Steve,

> by using varying distances you could drastically change the sound of anything safely <

"Safely" is a bit optimistic. Whenever one source arrives through different paths that are then combined, the result is comb filtering. Sometimes that filtering sounds good, but often it does not. I think of it as unintended EQ that sometimes gives a good result.

--Ethan

Ethan, by safely I meant only that if you recorded in mono then the sound you achieved through this method would not create problems in the mix. The purpose of my experiment, was to see if I could predict the frquency at which phase canellation would occur. I realize that this would not be one specific frequecy, but rather a range with the center at 180*.

I tried the same effect in stereo by combining 2 mics on each side and applying the same math. It is much more complex, and somewhat impractical, but none the less it was a fun exercise.

As I see it, since an eq operates by interrupting phase at a certain frequency this should work too. (sounds like it did.) Also since q is a ratio, theoretically then you should be able to vary the q of such an effect by changing the distances of the two mics. Both in reationship to the source, and to each other.

Ethan, if you could please explain the errors inherent with my exercise. I realize I am not a genious here, I am just trying to find interesting ways to get sounds, and to increase my understanding of the powers that be.
:c:

Steve,

> if you recorded in mono <

Just to be clear, you mean using two mikes that are mixed to mono before the recorder's input, right?

> then the sound you achieved through this method would not create problems in the mix. <

It might not create surprises when you do a mono check, but phase cancellation is what it is, and it can definitely cause an affected sound quality.

> > since an eq operates by interrupting phase at a certain frequency this should work too. <

Yes, equalizers work by shifting phase, but they do it in a more controlled manner than what you describe. And the phase shift of a given single EQ stage changes only that one band. When you use time delay instead of plain phase shift you create a comb filter, and that affects the start frequency as well as all higher frequencies.

> you should be able to vary the q of such an effect by changing the distances of the two mics. <

No, the Q of a typical filter is changed by applying feedback within the circuit.

I'm not arguing that mixing two or more mikes to mono can't give good results. Just that there are more direct ways to EQ a source.

--Ethan

Steve,
I have to side with Ethan on this.. IMO multiple mics smear the sound and deliver less impact or punch, because of the pahasing issues. When ever possible I prefer to use just one mic, choosing mic and mic pre to color the tonal content. Also I will attempt to alter the sound at the source (tune the drum , change the settings on the amp) before resorting to using eq.

Well now. THis was the info I was looking for. I didn't consider time delay. You got me.
Quote by ethan:
"Just to be clear, you mean using two mikes that are mixed to mono before the recorder's input, right?"

Yes.
Quote by ethan:
"It might not create surprises when you do a mono check, but phase cancellation is what it is, and it can definitely cause an affected sound quality."

Okay, considering the time delay, I can see your point. :p
Quote by ethan:
"No, the Q of a typical filter is changed by applying feedback within the circuit."

The thought here was that the distance between two mics would determine varying degrees of phase at certain frequencies, thus determining the amount of cancellation. Again, I have overlooked the time factor.
Quote by ethan:
"I'm not arguing that mixing two or more mikes to mono can't give good results. Just that there are more direct ways to EQ a source."

I will be the first to admit that my knowledge of electronic design and function is quite basic. I do understand basically how an eq works, but the main motivating thought here is the difference between a good eq, and a bad one.

In response to Kurts comment about dealing with a sound at the source, that is always my first area of attention, and where I spend the bulk of my time. I play, and understand the setup and tuning of drums, bass, guitar, most wind instruments, and miscellaneous percussion instruments.

I have found my best results to be yeilded from single mic setups, and it is certainly faster than playing with these elements.

I guess the whole purpose of this for me was to determine if the results yeilded would be better than that of an eq. I left out a few important factors, and I'm grateful that you pointed them out for me. This still doesn't change the fact that I was able to acheive the desired results I set out for. Also the amount of time is directly related to the distance. 1.13ft per 1ms. At that rate you should be able to have differences in distance from mic A to mic B of a little over 4 ft before most people would notice. Every 10.5 inches would be 360* of phase at 1khz.

Now the question! Why would you guys use mulit-mic setups after having said all of these things? The only other application I have used this type of setup for would be to a or acd a few mics to save time, and get the right mic placed for the job. I currently have only a couple of pre's to select from, which makes that part easy.

Also I have tried using two mics on a kick, or snare with good results, but not stellar by comparison to a single mic setup. This was another motivator for me, to maybe find the key to successful, and consistent two mic setups.

To sum It all up I have found a great way to illustrated the effects of phaseand time(Hmm.)to both my apprentice and to myself. I will say that the combination of the practical and theoretical here has been great, and also the input of the other pro's in this forum. :cool:

Thanks again for humoring a whim.

