MS mics placement

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Could you repeat the measurement at a frequency of 100Hz?
I could and maybe I will tomorrow, but I wouldn’t expect much from it because only the Line Audio omni is flat down to 100 Hz.
Then, you can go by rigorous measurements, empiricals, theory, and try to figure out what exactly is going on.
The theory should actually be pretty much covered by textbook knowledge regarding the basic principles of microphone construction. Only putting it together requires some effort and provides plenty of room for errors (at least for me). How well it works in practice is another question entirely.
 
When I set up mics for drums I use a series of impulses like rimshot on the snare or stick hits...
I'm still puzzled about than many still miss the difference between phase and time...

I do time align too don't make me wrong, but recording an -impulse- is just adding another sound source within space for multi source/mic instrument (like drum), everything is relative, adding just expand the complexity to me ?!?
How about the rim hit at drummer side or opposite just below the snare mic ? you already have 30cm difference, 1ms for the sound to get the snare mic (ok the sound in the ring travel faster but still...)

Back to the subject, since I had access to SF24 I prefer blumlein for coincident stereo micing, still it happens I set it up as double 8 for MS (rotating 45°)
I don't miss the MS setting of two 414 for example, SF24 is a 20sec setting ! and sound f... good
Probably I don't rec music that request the advantage of MS, like film post mixing where playing the S can focus according to image edit. It happened but It's not that often that I anticipate mix/aesthetic at rec to choos MS especially for that purpose.

Cheers
Zam
 
This kinda reminds me of this discussion:

https://groupdiy.com/threads/why-you-should-never-use-multi-pattern-mics.81157/


There are two schools of thoughts. One - if it sounds good, it's good, and then you can throw away all I stated in the thread above, and what Bert is saying.



Then, you can go by rigorous measurements, empiricals, theory, and try to figure out what exactly is going on.



Bert's claim is way beyond my current knowledge, or i just don't get what he is claiming.



One thing i know for sure is that ribbon/omni M/S setup is extremely complex, it will depend on frequency, angle of incidence, distance from the source, acoustics of the room, how the omni is placed. End adress omnis are not ideally omni-directional. They are still slightly directional, especially at close proximity. There will be diference depending on how you position the omni.

Ideally, you need identical frequency response for omni and F8. Otherwise the whole thing falls apart, and you get discrepancies throughout the frequency range.
And anyway, MS with omni mid is the least flexible/useful; all you end up with is two cardioid patterns aimed 90 degrees left and right - how often is that really useful?
 

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I'm still puzzled about than many still miss the difference between phase and time...

I do time align too don't make me wrong, but recording an -impulse- is just adding another sound source within space for multi source/mic instrument (like drum), everything is relative, adding just expand the complexity to me ?!?
How about the rim hit at drummer side or opposite just below the snare mic ? you already have 30cm difference, 1ms for the sound to get the snare mic (ok the sound in the ring travel faster but still...)

Back to the subject, since I had access to SF24 I prefer blumlein for coincident stereo micing, still it happens I set it up as double 8 for MS (rotating 45°)
I don't miss the MS setting of two 414 for example, SF24 is a 20sec setting ! and sound f... good
Probably I don't rec music that request the advantage of MS, like film post mixing where playing the S can focus according to image edit. It happened but It's not that often that I anticipate mix/aesthetic at rec to choos MS especially for that purpose.

Cheers
Zam
Using multiple mics at the same location for a comparison of phase difference with an impulse is just a way of comparing mics.
Using an impulse from the snare in recording is a means of aligning all the mics to the snare so you don’t get clouded snare hits when the mics are summed in the mix - the rim is just as good as anywhere to start - then you look at the actual snare hits on the skin to fine adjust further into the track. Only takes a couple of minutes to get a really tight drum mix.
Edit: it’s about getting rid of top end cancellations that in the mix cut the bite of the snare without killing the stereo image of the kit - cymbals and hats mics with the snare appearing in all these mics are an easy one as well as top and middle tom tracks - they’re all a slightly different distance from the snare and when you add all these together the snare can suffer - when you adjust them the snare comes good and these others don’t seem to be affected much at all.
 
