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[Issack Andrew]

And when a typical "calibration" is done- it is simple a SPL of 1 freq.

I am not aware (but could be wrong) of any apps that actually have a calibration curve for the particular internal mic in a "device".

And over time-due to the vibrations etc that a device is subjected to, that mic can change response.

So something that was accurate when new-is often not accurate a couple of years later.

People "assume" that the mic is flat.  Even AFTER putting a case around it.

That can change a good bit.  And different cases will change it differently.

DO NOT ASSUME that just because "something" shows up on a screen (phone-pad or computer) that it is accurate.

Yes it shows "something", but is it right?  When was the last time the mic was checked against another mic?

Even known expensive mics can develop problems.

I recently had a Earthworks M30 have a HF rolloff that was not correct.  I realized it when doing a multimic calibration and noticed that 1 mic was a bit different than the others.

DO NOT ASSUME!!!!

okay, okay .. I get it - DO NOT ASSUME!!! 

[Josh Millward]

okay, okay .. I get it - DO NOT ASSUME!!! 

Why?

...because IT DEPENDS.

Also, always test the tester.

[Irvin Pribadi]

It seems there are a thousand and one dB meter and RTA apps for mobile devices out there. These apps all suffer from the same problem:

Nobody knows the response curve of the microphone and supporting electronics on YOUR device.

The app vendor may have calibrated it for THEIR device. Is their device the same as yours? Probably not. If it's the same model, it's probably pretty close, but it's still not YOUR device. There will always be minor variations from device to device and from test to test. A dB reading could be several dB off from reality.

Unless you have calibrated your app under laboratory-controlled conditions, the only thing it's good for is impressing people with the pretty colors. And maybe identifying relative differences or the presence of a feedback spike (but it can't really tell you how bad it is). Same goes for a really expensive test microphone and RTA. If the two have not been calibrated TOGETHER, you really don't know much of anything.

And how do you know your pink or white noise generator is really generating a flat output?

The PC REW software I use minimizes device distortion by generating an internal calibration file.
A digital source is sent out in a closed loop chain back through the XLR-USB and PC audio. The test result is then made into a calibration file.

The analog reference mic in this case is the only one not in the closed loop chain. That said, a reference mic with a calibration file that is unique to the specific mic serial number is helpful to minimize possibility of skew.

[Luke Geis]

The honest answer is that there is no way to even test if the mic calibration file, heck even the mic itself is every truly accurate. The microphone is a precision device. The difference between precision and accuracy is the big issue. The mic will as best as we can tell, give you the same results day in and day out, when placed in the same spot of the same room under the same conditions. You can rely on it's result to be precise, but not accurate. The accuracy of the mic is what we are technically after. We cannot determine if the mic is truly flat from 20-20 within 1db. What we can do is acquire a measurement that when all conditions are the same, produces the same result.

Since even with the calibration file we cannot be assured of the mics accuracy, then we can only then rely on our ears and experience to tell us if what the mic says is true. A good mechanic can look at a bolt head and quickly guess if it is american or metric and what approximate size wrench will be needed pretty much every time. A good audio engineer can hear a tone and guess within at the very least a 1/3 octave, what that tone is. If you can't determine what 500hz. is, then having an accurate or precise instrument is nearly useless. How can you determine either result?

The RTA analyzer and any given mic and preamp combination should yield precise enough results that are accurate enough to do the task at hand, but are ultimately limited by the user. Since a mic, preamp and RTA analyzer are not able to be measured for accuracy, they will only be good for comparative measurements. The user must fill in the blanks. The user must have at least a decent idea of what the PA is capable of and what the room effects are. Then he can look at the data and confirm bias one way or the other. If you simply look at the data presented as absolute truth, you are in trouble. Even with a more advanced measurement system, the results cannot be assumed as being accurate. There is always going to be some amount of confirmation bias. What do you hear, what is the measurement system saying and do the results add up? Even the best measurement mic, preamp and software are going to give a result that the user must be able to properly conduct to acquire and compare against. The user must already have a pretty good idea of what is right though, or the measurement is useless.