Steve,
Usually less is more.. but nothing is set in stone. I will often use a mic on the top and bottom of a snare. But I check to make sure that they are in phase. I also usually gate the bottom mic , keying it off of the top mic. This helps keep the kick out of it and avoids the out of phase issues with that.

I also have used multi mic set ups on some guitar amps. Kenny "Blue" Ray's rig is a perfect example.. He uses open backed amps most of the time so I will throw up a couple of 421's on the front, left-right and the put a U87 sideways behind the amp in figure 8 pattern. After checking phase, this not only yields a stereo image but also provides a great deal of a sense of "depth" to the sound. Try it sometime. It's killer!

I regards to eq, (I can tell you are looking for alternatives) one trick I like is the use of a compressor with an eq through the side chain. By boosting certain frequencies, you can achieve a cut in that same region.. eq without the phase shift and time smear usually associated with eq circuits....

Originally posted by Ethan Winer:
Yes, equalizers work by shifting phase, but they do it in a more controlled manner than what you describe.

:confused: Ethan, all this time I thought phase shift was the undesirable result of the filter components rather than the actual method used. Some analog EQ makers boast of having little or no phase shift. I am confused, do you mean comb filters?

--Rick

Steve,

> the main motivating thought here is the difference between a good eq, and a bad one. <

I'm sure I'll get flamed for this, but as far as I'm concerned any decent EQ you care to use is fine, hardware or software. There are only a few things that distinguish any audio component:

1. Frequency response
2. Signal to noise ratio
3. Distortion

[I hope I didn't forget anything obvious!]

There are lots of types of distortion, and lumping them together is only for brevity. Same for noise, because modulation noise on analog tape comes and goes with the signal and so is very different from "normal" noise. And the same for frequency response, because severe anomolies like ringing are more damaging than an overall deviation from flat. But it's important to keep the basics in mind, because these really are the only ways in which any audio gear differs.

> I guess the whole purpose of this for me was to determine if the results yeilded would be better than that of an eq. <

I don't see how it could be. I never understood why people will pay $1000 more for a microphone that has a pleasing presense boost. You can do exactly the same thing with EQ! It's true that frequencies can be lost or severely attenuated due to poor mike placement, and in those cases it's difficult or impossible to recover them later with EQ. But otherwise, EQ from mike choice is identical to EQ in a box. And EQ from mike placement is the same as EQ in a box that also affects the proportion of reflections captured. So mike placement also changes the ambience too.

> This still doesn't change the fact that I was able to acheive the desired results I set out for. <

Absolutely! There's no one right way to do anything in audio. As they say, "If it sounds good, it is good."

> Every 10.5 inches would be 360* of phase at 1khz. <

Yeah, but with a different amount of shift at all other frequencies. So unless you're recording a theremin sustaining a single note that never changes, you can't count on a given distance to equal some set amount of phase shift.

> Why would you guys use mulit-mic setups <

Only two reasons I ever use two mikes: For stereo, and to capture ambience. Even if you plan to have a guitar amp end up as mono, you can get a big sound by mixing a close-up mike with another mike that's farther away. Personally, I always use two separate tracks, so I can pan the close and far mikes somewhat differently if I want.

--Ethan

Rick,

> all this time I thought phase shift was the undesirable result of the filter components <

Yes, and that's a huge misconception. As proof that phase shift is absolutely benign, think about the finest recording you have ever heard. If it used any EQ, that EQ applied phase shift.

> Some analog EQ makers boast of having little or no phase shift. <

I don't think that's possible. Some digital EQs make that claim, though I've never heard one.

> I am confused, do you mean comb filters? <

All EQ - except the digital ones that use special trickery - work by combining a direct and phase-shifted version of the same signal. The whole notion of "time smear" caused by phase shift is just wrong. To me, "smear" is too much ambience, or disjointed reflections that confuse the stereo image. Sometimes people report hearing what they assume are phase changes when they boost an EQ. But I'm sure what they're really hearing is slight comb filtering that was already present, and is just made more obvious by the very act of applying EQ!

I'm not sure what you mean by comb filters. Those are created with either a large amount of phase shift [phaser effects] or with straight time delay [flanger effects]. The only real difference is effects that use phase shift have fewer teeth in the comb, so to speak.

--Ethan

Not a flame, but I submit that the kind of EQ lift that one would get from using a C12 has a different character than that of an EQ, partly because of the lack of phase anomalies. And different eq's will exhibit different qualities even with exactly the same settings. A 1k cut on a Neve/Amek eq with a given q factor is going to sound different than the same settings on a Speck ASC.. not that one is better than the other, the will just have a different color or flavor. Mics, pres, eq’s and compressors all have different sounds.