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This stuff is fascinating. In the end, it reminds me of food science. Should you add salt before cooking your eggs? Science says one thing, but my mom's eggs are always way better than mine, no matter if she adds salt first or not! Similarly, people speak about how when Al Schmitt would sit behind the console the music sounded better. Recording music is about artistic sensibilities and the science only gets us so far - or it just confuses things sometimes...
 
Wow, thanks for all the brilliant folks’ take on recording music. So much technical knowledge in here.
I just wanna jump back a bit and leave a comment based on hundreds of organ concerts/recordings I’ve done.
The OP was wanting a feedback on their setup proposition.
Mine would be: yes experiment, and by all means use 2 or more methods on one bar and choose or blend in post.
But by all means, if you’re recording organ, the AB omni pair a couple feet apart has worked better than everything I’ve tried, MS included.
And I’ve also had GREAT recordings with my R-88 in Blumlein, blended with the AB pair for low-end support and a nice, realistic rendering of the acoustical space. So in theory, that could be a problem. But don’t forget, the job of post mixing is to fix problems and/or enhance reality/unreality. So the blend ratios I’ve ended up choosing have never been problematic. Are they the best? No such thing! Whatever works for a given instrument, player, venue, etc. is variable and theory goes quite far but not all the way.

I also say, in the context of that original post, the phase differences, and perfect theoretical techniques may not end up as relevant as one might hope, once on the ground, or in the cathedral.

In short: USE THE EARS, not the eyes or textbook as much. That’s what your audience is doing (hopefully), hehe!
:)
 
This stuff is fascinating. In the end, it reminds me of food science. Should you add salt before cooking your eggs? Science says one thing, but my mom's eggs are always way better than mine, no matter if she adds salt first or not! Similarly, people speak about how when Al Schmitt would sit behind the console the music sounded better. Recording music is about artistic sensibilities and the science only gets us so far - or it just confuses things sometimes...
Too true - in the end I use my ears and from experience know what works for differing scenarios - each studio space is different as well so there are no formulae, just educated start points to work from - time is money and you can’t have 3 hours of setup for a 1/2 hour take. For example having the overheads coincident X/Y and the same distance from the snare top as the floor Tom and ride mics is a fast reliable setup for phase coherence. Also by editing the audio and inserting silence between hits on less used items helps reduce the clutter for a multi-mic setup.
The other thing is using the overheads as primary and bringing up the other mics into the spread in pan position and in mono.
Also I’ve used the 3 or 4 mic setup for drums where each mic (except the inside kick mic) is equidistant from the snare - 1 overhead, 1 side of the floor tom pointing across to the snare, 1 back of the kick at a distance - all at the same distance, the inside kick mic with close proximity to the skin and getting not much snare spill with an isolator on the kick body adjacent to the snare. No close miking and no clutter. Some rooms are very reflective and the less mic’s the better.
 
This stuff is fascinating. In the end, it reminds me of food science. Should you add salt before cooking your eggs? Science says one thing, but my mom's eggs are always way better than mine, no matter if she adds salt first or not! Similarly, people speak about how when Al Schmitt would sit behind the console the music sounded better. Recording music is about artistic sensibilities and the science only gets us so far - or it just confuses things sometimes...

Wow, thanks for all the brilliant folks’ take on recording music. So much technical knowledge in here.
I just wanna jump back a bit and leave a comment based on hundreds of organ concerts/recordings I’ve done.
The OP was wanting a feedback on their setup proposition.
Mine would be: yes experiment, and by all means use 2 or more methods on one bar and choose or blend in post.
But by all means, if you’re recording organ, the AB omni pair a couple feet apart has worked better than everything I’ve tried, MS included.
And I’ve also had GREAT recordings with my R-88 in Blumlein, blended with the AB pair for low-end support and a nice, realistic rendering of the acoustical space. So in theory, that could be a problem. But don’t forget, the job of post mixing is to fix problems and/or enhance reality/unreality. So the blend ratios I’ve ended up choosing have never been problematic. Are they the best? No such thing! Whatever works for a given instrument, player, venue, etc. is variable and theory goes quite far but not all the way.

