# miniDSP DDRC-22D Dirac Live® 24/96 Room Correction Audio Processor Review Discussion Thread



## AudiocRaver

*miniDSP DDRC-22D Dirac Live® 24/96 Room Correction Audio Processor Review Discussion Thread*​[img]http://www.hometheatershack.com/gallery/file.php?n=23178[/img]



miniDSP Website
DDRC-22D MSRP: $899 USD
Available Direct From miniDSP



*by Wayne Myers*

*Introduction*

The recently introduced DDRC-22D, DDRC-22A, and just announced DDRC-22DA are miniDSP's entries into the room correction market making use of Dirac Live room correction technology. The Dirac Live technology has been available for PC- and Mac-based systems for years from Dirac Research, founded in 2001. The technology is also licensed for use in cars, smart phones, and other audio products. It is a thoroughly implemented and robust technology which uses mixed-mode filters, making time and impulse response correction possible. The miniDSP entries are two channel models, standalone products which can be inserted in a home or professional system at various points in the signal chain to implement room correction. miniDSP is a Hong Kong based company specializing in digital signal processing products for audio applications.


*Description*

The DDRC-22D which I received for evaluation came boxed with a power supply, a UMIK-1 calibrated microphone with cable and accessories for room analysis, a nifty collapsible mic stand, two 6-foot long TOSlink cables, a USB cable for connecting the DDRC-22D to a PC running the Dirac Live software (only necessary during calibration), and a pair of rack-mount ears. All software is downloaded from the miniDSP web site.


*miniDSP DDRC-22D Dirac Live® 24/96 Room Correction Audio Processor Review.*​


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## Sonnie

Nice review Wayne... very thorough. :T

Mine is still in the box... gotta get it out soon and get it setup for two channel and see how it does on the Ultra Towers. They sound marvelous with XT32 right now, and I will be able to switch back and forth for a direct comparison. XT32 will be running via the HDMI output of my 105 and the DRC via digital cabling to a different input. Should be fun experimenting and comparing. :bigsmile:


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## AudiocRaver

This thread is now open for comment or discussion.


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## theJman

Incredible review Wayne! Detailed, articulate and well thought out.

One question... the NSM 5 speakers you refer to, are they from NSMT by chance? I've looked longingly at some of their speakers for a few years -- as well as the stuff from their sister company, Role Audio -- but the prices were always a little stiff for me. In particular, the Model 30M; I'm drawn to concentric drivers for some reason. If NSMT had a matching center it would have been difficult for me to fight that money-vs-want battle, but ultimately I went with the Bag End M6. Easier on the wallet, yet uses a driver that's virtually identical.


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## AudiocRaver

Yes, the company now goes by an NSMT, and the little model 5 has been discontinued. It was like the model 10 only smaller. I was really hungry for the role Role Kayak, at about twice the price, but the model 5 looked like it was the same design. That was my first speaker purchase, about 10 years ago now, specifically with imaging and sound stage in mind.


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## Flak

We think that the review was very well written and based on sound principles and know-how... and we are certainly very pleased by the fact that Wayne Myers recommends it (even enthusiastically) and says that "Dirac Live delivers, delivering great sonic results with minimum trouble at a reasonable cost from a company that stands ready to keep you happy"
Nonetheless, because we endeavour in seeking perfection in the digital-acoustic domain, we are not 100% happy with it as we believe (as hinted in the review) that given more time to experiment, and tweak it further, he could get even better results 

The Dirac impulse response optimization aims at overcoming the usual limitations of digital room correction systems and one of these limitations (that a well-designed impulse response optimization not only avoids but in fact can also improve over the original system) is the preciseness of the imaging.

The position and definition of a phantom source between the loudspeakers is determined by the degree of similarity of the sounds at our two ears... i.e. if the frequency responses of the left and right loudspeakers are different imaging gets poor and the exact position of a voice, for example, is hard to pinpoint. 

So the first step is to use the same target response for both speakers and that's where most room correction systems stop. 
Now, if the filters applied to the left and right speakers have different frequency responses, they also have different phase responses (or impulse responses) but unless the system actually optimizes the impulse response, or phase response, then these differences are not matched to the speakers' and room's problems and thus they are likely to deteriorate the imaging. 

But Dirac optimizes the phase response and impulse response to also make these crucial aspects better than the original system. 
The difficulty is to get this done not just in one point in space but in the entire listening area. 
In effect, this means that at high frequencies the phase response is not affected, because the room-reflection pattern at high frequencies is completely different just decimeters apart. The early parts of the impulse response should however be improved and so should the entire low-frequency range. 