I remember about ten years ago, Julian Hirsh was embroiled deeply in a controversy where he stated that there would be no difference in the sound of different power amps. Well we all know that there is a big difference of a Peavey CS400 and a Bryston amp. The same thing applies to mics, pres comps etc. All the more reason to have a bunch of different types on hand. And not just high end stuff either. Perhaps at some point, that Mackie pre is going to be the best thing for a part. It's all part of the pallet of colors we have available to us in the world of audio and just when you think something is worthless, it will turn out to be just the thing for a certain application..

Kurt,

> Not a flame, but I submit that the kind of EQ lift that one would get from using a C12 has a different character than that of an EQ, partly because of the lack of phase anomalies. <

I can handle very hot flames, as long as things remain civil, so never worry about that.

And I submit that the only thing that defines "character" is the amount of EQ, its frequency, and its Q. I mean, what else is there?

> different eq's will exhibit different qualities even with exactly the same settings ... A 1k cut on a Neve/Amek eq with a given q factor is going to sound different than the same settings on a Speck ASC. <

Yes, this is the crux of it. I can't see how that's possible, unless maybe you're close to overloading an internal stage or something like that. Again, what would vary from one EQ to another?

> Mics, pres, eq’s and compressors all have different sounds. <

To be sure, but whatever is different is definable and can be easily understood. Not much about audio basics has changed in a many years.

> Perhaps at some point, that Mackie pre is going to be the best thing for a part. <

I love my Mackie 1202, and it's not even one of the newer VLZ models. I enjoyed very much the recent report I saw the other day about the blind test where all those famous engineers chose the Mackie as best. Never underestimate the truth-revealing powers of a blind test!

--Ethan

Originally posted by Ethan Winer:
Kurt,

> different eq's will exhibit different qualities even with exactly the same settings ... A 1k cut on a Neve/Amek eq with a given q factor is going to sound different than the same settings on a Speck ASC. <

Yes, this is the crux of it. I can't see how that's possible, unless maybe you're close to overloading an internal stage or something like that. Again, what would vary from one EQ to another?

--Ethan

Actually the reason I sited those two EQs is I happen to have a pair of both of them here. I can very easily record a comparison and post it. I assert that there will be a difference. I have heard this difference termed as the "transfer function..

Mic coloration vs. EQ:

(Gee, Ethan, I still have the bruises from the last time we tangled on this one...)

I suppose in theory you COULD painstakingly create an EQ that duplicates a microphone's frequency response, but that would only work for a source at a fixed volume, distance, and angle from the microphone. The mic's presence peaks, etc. dynamically change as sound hits it from different angles and distances (pressure gradients vs. velocity gradients as Stephen explained). So, for instance, the early reflections of a small recording space would be EQ'd by a mic quite a bit differently than the direct signal.If you could come up with an EQ that could dynamically model all those factors, that would be one pretty sophisticated (and expensive) EQ . Vocalists would develop whole new sets of skills (from working a mic to working an EQ...)

I would guess that's a lot of the reason why mic modellers are only marginally useful, if at all, in replacing the real thing.

But then again, what the hell do i know...

Assuming a "real-world" environment:

1) PHASE is the measure of specific point on a waveform relative to a specified reference point (i.e., a time or a point in space).

2) PHASE SHIFT is a term usually used to describe a change in a waveform's phase (see #1...a relative measure).

3) PHASE SHIFTING will always occur when a signal is delayed in time.

4) In an echoic acoustic space (i.e., *NOT* an anechoic space) signal delay is always created whenever there can be more than one path between the sound source and the sound receiver (e.g., ear or microphone).

5) A COMB FILTER is created whenever a signal is combined with a delayed version of itself.

6) A COMB FILTER is characterized by regular and periodically alternating peaks and valleys at various frequencies in a system's frequency response.

7) In an echoic acoustic space, a large number of comb filters are naturally created whenever a source's original acoustic wave can combine with a reflected version of that wave. This filtering occurs whether one is using zero, one or an infinite number of microphones.

8) A frequency-selective audio filter is a device that can increase or decrease the size of one frequency (or group of frequencies) relative to others.

9) In audio systems, time delay (& hence, phase shifting...see #3 above) is a NECESSARY component to realizing a filter.

10) All audio filters, whether analog or digital, achieve their aim through specific application of phase-shifting/time-delay.

Regards,
Ferd

One of Ferd's points brings up a question I always meant to ask:

Regardless of what they call it, can there really be a "phase-linear" EQ, and if so, how does one get filtering without phase?

Originally posted by littledog, jr.:
Regardless of what they call it, can there really be a "phase-linear" EQ, and if so, how does one get filtering without phase?

