How does diaphragm size/polar pattern relate to mic applications?

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Small diaphrams are more articulate..better to capture high frequencys details.
Large, were made for the human voice and instruments in that range for a "warmer" sound.

Experiment .....what you like is right.

Most will tell you carnoid for vocals is best, Paul McCartney told me he liked Omni.I suggested he go with what he liked ....and He's got a pritty good ear.
 
Thank U Harvey!

I just finished reading this thread on page 35 in 2 days, and what a wealth of info. Thanks you Harvey, Chris F., and all others who contributed because alot of this stuff really isn't readily available anywhere else I know of. I couldn't tear myself away from the computer, and I'm going to print this and read it again. I found this thread and it became the reason I actually joined the BBS instead of just reading from the outside.
Anyways, I do have a question. A while back you mentioned a mixing technique known as "fattening". Where you add a few milliseconds of delay, then pan both tracks out. The problem you mentioned arises when you switch the mix to mono, and the tracks disappear. Could this be remedied by keeping a third track in the middle, and panning two delay tracks out, possibly with all three tracks a few milliseconds apart? Or would that just be too much? This might be the wrong forum for this question, but I figured since I was already here...
 
You're probably better off having the singer sing it three times and slightly pan the two extra tracks, keeping them fairly low in the mix.
 
Thanks. I was looking to fatten vocals without losing them in the mono mix. The three vocal takes seems like overkill. I'll probably just try some new mike techniques I picked up from this thread. This newfound knowledge is like a new lightsaber. It's pretty cool, but I'm almost afraid to use it...
 
Recoding in home studio enviroment

Hello Harvey;
I Apologize if my inquiries are dumb. I just found your thread this morning and find it absolutley "AWSOME"! I am currently in a band situation where we practice in a finished basement and I am trying to set it up for cutting a demo. I have both Cubase SL3 and Reason 3.0. But we only have some low end hardware ( Peavey 6 ch Reference mixer, 100 watt pa amp (4CH) ,1 Shure SM58, a Shure Prologue 12L,Two StageWorks uni-directionals and a Shure/radioshack omni-directional.) My Bass amp has a balanced DI out which can be set pre or post eq. To make a long story short. Is it possible to get a good recording with all instruments playing together? I know the drums have to be isolated. But what about the guitar amps/bassamp? Your thoughts?????? I would have to say I am a novice at best, but I know what I'm aiming for. Is My Equip above too low end? We also have an issue with vocal feedback. (I have a feeling it may be we play too loud for the room (20'X20'). Plus I think the room has a lot of reflections. is there a way to suppress these without significantly modifying the room? Hope I'm not asking too much to start. But thank you for the education.










Harvey Gerst said:
Drums - the "king" of rock instruments; so many different sounds, so many different textures, so many different sources.

Where do we start? How many mics do you "really" need to record a set of drums? One? Two? Three? Four? Eight? More? And what are the best mics for drums? Ribbons? Condensors? Dynamics? Small mics? Big mics? Where do you put them for best sound?

How big a room is right? What's too small? Is there a such a thing as too big a room? If drums require a big room, why do many major studios use a small drum booth?

If some of these questions sound familiar to you (because you've asked them before), then this next section is for you. I'm gonna dispel some myths, and show you how to get a good drum sound in any room. It's gonna be a big section, but it has to be. It's similar to miking a guitar; classical guitar is different than steel string guitar is different than electric guitar.

We'll cover drum tuning, music styles, mic selection, room considerations, and mic placement - all in detail.
 
RivDBassman34 said:
Hello Harvey;
I Apologize if my inquiries are dumb. I just found your thread this morning and find it absolutley "AWSOME"! I am currently in a band situation where we practice in a finished basement and I am trying to set it up for cutting a demo. I have both Cubase SL3 and Reason 3.0. But we only have some low end hardware ( Peavey 6 ch Reference mixer, 100 watt pa amp (4CH) ,1 Shure SM58, a Shure Prologue 12L,Two StageWorks uni-directionals and a Shure/radioshack omni-directional.) My Bass amp has a balanced DI out which can be set pre or post eq. To make a long story short. Is it possible to get a good recording with all instruments playing together? I know the drums have to be isolated. But what about the guitar amps/bassamp? Your thoughts?????? I would have to say I am a novice at best, but I know what I'm aiming for. Is My Equip above too low end? We also have an issue with vocal feedback. (I have a feeling it may be we play too loud for the room (20'X20'). Plus I think the room has a lot of reflections. is there a way to suppress these without significantly modifying the room? Hope I'm not asking too much to start. But thank you for the education.