[Branimir Bozak]

Excellent comment!

[Ivan Beaver]

The user must already have a pretty good idea of what is right though, or the measurement is useless.

I have said for years: If you don't know the answer BEFORE you measure-how do you know the MEASUREMENT is correct?

You should have a general idea of the expected result.

I can tell you MANY cases in which an expensive-top level piece of test gear has given me wrong answers (because it was defective, but worked the previous time I used it).

But when something seems "wrong, you MUST dig a bit deeper to see if it is the test gear (or setup) or the device you are actually measuring.

Just because something shows up on a computer screen DOES NOT mean it is accurate.

It WILL be accurate for the particular test gear used, the particular setup used, the measurement position and the parameters entered into the program. 

But any one of those could be wrong and therefore give wrong answers.

That is why I carry multiple setups when I can, and ALWAYS carry multiple mics.

Yes, I have had Earthworks mics go bad and roll off the high end.

[Kevin Maxwell]

I can tell you that the pink noise generated by an X32 or even an M32 is not flat. And as someone else already said do you know if your pink noise is flat to start with. Even if it was it is not a good idea to EQ with an RTA to pink noise. With Smaart yes use pink noise. For me an RTA is useful to confirm that the frequency they you are having a problem with is the frequency (number) that you think it is.

BTW the TruEQ graphic EQs (dual and stereo) in the X32/M32 mixers are more like using a parametric in some ways. If you cut multiple frequencies next to each other it sums the result and you don’t get the dips and peaks between frequencies.

Here is a cut and paste to one of my methods of EQing a sound system.

Here are the basics. This the way I tune a system.
 
This is assuming that everything else in your system is set up properly and we are talking about overall system equalization. My method also assumes that you have a 2nd channel of EQ to insert on a vocal subgroup. This doesn’t even take into consideration on how to EQ the monitors. That will cost extra.

The technique that I use for EQing a system is – I EQ the system for linearity, in other words what goes in is what comes out. Or as close to that as I can get. I use a software program called SMAART. This can be done by ear also but not as quick and as accurate as when using SMAART. I will assume that you do not have SMAART.

So to do this by ear - I would usually play a bunch of different tacks from different CDs that I am very familiar with the way they sound. What you are trying to do here is to get the system to accurately reproduce the way the CD sounds. While playing the CD I then would listen for the things that don’t sound quite right. I like to only cut frequencies when doing system EQing. To pinpoint the offending tone sometimes it helps to boost the suspected offending frequency when hunting for the right one. So boosting the frequencies to make the bad sound stick out more. Sometimes you find that it isn’t the one you thought and you need to try another one. This means bring up the control of that frequency and if its not the right one bring it back down, when you find the frequency you are looking for you would then cut that frequency, how much depends on what it sounds like. I like to be conservative but you can get the feel rather quickly as to how much of a cut to make. When you are all done using this method you should hopefully find that you haven’t hacked the EQ to death. Also try hitting the bypass switch to see the difference with the EQ in or out of the system. It may be a very minimal difference.

I then insert (on the vocal subgroup) an EQ and EQ that subgroup for gain before feedback. The way I do that by ear is to have a vocal mic on stage that is on thru the system (thru the vocal subgroup) and I put another mic into another channel thru the vocal subgroup back at the mixer. I then, while using my voice at a decent level, slowly bring up the mic on stage till it starts to slightly ring (while I am making various noises and talking) I then find that frequency and cut it a bit and continue this till I start to get multiple frequencies ringing at the same time. This is usually the point at which you can’t get any farther without hacking the EQ to death and screwing up the sound. All this while I am paying attention to how my voice sounds. This is to give your vocal mics the best GBF (Gain Before Feedback) that you can reasonably expect. If you do this without exciting the system with your voice you will be surprised at the frequencies that pop up when a person gets on stage and talks into the mic. I find most mic’ed instruments don’t usually have a problem with gain before feedback and playback (CD) and instruments don’t need the additional EQ that the vocals do.  When more EQs are available you can breakup what needs to be EQed for GBF and do them separately (or in subgroups).  If you try to do a best gain before feedback EQ on the whole system you take the life out of playback and a lot of instruments. Now of course this is assuming you have that kind of flexibility to the system.

