"Where does the tone come from in a microphone?"

[abbey road d enfer]

Simpson website went out around 2016: https://web.archive.org/web/20120712235539/http://www.simpsonmicrophones.com/

I guess the reason Andy Simpson took heat was because there was a lot of BS in the presentation of his microphone.
A horn-loaded mic provides a quite narrow directivity pattern, which results in increasing significantly the critical distance, i.e. the distance at which reverberated sound equals direct sound, so, in comparison with a typical card or hypercard mic, it will sound more "present". Naming that "spectral-masking" is on the fringe of deception.
Now, this property is met only when the dimensions of the horn are comparable to the wavelength, so, in practice, this mic woud offer this property only down to 1kHz, and below would tend to omni.
In addition, when transitioning from "above the cut-off frequency" to below, there are significant frequency response variations, that require specific multi-band EQ.
Of course, horn-loading produces significant damping of the diaphragm, so the capsule does not need the usual complex damping that standard dynamic mics requires. That would make the capsule easier to make.
The same could be said for a condenser mic, which could take advantage of the inherent gain of the horn to reduce the sensitivity of the capsule by increasing the distance between diaphragm and backplate, which would reduce one cause of non-linearity. That may be why the litt claims reduced IMD.

 

[MicUlli]

Probably "minimum phase" behaviour of a microphone is hard to accomplish. The tonal influence is certainly completely overestimated and gets pronounced only in a stereo setup where the two paths have different group delay. The only mic principle that has minimum phase intrinsically is an omni SDC...

 

[Co.]

You may find that such an EQ has limited practical utility, because EQ is static and frequency response varies dynamically.

You mean, the frequency response of a microphone varies dynamically?
I thought most well designed mics do operate as a linear system within a pretty wide range, between noise floor and upper SPL limit, defined by an acceptable percentage of THD.

 

[soliloqueen]

I hope thls post somehow adds to this discussion and is not a distraction from it.

Andy Simpson (Simpson Microphones) once told me (and I'm paraphrasing from statements made years ago so please don't hold him or me too accountable...) You never see a microphone company post the phase response of their microphones, and for good reason. It tends to be horrible.

To change that, Andy designed his own microphone which aimed for best phase response. This gave it an unusual frequency response that you would compensate for in post-production by using (phase linear?) EQ. The microphone pattern was very directional, which made it difficult to use in stereo for instance, but I've never heard such clarity/transient response from anything else that comes close to his microphones. Comparing a recording of an orchestra I made with his mics and some high quality conventional mics (B&K, Schoeps), it was a real revelation as to what is possible. I could hear very clearly, very deeply to the back of the orchestra in great detail in a way the other mics couldn't convey. And the whack from percussion was stunning in its realism.

This is a discussion of the mics from way back when. Andy took a lot of heat for his ideas. Andy Simpson's New Microphone
Attached are two pictures of his microphone, profile and then inside the horn.

I had this exact idea after reading the rest of this thread only to come back and see someone else had already thought of and done this years ago. Interesting thought experiment

 

[Barry Hufker]

It is my understanding the Simpson microphone uses a moving coil. And not to take the thread on a sidetrack but a quick elaboration: Andy did have a lot of trouble convincing people of the usefulness/need for his design. It is my guess the high price was to keep the mic out of the hands of people who couldn't/wouldn't use it correctly and would cause it to then have a poor reputation. I didn't pay anywhere near the requested price for these mics. They were in the reasonable ballpark of any other mic, even with a custom design and manufacture. Andy wanted to educate people about the mics but he had difficulty getting to the right people to make that happen. It's too bad it all fell out the way it did. In my experience he's a great guy and very down to earth. I think he was surprised by the lack of acceptance and it eventually took a toll on the whole concept, which I think is a useful/important one.

EDIT: I am not technical enough to know exactly how the mic works (or why it does). It was very hard to place because of its directional pattern. I don't know if any array of them would be superior to a single mic. I wish I had more time to really experiment with them the way I ought to because I think they deserve it. Unfortunately when I have something to record, time is precious and guaranteed results must be delivered...

