MS mics placement

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Doesn’t seem to be that much of a level difference on the scope between 100Hz and 1KHz of the dynamic relative to the condenser - is that enough to give a 90deg shift?
I haven't mentioned amplitude, just slope. You should be able to read about 72° difference at 100Hz.
180° is 5 ticks. Shift between condenser and dynamic is 2 ticks.
180 x 2/5=72°, which corresponds to a slope of about 4.5dB/octave. which is precisely what I can see on the frequency response.
 
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I haven't mentioned amplitude, just slope. You should be able to read about 72° difference at 100Hz.
180° is 5 ticks. Shift between condenser and dynamic is 2 ticks.
180 x 2/5=72°, which corresponds to a slope of about 4.5dB/octave. which is precisely what I can see on the frequency response.
Fair enough - so where is the correction occurring? In the condenser electronics or the dynamic’s transformer?
Edit: from the original phase difference between the two types of generator elements.
 
I know there’s considerable mechanical and acoustic chamber damping employed to correct waveshape and directionality in dynamic mics - can this be uniform enough across the spectrum to even out the phase difference though?
 
Fair enough - so where is the correction occurring? In the condenser electronics or the dynamic’s transformer?
Edit: from the original phase difference between the two types of generator elements.
Neither. It comes from the fact that the diaphragm's damping and its mass counteract the sound pressure. The higher the frequency, the higher the counter-forces. These forces result in slowing down movement, which results in phase lag.
A similar thing happens in loudspeakers.
Considering only the dphi/dt would suggest a 6dB/octave rising frequency response.
Actually, a box-less speaker response is constituted of 3 segments, one at 12dB/octave in the so-called compliance-controlled region, a 6-ish dB/octave one that more or less obeys the dphi/dt formula, and one flat-ish, the mass-controlled region, until other factors kill the HF response.
 
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Then I must be doing something wrong. The setup now is with a Shure SM57 and a Haun cardioid condenser. Note that the frequency responses in this position are without a reference microphone because I’m too lazy to put one up.

Many thanks for this second measurement. As I assumed when I asked you to make a measurement at 100Hz, in the first frequency decade it is obvious that the signals from dynamic and condenser microphones are not in phase precisely because of the mechanical-electrical conversion methods of certain types of microphones. In this sense, what is written in the cited book and other sources is correct, that for stereo recordings a ribbon or dynamic-condenser combo may not be a good solution. In the second and third frequency decade, this phase difference disappears due to the electro-mechanical damping of the moving system in the case of ribbon/dynamic microphones and other effects.
 
And with MKH series mics there’s plenty of low frequency directivity to be heard. Anyway, it’s a red herring to speak exclusively of the omni.
 
As I assumed when I asked you to make a measurement at 100Hz, in the first frequency decade it is obvious that the signals from dynamic and condenser microphones are not in phase precisely because of the mechanical-electrical conversion methods of certain types of microphones.
Maybe I still don’t get it. Why is it so important to attribute a phase difference to “the mechanical-electrical conversion methods”?

Neither signal processing nor analog circuitry were ever my strong suit. But let’s say I have one high pass with a corner frequency of 50 Hz and one with 100 Hz (or two at the same frequency, but one with second order and the other with third). Then the phase of those two at a suitably low frequency will be different, won’t it? If that’s the case, then it holds regardless of whether it’s a mechanical filter, an RC high pass, or an IIR filter.
 
Neither. It comes from the fact that the diaphragm's damping and its mass counteract the sound pressure. The higher the frequency, the higher the counter-forces.
Maybe I still don’t get it. Why is it so important to attribute a phase difference to “the mechanical-electrical conversion methods”?
Have you tried testing the phase relationship by putting a scope at the voice coil terminals of the dynamic (mic transformer primary) as opposed to the transformer output which appears on the pins of the XLR
 
Have you tried testing the phase relationship by putting a scope at the voice coil terminals of the dynamic (mic transformer primary) as opposed to the transformer output which appears on the pins of the XLR
No I haven’t. But you can surely find some dynamic microphone to test it yourself? I’m not sure how a transformer or any other impedance converter would change the principle.
 
I’ve been saying all along that I think that you will get a different result by measuring directly at the coil of a dynamic mic, Abbey road seems to be adamant that all the phase correction is done by the mechanical damping of the dynamic. I’m not so sure. It’s easy enough to open the 57 and put a scope on the A and B terminals of the capsule - the transformer is glued into the back body of the mic.
 
The big picture seems to be missed by some commenters.

The mid channel is needed in order for the sides channel to work properly.

And in order for the side channels to work correctly, the mid channel needs to best represent signal that is common between the sides.

A mismatch is just not going to accurately create this context, and will fail for *multiple* reasons, in multiple ways. You can just use your ears on that one. Or test, whatever, but it’s very obvious if you *listen*.
 
