Before all, please, note : some audiophiles may not prefer to parallel microphones because, although made by the same manufacturer and processes, there are tolerances which may lead to slight non linearity. I hope this is negligible.
I have made a multiple microphone with sixteen Electret Condenser Microphone capsules.
I still wait for a 500K dual potentiometer which has been ordered. The device was tested without such with direct wire connections instead.
The document is not finished, there are more explanations to be made and pictures, which would probably be made once the dual potentiometer is installed as well as labels.
The document will be updated.
I would like to publish this document now, although not fully completed, because the project has been an obsession and a dream which I have been able to make a reality.
Here is the document :
Multiple Microphones - Google Drive
I have made a multiple microphone with sixteen Electret Condenser Microphone capsules.
I still wait for a 500K dual potentiometer which has been ordered. The device was tested without such with direct wire connections instead.
The document is not finished, there are more explanations to be made and pictures, which would probably be made once the dual potentiometer is installed as well as labels.
The document will be updated.
I would like to publish this document now, although not fully completed, because the project has been an obsession and a dream which I have been able to make a reality.
Here is the document :
Multiple Microphones - Google Drive
Here is a best introduction to microphone preamplifier design:
http://www.thatcorp.com/datashts/AES129_Designing_Mic_Preamps.pdf
http://www.thatcorp.com/datashts/AES129_Designing_Mic_Preamps.pdf
The joint Texas Instrument Application note is good reading.
A good point on parallelling 16 microphones is about noise.
This gives a better Signal / Noise ratio.
Signal ( correlated ) increases x 16 while noise ( un-correlated ) increases x 4.
So you get S/N x4 ( +12dB ).
This is only true for pressure waves in front, for pressure waves angled the signals don't simply add, they are phase shifted an effect which is more at higher frequencies.
16 amplifiers would make no difference. These behave like current sources, when paralleled they add as would do adding 16 amplifiers outputs.
The Op-amp used by Texas Instrument is perfect for this electret mic, as calculated in their detailed AN. Even better from the noise point of view whith 16 // electrets.
Acoustic of 16 //. The pick up pattern will be peculiar, with many lobes in the high frequencies.
You can experiment with various geometries for various effects.
The mics in line, close together.
The mic in line, spaced.
The mics closely grouped 4 x 4
The mics as close as possible. Hexagonal, honey comb like.
You should get strong directive effects. Have fun.
A good point on parallelling 16 microphones is about noise.
This gives a better Signal / Noise ratio.
Signal ( correlated ) increases x 16 while noise ( un-correlated ) increases x 4.
So you get S/N x4 ( +12dB ).
This is only true for pressure waves in front, for pressure waves angled the signals don't simply add, they are phase shifted an effect which is more at higher frequencies.
16 amplifiers would make no difference. These behave like current sources, when paralleled they add as would do adding 16 amplifiers outputs.
The Op-amp used by Texas Instrument is perfect for this electret mic, as calculated in their detailed AN. Even better from the noise point of view whith 16 // electrets.
Acoustic of 16 //. The pick up pattern will be peculiar, with many lobes in the high frequencies.
You can experiment with various geometries for various effects.
The mics in line, close together.
The mic in line, spaced.
The mics closely grouped 4 x 4
The mics as close as possible. Hexagonal, honey comb like.
You should get strong directive effects. Have fun.
I shall try to make an answer to all concerns made before this answer.
1. An updated, still, not ready file has been uploaded.
2. I still wait for the dual 500K potentiometer.
3. Paralleling microphones increase sensitivity and signal to noise ratio. Such sound may not be available from a single microphone with high gain.
4. The problem with paralleling any microphones, say, professional ones, is the microphones are not exactly the same which affects the non linearity error. However, when two microphones are played on two different stereo channels, any difference between their phases will move the sound in a 3D way.
5. This project does NOT need separate mixer, because, the microphone output is already added ( mixed ) by their parallel configuration. However, a better way may be to buffer each microphone with a buffer amplifier and then to add their voltages with an adder. Thus, 17 amplifiers are needed to achieve this. In this project, the transistors ( microphones are buffered by a JFET internally ) are just paralleled and the current of each is added to make a general current which is then converted to a voltage by a current to voltage converter, a. k. a. a transconductance or transimpedance amplifier.
