 to 12 stereo guitar tracks. Two similar rhythm guitar tracks (for fattening) and a lead track are usually more than enough. Most engineers (myself included) are frustrated producers. When you have all those tracks available, the temptation is to "use 'em all." My advice? Don't.
DON'T LOSE THE FEEL
The basic "groove" of the song is important. If you cover up the groove by adding more and more stuff, you stand a serious chance of messing up the song. If the groove isn't there, all the extra things you add won't help.
PERFECT DOESN'T ALWAYS MEAN GOOD
Nirvana, Smashing Pumpkins, Green Days, and Tripping Daisy proved that you don't have to be an Eric Clapton to have a hit record. Do what you do best. If the lyric is the most important thing in your songs, you don't need a killer guitar solo (or any solo for that matter).
IF YOU OVERPOLISH, YOU MAY LOSE THE EDGES
Sometimes a group will work for weeks in the studio, eliminating every fret rattle, adjusting the volume of each note in a solo until it's perfectly balanced, or actually punching in every line of the vocal, line by line. Sometimes it's better, but not usually. Most often, the life goes out of the song and you lose the emotional impact in the quest for perfection. If the group is solid in the studio, it comes through on the tape and it's fun. If it's overpolished, it can end up sounding cold and sterile. "The operation was a success, but the patient died."
Let me add one more important fact that should never be forgotten:
Without a great performance, by really great, talented musicians, none of the stuff in this thread is important.
It's never been about gear. It's about capturing a magical moment in time. And if the performance is truly memorable, you can do that with a $30 Radio Shack mic and a cassette recorder.
Q. ... advice on what mic/preamp to use if vocals are going to be distorted?
First of all, the distortion should be added during mixdown, not during the
recording stage. That way, you have more distortion options later. As far as mic
choices, it's impossible to tell without hearing your voice doing the material.
"Industrial" is a genre; it's not just one group, or one singer.
If you were in my studio, I might try a Shure SM-7, or a Beyer Soundstar, or a ribbon mic, or a large condenser mic on your vocals, but which mic would depend on the sound we were going after for that particular song.
I'd also be thinking ahead to the "kind" of distortion to add to the track during mixdown, so I'd be playing with the eq to insure intelligibility when I did add distortion.
Or maybe I'd leave your clean vocal alone and run the distortion to an empty track, so I could control the ratio between clean and distorted.
I'd record the quiet vocal stuff on a separate track and the loud vocal stuff on a different track, so I have even more mixing options later.
Final (?) Comments
Brad on Pro Sound Web gave Harvey a suggestion about a new mic pre thread. Harvey has taken the challenge and is inviting some real preamp heavyweights to take part. This new one looks to have a real big future and promise to turn into another one equal to the Big Mic Thread. I'm going to try to link here to the thread: Mic Pre Thread
I hope it draws the quality input and questions this Mic thread has and also lives just as long and fruitful life as this one.
Q. ...my question is about the concept of reverberation radius. Your PDF "Microphone" file mentions: "the larger the room and the less reverberant it is, the greater will be the reverberation radius.
The first part makes sense, large room, long path-length, larger radius. However, the reverberance (as I understand it) is only dependent on the attenuation from the walls, and how acoustically reflective they are...
So I guess I don't see how amplitude ties into radius.
I don't remember writing that, but what happens is this:
If you have a less reverberant room, it means that some of the signal (that is normally reflected) is being absorbed. The absorption acts like a mechanical delay, further increasing the reverberation radius.
The absorption pretty much "tricks" the wave into thinking the room is larger than it really is, by slowing the wave down as it passes thru the absorbing stuff twice, once going in, and then coming back out.
It's similar to an acoustic lens for a tweeter; the boundary layer of air molecules at the lens surface is denser than the normal air.
The sound wave front comes off the tweeter basically flat, but the outer edges of the lens are wider, so the center goes straight ahead while the lens curves the wave front by progressively slowing part of it down thru the denser molecules.
This bends the wave and it then follows the new curve, spreading out wider than it would without the lens.
Same idea; going thru the denser absorbing material twice has the same effect - making the wave appear slower by delaying the sound, thereby increasing the apparent radius.
Does that make a little more sense now, Chris?
Very clear, thanks. My background is in optics and electro-magnetics and I've been trying to draw analogies between acoustics and those fields. From what you explained, sound absorbing material is not just a bad reflector, but it includes change of media density, changing wave speed.
I had previously thought the optical analogy of a sound aborber was a "cloudy mirror", but from what you've described it's more like a cloudy mirror submersed in water.
Could you provide some other ideas about relationships between these fields? Although electro-magnetic waves and acoustic waves are probably very different in some ways (especially concerning polarization), I bet there are quite a few places where they similar (diffraction, transmission, absorption, reflection).
One curiosity is an analogy for non-linear optical materials. In the world of light, those materials have a "fast" and a "slow" axis, depending on the incident polarization. Is there an acoustic equivalent? I don't understand acoustic polarization (if there is such a thing) well enough to even guess at this one...
Yeah, kinda, but I think things may get too heavy for this kinda thread.
When a flat front sound wave is forced thru a vertical slit (that is smaller than the wavelength), the sound is bent into a curved wave front in the horizontal axis, but not the vertical axis. It's kinda like polarization, but not exactly.
Light waves can be described as being made up of either particles or waves, depending on what you wanna do with the light, but sound waves stay pretty much waves from a physics standpoint.
And then you have the "quantum physics" angle when you hafta deal with someone like Dolly Parton; You can measure her, or listen to her music, but you can't do both at the same time.
I have two papers that may be of interest to the physicist/EE with the maths to handle them.
Except for the high frequency regions, they are pretty much definitive with respect to the models of single and dual diaphragm mics and can predict their performance and patterns over the full range of design choices. They are generalized to acoustic point sources so as to include a full treatment of proximity.
Is there a place where they could be uploaded for access by this group? ...
It may be an idea to visit the Tech Talk forum at http://www.recording.org/cgi-local/ubb/ultimatebb.cgi - you may have to register (if you haven't been there already). There are guys there that host buckets loads of tech info on there own sites, some guys are working towards a DIY large diaphragm capsule at the moment, and there is the Group DIY site which is separate but linked in with Tech Talk.