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
I agree on GML preamps for most of my critical acoustic recordings. An interesting aside, I worked with George on the "To Make Me Who I Am" album by Aaron Neville (I co-produced and engineered a handful of songs and George mixed one of mine and a few others). In conversation with George, he mentioned that his favorite mic pre (as used on Linda Rondstadt, and many others) is the Mastering Lab preamp. He said he uses his own preamps when the ML preamps are not available or in large enough supply. He also said his favorite vocal compressor, after his own, of course, is the dBx 165A, which happened to be the compressor I used on that project.

As a collaborator, he is extremely sensitive to people's feelings around him, acutely aware of the musical arrangement (to the point of noting chord subs and details in the arrangements), and curious about the recording techniques and equipment used during the production. It was a really enjoyable experience for my nerdy and musical self!
 
I agree on GML preamps for most of my critical acoustic recordings. An interesting aside, I worked with George on the "To Make Me Who I Am" album by Aaron Neville (I co-produced and engineered a handful of songs and George mixed one of mine and a few others). In conversation with George, he mentioned that his favorite mic pre (as used on Linda Rondstadt, and many others) is the Mastering Lab preamp. He said he uses his own preamps when the ML preamps are not available or in large enough supply.
Believe me, I have a lot of respect for GM, but he sometimes has his quirks.
He came to Paris for a kind of masterclass and used a number of "The brick" preamps. I personally think it is a pile of garbage; measurements supported what was at first an impression. I haven't kept these measurements but one of the things that shocked me was the unduly low input impedance of less than 500 ohms and a poor EIN.
 
I’ve been saying all along that I think that you will get a different result by measuring directly at the coil of a dynamic mic, Abbey road seems to be adamant that all the phase correction is done by the mechanical damping of the dynamic. I’m not so sure. It’s easy enough to open the 57 and put a scope on the A and B terminals of the capsule - the transformer is glued into the back body of the mic.
A transformer, in its domain of validity, introduces no significant phase-shift. The voltage in the primary results in a current that lags 90°, then the flux variations produce a voltage that leads by 90°, for a net result of 0°.
 
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.
Boom - this is pretty much one of my standard concert setups. Always sounds deep and wide, full freq. But not on the same bar, just same plane for me. My bar is not that long, hehe! I use stands for the flanks. It’s amazing what omni flanks do. And the 4 together (my main usually is 20-40cm apart, depending, sometimes with a slight toe-ing outward of the main pair, say 5 degrees) allow for plenty of flexibility. My new ultimate fav for live concerts now might be becoming a Soundfield for main and the pair of omni flanks. It’s again, plenty flexy in post!

Another variant on that I use is the main pair as sub-cards in NOS ( i know! ), with the wide omni flanks.
Keep it coming, folks, this is rad!
 
Many thanks for this second measurement. As I assumed when I asked you to make a measurement at 100Hz, in the first frequency decade it is obvious that the signals from dynamic and condenser microphones are not in phase precisely because of the mechanical-electrical conversion methods of certain types of microphones.
I maintain that the phase difference is due solely to the slope of the frequency response at a particular point, irrelevant of the transduction method.
In this sense, what is written in the cited book and other sources is correct, that for stereo recordings a ribbon or dynamic-condenser combo may not be a good solution. In the second and third frequency decade, this phase difference disappears due to the electro-mechanical damping of the moving system in the case of ribbon/dynamic microphones and other effects.
If you mean that a dynamic capsule has a hi-pass response at LF, which a condenser mcapsule has too, but at a much lower frequency (typically about 3Hz instead of 50-100), I would agree.
So the concern of mixing dynamic and condenser mics may be justified. However, in most cases M-S arrangements rarely use dynamic mics. Most users of M-S arrangements use mics that have an extended LF response, such as ribbon or condenser types. There is still a debate about the LF response of ribbon mics.
LF content in the S signal is usually not terribly important, since the bass signals are usually centered, to the point that many SE's insert a HP filter in the S signal for stabilizing the LF image.
 
Maybe I still don’t get it. Why is it so important to attribute a phase difference to “the mechanical-electrical conversion methods”?
Maybe because it's true? I will try to explain it.

First, imagine a clock pendulum or a children's swing.

1694197792145.png

The maximum velocity of the pendulum is at point B, at points A and C it is zero. So the highest voltage that the ribbon gives is in position B (and it is in motion).
With a condenser microphone, the voltage is highest when the membrane is closest to the back plate (position A for example), and lowest when it is furthest away (parameter d, position C for example). It should be clear from this that the voltage from the condenser microphone and the ribbon microphone cannot be in phase and in some conditions they are shifted by 90 degrees.

Below are the formulas for both microphones from Beranek's book on acoustics.

