vocal levels for recording

theron3 wrote on 9/2/2001, 8:52 PM
It is my understanding that one should aspire to achieve 0db at peak on recording vocals. I've done this in the past an I get a great take only to realise that it is clipped/distorted when I take some time off and come back to listen again.
In the recent past, I simply sigh and cut another vocal. Now I'm faced with engineering for another vocalist. He is willing to help me out on a few tracks and I don't want to ask him back in the studio cause I can't get my levels straight. Any tips on how I can optimise his time and my recording quality?
theron3

Comments

PipelineAudio wrote on 9/3/2001, 6:25 AM
If I reach back too basic here forgive me, but I would rather the concepts be understood than someone try a record by numbers thing :)

I am going to assume that you are using 16 bit converters here. If thats the case strive for a top end of TRUE peak between -1.0 db and -0.3 db. If this were 24 bit then more like -6.0 db is in order.
The concept is this you need to record loud to minimize the signal to error ratio ( I think " signal to noise ratio " as a term is misleading in digital audio). If you record too quiet, there will be too much zipper noise in the lower level passages. This happens because the number of bits representing the sound is greatly reduced, thereby having drastic level differences between one "step " and the next as bits are toggled on and off.
There are 65,535 levels represented by the sixteen bit sample. Because of the exponential nature of these, more are available at the top than at the bottom. When you reach the bottom four bits, there are only 32 levles to represent an area of theoretically 24 db! You dont want much of this going on.

I suspect however, that the problem could be in your metering. If you are not getting past a 0 peak, there shouldnt be any digital distortion . Do not use Vegas' meters if you can help it as these will be AFTER the analog to digital converters. many ADC's wil not pass values above zero, so may look OK when in fact they are not.

You could be getting distortion in the analog realm. A compressor teamed with hopefully a PEAK limiter is usually called for when recording vocals, ESPECIALLY in the 16 bit digital realm. Be sure your meters are viewing peak and not VU or RMS, as these will be well below the peak readings and lull you into security.

Gain staging is very im[portant here. You have to be sure that each device in your chain (yes INCLUDING the mic itself) is operating at its correct range of levels. A stephen Paul modified Nuemann M-49 will not sound clean if the mic pre is saturated. A focusrite Red-8 will not be clean if it is overloading the input of your compressor and so on.

What mic, mic pre, conveerter/soundcard combo are you running ?
theron3 wrote on 9/3/2001, 1:39 PM
Thanks for the info. Well, let's see. I use a sure sm-48 going into a eurotrackmx 602a. At times I will put some reverb or chorus effect on the signal via the aux send and return on the mixer. The signal then goes from my mixer to the soundblaster live 5.1. I'm looking at the meter that is on the track fader. I don't know what else to tell you. Is this a bad setup?
theron
PipelineAudio wrote on 9/3/2001, 4:09 PM
In the case of that eurorack, there are two EXTRA gain stages at least. This could be where the problem is coming from.

Best bet is to set the channel fader at unity ( unless there is DRASTIC EQ going on. Use the mic pre gain to set the mic level to the channel...This will be the critical adjustment. get it loud enough so you arent boosting hiss with the other two faders but be SURE you are not distorting it here. When these levels are set correctly THEN use your master fader(I am assuming this is the path to your soundblaster) as simply a volume control. Try to see true peaks between -1db and -0.3db. If its still distorted after this then we gotta do some detective work :)
Rednroll wrote on 9/3/2001, 9:21 PM
Pipeline, sorry to say but I have to disagree with most of your information and advice on setting a good level. Especially the information about 24bit vs. 16bit info. In either case no matter what the bit rate is if you try and set a level at
-0.3dB on a vocal take, you are bound to get distortion due to overload. It also does not matter if you are recording 24bit or 16 bit in Vegas, the true bit depth is dependant upon what your sound card can support. (ie you don't get 24bit resolution by recording at 24bits in vegas if your soundcard only has a 16bit A to D converter. This is hardware dependant and relies on your driver of your soundcard, which vegas works together with. Seeing Vegas uses your soundcard driver, this is what the input metering of Vegas is based off of. Try arming a track in Record and then opening your soundcard mixer and adjust the gain of your soundcard mixer.....look your input fader also moves in Vegas. Anyways....enough mumbo jumbo, here's how to properly set a vocal level.