Yes, there really can be a "phase-linear" EQ (actually, "linear-phase"). Many of the "standard" filters used for EQ sections (etc.) have a phase response that varies with the inverse of a term (or squared term). You could call this "hyperbolic-phase". Linear-phase filters, however, shift the signal's phase(relative to the input signal) changes in proportion signal frequency as it passes through the filter.

Linear-phase response filters are implemented with what's called a "FIR" filter ("FIR" stands for Finite Impulse Response). One of the characteristics of this filter is the proportional phase change (with frequency) mentioned above, due to the fact that the FIR implementation utilizes a delay line as its foundational mechanism.

This does NOT mean that phase-shift does not occur in a linear-phase filter. In all causal filters, we need phase-shift to do the job!

Regards,
Ferd

> Some analog EQ makers boast of having little or no phase shift. <
I don't think that's possible. Some digital EQs make that claim, though I've never heard one.

The Nighpro EQ's (which are analog) do...I'm not sure how they do it, but they definitely sound different from any other analog EQ I've ever heard...

> different eq's will exhibit different qualities even with exactly the same settings ... A 1k cut on a Neve/Amek eq with a given q factor is going to sound different than the same settings on a Speck ASC. <
Yes, this is the crux of it. I can't see how that's possible, unless maybe you're close to overloading an internal stage or something like that.

I'm not sure exactly how it's possible either, but it certainly is...most of the analog EQ's I've heard sound very different from one to the next, even with relatively minor boost or cut. Are you saying that all EQ's should sound the same with the same settings?

[quoter]I enjoyed very much the recent report I saw the other day about the blind test where all those famous engineers chose the Mackie as best.

I didn't see that report...did they choose the Mackie EQ's or preamps? Preamps wouldn't necessarily suprise me, but the EQ would...

-Duardo

:) :D

The product mentioned does have some, though small, amount of phase shift.

[[url=http://[/URL]="http://www.nightpro…"]Read the specs[/]="http://www.nightpro…"]Read the specs[/]

--Rick

Kurt,

> I can very easily record a comparison and post it <

Sure, that would be great. If you post a short mono Wave file - not MP3! - of both equalizers processing pink noise that would be the most useful. Then I could analyze both and see what's different.

I have no doubt that two EQs set the same could sound different. But I don't see how that difference could be due to anything other than variations in the panel labeling. That is, you set both to have a Q of 1.0 at 500 Hz but one is actually giving you a different Q and a different frequency. It doesn't take much deviation to make an audible change.

> I have heard this difference termed as the "transfer function" <

I have no idea what transfer function means in that context. Transfer function is a general term that simply means the difference between what you put into something versus what comes out.

--Ethan

Dog,

> I suppose in theory you COULD painstakingly create an EQ that duplicates a microphone's frequency response, but that would only work for a source at a fixed volume, distance, and angle from the microphone. <

Yes, the off-axis response of mikes can be very different. But assuming you are singing into the front of the mikes from a fixed distance in a dead vocal booth, those factors should be minimized. The real issue for me is I can start with any decent mike and add some EQ to make the sound close enough to what I want.

Stop me if you've heard this one before: This same issue came up a few years ago with a friend in his studio. He needed a better vocal mike and I urged him to get an AT 4033, which is what I have. Instead he paid twice as much and bought a Neumann TLM-103 because it has a slight presence peak. So we put the two mikes side by side going to separate tracks, and I sang [badly] into each from the same distance. When we played back the tracks, sure enough the TLM sounded a little brighter. So we EQ'd the 4033 track by setting his O2R's EQ as close as we could to match the Neumann's published curve. Once we did that neither of us could tell a difference.

Is there a situation where you simply cannot EQ one mike to sound just like another? Of course. But in the overall scheme of things I see no reason to pay an extra grand [I'm starting to sound like Kurt here] just to get a particular presence peak that can be easily duplicated more or less with EQ.

--Ethan

Duardo,

> The Nighpro EQ's (which are analog) do <

I don't believe anything from them. These are the same guys who sell a $3000+ EQ with the slogan "The Magic Is In The Air." As I recall, you and I both debunked that claim a few months ago in the 24/96 thread elsewhere.

> Are you saying that all EQ's should sound the same with the same settings? <

Yes, that's exactly what I'm saying. And when they don't sound the same, the difference - whatever it is - is easily explainable using basic audio test gear. Inductors ring, circuits exhibit varying amount of level-dependant distortion, precision resistors and low-tolerance capacitors vary, and so forth. Also, with some EQ designs the Q changes as you vary the amount of boost and cut, where with others the Q stays the same. All of these easily explainable factors can and do affect the sound of an EQ circuit.

