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DIY 3D Speaker Scanner - the Mathematics and Everything Else

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NTK

NTK

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@Kessito From Erin's D&D 8c picture, it looks like the spin axis aligned with the speaker CG (center of gravity). It may be solely for stability and practicality reasons. CTA-2034 specifies its spin axis to be 0 - 5 cm behind the front baffle. I assume that would be closer to where the acoustic center is located.

Theoretically, it shouldn't matter. What matter is where we choose the expansion center in our calculations, and we can easily adjust (and readjust however many times to our hearts' content) that by shifting the origin of the coordinate system. No moving anything or re-measurement is required.

I am much slower than I had hoped (over-estimated myself again :facepalm:). I am going to try posting some code good enough for initial tests by this weekend.

D and D 8c.jpg


CTA-2034.PNG
 

dc655321

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I am much slower than I had hoped (over-estimated myself again :facepalm:). I am going to try posting some code good enough for initial tests by this weekend.

Don't you just hate it when day-jobs, life, etc gets in the way of play time?!? Ugh! ;-)
 

mwmkravchenko

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I am still (slowly, very busy times) progressing on the subject.
I have worked out a decent stable multichannel measurement application in matlab and have thought a lot about the best data structures.
I also have ordered a bunch of these in 1meter lengt: (3mm threaded strips)
View attachment 118364
they have very convinient 5.02mm hole spacing for very exact mic placement and calibrator source placement.
Now I am planning to build this:
View attachment 118370

2x 15 mics spcaced 50.7mm (10holes), I will use the31th channel for a reference mic and the 32th channel for timing/adda convertor reference

The mics will be spaced ca 20cm apart in the strips, the combi of the linear actuator for up/down movement of the whole array and the servo for rotation will make almost any position possible.

the strips with mics wil weight around 1kg, so very easy to position with low force motors

I will mount the mics inside the holes (have to drill the holes bit bigger) and have the strips pointed sideways to the DUT. This way there is only a 2mm wide, 5mm deep strip in the way for the sound to reach the second strip with mics, which is more "invisible" to the soundwaves at 20k than a 1/4 inch mic.
Top view of the strip with the mic in place:
View attachment 118371

I am planning to connect the analog differential mems mics directly to this: https://tascam.com/us/product/ml-32d/top 32channel in-out dante convertor that I alread own, just because dante is very convinient and I dont have to buy anything this way.

I jus came accross this very nice paper, which is highly relevant;
EDIT: link not working, updated link: https://drive.google.com/file/d/1JdUX7GXndH9DFTqxukMX2gebv0prMMQU/view?usp=sharing

One question that maybe amir or someone else who has seen the klippel in action can answer, does the klippel scan the cylinder according to A or B on the picture below? (top view/ DUT seen from above)

View attachment 118369
@NTK; once you find the time to make a a bit more progression with the Matlab ( I am especially lost on the lms coefficient fitting part) I wil start doing simulations in matlab. The quoted articel has a lot of good matlab code examples at the end, I have to look into this to see if it is usable.
though my progress is slow, I am highly enjoying the thought gymnastics in this project :)

regards,
Kees

If I am thinking about the measurement methods that we use here. We reference off of the front baffle and use this as the zero plane of reference.
Those strips are kind of interesting. I have thought also about another potential route to go with your Mems mics idea. Aluminum backed PCB is a solution to both holding microphones in the correct places and setting up the wiring as PCB traces. Used on edge in the shape of arc segments they would have a very tiny diffraction profile. Also make wiring up anything a rather trivial task. Mems mics are almost all SMT and that is not so easy to work with if you don't have a PCB.

Secondary idea is an array of microphone capsules. PRIMO make very high quality capsules. High enough quality for Earthworks to use them. And the capsules are pretty reasonable in price and very flat in response.
 

mwmkravchenko

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I forgot to mention. If any of you guys comes across a JAES paper that you want let me know. I am a member and I can get them for free.
 

Kessito

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Yes you are right about the smt mics being difficult to handle, but with a magnifying glass and some patience it can be done.

