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16th May 2018, 06:27 AM  #121  
diyAudio Member
Join Date: Nov 2010

Quote:
Some kind of extension of FRD with r, theta, phi data. Anyone know details of the HolmImpulse format that Earl uses? He hasn't answered my last few questions. Best wishes David 

16th May 2018, 07:37 AM  #122  
diyAudio Member
Join Date: Jan 2014

Quote:
Code:
# First sample number in file: 10000 # Last sample number in file: 10000 # Samplerate: 48000 # ## sample;0 (Import: woofer_mp.frd) ;1 (Import: cd_mp.frd) 10000;5.6496302941e007;2.2383797616e005 9999;5.77869767557e007;2.21340868047e005 9998;5.37885862763e007;2.2304279733e005 I think ARTA has an automated polar measurement thing, see AP9 in ARTA Support 

16th May 2018, 04:25 PM  #123 
diyAudio Member
Join Date: Dec 2004
Location: Novi, Michigan

I have never used a USB microphone so I can't comment. I do use a Behringer USB sound card and have only fleeting trouble with time sync. It can completely loose sync at times, but 80% of the time it is fine. I have never seen a slow drift in time sync with my card.
It seems to me that 400 points is an absurd goal. I do full horizontal polars with 13 points, doing verticals would add only about another 10. A full spherical scan should be doable with < 50 points.
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16th May 2018, 04:31 PM  #124 
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Join Date: Dec 2004
Location: Novi, Michigan

I import Holm impulses. The windowing should be done in the analysis software NOT in the data acquisition software. What is shown above is correct, except that 20000 samples is almost 10 times too many.
I once saw Holm drop a few samples  only once  so my software checks the sample numbers for continuity. Have not seen it happen again in hundreds of runs. But it was catastrophic when it happened.
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16th May 2018, 09:04 PM  #125 
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Join Date: Sep 2010
Location: Amsterdam

Hi Guys,
I don't have any experience with octave, but with Matlab you certainly can do everything that' s needed. I wrote my own measurement suite (like holm, but with more specialized things for my work) in matlab, it' s relatively easy to make the mathematic side of the things in matlab (much more easy than in python or C# etc.), only a nice interface is a lot of work. Most users including myself use Matlab without interface, which is not too user friendly if you didn't make the code yourself. I think that the thing with octave is that you don't have nice ASIO sound card driver support etc, but I'm not sure, maybe someone has experience with this. I guess that octave is pretty much codecompatible with matlab as long as you don't use special functions from matlab (like libraries to communicate with databases), but again I am not sure. If the Matlab route is chosen (or octave is compatible enough), I will gladly donate the part of my code which does the actual measurement (through swept sine method just like holm/rew), as I like this project. I could also help implement the math in matlab, but I would need some pseudo code description as I am not at all at home in the mentioned math, and don't have the time to get into this. cheers, Kees 
17th May 2018, 01:05 AM  #126 
diyAudio Member
Join Date: Dec 2004
Location: Novi, Michigan

My 2 cents is that the data taking and the analysis have to be two separate implementations/ apps. The user must be free to use whatever hardware and software they deem satisfactory to get the N impulse responses. Then the analysis software must read these N signals and compute the best possible fit to whatever points the user chooses. I can do this now with my software so its perfectly doable. Once you have the analysis code it makes no difference to the code how many points or where they are. It just gives you the best fit that it can get from the data supplied. Some point locations and numbers will yield better results than others, so some work on "optimization" (like I did, should be done.)
I use this convergence of results all the time to find the number of modes required. When the results cease to change as N is increased it means that it has converged. You don't need any more points than modes as its a linear problem of N unknowns with N data points. I find 1215 points is this convergence at 10 kHz. If we add M points around the axis then we have M x N points and potential modes, but M would never be more that say 6 at best, so 6 x 15 = 90 is the maximum points that I could ever see in even the worst possible case. I would bet its a lot lower than that.
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17th May 2018, 02:43 AM  #127  
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Join Date: Nov 2010

Quote:
I have taken Klippel's numbers to provide an order estimate. Their poster and presentation both show >1000 points for maximum details, between 1000 and 100 for accurate work. The presentation shows two case studies. One is 4000 points, presumably as an extreme example. The second is 500 points. So 400 points doesn't look absurd to me, at ~10 seconds/data point that's about an hour measurement, not absurd for time. Whether it's absurd for other reasons I don't yet know. Until I have it clear in my mind I am inclined to assume Klippel have had some experience and their numbers are in the ballpark. Quote:
So we should be OK. Quote:
Best wishes David Last edited by Dave Zan; 17th May 2018 at 02:46 AM. 

17th May 2018, 03:39 AM  #128  
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Join Date: Nov 2010

Quote:
I am currently at work on this but it's one of the bits that's not clear to me. Quote:
I assume that resolution in both axes should be similar and the limited literature I have read usually takes M about the same as N. So I think more like 15 x 15, I took N to be 20, hence my 400 ballpark number. That's to check that measurement time is feasible, practicality of file formats and so on. As I noted in the previous post, it's more or less in line with Klippel's numbers so I expect it's not too unreasonable but I don't have the details clear. Have you looked at the Klippel presentations? Why do you think their number is so much more than your bet? Best wishes David Last edited by Dave Zan; 17th May 2018 at 03:42 AM. 

17th May 2018, 04:31 AM  #129  
diyAudio Member
Join Date: Apr 2010

Quote:
The reason the acoustic timing reference works properly with e.g. MSO is that all complex summations in that software are done with the mic in the same position (assuming the project has been set up correctly). One cannot do complex summations of data taken at different measurement positions without error when using an acoustic timing reference in REW. In order to do such complex summations correctly when measuring with REW, one must use a true loopback timing reference. 

17th May 2018, 05:06 AM  #130 
diyAudio Member
Join Date: Feb 2003
Location: ..

not up to the math but I can search:
'python spherical harmonics' turns up SHTOOLS package Fortran/Python, and this page gives some gridding refs Implementation details  SHTOOLS  Tools for working with spherical harmonics 
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