> I didn't see that report...did they choose the Mackie EQ's or preamps? <

That test was for mike preamps only. It's somewhere here on RO. I just saw it two days ago. I believe the source was a Mackie press release, but it seemed credible as I read it.

--Ethan

Ferd,

Great posts. Thanks. We always need hard facts, but especially in threads like this one.

--Ethan

So far this thread has become a great collection of info. After taking a look at all the posts, I have to say it looks like ferd has the greatest understanding of how an eq actually does the job. No disrespect to ethan or kurt, but it seems to me that all of the theories I came under attack for initially are at least a little correct.
I have to ask Ethan and Kurt:
If an eq creates a shift in phase ime, which the difference in time technically being a "comb filter", how then is this two mic idea any different in relationship to audio quality?
Would not real time, and real sounds be better than a signal from an eq, considering that these anomalies are inherent already in the acoustic space you record in?
What would be wrong with deliberately manipulating them?
Ethan, you stated that this technique yeilds good results sometimes, but wouldn't it be good to be able to duplicate and manipulate those results intentionally?
Yeah, it might be done with an eq, but are you toying around with eq's while initially micing an instrument?
Kurt left instructions for a great way to mic an open back combo, all of the phase issues will be present in varying degrees at varying frequencies, time delay between three mics as well! But it can be understood, predicted, and changed one way or another to vary the results, as kurt had mentioned.

So this brings me back to my original hypothesis.How then is this two mic setup different than an eq? I don't mean to harp on this, but I have a few really cool ideas for multi mics, and I wan't it down to a science!

(please don't think I'm being snotty here, I just want the facts, and some kind of definite conclusion, and I mean no disrespect to anyone!)
:p:

Steve,

> please don't think I'm being snotty here <

Of course not. Please challenge anything I say, and don't be shy about it. I will tell you that I've designed and built plenty of EQ circuits, so I have a pretty good idea of how they work. :)

> If an eq creates a shift in phase ime, which the difference in time technically being a "comb filter", how then is this two mic idea any different in relationship to audio quality? <

First, an EQ circuit does not usually create a comb filter. That's a special case. There are two main differences between a circuit that creates a comb filter and doing that with mike placement in a room. One difference is the reflections off surrounding surfaces in a room typically have high frequency losses due to absorption of those surfaces. So the comb depth is less at higher frequencies when compared to creating a comb filter electronically.

Another important difference is those reflections can also contribute a fair amount of ambience to the sound, depending on how far the second mike is from the source. That is, you get comb filtering and you also get additional short echos.

> you stated that this technique yeilds good results sometimes, but wouldn't it be good to be able to duplicate and manipulate those results intentionally? <

Sure, and that's exactly what you do when you add an EQ to a track. You are intentionally changing the frequency response in a controlled manner. But again, using an EQ is not necessarily the same as using multiple mikes because the extra mike also picks up reflections and echoes.

> How then is this two mic setup different than an eq? <

Two mikes pick up additional ambience and "room tone" that an EQ cannot create.

> I don't mean to harp on this, but I have a few really cool ideas for multi mics, and I wan't it down to a science! <

I applaud you for wanting to understand how this stuff really works. I wish more folks had the desire to truly understand, rather than just accept everything they read in magazines!

--Ethan

Thanks, Ethan. I think there is one thing I forgot to mention. This idea I have uses much less than an arms breadth of distance from mic to mic. This is because placement will be done with a headset on. The knowledge of distance related to frequecy, as I set out to map here, is only as a guide. Use your ear to hear the sound you don't like, and use your brain to decide how to correct it. I may post this, but either my self or,(more likely)my assistant(who has not been to school either)will chart distance ime(ms), and phase in 90* intervals @ 12 octave starting at 50 hz. If it proves interesting enough, I will see about making a graph. I think this will put the whole debate into less speculative terms.

Also you had mentioned combining a close mic with one on the other side of the room. I will work out the math for this one and place a comparison as well.

Thanks Ethan. :cool:

Steve,

> The knowledge of distance related to frequecy, as I set out to map here, is only as a guide. <

Also, understand that even at midrange frequencies the wavelengths are very short, so a tiny distance yields a large amount of phase shift. For example, at 1 KHz the waves are about a foot long, so 6-7 inches is all it takes to "reverse" the phase. At one foot the cycle is complete again, and at 20 inches it's reversed again.

> you had mentioned combining a close mic with one on the other side of the room. <

Yes, though by the time a mike is that far away the phase shift is nowehere near as important as the large amount of added ambience.

--Ethan

OK...I'm chiming in here.