 

[soliloqueen]

It is my understanding the Simpson microphone uses a moving coil. And not to take the thread on a sidetrack but a quick elaboration: Andy did have a lot of trouble convincing people of the usefulness/need for his design. It is my guess the high price was to keep the mic out of the hands of people who couldn't/wouldn't use it correctly and would cause it to then have a poor reputation. I didn't pay anywhere near the requested price for these mics. They were in the reasonable ballpark of any other mic, even with a custom design and manufacture. Andy wanted to educate people about the mics but he had difficulty getting to the right people to make that happen. It's too bad it all fell out the way it did. In my experience he's a great guy and very down to earth. I think he was surprised by the lack of acceptance and it eventually took a toll on the whole concept, which I think is a useful/important one.

EDIT: I am not technical enough to know exactly how the mic works (or why it does). It was very hard to place because of its directional pattern. I don't know if any array of them would be superior to a single mic. I wish I had more time to really experiment with them the way I ought to because I think they deserve it. Unfortunately when I have something to record, time is precious and guaranteed results must be delivered...

It's pretty simple really. It's a natural sort of inversion of what I said earlier in this thread. If you can't produce an extremely directional sound in a laboratory I suppose you could produce an extremely directional microphone instead to get the same control over the phase of the recording, which is why I was thinking about it, but I'm not sure of the practical applicability of this and I agree with others that the terms he's quoted as using are pretty flowery. Not to say it's not a real concept, because it is, for the reason I just outlined, but maybe people took issue with his approach.

 

[abbey road d enfer]

You mean, the frequency response of a microphone varies dynamically?
I thought most well designed mics do operate as a linear system within a pretty wide range, between noise floor and upper SPL limit, defined by an acceptable percentage of THD.

I used the word "dynamically" in order to avoid a lengthy enumeration of all the reasons that make the response of a mic vary. And in the context of Minimum-Phase, response is not only frequency response, it's also amplitude response (non-linearity) and phase response.
Many factors there, temperature being the most prominent, humidity, ambient pressure, ageing of material, even te orientation of the diaphragm. They result in changes in response that cannot be handled by a fixed EQ, however sophisticated. It would take a fully adjustable EQ with a specific HI.

 

[abbey road d enfer]

It was very hard to place because of its directional pattern.

It's probably the main reason it didn't catch.
The same issue existed in PA. Using very directional speakers resulted in part of the audience being disdavantaged, when usinh wide pattern speakers resulted in spillage, reflections and wasted energy.
The solution came with the advent of arrays. Arrays are constituted of narrow-pattern elements that can be assembled to provide the desired coverage. Achieving this requires a certain number of criteria.
This happened mainly because the ergonmics have benefitted of nearly 30 years of development, and DSP power is commonly available.
The same would apply to microphone arrays. I believe the Simpson mic didn't conform, because the conical pattern is not the ideal construction brick and the cost was prohibitive.
Microphone arrays have gained a surge of interest, due mainly to the cheapness of good elements (electrets and MEMS) and the availability of DSP resources.

I don't know if any array of them would be superior to a single mic.

Using horn-based arrays for large-stage recording may be an interesting research direction. And it certainly woud take a lot of convincing and education to catch on.

 

[Marik]

I might yet persuade Marik to start checking for MP so he might come up with evidence to the contrary I've provided some hints on where to look

I am still to find a way to organize measurements in that direction, let alone to get convinced in the significance of the measurements and their practicality, at all. So far I was unable to get any meaningful correlation between microphone sonics and measurements. Another problem is a difference in frequency response between measuring the impulse response and just sweeping the bandwidth. That alone might already be a good indication of non MP of the system.

Intuitively, I feel the on axis is the least affected area and the whole problem is off axis. In a classic resistance controlled cardioid system we DO HAVE two paths, where the second one is highly non linear, frequency dependent, and on top, prone to chamber resonances. As a result the cancellation is also highly nonlinear. I do not have good measuring methodology to prove one way, or another and can judge the results rather indirectly. That is, because of the way linear response formed in both mass and stiffness controlled systems, they do take the EQ very well. That might be a good indication of their good phase integrity. Indeed, one might say the ribbon Fig8 does have two paths, however, the second is completely linear and the shorter front to back path, the less phase anomalies.

On the contrary, the resistance controlled cardioid pattern is a total mess. The Braunmuhl Weber arrangement is even messier because of its contradicting parameters of the rear diaphragm... the CK12 is better because of its additional chambers, which split the damping/delay functions, which in turn takes care of some chamber resonance problems.