6. This microphone has a built in preamplifier and can be directly connected to an amplifier input or an amplifier microphone input or a PC. Phantom power is not necessary. Any phantom power provided by a microphone phantom power input is ignored by the output capacitors. This microphone has three outputs, two of which are the same, just provide a different socket :
A. 1 / 4 inch mono output of the preamplifier
B. 3.5mm dual channel mono ( stereo with the same channels ) output, the same as A.
C. Output directly from the microphones without any preamplifier ( phantom, power is not necessary and will be ignored when present ). Please, note, the switch which switches the output capacitors on or off must be in on ( 30uF ) position.
I hope this answers the questions.
1. An updated, still, not ready file has been uploaded.
2. I still wait for the dual 500K potentiometer.
3. Paralleling microphones increase sensitivity and signal to noise ratio. Such sound may not be available from a single microphone with high gain.
4. The problem with paralleling any microphones, say, professional ones, is the microphones are not exactly the same which affects the non linearity error. However, when two microphones are played on two different stereo channels, any difference between their phases will move the sound in a 3D way.
5. This project does NOT need separate mixer, because, the microphone output is already added ( mixed ) by their parallel configuration. However, a better way may be to buffer each microphone with a buffer amplifier and then to add their voltages with an adder. Thus, 17 amplifiers are needed to achieve this. In this project, the transistors ( microphones are buffered by a JFET internally ) are just paralleled and the current of each is added to make a general current which is then converted to a voltage by a current to voltage converter, a. k. a. a transconductance or transimpedance amplifier.
6. This microphone has a built in preamplifier and can be directly connected to an amplifier input or an amplifier microphone input or a PC. Phantom power is not necessary. Any phantom power provided by a microphone phantom power input is ignored by the output capacitors. This microphone has three outputs, two of which are the same, just provide a different socket :
A. 1 / 4 inch mono output of the preamplifier
B. 3.5mm dual channel mono ( stereo with the same channels ) output, the same as A.
C. Output directly from the microphones without any preamplifier ( phantom, power is not necessary and will be ignored when present ). Please, note, the switch which switches the output capacitors on or off must be in on ( 30uF ) position.
I hope this answers the questions.
A very important application is the ability of this microphone to amplify the sound so well, so, the microphone can be positioned anywhere and the speaker does not need to be in front of the microphone. This is excellent for Skype communication where the speaker can be anywhere in the room doing anything and the microphone can be anywhere and does not need to be close to the person.
Of course, when the microphone is connected to an amplifier with speakers, high gains may cause a microphone feedback. This is not a problem with a PC or headphone where there is not any speaker to " talk back " to the microphone and thus cause a microphone feedback.
For guitar and singing, the same applies : the microphone can be away from the singer and guitarist. One application may be where the guitar and the voice are picked by a single microphone which may be in the middle or closer to the quieter guitar to get the voice and the guitar equally.
Also, voice may be recorded at higher acoustic quality when the microphone is away from the singer.
Of course, when the microphone is connected to an amplifier with speakers, high gains may cause a microphone feedback. This is not a problem with a PC or headphone where there is not any speaker to " talk back " to the microphone and thus cause a microphone feedback.
For guitar and singing, the same applies : the microphone can be away from the singer and guitarist. One application may be where the guitar and the voice are picked by a single microphone which may be in the middle or closer to the quieter guitar to get the voice and the guitar equally.
Also, voice may be recorded at higher acoustic quality when the microphone is away from the singer.
Microphone arrays
325 microphones
Super-Microphone Picks Out Single Voice in a Crowded Stadium | WIRED
Another microphone array
Precision Acoustic Detection – Squarehead Technology – Squarehead Technology enables the use of precision audio capture of acoustical events. Filtering out and reducing unwanted noise from areas of no interest, Squarehead achieves better range and clearer sound compared to conventional systems. Squarehead is one of the leading solution providers for acoustic drone detection.
325 microphones
Super-Microphone Picks Out Single Voice in a Crowded Stadium | WIRED
Another microphone array
Precision Acoustic Detection – Squarehead Technology – Squarehead Technology enables the use of precision audio capture of acoustical events. Filtering out and reducing unwanted noise from areas of no interest, Squarehead achieves better range and clearer sound compared to conventional systems. Squarehead is one of the leading solution providers for acoustic drone detection.
Sixteen 1/4" capsules in tight array is a 1" mike, which will be non-directional over the speech/Skype range.