I also say, in the context of that original post, the phase differences, and perfect theoretical techniques may not end up as relevant as one might hope, once on the ground, or in the cathedral.

In short: USE THE EARS, not the eyes or textbook as much. That’s what your audience is doing (hopefully), hehe!
:)
While i do agree to a certain extent, there's an issue with this approach when it comes to the gear.

I mostly go with this approach when i'm creating art, and suspend my analytical, technical side.

When i make gear i just have to go the other way around.

Al Schmidt has the access to every piece of gear ever made, and all the experience one could have. However if he was to make a microphone he would get lost immediately. Gear is not made by producers ( for the most part ), but by very technical people whom use empiricals and learn from books. Dare i say real engineers? Here we mostly talk gear.

While the OP was more "right brain" we ended up in rabbit hole that can not remain in that domain.

In the end we should all agree if something sounds good or bad, but in order to get there we need to use science first. Once we nail something that everyone agrees sounds good, wouldn't it be nice to know exactly why it does, so that we can replicate the result and understand WHY something sounds good. This part is done by someone else for Al Schmitt so that he doesn't have to think about it. We have to do it on budget. But then we have this amazing comunity.
 
While i do agree to a certain extent, there's an issue with this approach when it comes to the gear.

I mostly go with this approach when i'm creating art, and suspend my analytical, technical side.

When i make gear i just have to go the other way around.

Al Schmidt has the access to every piece of gear ever made, and all the experience one could have. However if he was to make a microphone he would get lost immediately. Gear is not made by producers ( for the most part ), but by very technical people whom use empiricals and learn from books. Dare i say real engineers? Here we mostly talk gear.

While the OP was more "right brain" we ended up in rabbit hole that can not remain in that domain.

In the end we should all agree if something sounds good or bad, but in order to get there we need to use science first. Once we nail something that everyone agrees sounds good, wouldn't it be nice to know exactly why it does, so that we can replicate the result and understand WHY something sounds good. This part is done by someone else for Al Schmitt so that he doesn't have to think about it. We have to do it on budget. But then we have this amazing comunity.
That's what makes George Massenburg such a remarkable exception to the rule.
 
Funny Masseburg get here, from previous post I talk about SF24 for blumlein or MS, the go-to without exception is a GML 8304 pre... without any doubt so far !

As extended perspective about art and/or/vs science, the dichotomy between them is a very modern sentence...
most humanity history until recent time don't split that, but craftsman masterizing technical gesture and understanding of the nature of the (their) world, I have in mid someone like Vinci (he also write music)...but there is many others.
In other hand it was a time where -whole- knowledge may be integrated by a (brilliant) single brain...now there is so many to know that it's impossible to handle everything...welcome to the age of homo specialistus.

Cheers
Zam
 
This all reminds of me of my all time favorite mic array for orchestra, developed by Onno Scholze of Philips. By pure theory it should sound horrible due to combing, but it doesn't.

It's a pair of omnis only a foot apart on the same bar as a wide pair that are 10 feet apart. What breaks all the rules is that the two pairs are mixed together at equal levels; by the book this should result in unacceptable combing, plus there's little in stereo theory to support how fantastically real it sounds, both on headphones and loudspeakers - something few stereo arrays can claim.
 