In the lower voice region, there should be a clear improvement in imaging, manifested by a more well defined center voice position. 
At high frequencies, any difference would be mainly attributable to any remaining frequency response differences. 

We recommend that the microphone positions are spread out as "randomly" as possible within the measurement region and that the measurement region is not too small. The verification measurements, all excellent, show quite small differences between the 1 point and 9 point measurements, which indicate that the 9 point measurements were taken within a quite small region. If the region becomes too small, there is a risk of over-compensation. With a larger measurement region, more variations from the soundfield are captured, which typically yields a better correction (which may be contrary to many other systems) 

Another exercise would be to investigate the measured impulse responses to find any anomalies in the set-up. There are occasions when the algorithm has to make a compromise between imaging and reaching the target response, and there are unusual cases where you can't achieve both at the same time. The algorithm's first objective is getting the best possible frequency response and impulse response of each individual loudspeaker. If that conflicts with having similar impulse responses, the first objective is always prioritized. Typically, however, with some fiddling, for example re-measuring the speakers in a slightly different measurement pattern, or actually moving the loudspeakers a little, can give the algorithm enough flexibility to overcome such compromises.

In any case an excellent, indipendent, well informed and documented review,
nice job  Flavio


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## bkeeler10

Thanks Wayne for another thorough review. I've been interested in Dirac and have been curious about how it compares with other auto EQ solutions, including Audyssey. I do hope to hear how XT32 compares if you get the chance you are hoping to get.

I'm hoping some time soon HTS will get a hold of an Anthem MRX receiver and hand it over to you to take a look at their ARC solution.

Interesting comments also from Dirac, particularly in regard to random measurement positions over a sufficiently large area. I know that, in the case of Anthem's ARC, they recommend the following: "The first [position] must be at or just in front of the central seating position . . . positions 2 and 3 should be symmetric to the left and right of the center line, and the same applies to the remaining positions. If your room has less than five seating positions, measurements must still be taken from five positions at least 2 feet apart to ensure optimal sound."

My understanding from that and elaborations of people in the know is that, when Anthem says that positions should be two feet apart, they mean that no two measurement positions should have less than two feet of separation distance between them. 

I know this flies in the face of Wayne's experimentation and experience with Audyssey. So I find it quite interesting that Dirac also suggests a wider area of measurement. Of course, the secret sauce is different with each auto EQ solution so what is ideal for one may not be ideal for the others.


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## bkeeler10

Sonnie said:


> Nice review Wayne... very thorough. :T
> 
> Mine is still in the box... gotta get it out soon and get it setup for two channel and see how it does on the Ultra Towers. They sound marvelous with XT32 right now, and I will be able to switch back and forth for a direct comparison. XT32 will be running via the HDMI output of my 105 and the DRC via digital cabling to a different input. Should be fun experimenting and comparing. :bigsmile:


Please let us know about your experience with this too. :bigsmile:


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## Sonnie

I am definitely curious about mic placement. I know I have tried this with Audyssey in varying degrees but always come back to the single measurement spot for the best results. It is only critical for me for music, and my head is fairly stable in one position during music listening.


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## theJman

My Audyssey measurement process consists of 4 spots. The first is the PLP, as it always should be, then I do the secondary. I repeat that twice, so each of those seats get two passes. Seems a bit non-standard, in the grand scheme of things, but that's what works best for me. Since I live alone now there are only 2 seats being used when watching TV, so I didn't see the need to EQ at locations where no one ever sits. However, measuring once at each spot often resulted in sound that was a bit off to my ears. Measuring them twice helped even out the response I hear.


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## AudiocRaver

Flak said:


> We think that the review was very well written and based on sound principles and know-how... and we are certainly very pleased by the fact that Wayne Myers recommends it (even enthusiastically) and says that "Dirac Live delivers, delivering great sonic results with minimum trouble at a reasonable cost from a company that stands ready to keep you happy"
> Nonetheless, because we endeavour in seeking perfection in the digital-acoustic domain, we are not 100% happy with it as we believe (as hinted in the review) that given more time to experiment, and tweak it further, he could get even better results
> 
> The Dirac impulse response optimization aims at overcoming the usual limitations of digital room correction systems and one of these limitations (that a well-designed impulse response optimization not only avoids but in fact can also improve over the original system) is the preciseness of the imaging.
> 
> The position and definition of a phantom source between the loudspeakers is determined by the degree of similarity of the sounds at our two ears... i.e. if the frequency responses of the left and right loudspeakers are different imaging gets poor and the exact position of a voice, for example, is hard to pinpoint.
> 
> So the first step is to use the same target response for both speakers and that's where most room correction systems stop.
> Now, if the filters applied to the left and right speakers have different frequency responses, they also have different phase responses (or impulse responses) but unless the system actually optimizes the impulse response, or phase response, then these differences are not matched to the speakers' and room's problems and thus they are likely to deteriorate the imaging.
> 
> But Dirac optimizes the phase response and impulse response to also make these crucial aspects better than the original system.
> The difficulty is to get this done not just in one point in space but in the entire listening area.
> In effect, this means that at high frequencies the phase response is not affected, because the room-reflection pattern at high frequencies is completely different just decimeters apart. The early parts of the impulse response should however be improved and so should the entire low-frequency range.
> 
> In the lower voice region, there should be a clear improvement in imaging, manifested by a more well defined center voice position.
> At high frequencies, any difference would be mainly attributable to any remaining frequency response differences.
> 
> We recommend that the microphone positions are spread out as "randomly" as possible within the measurement region and that the measurement region is not too small. The verification measurements, all excellent, show quite small differences between the 1 point and 9 point measurements, which indicate that the 9 point measurements were taken within a quite small region. If the region becomes too small, there is a risk of over-compensation. With a larger measurement region, more variations from the soundfield are captured, which typically yields a better correction (which may be contrary to many other systems)
> 
> Another exercise would be to investigate the measured impulse responses to find any anomalies in the set-up. There are occasions when the algorithm has to make a compromise between imaging and reaching the target response, and there are unusual cases where you can't achieve both at the same time. The algorithm's first objective is getting the best possible frequency response and impulse response of each individual loudspeaker. If that conflicts with having similar impulse responses, the first objective is always prioritized. Typically, however, with some fiddling, for example re-measuring the speakers in a slightly different measurement pattern, or actually moving the loudspeakers a little, can give the algorithm enough flexibility to overcome such compromises.
> 
> In any case an excellent, indipendent, well informed and documented review,
> nice job  Flavio


Flavio,

Thank you for the detailed response, which of course is intended for all users and potential users, but I take it to heart as well. I wish I had a fraction of the experience and background that your team has in this fascinating area.

I was actually quite impressed by the way your comments illustrate precisely the leverage that a great DRC product like Dirac Live can offer a typical user. Your main suggestions (my interpretation): Try a larger mic pattern. And run several with variations to see which give the best result. The leverage is in potentially being able to get so much more out of a product with such simple operational changes.

Those who have experienced great soundstage and imaging will agree that any product that can get even close to them easily is a find. But, being the perfectionist as you are, you rightly insist that your product can get better than “close.” I appreciate your insistence on this point. As an aside, I am stunned to realize how few listeners have experienced that kind of sonic nirvana, and offer you this question, in jest, of course. How do they know when to stop? A matter for further contemplation, perhaps.

You advise that the wider, more randomly spaced calibration mic pattern can give better imaging results, and that this is contrary to what users might have experienced with other auto-correct products, all a result of the uniqueness of the Dirac Live algorithm. It appears that you are looking for a _ localized room sound_ in the vicinity of the LP, then let your algorithm work its wizardry to provide tighter imagery at the LP. We will simply have to trust that you have something special running “under the hood” to accomplish all this, and that it can consistently deliver a seamless soundstage presentation as well.

I am intrigued, partly from the angle that the best soundstage is usually made worse by DRC, although Dirac Live has already provided one data point to the contrary. Tony at miniDSP has generously allowed me to hang onto the DDRC-22D to experiment with. I will perform a more in-depth study of mic patterns & spacings and post my findings.


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## tesseract

Good stuff, Wayne. I am looking forward to hearing more about Dirac and how it compares to other DRC processors.


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## AudiocRaver

Sonnie said:


> I am definitely curious about mic placement. I know I have tried this with Audyssey in varying degrees but always come back to the single measurement spot for the best results. It is only critical for me for music, and my head is fairly stable in one position during music listening.


Every room tuning guy in the world will tell you you are crazy. 'Cept me.


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## bkeeler10

One single measurement eh? I should try that some time with my Audyssey Pro. It makes me take at least three, so I guess I will just leave the mic in the same spot for all three then.