Sixteen capsules spaced apart will of course beam and lobe.
A possible config which I have not seen is a vertical line array. Assuming it is "level" with the mouths, it will pickup mouth sound better than ambient room noise or reverberation. Off-plane it will lobe, but 2" spacing will not cause large errors in the speech band all-around on-plane, and the "near field" will extend several feet further into the room.
He's increasing signal more than hiss. Just a matter of more sail area. It "sounds louder" which is always "better". Simply gaining-up one capsule will gain-up the hiss in the same proportion, for no real advantage. (Anyway there is always a level control somewhere.)
The $1 capsules' hiss-level is low but not as low as an AKG 414 or similar. In my experience, the ambient noise in most rooms is similar to the cheap capsules, which is why they are popular for general pickup.
Using a fist-full of mikes seems un-thrifty in general engineering, but these capsule are a buck a piece so far cheaper per acre than other microphones.
That said, this is not a new problem and there are MUCH better solutions, if not so DIY-friendly. Space several mikes apart. Use a processor to compare the outputs in fine detail. You can determine the angles of the sound sources. You can delay the signals so that angle lines-up, sums, while other angles cancel. This technology is found in high-end cellphones and some PC microphones.
It's already been said, those systems are delaying each microphone with a dedicated dsp for each one to get them in phase and literally focus on one point. If you don't do that, you'll get a horrible comb filter effect which rips apart your upper frequency spectrum by the canceling effect of the phase difference. That means, without dsp'ing each channel you are fixed to just one 'focal point' and you have to adjust the array exactly, really precise. If the sound source moves (or the microphone because of a bit wobble of the mic stand or floor), you'll get phase problems and again the comb filter effect.
Your microphone mounting plate also changes the frequency response unless they are each exactly the same distance away from the edge because lower frequency sound 'goes around' objects up to a certain size.
I don't know what you are planning on doing with it but without going the extra mile (which is, in fact, one time across america in distance in comparison at this mic), you won't get the same result. I think you'll get in >99,9% of the cases a much better result with a low noise microphone in a parabolic reflector.
Your microphone mounting plate also changes the frequency response unless they are each exactly the same distance away from the edge because lower frequency sound 'goes around' objects up to a certain size.
I don't know what you are planning on doing with it but without going the extra mile (which is, in fact, one time across america in distance in comparison at this mic), you won't get the same result. I think you'll get in >99,9% of the cases a much better result with a low noise microphone in a parabolic reflector.
Every microphone is VERY SLIGHTLY different because of manufacturing tolerances. Thus, the response of every microphone is slightly different. Different parasitic capacitances and resistances, for example, will make different delays. Again, the difference is extremely tiny. Any microphone takes sound waves which are distributed everywhere in a room because of the path of the sound pressure wave ( air like a wind ) and because of reflection ( echo ). Thus, a powerful microphone would get the signal regardless on where positioned. Of course, there will be a difference in levels of the sound which goes directly into the microphone and from the back.
Every microphone capsule is omni directional. The overall result will be omni directional too as long as the MECHANICS ( mechanics of fluids ) of the sound wave can travel without stopped by capsules around. The MECHANICAL interference would be very tiny, negligible. The problem is not this. The problem is the omni directional microphones are NOT omni directional. They get the direct wave loudly and the non direct waves NOT SO quietly.
Again, the MECHANICAL hiss caused by the diffraction of the sound wave would be very tiny, probably, negligible. In this case, the capsules are NOT very tightly spaced. Even when spaced very tightly, the effect would also be very tiny, probably, negligible.
The effect of the tolerances has been mentioned. The other effects are supposed to be even lower. Again, the phase and other differences are very tiny, probably, negligible. Because the microphones are added, the overall effect is like talking to perfect microphones spaced probably micro or nanometers away. Again, the effect of the differences is supposed to be negligible. A better approach would be to use 16 buffers, one for each microphone, before the adder. The buffers are not expected to introduce significant differences, compared to the differences of the microphones.
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An updated document with pictures has been uploaded. The dual potentiometer has been installed. However, the power is so huge, even a slight increase in the volume creates a feedback. I am unable to test the microphone with a long wire and away from the amplifier. Two 9V batteries were used for a test which was OK. The microphone is extremely quiet. The sound response is OK. All this is tested by ears. I do not have scientific equipment for real tests. The potentiometers are not necessary for normal applications. Even without them, with the so calculated resistors, the sound is huge and no noise can be heard. The potentiometers may be used in a quiet environment to get low level sound, such as to listen to birds far away or to the neighbours next door.