A simpler explanation is that there is no phase lead in the ribbon mic, as Abbey has been patiently explaining.
The net output is not necessarily what is seen at the ribbon ends or voice coil ends voltage wise - the following circuit inside the mic can alter the phase of the signal.
By nature of the simple law of a wire travelling through a magnetic field there is a difference between that mechanism and a condenser diaphragm which works on proximity capacitance changes in a charged film surface approaching and moving away from a fixed potential backplane (or grounded film to charged backplane).
The voltage across the ends of the ribbon will be at maximum at the highest velocity point of travel, coming to zero at apogee, the positive air pressure as it reduces to zero at the peak of acoustic pressure results in the ribbon coming to a halt at max extension with zero electrical output, as the ribbon is fed now a negative air pressure it gradually increases in velocity in the opposite direction where it is at maximum at the midpoint between positive apogee and negative apogee, this is where it’s voltage is at maximum - this is where a condenser diaphragm has zero voltage output at the acoustic rest position, it’s + maximums are at positive and negative motion peaks, where the ribbon (or coil in a moving coil mic) will have zero voltage output - the ribbon now travelling in the opposite direction will produce a negative going (relative to the positive air pressure generated voltage) voltage, maximum at the acoustic pressure zero crossover point. This is true for Ribbon and Moving Coil mics. See pages 505 & 506 Handbook for Sound Engineers.
My point is that I think the lead obtained timewise in the ribbon or moving coil types is negated by the delay in current flow due to the inductance of the transformer used plus the inductance of the originating ribbon or coil and the 90 deg lag would render this to be the same phase as a FET condenser mic.
I never said the mic was out of phase, I’m talking about the elements of the ribbon or coil microphone that generate an output voltage being of a different phase to a condenser mic diaphragm, the internal microphone circuitry following can alter this. It’s not rocket science.
 
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Ideally, you need identical frequency response for omni and F8. Otherwise the whole thing falls apart, and you get discrepancies throughout the frequency range.
Ideally I would think you need identical frequency response regardless of the mic used and I would also think the MS matrixing into LR would not alter that response. Conversely you'd have a range of responses going from the M only (source @ 0°) to the full left/full right.
It occurs to me that, speaking of MS, we should definitely take into account the various ratios of M to S to obtain the desired pickup angle. Applying math to my SF12 (identical ribbon capsules, same preamp), a delta of 4,77 dB changes the angle by 30°.
 
Therefore I make a very last try – starting by GabrieleP's statement that the diaphragm is moved by air pressure. Most evident it is true – but it does not matter.​
Sure, it does not matter. I just wanted to get rid of the term "velocity" ;)
The point is how this mechanical movement is transformed into an electrical signal, which is the basic task of a microphone.​
That's exactly the point (as told also by RoadrunnerOZ in post #156).
In the dynamic microphone case (moving coil or ribbon) the resulting signal follows the velocity of the movement by law of induction. Thus if the exciting sound wave is of harmonic nature (sin or cos) the elongation is the result of integration from velocity, and apart from a constant, is cos or sin, and exhibits a phase shift of 90 deg. So far, that is basic math and proves the fundamental incompatibility for a combination of condensers and dynamic microphones in a MS-configuration, which by its nature is phase sensitive. Thus the problem is simply a matter of the conversion method, and not at all by any effects on the acoustical side.

How far this basic incompatibility can be overcome by structural manipulations in design of different diaphragms for the 2 types is a matter of arguing and mostly lacks a stringent base for proof.​
After 158 posts it is clear to me that the point made above is just the first in a sequence of points and not necessarily a stringent incompatibility. So... yes, the conversion method is different but, as shown by molke (and also as experienced in person), this doesn't necessarily means a phase shift between the outputs of the two microphones.
And if there IS any shift, is it significant? Can it be measured or mathematically calculated?

Something happens after that first point that can not be ignored.

I like to understand things I do, but then... if it sounds good...
 
From experience and a lot of trial and error with twin miking everything from vocals to violins, electric guitars to vibes - there are no set “this will always work” formulae for two mics - some pair nicely on an acoustic guitar 🎸- depends on the sound of the guitar - some don’t. Same goes for M/S - some mics work together and some don’t - they just sound weird. Try what you’ve got available and if that doesn’t work try a different array format.
Some mics just don’t sound good with certain voices and if one of those is the core for an M/S setup then that setup is not going to work for that vocal ensemble. That’s why I like a matched pair for M/S but you need pattern selectable ones that have fig 8.
 
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