Putting some packing blankets on two adjacent walls can help deaden some of the reflections. Playing softer (forcing the mics in closer) can also help in reducing reflections and feedback.

You asked, "Is it possible to get a good recording with all instruments playing together?" Short answer, with what you have, and what you know, is probably no.

It IS possible to get a "decent" (but not great) recording with what you have, once you learn a little more about how it all works. If you need a recording right now, see if you can find a engineer at one of the nearby studios to come out and do a setup, or show you some tricks, maybe for a case of beer.

If you don't have the "gear" (the good stuff to work with) or "ears" (knowing how to use it all), then "years" (experience in working around problems) is all that's left.

Find someone with some "years".
 
Hello again Harvey;
Thank you for your response to my previous question. I have been doing a lot of reading (particularly this thread), and some research at some of the main manufacterer web sites. I have learned a tremendous amount and can't wait to apply it.
Unfortunatly, I am having a problem with seeing your drawings and diagrams. they keep getting blocked out (little white square w/ RED "X" in it) Is it my computer or what format do you send your drawings in?
I also have a question on Mic identification. If there are no names or markings on the exterior of the mike (worn off or whatever). Is there another way to ID it to determine the type of Mic it is (Dynamic,Omni, Etc).I was given a mic from a vocalist in another band, and I have this particular issue. Lastly this may seem like a very dumb question. But considering I'm new, Could you briefly explain what "SPL" is, and what exactly a "low-Z" Mic does as opposed to a standard dynamic Mic? Thank you For your knowledge. :eek:
Harvey Gerst said:
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:

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 colume since they're now radiating into one half of a hemisphere. If you put the speaker at the juction 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 interferring 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 nottom 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:

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 inreased chance of phasing problems and more susceptable 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 emphesize 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 inreased chance of phasing problems and more susceptable 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 emphesize 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.

Hey, we're nearing the end of this whole mess - just a few more things to clean up, and then we're done!!!
 
RivDBassman34 said:
Hello again Harvey;
Thank you for your response to my previous question. I have been doing a lot of reading (particularly this thread), and some research at some of the main manufacterer web sites. I have learned a tremendous amount and can't wait to apply it.
Unfortunatly, I am having a problem with seeing your drawings and diagrams. they keep getting blocked out (little white square w/ RED "X" in it) Is it my computer or what format do you send your drawings in?
I also have a question on Mic identification. If there are no names or markings on the exterior of the mike (worn off or whatever). Is there another way to ID it to determine the type of Mic it is (Dynamic,Omni, Etc).I was given a mic from a vocalist in another band, and I have this particular issue. Lastly this may seem like a very dumb question. But considering I'm new, Could you briefly explain what "SPL" is, and what exactly a "low-Z" Mic does as opposed to a standard dynamic Mic? Thank you For your knowledge. :eek:
Try this to get all the pictures:

http://www.ITRstudio.com/MIC_CHAT.PDF

Usually, a picture of the mic posted in this forum will identify most common mics.

"SPL" stands for "Sound Pressure Level" and it's a measurement of how loud something is. Knowing how loud a sound the mic can handle before it distorts is very important.

Finally, "low-Z" doesn't apply to just dynamic mics; it's a measurement of impedance from a source to determine how well the signal can be transfered to another piece of gear in the signal chain.
 
Last edited:
Harvey Gerst said:
"low-Z" doesn't apply to just dynamic mics; it's a measurement of impedance from a source to determine how well the signal can be transfered to another piece of gear in the signal chain.
I realized that answer may not help you very much, so lemme try again: Impedance is just a fancy name for resistance, but it's resistance in action; it can change depending on frequency and motion. It's important to know when you're hooking up different electronic gear.