During your vocalist warmup raise the input fader in vegas until it peaks between -9dB to -6dB and leave it there. This will allow a minimum of 6dB amount of headroom before clipping and distortion will occur. This now also means the vocalist can get 6dB louder before clipping will occur.....quick lesson..."6dB means TWICE AS LOUD". A vocalist will usually get 3 to 6dB louder after the warmup....that's why I said
-9dB...just to be on the safe side. Now to address the bit depth b.s. Pipeline was getting into. Last time I checked 16bit recording still had a 96dB signal to noise ratio, it's 6dB per bit...ie 16bitsx6dB=96dB. Now if you record at a level of
-6dB, you are still using 15 out of the 16bits....which is still 90dB of signal to noise ratio. Which is pretty damn good by the way. And the noise they're actually talking about is the "noise" or "distortion" which is added due to "quantization error", which happens when converting an analog signal to a digital number. Now in a 24bit system there is a (24x6dB)=144dB signal to noise ratio. The higher the signal to noise is due to there is less "quantization error" because there is more bits (ie more numbers) to represent the level of the analog signal, therfore there is less error when converting from Analog to Digital. Point is 16bit or 24bit, you want to give yourself enough headroom to avoid distortion. In either case you want to set the level at -9dB to -6dB. This will give you a very high bit resolution and yet still give your vocalist some room to get louder and not cause any distortion. Distortion is very unforgiving in the digital domain, most sound card drivers now have a "soft clip" function to help avoid this, but still, once you do get distortion it does not sound very good as you've already learned by ruining some takes. I would rather live with a 90dB S/N ratio than distortion.

Hope this helps,
Brian Franz
Rednroll wrote on 9/3/2001, 9:46 PM
One other thing, seeing that you're also using a mixing board going into your sound card, it is always a good idea to make sure you're metering correlates between your computer and your mixing board. Pipeline is correct, that you may actually be getting distortion coming from your mixing board mic/pre or fader or buss stage. The best thing to is if you have an oscillator. Run a 1Khz tone through your mixer and out the buss...or what ever output is connected to your sound card. Make sure that the meter on your mixing board matches the meter in Vegas. In my personal studio, I use a Yamaha 03D. I run a -6dB 1Khz tone out my busses, so now my meters read -6dB coming from my mixer. Now I adjust the inputs of Vegas until the meters read -6dB, so now the readings on my mixer match the meters in vegas. Now I never touch the input settings in Vegas again. I arm a track in Vegas, I set my mic channel fader on my mixer to 0dB and then when my vocalist warms up, I adjust the mic/pre-amp until it read -9dB to -6dB in Vegas. The mic/pre will usually be about 12 o'clock, which is also the optimal setting for your mic/pre stage. Another thing, try not to add things like "Reverb" or "EQ", when recording a vocal. It is not necessary to add "EQ", when recording on a digital system, especially if you're not an experienced engineer....it is much easier to add EQ later, than trying to remove bad EQ later. Save the EQing for the mix. That goes the same for Reverb. NEVER record Reverb during Vocal takes...always record dry...Add the reverb when it comes time to mix.
PipelineAudio wrote on 9/3/2001, 10:07 PM
"information about 24bit vs. 16bit info. In either case no matter what the bit rate is if you try and set a level at
-0.3dB on a vocal take, you are bound to get distortion due to overload"

actually by the very definition of overload, this is not true. If you are at -0,3 you are NOT overloading.
if you have a peak limiter set to guarantee nothing going past this, then there will be no DIGITAL overload. But you are right to err on the side of caution, since digital distortion would be a lot nastier than SOME quantization error.

"It also does not matter if you are recording 24bit or 16 bit in Vegas, the true bit depth is dependant upon what your sound card can support. (ie you don't get 24bit resolution by recording at 24bits in vegas if your soundcard only has a 16bit A to D converter"

This goes without saying ( I hope ). Anyone not understanding this is in many other grave dangers.

"Now to address the bit depth b.s. Pipeline was getting into. Last time I checked 16bit recording still had a 96dB signal to noise ratio, it's 6dB per bit...ie 16bitsx6dB=96dB. Now if you record at a level of
-6dB, you are still using 15 out of the 16bits....which is still 90dB of signal to noise ratio. Which is pretty damn good by the way. "

how is this any different than what I said and how is it in any way b.s. ?