This project is completely analogue without any compensation. The same can be done with individually measured microphones and with either digital ( preferably ) or analogue compensation for the differences. I have tried to answer some of the questions but I am unable to provide experiment data which requires sensitivity and accuracy.
The only thing I can say is when the microphone is plugged in an amplifier, there is no audible noise. When spoken into the microphone, there is a huge signal. Again, the microphone must be far away from the speaker to avoid feedback. I am unable to test the microphone more than 2 meters away from the speaker. I do NOT have equipment which has feedback cancellation. Attention must be paid, of course, not to saturate the amplifiers when the signal is too loud.
Again, the only test I have done is : plug the microphone into a non microphone input of an amplifier, see whether the gain can be increased. May only be slightly possible because of the feedback. No talk, no noise. Talk, excellent signal. As you can see, this is not a scientific test.
Also, as you can see, these are simple electret microphone capsules and not a $50 Panasonic. They are 10V max rated and not 48V. Those who want to see the effect of adding microphones can do a simple experiment : Take a few, say, four the same microphones, position them next to each other and plug them to an adder ( mixer ). Adjust the levels to be the same. Because of manufacturing tolerances, the microphones are not the same, yet, the difference is very tiny. Provided the mixer channels are exactly the same, the microphone differences will be the only differences.
Most likely, however, these differences cannot be observed ( heard ) because these differences are negligible. Most likely, accurate equipment may be possible to find any effect, although they is doubtful. In case any effect is observed, this effect would be the same as when four exactly the same microphones plugged to exactly the same mixer channels are positioned slightly differently in space, say, a few millimeters closer or away from the source of the sound. The same applies to the added signal of 16 microphone capsules. When the capsules are close to each other, the effect is lower. Again, this difference is negligible. When a rock band uses four microphones to sing, each of them would have more delayed or advanced signal then any of the others.
In the worst case, the phase differences would add the effect of 3D sound : each of the microphones would give the effect of a different position in space. When one person sings to the four microphones, the effect would be as four persons sing the same but in a different position in the room. Again, this is purely theoretical. The differences are negligible for a human to hear them and interpret them. And when other problems are added, such as echo, reflections, etcetera, the differences are even more insignificant. I have mentioned the effect from a theoretical point of view.
Practically, the differences cannot make a difference. Again, as mentioned, possible improvements may be :
1. To buffer every microphone and then add the signals.
2. To supply the microphones from a low noise DC voltage reference with good capacitors before and after the voltage reference. Buffered or amplifier low noise zeners can be used too.
Larger audio capacitors may be needed to reduce the Zener noise. Low noise zeners and voltage references do have their noise though. Because their output impedance is very low, large, extremely low ESR and ESL audio capacitors are a good idea. An array of capacitors with X7R's and other low ESR capacitors may give a better noise reduction. Some add low value resistors in series. This may not be a good idea.
A document, called Errata.doc has been published here : Errata - Google Drive
The document correct errors I may have made in various projects I have worked on.
The document is to be updated.
Please, report all errors and problems.
The document correct errors I may have made in various projects I have worked on.
The document is to be updated.
Please, report all errors and problems.
The document has been updater with improvements, error corrections and more explanations and ideas.
The new chapter is called Errata and Improvements. This is a different errata, only, for this document and not the one, posted in the previous post.
The link to the document is the same : https://drive.google.com/drive/folders/1vgZMFulGjXqXrpkz5W6OeFOUqL1NC7nk?usp=sharing
Please, after downloading the document, search the document with the word " Errata ". The search must bring you to a chapter name " Errata and Improvements ". All of the new stuff is in this chapter.
Thanks.
The new chapter is called Errata and Improvements. This is a different errata, only, for this document and not the one, posted in the previous post.
The link to the document is the same : https://drive.google.com/drive/folders/1vgZMFulGjXqXrpkz5W6OeFOUqL1NC7nk?usp=sharing
Please, after downloading the document, search the document with the word " Errata ". The search must bring you to a chapter name " Errata and Improvements ". All of the new stuff is in this chapter.
Thanks.