There are a couple of reasons the mems are more attractive than electret imo;
1- price, I have bought 100 pcs knowles ellen mics, costs 71 cents/ piece
2- size, far far smaller than a 1/4 capsule
3- consistency, +-1dB sensitivity spec (verified by me)
4- basically class1 freq resonse ( altough i do plan to cal every mic in the array for +-0.1dB , +- 0.5deg
5- response goes up to 40khz easily, so phase info is good at 20khz
6- durabillity, the new coming infineon model is ip58(a meter under water!), and can withstand 1000g(!) Try that with your electret ( accidentally drop is or bounce into it, whoops there goes your diaphragm..)
7- ease of backend; differential output, so direct in to a line input of a decent ad covertor, no ultra high impedance ( and thus very sensitive for noise) circuity nessecary) input inpedance must be >=10kohm though, so no mic pre ( which is usually 6k8 or 3k4 because of phantom power load resistors)

I have a lot of experience with electrets and real condensers; mems makes me happy ;)

I am pretty confident that electrets will slowly fade out, mems is just so much better, there are slowly emergibg class one spl meters, the only reason why there isnt a real class compliant spl meter available is because the iec/ansi standard demands electrical signal inserted directly on the preamp with a dummy capacative load, which cannot be done with a mems, so the standards need to be updated.
The thing with mems; you need more sensitivity/ lower noisefloor? -> just add more mics in parallel.
I have made an environmental sound measure station with 4 mems mics in the tip of a 1/2inch mic; 2 in parallel en 2 channels so you can verify the performance easily ( is there a difference between the channels?). This is something you just cannot do with electrets because its too large an too expensive

Of course putting it on a pcb would be the best solution, I fully agree, but for now we dont no what tthe best config would be and I want to be able to play aroud with it ( maybe make a square array or whatever I come up with in the future)
Also; making pcbs is ( big) money-out-of-pocket,
Something I want to minimize at the moment ( thankyou, covid)
Once we have a finalized design for the array we can do a group buy for a pcb version ( and then have some preamplification, psu and maybe even ad-> dante or usb onboard, that would be awesome
 
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Kessito

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@Kessito From Erin's D&D 8c picture, it looks like the spin axis aligned with the speaker CG (center of gravity). It may be solely for stability and practicality reasons. CTA-2034 specifies its spin axis to be 0 - 5 cm behind the front baffle. I assume that would be closer to where the acoustic center is located.

Theoretically, it shouldn't matter. What matter is where we choose the expansion center in our calculations, and we can easily adjust (and readjust however many times to our hearts' content) that by shifting the origin of the coordinate system. No moving anything or re-measurement is required.

I am much slower than I had hoped (over-estimated myself again :facepalm:). I am going to try posting some code good enough for initial tests by this weekend.

View attachment 118427

View attachment 118428
Nobody's blaming you anything, we all know how it is!
Please do check the paper I linked ( the google drive link) there is a lot of working code in the last chapters which maybe easy for you to do some copy pasting.
 
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mwmkravchenko

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Yes you are right about the smt mics being difficult to handle, but with a magnifying glass and some patience it can be done.

There are a couple of reasons the mems are more attractive than electret imo;
1- price, I have bought 100 pcs knowles ellen mics, costs 71 cents/ piece
2- size, far far smaller than a 1/4 capsule
3- consistency, +-1dB sensitivity spec (verified by me)
4- basically class1 freq resonse ( altough i do plan to cal every mic in the array for +-0.1dB , +- 0.5deg
5- response goes up to 40khz easily, so phase info is good at 20khz
6- durabillity, the new coming infineon model is ip58(a meter under water!), and can withstand 1000g(!) Try that with your electret ( accidentally drop is or bounce into it, whoops there goes your diaphragm..)
7- ease of backend; differential output, so direct in to a line input of a decent ad covertor, no ultra high impedance ( and thus very sensitive for noise) circuity nessecary) input inpedance must be >=10kohm though, so no mic pre ( which is usually 6k8 or 3k4 because of phantom power load resistors)

I have a lot of experience with electrets and real condensers; mems makes me happy ;)

I am pretty confident that electrets will slowly fade out, mems is just so much better, there are slowly emergibg class one spl meters, the only reason why there isnt a real class compliant spl meter available is because the iec/ansi standard demands electrical signal inserted directly on the preamp with a dummy capacative load, which cannot be done with a mems, so the standards need to be updated.
The thing with mems; you need more sensitivity/ lower noisefloor? -> just add more mics in parallel.
I have made an environmental sound measure station with 4 mems mics in the tip of a 1/2inch mic; 2 in parallel en 2 channels so you can verify the performance easily ( is there a difference between the channels?). This is something you just cannot do with electrets because its too large an too expensive