1. Every piece of gear sounds differrent even two models of the same eq ( This gets pronounced with age, as caps dry out etc.). The actual electronic components (that is: the Transformer, capacitors, tubes and or transistors, op-amps, etc) all have an affect at helping to create the sound we hear coming out of said piece. Different collections of different components will have a very perceptibly different affect on the sound coming out of said piece of gear. Forget theoretical ... this is real world. A trident vs. an neve vs. a SSL e-series vs an SSL g-series, etc, all sound significantly different (even if you you were to try and use tones/pink noise/meters and scopes to achieve the same eq curve/ect.).
2. The earlier posts regarding FIR and IIR types of EQ (found in digital) is correct. There are EQ's with zero phase shift.
3. Regarding Multi-Mic recording, and it's use to acheive a sound sans eq.
This is an old timetested approach that many engineers (myself included) have done over the years. For example: Snare. In addittion to a Top and Bottom mic, one can tape a 451 (with a pad ) to the side of the top sm57 (that way the sm57 holds it...one clip & stand). "We'll" do this to get more top to the snare as opposed to adding eq on the board. Of course what most don't do to get this really right is to meticulouslly get the phase between the 451 and 57 just right. This can be done really fast in the control room with cans before you gaffer tape the 451 to the 57. Listening in summed mono with your cans, with each mic and relative equal gain, flip the phase of the 451 180 degrees and while listening to yourself talk into them, slde the 451 up & down alongside the 57 until you get the most cacellation. Now your capsules are the closest to being in phase. When you're getting your snare sound, you can add a touch of 451 for more top.
I use two or more mics all the time to create a depth. I many times use headphones to do this. When micing for stereo image (like a piano) this is a great way to go. I've been doing it for so long though that I've gravitated away from this, and now only put the cans on when either I have a long time, or need something to be critically tight....OH's...I still usually fine tune this placement by having the player lightly hit the kick, snare, toms, cymbs, hat, etc...to get the exact stereo/frequency and drums-to-cymbal balance I desire.

Also...if you read my post about the Little Labs IBP....you see there's now a device to correct these phase anomalies after the fact in the mix if the mics are printed separately. And to "beat" reality. Especially with micing a bass and using the DI.

The "Black Album" by Mettalica is a good example of multi-mic. Three mics on each drum, etc.

On the Other hand...just as often I'm into the single-mic/perfect-place scenario.
It just depends on the mood, and what needs to be done to capture and create the sounds right.

But multi-micing can be used to "eq" very effectively, it' s just an order of magnitude more difficult be yields very good results. All things sound differrent...heck even our own two ears aren't matched....so why would you think anything else is (except maybe for the digital equivalent...I'll give you that two FilterBank EQ's sound the same)

Thanks, Recorderman. I am growing increasingly more assured that I am not insane.

Ethan posted about the size of wavelenghts. For those of you reading this that dont know, the way to figure it out is like this.(keep in mind this varies with, altitude, temperature and humidity.)
Speed of sound is 1130 fps. Divide 1103ft by 1ft, and you get 1130, or 1130 hz. Each complete wavelength is 360* of phase, or one complete cycle.(see also hertz!) Therefore a 1000hz tone has 360* of phase over 10.5 inches. 1/2 that distance, or 180* of phase occurs at 5.25 inches. If you do the math, sound travels at 13.56inches per every 1/1000th of a second. (1ms)
If you offset two identical tracks(true copies) you will start to hear an audible comb filter effect created by timephase delay at about 2ms, or just slightly less than that, as the time delay increases the perceived comb effect starts to drop in pitch, then it also depends on the sound in question, as to when the effect is percieved to the human ear.

I would like to say that at a difference of less than 1 foot, it would be almost impossible for one to perceive a "comb filtered effect". Rether what one would hear would be an eqing with a q that spans over wavelengths that could acheive 90* of phase or greater over that one foot. That makes 282hz the first frequency to be effected by a one foot distance at 90*. (I am using 90* because a lesser amount may be present such as 45* at 141hz but not perceptable enough to worry about.) The effect would be audible on all frequencies above this, increasingly more so from approx 280 and up. The center frequency here would be 565hz where 180*of phase would cause cancellation, and again in varying amounts up from there, decreasing it's effect the farther it gets from the center. I used this technique to cut mids out of a guitar amp, purely as an experiment, it worked quite well, and had a pleasing tone both dirty, and clean. Two sm57 were the mics I used. I could not hear any difference in room ambience being substantial in this exercise. Simply move the mics closer together to raise the center frequency of the "eq", while simultaneously changing the "Q".

I really didn't mean to open up a "can o worms" it was just a neat experiment for myself, and my assistant engineer.

I have a question born out of pure ignorance!

Ethan, you said that Q is controlled by creating feedback within the circuit. Would I be able to vary this by feeding some signal from a mic back into itself through an aux?