The advantages to low impedance devices are longer cable runs without signal degradation, and less outside interference (such as hum and noise). Other tricks (such as balanced cables and/or differential amplifiers) can help to reduce noise as well.

Low impedance devices include things like most mics and loudspeakers.

There are also high impedance mics, designed for shorter cable runs (like a harmonica mic). High impedance devices include things like most electric guitars.

Impedance is important when trying to match different pieces of equipment to work together. It's like trying to match automobile wheels to a particular engine and transmission. A car with a 500 horsepower engine and 10" wheels may not be the best choice for freeway driving.

In general terms, low impedance mics (50 to 600 ohms) work best into a load about 5 to 10 times greater (1000 to 5000 ohms). High impedance guitars and mics work best into loads of about 250,000 ohms to 1 megohm.
 
I know this thread is old, but noticed there was no discussion on transformers, FET, tubes, transformerless, etc. and how they effect a mic. I am pretty ignorant in this area and I think it might be pertinant info for others. Thanks.
 
bubbagump said:
I know this thread is old, but noticed there was no discussion on transformers, FET, tubes, transformerless, etc. and how they effect a mic. I am pretty ignorant in this area and I think it might be pertinant info for others. Thanks.
Good point.

Condenser capsules are very high impedance, in the megohm range. That means they hafta work into an even higher impedance. Now tubes love working up in that range and up until transistors came out (the FET in particular), tubes were pretty much the only way to go.

Now, the signal output of most tubes rides on top of a DC voltage at the plate, so you need a way to get rid of the DC and keep the signal. Enter the transformer. Transformers block DC, but they're not always linear, but many people found those nonlinearities very pleasing. And of course transformers made balanced lines very easy to produce, for long cable runs. But you needed lotsa voltage to run the tube's heater and plate, plus power to charge the capsule, all of which required a separate, outboard power supply.

Enter the FET (Field Effect Transistor) which could accept a high impedance input and run a small balancing circuit, either thru a transformer, or direct. All you needed was enough voltage to charge the capacitor and the rest of the electronics could be run off of that as well. That was usually around 70 to 90 volts or so. But there was a snag in using those voltages and they had to come up with a solution.

70 volts or more isn't allowed (for runs thru walls) in certain countries without using metal conduit; the maximum you could run without using conduit was 48 Volts. That's where the 48 volt standard came from.

Neumann came up with a very clever scheme to run the 48 volts DC on both audio lines (pins 2 & 3). Since dynamic mics coils weren't tied to pin 1, they wouldn't even see the voltage (hence, "phantom", or invisible power).
And transformers would keep the DC out of anything that might be harmed by it. All in all, a very clever solution to a vexing problem.

Now some people said, "hey, why the hell do we even need a transformer at all? There are other ways to block DC and make balanced circuits." Removing the transformer removed a lot of cost in mics that used expensive transformers, and opened the high end up a bit more than using cheaper transformer.

And, with the development of electrostic films, they said, "hey, we don't even need the 48 volts anymore: we just need a small battery to run the other electronics in our electret mics." Others said, "screw the battery; we'll run our electret electronics off the 48 volt line."

So which is better, or best? As with all other things audio, it depends. On the build quality, application, and a thousand other factors.

But that's the reason behind all the different types condenser mics on the market today - all different solutions to the problems of high capsule impedance, how to power the thing, and how they get balanced outputs.
 
:) Hello Again Harvey;
and thank you so much for your replies and help. Just a few hours appliing the knowledge you have given has been very useful. My band mates are pretty impressed. (A little good knowledge goes a long way ). Since my last post, we have upgraded significantly, The drummer now has a seven peice mic kit on the drums ( made by Samson. 2 condensors, a kick, snare, and 3 toms). The Rythm Guitarist also picked up a good mic, (I believe also a Samson). I will be getting a Marshall MXL990. (It's coming free w/ an interface I purchased to expand our multi-track capabilites). What I want to ask is the marshall more suited for vocals, or can it be a good room mic? I was also going to try using it to record my acoustic/electric bass. I have seen more good info on it than bad. Your thoughts? Also, do you know of any decent mixers below $450.00 w/direct outs? (Behringer is supposed to be releasing one this month. But I have heard a lot of mixed feeling on this brand in extremes. Some like it. Others are offended if you use it's name and Recording in the same sentence. Just briefly, It's a 16 channel w/8 direct outs. 8 sub outs. 4 Aux sends per channel (pre/Post fader). 10 XLR ins. a C/R out. Phantom power, 3 band british EQ W/ mid sweep,and built in effects. (Not too worried about this feature). Too good to be true? or good bang for the Buck? Have you used Behringer in the past? I know it's been awhile so I've been saving these up. I really have to learn to key down, or post more frequently. At any rate, as always, your input is very helpful. Also I hope your back is doing well. Thank you again for your help and Knowledge.
Harvey Gerst said:
I realized that answer may not help you very much, so lemme try again: Impedance is just a fancy name for resistance, but it's resistance in action; it can change depending on frequency and motion. It's important to know when you're hooking up different electronic gear.