Of course putting it on a pcb would be the best solution, I fully agree, but for now we dont no what tthe best config would be and I want to be able to play aroud with it ( maybe make a square array or whatever I come up with in the future)
Also; making pcbs is ( big) money-out-of-pocket,
Something I want to minimize at the moment ( thankyou, covid)
Once we have a finalized design for the array we can do a group buy for a pcb version ( and then have some preamplification, psu and maybe even ad-> dante or usb onboard, that would be awesome


I am learning more about modern MEMS mics. I looked a few years ago and decided that they were not fully baked. Thanks for the instruction. Looks like a viable option especially if they are calibrated. I think you have the means and so do I.

As for PCB's JLCPCB is the place to go. Fast and very high quality with a very low cost.

Which Knowles MEMS are you suggesting? And I am wondering what amplification if any is required to interface with them. TIme to hunt down some data sheets and application notes! And Infineon to! I will check them out. I am guessing the the little entrance to the MEMS unit will be the direction they have to be pointed at the Loudspeaker under test I am guessing.
 

mwmkravchenko

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@Kessito From Erin's D&D 8c picture, it looks like the spin axis aligned with the speaker CG (center of gravity). It may be solely for stability and practicality reasons. CTA-2034 specifies its spin axis to be 0 - 5 cm behind the front baffle. I assume that would be closer to where the acoustic center is located.

Theoretically, it shouldn't matter. What matter is where we choose the expansion center in our calculations, and we can easily adjust (and readjust however many times to our hearts' content) that by shifting the origin of the coordinate system. No moving anything or re-measurement is required.

I am much slower than I had hoped (over-estimated myself again :facepalm:). I am going to try posting some code good enough for initial tests by this weekend.

View attachment 118427

View attachment 118428
I am still astonished you are this far along in this endeavour. Take the time you need to keep it fun.
 

Kessito

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I am learning more about modern MEMS mics. I looked a few years ago and decided that they were not fully baked. Thanks for the instruction. Looks like a viable option especially if they are calibrated. I think you have the means and so do I.

As for PCB's JLCPCB is the place to go. Fast and very high quality with a very low cost.

Which Knowles MEMS are you suggesting? And I am wondering what amplification if any is required to interface with them. TIme to hunt down some data sheets and application notes! And Infineon to! I will check them out. I am guessing the the little entrance to the MEMS unit will be the direction they have to be pointed at the Loudspeaker under test I am guessing.

Yeah actually you had me thinking bout making a pcb at jcpcb ; a strip of 350*6mm 4 layer cost only a couple of dollars if you but 50pcs. Makes a big difference in all the wiring that you dont need to do plus you can add a gain buffer and decoupling cap.

Is there someone here who can draw a pcb?
(if not, I know some good guys, but it wil cost me i owe you's ,)



I have these mics atm: https://www.knowles.com/docs/defaul...000940c19.pdf?Status=Master&sfvrsn=1a3871b1_0

And this is the new infineon which is highly interesting: https://www.infineon.com/dgdl/Infin...N.pdf?fileId=5546d46276fb756a01771566ff816022

You dont need to point the hole at the source; it is 0.6mm, 90 degree of axis wil be almost 0 at 24kHz, I feel that the narrow barrier of having the pcb sideways is more important for the row of mics behind the first row.
 

mwmkravchenko

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Yeah actually you had me thinking bout making a pcb at jcpcb ; a strip of 350*6mm 4 layer cost only a couple of dollars if you but 50pcs. Makes a big difference in all the wiring that you dont need to do plus you can add a gain buffer and decoupling cap.

Is there someone here who can draw a pcb?
(if not, I know some good guys, but it wil cost me i owe you's ,)



I have these mics atm: https://www.knowles.com/docs/defaul...000940c19.pdf?Status=Master&sfvrsn=1a3871b1_0

And this is the new infineon which is highly interesting: https://www.infineon.com/dgdl/Infin...N.pdf?fileId=5546d46276fb756a01771566ff816022

You dont need to point the hole at the source; it is 0.6mm, 90 degree of axis wil be almost 0 at 24kHz, I feel that the narrow barrier of having the pcb sideways is more important for the row of mics behind the first row.