Please dont get an image in your mind of, me trying to eq this way oscillating my board and speakers into oblivion! I have actually done this before with effects to create a "Flangy", or "metallic" sound in my reverbs intentionally. You just cant get too happy with the amount of feedback or.... :eek: :d:

Just for the record, everyone who knows me insists I am just a little insane, so you wont upset me for calling me crazy!

Originally posted by RecorderMan:
OK...I'm chiming in here.
(major snippage)
...2. The earlier posts regarding FIR and IIR types of EQ (found in digital) is correct. There are EQ's with zero phase shift...

Hey RM (Happy 4th!):

I had a couple quick questions about the "zero phase shift" EQs:
What manufacturer/models?
Have you used them?
What did you think?

Thanks!
Ferd

Originally posted by Ferd Berfel:

Originally posted by RecorderMan:
OK...I'm chiming in here.
(major snippage)
...2. The earlier posts regarding FIR and IIR types of EQ (found in digital) is correct. There are EQ's with zero phase shift...

Hey RM (Happy 4th!):

I had a couple quick questions about the "zero phase shift" EQs:
What manufacturer/models?
Have you used them?
What did you think?

Thanks!
Ferd Sony EQ's on the OXFORD. Very good. The absense of phase shift makes them harder to "hear"...you need to boost &/or cut more that you're used to to acheive the same degree as one would with "norma" eq's (ones that exhibit phase shift).
Give me a 1081...warts and all.

Welcome back RecorderMan,
Ethan, getting back to your comment on using mic A + eq (4033)to mimic the freq response of mic B (103). I don't doubt you were able to achieve somewhat similar results, but my thinking is that any time you send a signal through more electronics (ex. adding an eq to the path) you increase the noise and diminish the "bigness" of the sound. Also, the mic-modelling software does this very sort of thing, and I have never been convinced by the results of mic modelling. I do, however, admit that sometimes I have spent 10 minutes eqing a guitar track, getting it perfect, then realized I was using the on the bass track (the brain percieves what it wishes to...).
Doc

Originally posted by Doc@BeefyTreats.com:
I do, however, admit that sometimes I have spent 10 minutes eqing a guitar track, getting it perfect, then realized I was using the on the bass track (the brain percieves what it wishes to...).
Doc

:) LOL! Doc, I have been there, you're not alone.

--Rick

RM,

> A trident vs. an neve vs. a SSL e-series vs an SSL g-series, etc, all sound significantly different <

I do not dispute that! What I'm saying is that whatever differences there are, they all can be easily measured and understood. My point is to dispell magical thinking, not refute that different gear sounds different. I stand by my earlier statement that different brands of EQ that use the same basic filter type probably should sound the same. But when differences exist - whether intentionally or not - there should be no mystique about why.

--Ethan

Steve,

> you said that Q is controlled by creating feedback within the circuit. Would I be able to vary this by feeding some signal from a mic back into itself through an aux? <

Maybe. Okay, probably. But it's tricky to set that up with mikes.

--Ethan

Doc,

> I don't doubt you were able to achieve somewhat similar results <

No, they sounded exactly the same. Because I have such an interest in this subject I was paying very close attention to the results. Doug and I went back and forth several times. Of course, this was with my voice, not a snare drum or trombone section.

> any time you send a signal through more electronics (ex. adding an eq to the path) you increase the noise and diminish the "bigness" of the sound. <

I don't see why that should be the case. First, room noise usually exceeds by a large amount the noise of mike and preamp circuits. Second, to me "bigness" is a function of ambience in a room or added electronically. There is no doubt that some circuits make a signal sound less clean, but that's a function of distortion and perhaps a poor frequency response. Distortion is a much bigger issue than many people realize. Harmonic distortion is not nearly as bad as IM distortion which adds new components that are not musically related. And whenever you have one type of distortion you also have the other.

> I have never been convinced by the results of mic modelling. <

I absolutely agree, but keep a few things in mind. You can't model an SM57 to sound like a C-451 because the really high stuff just isn't there no matter how much you boost it. Also, as I mentioned earlier, microphones can have very different off-axis response. They also have different proximity effects, and so forth. I'm not a mike expert, and others here can probably explain better how and why mikes respond to sound differently. Yet again, my main point is that whenever there are differences, they can always be measured and understood.

> I do, however, admit that sometimes I have spent 10 minutes eqing a guitar track, getting it perfect, then realized I was using the on the bass track (the brain percieves what it wishes to...). <

Yes, and this is extremely important because it shows that often we don't really hear what we think we hear. This is why blind testing is always mandatory!

--Ethan

Originally posted by Rick Hammang:
LOL! Doc, I have been there, you're not alone.