PipelineAudio wrote on 9/3/2001, 10:09 PM
rednroll do you record straight to vegas a lot?
I usually transfer the tapes when I am done recording. I am sometimes experimenting with going straight to disk. It seems like you get a LOT of takes this way, then its hard to get rid of the unused ones without ditching potentially good ones. How do YOU handle this ?
Rednroll wrote on 9/4/2001, 12:55 AM
It didn't differ, with what you said, The b.s. I was refering too, had to do with setting levels different if you're recording at different bit depths.....I just further explained what you had said and pointed out, that this should have nothing to do with setting levels.....if you're setting levels differently according to bit depth, then you better go back to that recording class 101 you referred to, because obviously you missed something. And nowhere did I see it mentioned that you where using a limiter or ...in fact if you are then that's even worse, because now you're completely compressing the vocals and taking all dynamics out of it at the loud parts....which sounds a lot worse than only using 10 bits out of a 16 bit system. Why would you compress the vocals range that hard (ie 10 to 1) just so you can use all 24 bits of your 24bit system? I have never used a limiter when recording vocals....over compression is a no no for any vocal performance when recording. And that's what a limiter is...it's a compressor with a very high compression ratio. Leave yourself more headroom and throw away the limiter, which is only making more noise in your signal path and sucking the life out of the artists vocal performance because you killed the dynamics and added overcompression pumping sounds, plus 8 more amplification stages of noise that are built into that limiter.

When I record I do record directly to vegas, and I also run a backup DAT. This is a professional standard when recording to a hard disk system. Also when you're done editing in Vegas you can use the broom in the media pool, which get's rid of all unused media...thus your unused takes....and then you save the vegas project and it saves only the takes you used. Back this up to a CDR and plus you have a DAT backup of ALL the takes if you ever need it.
PipelineAudio wrote on 9/4/2001, 1:21 AM
"It didn't differ, with what you said, The b.s. I was refering too, had to do with setting levels different if you're recording at different bit depths.....I just further explained what you had said and pointed out, that this should have nothing to do with setting levels"

the logic here is that at heigher bit depths you have more room to spare on the bottom as well as the top.
If I am at 96 db (the theoretical 16 bits) then we should be OK, assuming that the 16 bits used all these years are OK. If I am at 24 bits then yes, from the absolute zero of the top end of a 24 bit syustem I will be down quite a few Db's and still it will sound better than using HEAVY compression to force a 16 bit sytem into using the last 3 or 4 bits. 24 bit converters give us the freedom to record at a nice high signal to error ratio, while still giving us plenty of headroom.

"And nowhere did I see it mentioned that you where using a limiter or ."

from me first post"A compressor teamed with hopefully a PEAK limiter is usually called for when recording vocals, ESPECIALLY in the 16 bit digital realm."

"in fact if you are then that's even worse, because now you're completely compressing the vocals and taking all dynamics out of it at the loud parts"

come on now, as if everyday, most if not all vocals werent recorded with at least SOME compression. Guys who use 2" analog will SWEAR how a compressor behaves differently coming off tape than going onto it. many an LA-2A sees use on the INPUT stage of a recording chain. As per the limiter, I am talking about a PEAK limiter. Not one set down low and slow either. I mean at the last half a db to catch those pieces of transient NOTHING that would never matter at all. And this is coming from a guy who HATES to "hear" compression.

"plus 8 more amplification stages of noise that are built into that limiter"

This is a legitamate concern, but not if its catchin transient peaks of around 2-5 miliseconds. How much more are you gonna turn it up to make up for those? not much!

"When I record I do record directly to vegas, and I also run a backup DAT"

Is this a data dat or an audio DAT ?

"This is a professional standard when recording to a hard disk system"

Im NOT even gonna touch this, come on!

msrpro wrote on 9/4/2001, 7:45 AM
The standard is -12db when recording in digital.
Rednroll wrote on 9/4/2001, 2:09 PM
The Golden rule of recording is to use the shortest path from the microphone to your recording destination.....always has been....always will be. Anytime you add another device( ie a limiter daisy chained with a compressor) in the channel path, you add noise from that device to your signal. Seeing that you are concerned about 24bit resolution so much, and using every bit, you should be a little more concerned with signal to noise ratio from amplified devices. Even if that limiter isn't triggering, you are still adding the noise from the limiters input and output amplifier stages. Using a conservative compressor is a common practice in recording vocals, there is nothing wrong with this, as long as you are not over compressing. Adding a limiter after the compressor is really a dumb practice, and it makes me cringe hearing you teach this to a newby just so you can get as close to 0dB as you can....like I said...you can record at -6dB and still be using the same amount of bits that you are using with your limiter in the signal path....so what are you really gaining? Noise!!!!