Best, M

 

[Marik]

This is excellent. Part of the reason why the ribbon doesn't have the frequency response extension could be that the physical size of the ribbon means that you get phase cancellation @ high frequencies.

No, the ribbon physical size has much less influence than cancellation due to front to back path distance. But there are ways to greatly extend the top extension. For example, our MF65 ribbon is +2dB at 25kHz (!!!). Obviously, it goes quite a bit higher than that. The only reason it is not shown in the graph is that was the highest frequency the system was certified. But its design with 6 magnets per motor already starts with the shortest front to back path ever seen a ribbon microphone, so that kind of extension is easy to achieve.

Best, M

 

[Marik]

I hope thls post somehow adds to this discussion and is not a distraction from it.

Andy Simpson (Simpson Microphones) once told me (and I'm paraphrasing from statements made years ago so please don't hold him or me too accountable...) You never see a microphone company post the phase response of their microphones, and for good reason. It tends to be horrible.

To change that, Andy designed his own microphone which aimed for best phase response. This gave it an unusual frequency response that you would compensate for in post-production by using (phase linear?) EQ. The microphone pattern was very directional, which made it difficult to use in stereo for instance, but I've never heard such clarity/transient response from anything else that comes close to his microphones. Comparing a recording of an orchestra I made with his mics and some high quality conventional mics (B&K, Schoeps), it was a real revelation as to what is possible. I could hear very clearly, very deeply to the back of the orchestra in great detail in a way the other mics couldn't convey. And the whack from percussion was stunning in its realism.

This is a discussion of the mics from way back when. Andy took a lot of heat for his ideas. Andy Simpson's New Microphone
Attached are two pictures of his microphone, profile and then inside the horn.

Barry,

At the time I was also working on a similar concept based on a ribbon transducer. I tried a few different topologies, including loading the front, only, and also, splitting the load between front and back of the ribbon with different ratios, as well as putting at the back infinite load. With Andy we were privately discussing many aspects of the design. The concept itself is quite attractive, especially for lower output dynamic transducers, which we both used (his was moving coil, as already mentioned) where the large size horn loading worked as an acoustical transformer. In Andy’s design the horn was effectively cutting out off-axis junk, making the mic look very low distortions and (because of acoustic transformation) very present. Besides pick up pattern and funky freq. response there was another problem which is... practicality of the design. Indeed, that was possible to improve the pattern, however, the horn should've been quite a bit larger and wider. The sheer size would totally negate, or contradict any rules of stereo techniques, or ease of setup. So finally, I had no choice, but drop it... however, there is something else pretty interesting coming out pretty soon, so stay tuned!

Best, M

 

[soliloqueen]

There is probably some smarter way to do this with like, acoustic metamaterials or something. Maybe I'll mess around with it later.

 

[abbey road d enfer]

There is probably some smarter way to do this with like, acoustic metamaterials or something. Maybe I'll mess around with it later.

Probably, but don't forget teh basic rules of physics. Achieving low frequency directivity with arrays implies large dimensions.

 

[soliloqueen]

Probably, but don't forget teh basic rules of physics. Achieving low frequency directivity with arrays implies large dimensions.

No way around good old physics!

 

[Matador]

I really liked the video (and many of his others). The hard part is that trying to control for single variables in a complex system is difficult.

In my experience, room treatment/size, and positioning of the mike relative to the sound source dominates the captured 'tone' far more than capsules, head basket, head amp, tube vs. solid state, etc, which adds a ton of caveats to his "conclusions". What would have been more useful is not answer "where does tone come from", but "how does the mike translate the performance into a medium", of which "tone" is a (smaller) part. This also would have applied equally to his other videos as well, which seem to be trying to answer which parts of a thing are "most influential" to the overall end-result.

 

[Tubetec]

I second what Mat says ,
Atempting to mic something up in a room with parrallel walls , unfavourable resonances and flutter echo is gonna sound like crap , a higher quality mic is only likely to make matters worse ,
If a sound source in room sounds good to your ear then in all likelyhood it will record ok , if it doesnt your left fighting , attempting to compensate electronically for fundamental acoustic flaws .