(For church)
If you were doing the church system using lavaliere mics on the preacher/speaker and handheld or in stand mics like the SM58 for the singers, I would route each of these types of mics to its own sub group. In other words you would have a wireless lav sub group and a singing sub group, and the other instruments in whatever sub groups you have leftover. It depends on how many sub groups that you have. I then EQ each vocal sub group separately. Then the channel EQ on the mixer can be used for tonal shaping for each mic.

(for theater)
If you were doing the traditional theater system using mics on the apron of the stage (apron mics) and mics hanging over the stage (overhead mic) I would route each of these types of mics to its own sub group. In other words you would have a wireless sub group, an apron sub group, an electrics (overheads) sub group etc.. It depends on how many sub groups that you have. I then EQ each sub group separately. Then the channel EQ on the mixer can be used for tonal shaping for each actor.

I find that when I EQ a system with SMAART I can do it quicker than I can with just my ears and I think I get a more consistent sound. And I even have a technique for EQing the vocal inserted EQs using SMAART.

[Luke Geis]

To simplify, I would simply make sure that the PA is set in the best position it can be utilizing physics to guesstimate a probable outcome. If you cannot help where the PA must be placed then no amount of EQ will fix the majority of problems anyway. So at that point simply EQ them to sound as best they can to you. Spend some time and learn about the physics of sound, how crossovers work, phase and time alignment to further your knowledge on what you can do to fix issues.

I used to EQ the hell out a PA to make it sound the way I THOUGHT it was supposed to. After many years I learned that if I did my homework first, utilized the physics the best I could and didn't try and play god with EQ, I worked half as hard to end up with better results. Less EQ is technically better. The goal is to try and make the PA sound balanced and pleasant without overdoing it. I find that if you have to change more than 6 frequencies more than 6db, you either set things up wrong, or are trying to play god. At some point you have to accept what the PA is and isn't and how it will sound in the different environments. You are really just trying to make the PA sound the same no matter where you go; the easiest way to do this is to NOT EQ it too much. There are other ways to fix issues in sound quality. Using EQ on groups, mic technique and even mic choice can really take you from here to there. You just have to play around a bit.

[Ivan Beaver]

I used to EQ the hell out a PA

As did most of us and many people still do.

The problem is that eq CANNOT fix TIME issues OR amplitude issues cause by TIME problems.

All an RTA shows you is amplitude.  There is no way of knowing if the peak or the dip is caused by time related issues.

Only something that shows phase (and understanding how to read it) will do that.

NOW for the really SAD part.  I know a number of people who own Smaart, and they talk about the phase issue etc.

HOWEVER-they ONLY use the RTA-NOT the transfer function!!!!!!!!!!!!

Somehow they believe that because you can see the phase response in Smaart, that "also" applies to the RTA section.

WRONG!!!!!!!!!!

Another case of having a "little" bit of understanding and completely misapplying it and telling others who then "pass it on".

Misinformation is one of the biggest problems in our industry.

[Issack Andrew]

I can tell you that the pink noise generated by an X32 or even an M32 is not flat. And as someone else already said do you know if your pink noise is flat to start with. Even if it was it is not a good idea to EQ with an RTA to pink noise. With Smaart yes use pink noise. For me an RTA is useful to confirm that the frequency they you are having a problem with is the frequency (number) that you think it is.

BTW the TruEQ graphic EQs (dual and stereo) in the X32/M32 mixers are more like using a parametric in some ways. If you cut multiple frequencies next to each other it sums the result and you don’t get the dips and peaks between frequencies.

Here is a cut and paste to one of my methods of EQing a sound system.

Here are the basics. This the way I tune a system.
 