Sorry, but I've done many 3D-2/ISDN sessions in my recording history and it is "common practice" to run a back up DAT. If you don't know what 3D2 means, that means I worked with many professional studios on a National level, and everyone of them runs a DAT backup when recording to hard disk. Maybe you should listen and learn something and be thankful, instead of pretending you know everything. By the way, it's an "audio DAT"....any other pro on this forum will tell you the same info, on this practice. Obviously you've never worked with any other studio outside of your ghetto basement to know any better. It's engineers like you that pros receive material from and we cuss for the next hour because the vocals are pumping from over compression and limiting, and we have to try and immulate your bad recording techniques to do a vocal punch in.

Oh by the way....0dB in 16bit or 24bit....is still 0dB, and is where clipping will occur in either case, so I don't know what you're point is about more headroom. More headroom yes!!!, but still clipping occurs at 0dB. You keep recording at your -0.3dB level with a brickwall limiter, you will never make it out of your basement studio with practices like this.
PipelineAudio wrote on 9/4/2001, 5:38 PM
"Seeing that you are concerned about 24bit resolution so much, and using every bit, you should be a little more concerned with signal to noise ratio from amplified devices"

Maybe you should go back and reread these posts from, the beginning. I sadi that at 24 bits, squishing the signal wasnt as necessary, as you could still be using all 16 bits and have QUITE a bit more headroom of 8 more bits.

And maybe I was confusing the issue, but I mean a compressor say like the Blue DBX 160 SL that also has a limiter in the output chain. Compared to the noise found in most desktop PC soundcards, this contribution is minimal, in fact I would argue the contributed noise would be UNDETECTABLE using most of these cards.

"Maybe you should listen and learn something and be thankful, instead of pretending you know everything. By the way, it's an "audio DAT"...."

I was asking this as I need to find an economical DATA backup solution as my hard disks get full if were doing many projects at once. I am looking at exabyte and TapeStor mostly.

So let me get this right, a DAT is a TWO track format, all these pro projects you do are two track? Funny...hmmm

"Obviously you've never worked with any other studio outside of your ghetto basement to know any better. It's engineers like you that pros receive material from and we cuss for the next hour because the vocals are pumping from over compression and limiting, and we have to try and immulate your bad recording techniques to do a vocal punch in. "

Oh sorry, I only ran Vintage Recorders for 6 years, where did you say YOU worked? If you dont know what studio THIS is, well, sorry, go back to Full Sail, or the Conservatory or wherever it was you learned to mop bathrooms.

And BTW most of the pros I know wont touch a computer at all, not vegas, not Pro Stools, not anything. MAYBE to edit, for a few of the open minded ones. Ditto for digital, though a few would use our Sony PCM-3348's from time to time but most were still firmly analog, who were these pros you work with that would send their stuff across ISDN lines???

And further, I dont mind to chat on the substance of our work/art and I sent theron an e-mail indicating that he would be OK using either of our suggestions and that we were just splitting hairs, but this " my dad is bigger than your dad" stuff is LAME.

althoff wrote on 9/4/2001, 7:55 PM
Gee, looks like someone's doing a "Victor" here... *laughs*

Any Vegas Forum oldtimers probably know what / who I'm talking about =)
PipelineAudio wrote on 9/4/2001, 9:18 PM
hey I MISS good old Vic/irvin/ twin chinese boys, etc whoever he is, but alas, we have lost him to Nuendo :(

Its guys like him and mixerman that remind people to stay on their toes, and keep making things better. Thugh Vic harriman mostly did it in a way that was SO insulting as to be counterproductive!

But reading that stuff was FUNNY!
theron3 wrote on 9/4/2001, 9:46 PM
So, I'll stop adding reverb while I'm recording (old habit from years ago in a box apartment, large doses of ignorance and a new effects toy), I'll shoot for -3db as my safe zone and see what that gets me. Is that the gist of this can o'worms I opened?
Thanks all for there advice and passion for the muse.
Theron
Wondering wrote on 9/5/2001, 2:10 AM
Me too using -12dB as the reference for digital.
What's all this about 0dB, -0.1dB, -0.3dB, ...etc???
Me found out that anything above -12dB = distorted.
PipelineAudio wrote on 9/5/2001, 3:16 AM
If you have something that works for you, you know the saying : if it aint broke....

Its all a compromise between two distortions, quantization distortion the lower you go, and digital overs the higher you go. Truly if no transient or peak is going over -0.3 though there should be no distortion. If you have RMS meters and theyre reading around -12dB then great, I bet there sre still peaks much higher than that though. I certainly dont mean to squash ALL of the signal to the last 3 or 4 dB, even though that IS how CD's seem to be lately.
althoff wrote on 9/5/2001, 5:32 AM
The funny thing with Victor Harriman / Irvin Gomez was that by being a total bitch about Vegas in the forums, he made people jump in and defend and support the Sonic Foundry crew wholeheartedly. We probably wouldn't have done that if he hadn't said anything...