The advantages to low impedance devices are longer cable runs without signal degradation, and less outside interference (such as hum and noise). Other tricks (such as balanced cables and/or differential amplifiers) can help to reduce noise as well.

Low impedance devices include things like most mics and loudspeakers.

There are also high impedance mics, designed for shorter cable runs (like a harmonica mic). High impedance devices include things like most electric guitars.

Impedance is important when trying to match different pieces of equipment to work together. It's like trying to match automobile wheels to a particular engine and transmission. A car with a 500 horsepower engine and 10" wheels may not be the best choice for freeway driving.

In general terms, low impedance mics (50 to 600 ohms) work best into a load about 5 to 10 times greater (1000 to 5000 ohms). High impedance guitars and mics work best into loads of about 250,000 ohms to 1 megohm.
 
Too many questions that probably don't belong here. There's a thread below about the 990. It's an okay mic for some things. The Behringer questions belong in the Recording Forum or Other Equipment Reviews Forum. (I don't really know anything about Behringer mixers.)
 
I would like to join in after discovering this forum and this thread today. This must be one of the greatest resources I have stumbled upon regarding microphones and recording on the net. Thanks so much for sharing this, Harvey. Really looking forward to reading more (have just made it to page 7 so far, so there are still much to learn :) )

Could you give me some advice on what microphones to buy, that will fill in the most obvious gaps in my modest collection? I have around 1000$ to spend on microphone(s).

In november I will record a album with my band, and besides the usual drums, bass, guitars (acoustic/electric), keyboard/organ and vocals, we will also use bells, xylophone and musical saw.

We already have:

1 x Rode NTK
2 x Rode NT5
1 x Audix D1
2 x Audix D2
1 x Audix D4
1 x Audix D6
2 x Shure SM57
1 x Sennheiser e609

Any help would be much appreciated. :)
 
wernerr said:
I would like to join in after discovering this forum and this thread today. This must be one of the greatest resources I have stumbled upon regarding microphones and recording on the net. Thanks so much for sharing this, Harvey. Really looking forward to reading more (have just made it to page 7 so far, so there are still much to learn :) )

Could you give me some advice on what microphones to buy, that will fill in the most obvious gaps in my modest collection? I have around 1000$ to spend on microphone(s).

Any help would be much appreciated. :)
The next Sticky down (The Microphone FAQ) has most of the information you asked for. This thread is mainly aboout mic theory which will apply to all mics.
 
Harvey,

I got on this forum because I wanted to start doing some recording. I have been bewildered by the variety of choices there are to make. This thread has been instrumental in helping me move forward. It gave me a place to start selecting mics and also the other gear I need. Your work here will live on in whatever I accomplish in this new pursuit. I hope that feels like a tiny piece of immortality for you because that's what it is. I doubt I'm alone in feeling that way. I'll probably never understand how much I have learned.

My heartfelt thanks
 
Harvey Gerst said:
It's all in this thread. Figure 8 (pure pressure gradient, or velocity) has the most proximity effect; omni (pure pressure) has none. Cardioid is a 50/50 blend of omni and figure 8. These multi-pattern mics simply let you dial in the level of the back diaphragm to create all the patterns between figure 8 and omni.