This is assuming that everything else in your system is set up properly and we are talking about overall system equalization. My method also assumes that you have a 2nd channel of EQ to insert on a vocal subgroup. This doesn’t even take into consideration on how to EQ the monitors. That will cost extra.

The technique that I use for EQing a system is – I EQ the system for linearity, in other words what goes in is what comes out. Or as close to that as I can get. I use a software program called SMAART. This can be done by ear also but not as quick and as accurate as when using SMAART. I will assume that you do not have SMAART.

So to do this by ear - I would usually play a bunch of different tacks from different CDs that I am very familiar with the way they sound. What you are trying to do here is to get the system to accurately reproduce the way the CD sounds. While playing the CD I then would listen for the things that don’t sound quite right. I like to only cut frequencies when doing system EQing. To pinpoint the offending tone sometimes it helps to boost the suspected offending frequency when hunting for the right one. So boosting the frequencies to make the bad sound stick out more. Sometimes you find that it isn’t the one you thought and you need to try another one. This means bring up the control of that frequency and if its not the right one bring it back down, when you find the frequency you are looking for you would then cut that frequency, how much depends on what it sounds like. I like to be conservative but you can get the feel rather quickly as to how much of a cut to make. When you are all done using this method you should hopefully find that you haven’t hacked the EQ to death. Also try hitting the bypass switch to see the difference with the EQ in or out of the system. It may be a very minimal difference.

I then insert (on the vocal subgroup) an EQ and EQ that subgroup for gain before feedback. The way I do that by ear is to have a vocal mic on stage that is on thru the system (thru the vocal subgroup) and I put another mic into another channel thru the vocal subgroup back at the mixer. I then, while using my voice at a decent level, slowly bring up the mic on stage till it starts to slightly ring (while I am making various noises and talking) I then find that frequency and cut it a bit and continue this till I start to get multiple frequencies ringing at the same time. This is usually the point at which you can’t get any farther without hacking the EQ to death and screwing up the sound. All this while I am paying attention to how my voice sounds. This is to give your vocal mics the best GBF (Gain Before Feedback) that you can reasonably expect. If you do this without exciting the system with your voice you will be surprised at the frequencies that pop up when a person gets on stage and talks into the mic. I find most mic’ed instruments don’t usually have a problem with gain before feedback and playback (CD) and instruments don’t need the additional EQ that the vocals do.  When more EQs are available you can breakup what needs to be EQed for GBF and do them separately (or in subgroups).  If you try to do a best gain before feedback EQ on the whole system you take the life out of playback and a lot of instruments. Now of course this is assuming you have that kind of flexibility to the system.

(For church)
If you were doing the church system using lavaliere mics on the preacher/speaker and handheld or in stand mics like the SM58 for the singers, I would route each of these types of mics to its own sub group. In other words you would have a wireless lav sub group and a singing sub group, and the other instruments in whatever sub groups you have leftover. It depends on how many sub groups that you have. I then EQ each vocal sub group separately. Then the channel EQ on the mixer can be used for tonal shaping for each mic.

(for theater)
If you were doing the traditional theater system using mics on the apron of the stage (apron mics) and mics hanging over the stage (overhead mic) I would route each of these types of mics to its own sub group. In other words you would have a wireless sub group, an apron sub group, an electrics (overheads) sub group etc.. It depends on how many sub groups that you have. I then EQ each sub group separately. Then the channel EQ on the mixer can be used for tonal shaping for each actor.

I find that when I EQ a system with SMAART I can do it quicker than I can with just my ears and I think I get a more consistent sound. And I even have a technique for EQing the vocal inserted EQs using SMAART.

Wow, amazing advice on subgroup EQ!

I was using subgroups for Vocals on the x32 but never thought of doing GBF EQ on the subgroup instead of the master bus EQ.

I guess thats why I have issues with some instruments at times and not the vocals. I will make this change immediately for our next session.

Never used SMAART before. I'm going to do some study on it and try the demo before I decide if I can spend on it.

[ProSoundWeb Community]