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## AudiocRaver

bkeeler10 said:


> Thanks Wayne for another thorough review. I've been interested in Dirac and have been curious about how it compares with other auto EQ solutions, including Audyssey. I do hope to hear how XT32 compares if you get the chance you are hoping to get.
> 
> I'm hoping some time soon HTS will get a hold of an Anthem MRX receiver and hand it over to you to take a look at their ARC solution.
> 
> Interesting comments also from Dirac, particularly in regard to random measurement positions over a sufficiently large area. I know that, in the case of Anthem's ARC, they recommend the following: "The first [position] must be at or just in front of the central seating position . . . positions 2 and 3 should be symmetric to the left and right of the center line, and the same applies to the remaining positions. If your room has less than five seating positions, measurements must still be taken from five positions at least 2 feet apart to ensure optimal sound."
> 
> My understanding from that and elaborations of people in the know is that, when Anthem says that positions should be two feet apart, they mean that no two measurement positions should have less than two feet of separation distance between them.
> 
> I know this flies in the face of Wayne's experimentation and experience with Audyssey. So I find it quite interesting that Dirac also suggests a wider area of measurement. Of course, the secret sauce is different with each auto EQ solution so what is ideal for one may not be ideal for the others.


As you say, the secret is in the sauce. As the world of auto-correction / DRC progresses, it is clear there is so much more going on under the hood than averaging. Flavio has generously shared a wealth of insight into the workings of Dirac Live, yet I am sure there is much much more we will never hear about.

I for one am glad that he shared enough with us to knock me out of my somewhat stuck way of thinking about mic patterns, and look forward to implementing his suggestions for some in-depth testing. I make no apologies for the results already achieved, but am willing to see if those changes can get results closer to perfection. Naturally, I have a few other interesting mic patterns to try out, too - nudge nudge wink wink.


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## AudiocRaver

bkeeler10 said:


> One single measurement eh? I should try that some time with my Audyssey Pro. It makes me take at least three, so I guess I will just leave the mic in the same spot for all three then.


I would never go so far as to "recommend" it. But both Sonnie (XT32) and I (XT) have tried it with Audyssey and loved the result. That is only 2 examples, though, and it is for 2-channel with listeners who listen alone and do not move around much. A key to success: a thick, plushy blanket over the back of the chair for measurements and listening - it reduces reflections. 99.99999% of experienced setup people om the planet will tell you not to waste your time. But that's the thing - it takes almost zero time. So the temptation to try it once is almost too much to pass up. If you don't get a great result right off, then move on.

At Sonnie's place in January, we went through a full XT-32 Pro setup on his (then) Montis fronts, and did not like the way they measured or sounded. We then ran XT32 (min # of passes, 3??) with the mic at center of head and it sounded - and measured - GREAT. I believe that is where he left it.

EDIT: It is good for a quick "auto-correct sanity check" when a room or speaker setting is made.


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## Flak

Thanks Wayne, your interpretation of the logic behind Dirac Live and of our suggestions is 100% correct and I thank you for taking the time to evaluate them... and thanks to Sonnie, I need not say that we are eager to read your future review of Dirac Live vs. Audissey XT32 if it will come.

I think it is noteworthy that we have all been using ESLs also.... Martin Logans in your case and Sanders Model 10c in my case.
Those are a somehow special case as they are very coherent and have a pretty good impulse response to start with, together with a mainly directional behaviour which does help in managing reflections, so I consider an improvement in imaging a meaningful achievement.

Also in my opinion the evaluation of the results when a single person is listening in a single fixed position is very important, but may be an additional evaluation parameter of the results in other listening positions could be useful also, at least for those with different listening habits. 

I'll keep reading your articles and reviews with great interest... qualified and well documented side by side listening tests and measurements are quite scarce 

Ciao, Flavio


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## bkeeler10

Looking forward to the results of additional experimentation . . . and if I get a chance I will play with a single-point measurement with Audyssey myself.

On another note, I am very intrigued by the reports I'm hearing on Dolby Atmos for home. I'm wondering how an auto-EQ solution from Dirac could be implemented for every channel in a system like that. There are various difficulties I see arising. As I understand it, Dirac only operates on a PCM signal -- is that correct? If so, I see only two ways to insert Dirac into the mix. 

One would be to intercept the PCM signal, which would have to be done as it came out of the player. So far players will only bitstream for Atmos, so that isn't really an option. Plus you would have to have a box capable of receiving a PCM stream via HDMI with perhaps 11 (or eventually more) channels of information, which box would be running Dirac on all those channels, similar to the unit under review here. And then you would have to have an AVR capable of receiving more than eight channels of PCM via HDMI, which as I understand it is non-existent.

The other way would be to go from the player to the AVR as the Atmos spec is currently proposed and will be rolling out this fall (bitstream), have the AVR decode, route and perform D/A conversion, and then intercept the signal between the AVR pre-amp outs and the outboard amp(s). This would require a box with Dirac Live and A/D - D/A converters, similar to the analog version of the box reviewed here but with a whole bunch of channels.