Wholy crap bat man! They are tiny! I see your point on the doesn't really matter which way they are pointed. See what happens in Semiconductor design when you are 5 or 6 years behind the latest! I am agreeing with a very loud nod :) goodbye Primo! And having a multi mic array will be the best in terms of consistency and speed of doing a measurement.

I have been gathering a research file since I found this thread again (Thanks DDF Dave) I may have a few papers that are useful.

As for PCB I can ask around. They will make from even a sketch at JLC PCB. ( https://jlcpcb.com ) They also have a component sister company ( https://lcsc.com/?href=jlc-home&source=jlc ) and a assembly company. ( https://jlcpcb.com/smt-assembly ) Last time I check for some DSP boards it was to low in cost for me to abuse my eyeballs. And to think in the late 90's I was able to read off any IC or Transistor package information with no problems. Not any more! No affiliation to any of them. Just like what they do. And love their prices.

I have to find a multiplexer like the one you have. But with a higher sampling rate if I can. Plus I have to get my little head around the output of the mics. Digital or is it analog? I will read the App notes and learn some more. Thanks for the links.

Many years ago I was a decent electronics designer for audio purposes. Now I am dangerous enough to ask questions. So many years of not using it every day. It's amazing to me what you loose. I spend a lot of time designing drivers and that in itself is a huge filed of intense research for me. Why for I am so excited about this project.

Something just hit me for a good doh moment. Flexible PCB. If it is properly backed they are robust enough. I did a boat load of AMT design work 5 and 6 years ago. Part of that was chasing and then rejecting Kapton as a useable material. To much stiffness and breask up problems. Little note for anyone interested. Mundorf is the only company that actually uses Kapton or a competitive poly-amide. If it's yellow it's a Polyester. Reading one of the Knowles app notes sparked the idea. Either way. It is an option that may or may not have utility. I personally like the idea of a rigid self supporting PCB. Can be done in a number of ways. Either with a thicker conventional PCB or an aluminium backed one.
 
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Kessito

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Wholy crap bat man! They are tiny! I see your point on the doesn't really matter which way they are pointed. See what happens in Semiconductor design when you are 5 or 6 years behind the latest! I am agreeing with a very loud nod :) goodbye Primo! And having a multi mic array will be the best in terms of consistency and speed of doing a measurement.

I have been gathering a research file since I found this thread again (Thanks DDF Dave) I may have a few papers that are useful.

As for PCB I can ask around. They will make from even a sketch at JLC PCB. ( https://jlcpcb.com ) They also have a component sister company ( https://lcsc.com/?href=jlc-home&source=jlc ) and a assembly company. ( https://jlcpcb.com/smt-assembly ) Last time I check for some DSP boards it was to low in cost for me to abuse my eyeballs. And to think in the late 90's I was able to read off any IC or Transistor package information with no problems. Not any more! No affiliation to any of them. Just like what they do. And love their prices.

I have to find a multiplexer like the one you have. But with a higher sampling rate if I can. Plus I have to get my little head around the output of the mics. Digital or is it analog? I will read the App notes and learn some more. Thanks for the links.

Many years ago I was a decent electronics designer for audio purposes. Now I am dangerous enough to ask questions. So many years of not using it every day. It's amazing to me what you loose. I spend a lot of time designing drivers and that in itself is a huge filed of intense research for me. Why for I am so excited about this project.

Something just hit me for a good doh moment. Flexible PCB. If it is properly backed they are robust enough. I did a boat load of AMT design work 5 and 6 years ago. Part of that was chasing and then rejecting Kapton as a useable material. To much stiffness and breask up problems. Little note for anyone interested. Mundorf is the only company that actually uses Kapton or a competitive poly-amide. If it's yellow it's a Polyester. Reading one of the Knowles app notes sparked the idea. Either way. It is an option that may or may not have utility. I personally like the idea of a rigid self supporting PCB. Can be done in a number of ways. Either with a thicker conventional PCB or an aluminium backed one.