Maybe he WAS a SF employee after all? =)

We'll never know.
Rednroll wrote on 9/5/2001, 6:53 AM
Ok, Peace Pipeline.
Words of advice as far as DAT backup.
YES!!! Dat is a 2 track. You can lay mixes off 2 tracks at a time to DAT, by muting all the other tracks. A "2 pop"(ie a 2 second beep before the song starts) is added to the front of a mix and then layed off to DAT 2 tracks at a time with the "2 POP" in front of it. That DAT can then be carried to another studio and and recorded into that studios hard disk editing system, and then just lining up all the "2 pops" you have your multitrack back....so it doesn't matter if that other studio uses pro tools, Vegas,Nuendo, SAW, Cakewalk, Neve Audiophile, SSL editor....point is that everyone has a DAT player. If you then set all your faders at 0dB, you also have the original mix that was layed off. Exabytes, are troublesome, they fail alot and not every studio has exabytes, but like I said most every studio owns a DAT player. Also, when you are recording vocals, you run a DAT backup while recording to your hard disk. This is a backup method and also a safety precaution....or I suppose next you're gonna tell me you work on hard disk systems that have never froze up. Well when they do freeze up, you haven't missed a take..you lay it off from the DAT after you reboot. I have a strong background in the advertising field and the music industry, ISDN is used on a daily basis to record commercials everyday...it's much cheaper than flying someone from LA to Detroit, and much quicker. Jimmy Paige has laid guitar tracks being recorded in the U.S., when he was in England. These studios charge $150-$350/hr, and another $200/hr to do the ISDN hookup. So I guess these aren't kids studios I'm talking about. Do some calling around and ask some studios if they have a 3D2 hookup, and then ask the price, you can learn some things. Obviously there's a lot of technology out there being used on a daily basis that you haven't experienced. What was the name of that studio you ran again?
SonyEPM wrote on 9/5/2001, 8:51 AM
As far as recapturing from DAT at another studio goes, could you not avoid that real-time recapture process entirely by dumping off wave files to CD(s) and then lining those up in the studio multitrack? (Keeping of course a DAT backup in your Haliburton briefcase).

I agree, DAT is good (I have one), but its also an expensive piece of gear with moving parts that eventually break (mine did).

By the way (new subtopic)- has anybody found a DAT machine that can transfer audio at faster than 1x? I've heard they are out there but have only found Roarke data machines that don't work for audio. Just wondering...
Rednroll wrote on 9/5/2001, 9:47 AM
Yes, dumping to Wave is a good idea. Would save much transfer time going out and in. That would be in an ideal world where everyone was either MAC or PC based. What happens when the other studio owns AMS Neve Audiophiles who don't support the Wave format yet?, like studios I've done worked in? Maybe a better solution would be to save as individual wave files and then create an audio CD...that way you have the choice of either laying it off in realtime with a CD player or ripping it into your PC/MAC using your CDrom and favorite CD ripping software.
colin2 wrote on 9/5/2001, 2:50 PM
I've just read this entire thread and I am left exactly where I was before I read it. I don't have the faintest idea where to set my recording levels. I do have a hint now about who might have the penisimulator with the most sound fonts, and whose frazilak thwicket springles quickest, but as far as recording levels...I got nothing.

Incidently, I've been reading FAQs and tutorials all over the net in the last week or so (yes, I'm a complete novice), and I've discovered that there are two options: FAQs for people who are just barely capable of finding a web page or restarting their computer, and explanations for people with degrees in computer science and backgrounds in electrical engineering.

I've never run a professonal recording studio. I never worked with George Martin. Frank Zappa has never asked me to oversee the mastering of any of his work. But I'm not an idiot either. At least that's what my Mom says.

Could someone stoop to my level and please explain, in plain English what I'm looking at when I arm for record in Vegas? Is this not my recording level? If it isn't my recording level, then what is it?

Here is exactly what I am doing:

I am recording an acoustic guitar with an AKG C1000s small condenser microphone into a BlueTube mic preamp, which is fed directly to my soundcard. When I mention the BlueTube settings, assume they refer to high on the "gain," low on the "drive," which, according to the BlueTube manual, gives an unsaturated, clean sound (This also confused me a bit, since more "gain" on my guitar amp means more distortion. I guess "drive" is "gain" at the BlueTube factory?).