Ribbon:
1694197941876.png

Condenser:
1694198016564.png

I hope that there is no one here to relativize what is written in that book, just as what is written in Ballou's book and what those "ignorants" from DPA have concluded are relativized. I have nothing more to add here.

Neither signal processing nor analog circuitry were ever my strong suit. But let’s say I have one high pass with a corner frequency of 50 Hz and one with 100 Hz (or two at the same frequency, but one with second order and the other with third). Then the phase of those two at a suitably low frequency will be different, won’t it? If that’s the case, then it holds regardless of whether it’s a mechanical filter, an RC high pass, or an IIR filter.

I have nothing to object to that model.
 
Maybe because it's true? I will try to explain it.

First, imagine a clock pendulum or a children's swing.

View attachment 114333

The maximum velocity of the pendulum is at point B, at points A and C it is zero. So the highest voltage that the ribbon gives is in position B (and it is in motion).
With a condenser microphone, the voltage is highest when the membrane is closest to the back plate (position A for example), and lowest when it is furthest away (parameter d, position C for example). It should be clear from this that the voltage from the condenser microphone and the ribbon microphone cannot be in phase and in some conditions they are shifted by 90 degrees.

Below are the formulas for both microphones from Beranek's book on acoustics.

Ribbon:
View attachment 114334

Condenser:
View attachment 114335

I hope that there is no one here to relativize what is written in that book, just as what is written in Ballou's book and what those "ignorants" from DPA have concluded are relativized. I have nothing more to add here.



I have nothing to object to that model.
If i get this correctly, shouldn't the diaphragm/ribbon travel distance play a role? How much do both ribbon and condenser travel at say 1k, and compare that with the distance in placement between the two, and how they are oriented. Going purely by instinct, phase discrepancy could appear at very high frequencies. And that would be in anechoic environment, adding room acoustic to the equation, this effect is masked by all phase craziness going on with the room reflections. Very frequency dependent stuff.

Also even if it's omni, direction in which the diaphragm is pointing will play significant part. In the example above where omni is pointing downwards vs front/rear facing ribbon, i feel, phase interaction would be different if the omni was to be pointed to the ceiling or front/rear, side, or anywhere in between.
 
A transformer, in its domain of validity, introduces no significant phase-shift. The voltage in the primary results in a current that lags 90°, then the flux variations produce a voltage that leads by 90°, for a net result of 0°.
Until current flows and then falls in the transformer primary for the 1st 1/2 cycle you can’t get an output voltage on the secondary - if you looked at an impulse test there should be a significant time lag of around 180deg - in that case this signal at the transformer output of a dynamic or ribbon will lag timewise behind the signal from a condenser capsule by 90deg and no longer lead by 90deg, unless the condenser uses a transformer as well in which case the condenser mic output should lag by 90deg timewise. Lots of transformerless condenser mics out there.
Try comparing a looped impulse audio signal like a snare hit on the input and the output of an audio transformer - the output voltage should lag behind the input voltage timewise if observed on a scope.
There can’t be an output voltage without the magnetic induction occurring to energise the secondary winding, which relies on current flowing and collapsing then reversing in the primary winding then this induction can produce a voltage at the output - this takes time to occur. It’s misleading to say voltage “leads” the current at the output of a transformer - phase wise yes (looking at a sine wave on a scope would make you see that as all wavefronts look identical anyway), timewise no, the output voltage doesn’t travel back in time to be leading the current that created it where the current hasn’t even started to flow for the leading wavefront.
Unless I’m completely mistaken in all of this???

When putting dual mics at one location to get the sonic benefits of each I do a simple slap test recording to see if there’s a phase/time lag in one and then placement adjustment can be calculated by using the time difference between the two waves mSec/1000 x 1100ft. Recording engineers have been doing this for years - takes minutes to do.
 
First, imagine a clock pendulum or a children's swing.
In this example, the condenser sensor detects position, correct? And the magnetic sensor detects speed. Seems to me that for equal displacement amplitude, the condenser sensor has a flat fequency response and the magnetic a +6dB/octave response, accompanied with a 90° phase lead, since speed increases linearly with frequency for constant amplitude. Am I right?
Pursuing the comparison with microphones, dynamics should exhibit a +6dB/octave response, which they do only to a limited extent. So I imagine there is some mechanism that "flattens" the frequency response, and in turn reduces the phase lead.
I'm asking. You're the one with a PhD, I'm just a humble EE estranged from academia for more than 50 years.
Actually, molke's experiments seem to corroborate my understanding.
I think that considering a dynamic diaphragm with its moving coil reacts to sound pressure identically to a condenser diaphragm with a different tuning is quite dubious.
It seems to me the || (abs value) exclude any notion of phase...? In the absence of definition for variable u, it's difficult for me to understand it.
 
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