Yup tiny they are, we have to thank the 50biljon dollar wearables market for the awesome developments taking place.
In hear you regarding reading components; I used to solder 144pin multi connectors inside fx racks in dimly lit clubs ( have been in livesound a long time and also had some electronics education), my eyes are still ok, but sometimes I hear my dads voice coming out of my mouth; "why is there no decent lightning at this desk!@#@%@" :)
This is a good reason to have some pcbs made; I dropped 2 mics on the floor in a clumsy moment, never found them back, and my office isnt too messy.
The mics I linked are analog, differential output.
There is also the option of Pulse Density Modulation output, which has both advantages ( more immunity for rf noise) and disadvantages ( less general to interface).
I started of with the idea to use the minidsp mchstreamer, which is 16x pdm to usb. But I discarded this because;
1- I want 32 channels ( or even 64) , having multiple mchstreamers on one pc is possible with some jingling, but you loose absolute timing because of different jitter between the 2 boards, bad news if you are going to do this stuff which highly depends on phase info with it
2-i want a timing reference/filter and phase reference loopback signal, this is not possible witj the pdm
3- the above is theoretically possible with an xmos or dsp, but I have not found oftheshelf dev kits which can do this, and I am certainly not going to build a dsp board

I think that It would be nice to make a cheap pcb strip ( 400x6mm) with 4 analog mics, opamp buffer with 20dB gain, and power distribution/ dc decoupling. This will then output 4 balanced line level channels which can directly be used with any resonable decent audio interface.
maybe with an rj45 as connector on the pcb, because then everyone can use cheap oftheshelf shielded twisted pair cable ( cat 6/7) which works fine with balanced line signals.
Come up with a good system to interconnect x strips in different ways/ configurations
A strip with 4 channels will cost+- 20usd, which is excellent in my book.
I will use the tascam because I already own it and because it is good qualiity and highly reliable (good japanes engineering). You dont need esoteric stuff for good measurements but you do need stuff that doesnt drift with temperature ( I owned several motu stuff in the past that drifted more than a dB (!) when warm. I trashed it and threw it away ( really pissed of because it costed me 2 days of debugging) . Then I bought RME and never looked back.
The good thing with the dante stuff is that it is highly scalable, up to 512x512 channels.
Ps, you need higher than 96k samplerate? For what reason?
Cheers,
Kees
 
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mwmkravchenko

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Yup tiny they are, we have to thank the 50biljon dollar wearables market for the awesome developments taking place.
In hear you regarding reading components; I used to solder 144pin multi connectors inside fx racks in dimly lit clubs ( have been in lifesound a long time and also had some electronics education), my eyes are still ok, but sometimes I hear my dads voice coming out of my mouth; "why is there no decent lightning at this desk!@#@%@" :)
This is a good reason to have some pcbs made; I dropped 2 mics on the floor in a clumsy moment, never found them back, and my office isnt too messy.
The mics I linked are analog, differential output.
There is also the option of Pulse Density Modulation output, which has both advantages ( more immunity for rf noise) and disadvantages ( less general to interface).
I started of with the idea to use the minidsp mchstreamer, which is 16x pdm to usb. But I discarded this because;
1- I want 32 channels ( or even 64) , having multiple mchstreamers on one pc is possible with some jingling, but you loose absolute timing because of different jitter between the 2 boards, bad news if you are going to do this stuff which highly depends on phase info with it
2-i want a timing reference/filter and phase reference loopback signal, this is not possible witj the pdm
3- the above is theoretically possible with an xmos or dsp, but I have not found oftheshelf dev kits which can do this, and I am certainly not going to build a dsp board

I think that It would be nice to make a cheap pcb strip ( 400x6mm) with 4 analog mics, opamp buffer with 20dB gain, and power distribution/ dc decoupling. This will then output 4 balanced line level channels which can directly be used with any resonable decent audio interface.
maybe with an rj45 as connector on the pcb, because then everyone can use cheap oftheshelf shielded twisted pair cable ( cat 6/7) which works fine with balanced line signals.
Come up with a good system to interconnect x strips in different ways/ configurations
A strip with 4 channels will cost+- 20usd, which is excellent in my book.
I will use the tascam because I already own it and because it is good qualiity and highly reliable (good japanes engineering). You dont need esoteric stuff for good measurements but you donmeed stuff that doesnt drift with temperature ( I owned several motu stuf in the past that drifted more than a dB (!) When warm. I trashed it and threw it away ( really pissed of because it costed me 2 days of debugging) . Then I bought RME and never looked back.
The good thing with the dante stuff is that it is highly scalable, up to 512x512 channels.
Ps, you need higher than 96k samplerate? For what reason?
Cheers,
Kees


I agree with what you are saying here. I missed the 96 kilohertz. This is enough. Basically you want to go above and beyond what you are measuring in order to be confident in the measurements. Lots of tweeters are doing very strange things to get out to 20 kilohertz. Resonance being the most common go to method. And seeing what is happening in the next octave means 44.5 kilohertz bandwidth. I have limited experience with Multiple channel interfaces. What would you recommend?
 