I appreciate all the other accumulated knowledge put forward in this thread , but why get bogged down trying to measure ,when the naked ear in a room tells you theres a problem , test tones are far to simplistic a stimulus to replecate what happens with real world sounds .

I didnt actually even look at the video posted ,because I didnt like the way the thing was framed because its not all about the mic ,
Kinda reminds me of the video where Chuck and Kieth come to blows in the studio , Chuck has this big woolly bass end bottoming out the speaker cabs and creating a bad room interaction , Keith tells him the guitar isnt hitting the mic right , Chucky strikes out at him and shuts the opperation down

 

[Matador]

If a sound source in room sounds good to your ear then in all likelyhood it will record ok , if it doesnt your left fighting , attempting to compensate electronically for fundamental acoustic flaws .

Exactly: the best 251 is going to sound like crap in my tiled bathroom (pun intended).

If I had to just make up numbers based on my own experience, I would say 50% of the recorded sound is the sound source in the environment (what kind of room, it's size, and how it is treated, and the actual instrument/talent), and about 30% is mike positioning (close miking versus room miking or any combination therein), and maybe 10% is mike type (LDC, SDC, ribbon, etc). The last 10% is everything else (size of capsule, type of amplifier, transformers, mike preamp, etc), which happens to be the 10% focused on in his testing / his video. It's quite astonishing how far one can get with a clean preamp and an SM57, paying close attention to the room and mike positioning.

Knowing this, the best video would be to first hold the recording chain constant, and test out different types of rooms, treatments, etc, and show how they impact the sound, then test mike placement holding the room constant, etc, etc.

 

[Tubetec]

Ive got some great results over the years with 57and 58's for main vocal,
despite having mic cupboards full of high end condenser mics ,
Sure its lo-fi ,but with the right combination of talent ,room ,eq ,compression and tape path it can really grab you by the ears ,

 

[Icantthinkofaname]

I wouldn't describe it as dark. I'd say it means "unclear", coming from a resonance.
Can be good or bad, depending on taste and application.
A neve EQ is probably more "unclear" than a Calrec, but this can be nice.
Weird membrane resonances, that are filtered out with not so great electronics in a condenser mic are "unclear" in a way that probably nobody likes.

My problem with judging stuff that by ear is that to my ears many of these "smeary" mics sound just as fast as another mic, but just darker.

 

[ricardo]

It is totally possible, to have microphones with the same frequency response, but a completely different transient response. This of course is going to lead to a different representation as an IR.
So transient response and frequency response might be represented as an IR, but that doesn't mean, that transient response and frequency response is completely related.

Got some examples where you see this?

How is this all going to be represented (in an IR) considering proximity effect and off axis response? What is actually the practical value of talk about IR and minimum phase, related to the topic of the thread

Err..rh! Proximity and off-axis response are also MP, at least on the mikes I've tested

The practical value is what I posted earlier.

• If you improve a MP Amplitude response with analogue (ie usually MP) EQ, you also improve the Phase and Impulse Response.

This is why I was interested in MP from da late 70s.

My 'previous life' specialty was integrating analogue & digital EQ for better speakers & mikes. MP might explain mikes which respond 'well' to (analogue) EQ ... but I haven't found any LDC or SDC mikes which were non-MP. Haven't measured a ribbon since circa 1979 but see no reason why they should be non-MP

Gotta warn yus guys today, with DSP, there are EQs which are NOT MP .. especially those which use FIRs and/or have Linear Phase. Such evil devices will make your Transient (Impulse) Response and Phase Response WORSE as you try to make the Amplitude Response better.

More details in
Simple Arbitrary IIRs

A mic would have to have impulse response taken at various levels, on axis, off axis, etc, and then have some interpolation scheme that would at best be an approximation, in order to have some sort of *rough* mathematical model.

Calrec Soundfields (and my 21st century efforts) have response measured in 10 directions in production test and many more for experimental prototypes. Interpolation is only needed above circa 10kHz. There's a couple of articles on Spherical Harmonics & Tetrahedral mikes that explain the limits. And they are all MP

Textbook examples of minimum phase devices vs non, immediately reveal several common sub-systems in microphones to not be minimum phase by definition.

Wanna tell us what they are? You'll note I often say "necessary but not sufficient" in my pontificating I've investigated many of these conditions but it's possible I've missed some