--Rick

LMAO - haven't we all at one point or another? :p :p :p

Happy Hunting

Rod

Originally posted by Ethan Winer:
Dog,

> I suppose in theory you COULD painstakingly create an EQ that duplicates a microphone's frequency response, but that would only work for a source at a fixed volume, distance, and angle from the microphone. <

Yes, the off-axis response of mikes can be very different. But assuming you are singing into the front of the mikes from a fixed distance in a dead vocal booth, those factors should be minimized. The real issue for me is I can start with any decent mike and add some EQ to make the sound close enough to what I want.

Stop me if you've heard this one before: This same issue came up a few years ago with a friend in his studio. He needed a better vocal mike and I urged him to get an AT 4033, which is what I have. Instead he paid twice as much and bought a Neumann TLM-103 because it has a slight presence peak. So we put the two mikes side by side going to separate tracks, and I sang [badly] into each from the same distance. When we played back the tracks, sure enough the TLM sounded a little brighter. So we EQ'd the 4033 track by setting his O2R's EQ as close as we could to match the Neumann's published curve. Once we did that neither of us could tell a difference.

Is there a situation where you simply cannot EQ one mike to sound just like another? Of course. But in the overall scheme of things I see no reason to pay an extra grand [I'm starting to sound like Kurt here] just to get a particular presence peak that can be easily duplicated more or less with EQ.

--Ethan

I understand your answer, and the general principles you describe in this and other posts. But it seems to me that what it boils down to is your emulation scheme will work just fine as long as the source remains at an exact fixed location (...please, no "working" a mic) and a fixed volume (...and no getting really loud or really soft either!). And to top it off, we're only allowed to use a dead room...

I just don't think that has a whole lot of relevence to real world situations. Or musical ones... but i see your theoretical point. But in my particular case it's not going to stop me from lusting after gear that costs thousands just because in a highly constrained situation one can get a similar result with something cheaper.

Now, would you next like to try and talk me out of getting some mini traps in favor of some empty egg cartons? :D

This thread is becoming very interesting, and informative! I just hope that all the witty jabs, and tongue in cheek humor are in good fun! I really hate to see threads turn ugly over a difference in opinion.

If I could I would like to pull things back on topic a little. The initial reason I put this post up, was to update on a previous thread where opinions seemed to be more skeptical than not.
What do you mean? Intentionnally create phase cancellation in order to "eq" with a mic! :D
Another great idea(i think) is to feed the mic out to an aux and back into its channel. Intentional feed back to narrow the q!
Also we covered the aspects of phase ime delay, and now know that small distance differences between mics would not cause a noticeable comb filtering as was initially thought. In addition to this the thought occurs to me that using variable pickup patterns together, as well as different types of mic's may yeild some cool results too. My previous experiments were using two of the same mic's.

It seems to me, that if we keep our room in mind(as we always should)and use our memories to remember certain eq settings we like, a little experimentation could yeild some really cool results.
AND!
Most imporantly it's being done deliberately, and with a specific technique in mind, so that results can be duplicated reliably!

So guys, what do all of you think so far? I'm actually getting kind of exited to play around with this a bit more!

Steve.

Steve,
This is a civil discussion, absolutely, and I agree that it is one of the best threads going. Ethan rocks.
Ethan- to clarify, when I am using the phrase "bigness" I don't mean bigness in a spacial sense particularly, but more the way the word "big" was used in certain quarters in the early 90s; if somebody said, "Whoa, that's a frikkin' BIG guitar sound, man," they were saying the guitar sounded as if it was about to tear the speaker cones right off the monitors. My best examples would be Steve Albini's work with PJ Harvey on her 2nd album, or the first song on her "To Bring You My Love" album w Flood. There were also some huge guitar sounds on the first Frank Black album. Specifically, the bigness seemed to be the detail in the 150-700 hz range, and the authenticity of the guitar sound (a minimum of processing). Cheers, Doc

Originally posted by Ethan Winer:

> any time you send a signal through more electronics (ex. adding an eq to the path) you increase the noise and diminish the "bigness" of the sound. <

I don't see why that should be the case.

This is not ment to sound condensending....but his point is absolutlely true. Even if it doesn't fit into either some theoretical logic or your own experiance. The more things you run a "sound" through the smaller sounding it gets . Now you can fight this or augment it artificially, in a relative way (such as adding a sub-harm synth, aphex, tons of compression, whatever the situation)...but it isn't really the same. That's one of the reason's old vintage gear on average has an edge. That's why an 8078 or an API sound BIGGER than an SSL. Less components, larger cross-section of wire for the electrons to run down. That's why so many engineers strive to cut sans eq&comp...it gennerally keeps you on the bigger fuller side.