Harvey:

This is a little tricky to explain, so let's not confuse folks on this point:

The typical multi-pattern condenser has two back-to-back cardioid capsules and the signals are mixed to generate the patterns. Cardioid isn't quite a blend of figure-8 and omni. The building blocks here are the cardioid signals themselves. Omni is approximated by adding the two cardioid signals in equal amounts (A+B), the cardioid signal is just the front capsule (A), and the figure-8 is approximated by taking the difference of the two cardioid signals (A-B). Subcardioid is usually A + 1/2B and hypercardioid is A - 1/2B.

However, the omni setting of such a mike is not the same as a true, pressure omni mike where there is a single, pressure-only capsule. The difference is most apparent where there is something near the mike and/or where the mike is used up close, because there the velocity effects to which the cardioid capsules are subject will be most apparent. Also, the individual cardioid capsules will have more off-axis coloration than true, pressure omni capsules have.

By the way, Harvey, I have a story to tell you about what happened to that Auditronics 110 console you sold me! I'll send you a PM.

Otto
 
ofajen said:
Harvey:

This is a little tricky to explain, so let's not confuse folks on this point:

The typical multi-pattern condenser has two back-to-back cardioid capsules and the signals are mixed to generate the patterns. Cardioid isn't quite a blend of figure-8 and omni.

Hmmm, now I think I have created more, not less, confusion!

Mathematically, cardioid IS a blend of omni and figure-8. The response as a function of angle, a, for the three are:
Omni: f = 1
Figure-8: f = cosine(a)
Cardioid: = 1/2(1 + cosine(a))

So, in the perfect world of math, cardioid is the average of omni and figure-8.

Ahhh, behold the wonder of trig functions... this math works nicely in reverse, too: you can take two cardioid capsules, back-to-back, and reconstruct omni and figure-8. The front capsule, capsule A, has a response = 1/2(1 + cos(a)), while the back capsule, capsule B, has a response = 1/2(1 + cos(-a)) = 1/2(1 - cos(a)). So, if you add A + B, the cos parts cancel and you get f = 1 (omni), and if you take A - B, the constant parts cancel and you get f = cos(a) (figure-8).

In reality, however, there are other important things to consider. True figure-8 mikes are typically ribbon mikes with a very small, light ribbon element. True pressure omni mikes have a larger, diaphragm-type element, but both mikes tend to exhibit much less off-axis coloration than cardioid capsule mikes. Every mike has off-axis coloration, that just means that the angular response is a complex function that depends on frequency. Off-axis coloration is a big part of the "sound" of a mike. But good ribbon and omni mikes tend to have less variation from their ideal pattern with frequency. They tend to sound more natural and accurate to the original sound (which you may or may not like, but that's a different matter).

So, when we use two cardioids in this way, we don't quite get the same results for omni and figure-8 that we get from dedicated omni and figure-8 mikes. Functionally, you get the directional reponse, but in the details of the sound and the off-axis character, things can be very different.

One small rant: I wish mike manufacturers would get better about publishing proper data to quantify off-axis coloration. The last time I asked, David Josephson was the chair of the AES working group on mike characterization, but he was not optimistic about getting folks to agree to useful standards for publishing the necessary data so that we can see what mikes actually do off-axis, where much of the fun and mischief lies. If more of us bitch to the mike makers about getting this info out, perhaps they will respond. Can you help me out here, please?

This also leads me to one of my suggestions for something to try, if you haven't already. If you sometimes have the task of recording, say, several loud instruments in a single room at the same time, say, drums, bass and guitar, try miking the instruments with omni mikes. The nature of small rooms is that sound from the instruments only falls off so far with distance and then the reflected sound level is basically constant in the rest of the room and not as far below the direct sound level as we'd like. That's what makes it just about impossible to make a decent vocal recording in a small room while there are loud instruments playing.

So, you're stuck with bleed and probably a bunch of it. Rather than fighting it, try giving in. The possible advantages of using omni mikes in this case are: 1) without proximity effect, you can mike really close and the mike doesn't change the sound, though the location is still critical and the sound itself may change close up and 2) the inevitable bleed from other instruments will probably suck less than it would suck if you used a bunch of directional mikes on the instruments. When it comes time to mix down, the tracks will play better together.

Hmmm, I hope that reduces some of the confusion I might otherwise have created.

Otto
 
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