Kessito

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I agree with what you are saying here. I missed the 96 kilohertz. This is enough. Basically you want to go above and beyond what you are measuring in order to be confident in the measurements. Lots of tweeters are doing very strange things to get out to 20 kilohertz. Resonance being the most common go to method. And seeing what is happening in the next octave means 44.5 kilohertz bandwidth. I have limited experience with Multiple channel interfaces. What would you recommend?

Depends a little on what platform you are on (desktop or laptop ), the flexibillity that you would like to have, and the scalabillity and absolute quality.
the devices with best (by a great margin) quality drivers (so the software drivers that make your device communicate with your software) are without any doubt RME, you will connect this and never think about it again. Also theredigital clocks and jitter rejection are really good

You could choose i.e. an RME 8x adat interface https://www.rme-audio.de/digiface-usb.html (you could even connect several of these and they will be seen as one big onterface by the driver) and then connect several ad / da convertors in the configuration that you need (i.e. 8x analog out, 32x analog in, 8x AES EBU inout etc.)
The big advantage here is that you can use relatively cheap convertors for the non- ultra critical stuff regarding distortion and noisefloor (like these array measurments), and ultra high end stuff (octo dac etc.) for high quality distortion measurements and critical listening and xover listen tests etc.

OR you could go full DANTE network: run DANTE virtual audio card on your pc/mac:
https://www.audinate.com/products/software/dante-virtual-soundcard (costs 30dollar)
and then you open up a world of possible devices that will all be tight synchronized on very cheap standard ethernet infrastructure, this is what the pro-audio world does nowadays and what I am doing whith the tascam unit.
The big advantage here is scalabillity and connectivity.
The disadvantage is price (but a lot of cheaper stuf being released lately) and latency (5ms with virtual soundcard) but latency is irrellevant if you only measure with it
Check the dante site for all possible products that you can hook up:
https://www.audinate.com/products/dante-enabled (check in the category interfaces)
 

mwmkravchenko

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Ok I guess I will show my ignorance out front and centre. How can a box from RME that appears to have 8 channels, 4 in and 4 out be said to have 66? Do you have to daisy chain these things? Oh $700 Canuckistan clams.
 

somebodyelse

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Ok I guess I will show my ignorance out front and centre. How can a box from RME that appears to have 8 channels, 4 in and 4 out be said to have 66? Do you have to daisy chain these things? Oh $700 Canuckistan clams.
The digital ins and outs can be used for ADAT which can do 8 channels each, or SPDIF which can do 2. If you use them all for ADAT that's 32 in, 32 out. Add the 2 channels from the analog headphone out and you have 66 channels.
 

Kessito

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BOOM:
https://www.ak.tu-berlin.de/menue/publications/open_research_tools/aktools/

TU Berlin has a MATLAB toolbox which has almost ALL needed functionality onboard!!!

INCLUDING the transformating from magnitude/phase data to the spherical harmonics and back!! (basically you can already do everything but thesoundfield seperation)

this means that the only piece NTK has to add (modify some of these functions) is the part where the secondary soundfield spherical harmonics are created and subtracted from the direct part!

And the good news: this is all highly verfied/documented open source/open use university research code

SUPER EXCITED WITH TIS FIND!!!!!! :)

@NTK: no excuses left now for you ;)
 
OP
NTK

NTK

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BOOM:
https://www.ak.tu-berlin.de/menue/publications/open_research_tools/aktools/

TU Berlin has a MATLAB toolbox which has almost ALL needed functionality onboard!!!

INCLUDING the transformating from magnitude/phase data to the spherical harmonics and back!! (basically you can already do everything but thesoundfield seperation)

this means that the only piece NTK has to add (modify some of these functions) is the part where the secondary soundfield spherical harmonics are created and subtracted from the direct part!

And the good news: this is all highly verfied/documented open source/open use university research code

SUPER EXCITED WITH TIS FIND!!!!!! :)

@NTK: no excuses left now for you ;)
Dang :D
 
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