I've tried the following:

(1) BlueTube set to peak at 0db, Vegas level set to peak at -.01. Completely distorted.

(2) Blue Tube set to peak at 0db. Vegas peaking at
-3.0db. Mostly distorted.

(3) BlueTube set to -3db, Vegas set to -3db. Still quite distorted.

(4) BlueTube set to 0db, Vegas set to -9db. Still distorted.

(5)BlueTube set to 0db, Vegas to -12db. Slight clipping at loudest midrange/bass notes.

(5) BlueTube set to -3db, Vegas set to -12db. This works without clipping.

When I play back the above options, I can hear the distortion, but I can also see it in the squared wave forms, at least until I get to the last couple options. Then it's harder to see because the wave form is getting rather flat. Now I understand the obvious answer is to do #5 because it "works," but I also want to end up with the loudest possible undistorted recording without compression. BTW, does having a flatter wave form mean loss of the *range* of sounds or just a quieter signal?

There was mention of another type of meter. Could you explain what this is and, in simple terms, what it does differently? And where would it go? And, again, if the meter in Vegas isn't really monitoring the level of the signal, what is it monitoring?

Thanks, I really appreciate it. I hope I'm not the only one here who might benefit from the answer.
SonyEPM wrote on 9/5/2001, 3:20 PM
Simple version: tweak your settings so you get the loudest signal you can get without clipping in the hot spots. Your second #5 sounds about right.

Try slamming a monster chord and see if you clip- if so back off on the level until you don't clip on the Vegas input meter.

Others may have opinions-
PipelineAudio wrote on 9/5/2001, 4:07 PM
Colin, do you mean the Blue Tube from Tube Works ?
Many times we use that AS a distortion pedal!
If you set the drive low enough and the output high enough you can get a undistorted signal out, but you have to be real careful that you arent bringing up a LOT of noise that way. If your soundcard has -10 inputs you should be OK. When I was talking about using the Vegas meters, I was just saying that many soundcards will not pass levels above 0dB, they will pass the signal, but it wont show on the meters as past 0dB, so you could be clipping even though the meters on vegas show a maximum level of 0.0dB.

You dont ever want to see 0.0 dB or even -0.2dB. The way different apps and converters behave it is safer to think of -0.3 as zero. That said, thats where you would only want to see your HIGHEST peaks, not the majority of your signal, which is where I think me and Brian got off on the wrong foot. If your nominal level is around -12dB for most dynamic instruments, you will probably see spikes of 8-12 dB.

Accoustic guitar is very tricky, as many notes and passages come out very quiet, but then a few spots will JUMP a LOT louder. Sometimes micing the guitar farther away than normal will even it out a little, sometimes micing it too close will do the same thing, as you will distort the mic pre or even mic, yet this may not "sound " like distortion, more of a compression.

Accoustic guitar is a PERFECT example of how recording too low can really mess things up. Trailing notes get really funny sounding as they get quiet. The problem is, compressors used on accoustic guitars, can make them sound even weirder. Sometimes, to be safe, you have to mic an accoustic further up the neck than you'd normally want, and sacrifice some of the bass, in order to get a more balanced sound, or mic further away, or just use judicious amounts of compression.

I will surely get stomped for this, but here is my usual way of micing an accoustic. I use two AKG 414's in an M-S pattern. The mid mic is above the guitar and in front pointed JUST on the neck side of the soundhole, no more than a foot away, just below that is the S mic, you gotta mess with the placement till there is no " puffs" of sound coming out the soundhole, because that would wreak havok with my compressor. I set the compression in stereo link between the channels, at about 3 to 1. Attack time is 20 mS or so, as I dont want to kill any high notes, and bass takes a little time to build up, so Im usually safe here. If the release is set too fast, you will really 'hear" the compressor, which is not usua;lly what you'd want. As my compressor of choice has a peak limiter built into it, I will set that to just before zero on my tape recorder, using a test tone. I dont ever want to see the indicator light turn on on my limiter, but its good to know its there. I use the compressor's output to set it somewhwere between -9 and -6 db nominal to the tape recorder. I dont get any distortion this way, or any untoward compressor artifacts, or any detectable unwanted noise.

Remember digital distortion of the clipping kind ONLY happens when more than one consequetive sampe goes to or above 0dB. If you see nothing past -9db or whatever you were doing earlier and you hear distortion, it is somewhere else in your chain, or it is the meter ballistics themsaelves that are being tricky.