I suppose a third way would be to implement this on an HTPC with Dirac Live software installed. A program like JRiver or XBMC would have to decode the Atmos bitstream to PCM, and then Dirac Live would process each resulting channel. This seems like the cleanest solution, except that I would expect that the computer would also have to perform the D/A conversion and you'd be sending analog out of the computer. You wouldn't be able to send it digitally since, again, no AVR I'm aware of can accept a PCM stream with more than eight channels. And computers are rather noisy for analog signals.

A fourth way would be to have Dirac Live in an Atmos AVR. That would be nice . . . 

Getting a little off-topic here, but since Flavio is keeping track of this thread I thought maybe he could chime in on their thoughts about how this might be done.


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## Flak

Hi Bryan,

there are several possible solutions and many innovative (some even thrilling) products which in theory can be developed but Dirac Research is a research company, not a hardware manufacturer, so most of them require a partnership with manufacturing companies that are just as innovative... I'm confident that in the future we will see even more of them 

Flavio


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## Sonnie

10C's Flavio? Had those in my home with a pair of the Magtech amps... Wayne can attest... we were about ready to say that we thought we had heard the best speakers ever at RMAF, but when I got them in my home... about 3 days solid of Wayne and I trying to find the proper placement and we never could get them to sound right. HOWEVER... even "not right" was better than _just about_ everything I have ever heard in my room. At least the Montis did give me a little more flexibility to move my head an inch or two. :bigsmile:

Had this day planned out to work with my Dirac unit and a good friend passed away on us and we spent about half the day an hour away from home, so it pretty much scratched my play time. Maybe over the weekend. :T


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## tesseract

AudiocRaver said:


> A key to success: a thick, plushy blanket over the back of the chair for measurements and listening - it reduces reflections. 99.99999% of experienced setup people om the planet will tell you not to waste your time. But that's the thing - it takes almost zero time. So the temptation to try it once is almost too much to pass up. If you don't get a great result right off, then move on.


I've told numerous people this over the years, but have never been taken seriously. Especially by the leather clad furniture crowd, curiously.


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## bkeeler10

So even if your couches are cloth the thick cushy blanket makes a substantial difference?


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## Sonnie

I think if your seat back is cloth, you will likely be fine, although you could try the puffed up part on measurements. Try it with and without and then look and see if the reflections are any different in REW.

I use one of these from Walmart on all three of my front row leather recliners... well... actually the one on my seat is a bit thicker. They are folded so that there are four layers draped over the top half of the back.


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## AudiocRaver

bkeeler10 said:


> So even if your couches are cloth the thick cushy blanket makes a substantial difference?


Depends on your level of pickiness. I started experimenting with it while using a leather chair, and of course the difference is BIG there. We quickly heard the difference with Sonnie's leather theater seats. My 2-channel recliner is cloth, but fairly stiff, and when I paid attention to it, I could hear how the imaging and frequency response shift due to reflections from it, and clearly see it in measurements. You can even see a very small reflection off the plushy blanket, but I have not been able to hear any effects from it. It seems like a good standard way to go, unless your chair material is so soft that the kids and pets have to be routinely rescued from its folds.


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## Flak

Hi Wayne,
a note on a note 

I've read the following in your test:

"Note: Having the UMIK-1 hanging by its cord for measurements will introduce some measurement error, the mic being 90 degrees off its reference axis and reading 1 to 2 dB low at 9 kHz for a typical UMIK-1. This would lead to extra boost at 9 kHz in the correction filter set. My target curve has enough loss at that frequency that the brightness is kept in control"

If you want to be more precise you may use an utility by Robert Cohen to convert your UMIK-1 0° calibration file to a 90° cal file, it is accurate and you will find it here:
http://www.minidsp.com/forum/umik-questions/10088-translate-0-degree-calibration-to-90-degree

To use that "micdelta" utility with Win7...

- The 0° calibration file of your UMIK-1 should be copied inside the micdelta folder together with the other files

- Press and hold Shift and right click on the micdelta folder to open the command prompt at that location and click on Open Command Window Here

- Enter the following string replacing umik1_serXXXXXX.txt with the name of your calibration file:
micdelta.exe umik1_avg_0.txt umik1_avg_90.txt umik1_serXXXXXX.txt umik1_serXXXXXX_90.txt

- Enter

DONE 
Flavio


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## AudiocRaver

Flak said:


> Hi Wayne,
> a note on a note
> 
> I've read the following in your test:
> 
> "Note: Having the UMIK-1 hanging by its cord for measurements will introduce some measurement error, the mic being 90 degrees off its reference axis and reading 1 to 2 dB low at 9 kHz for a typical UMIK-1. This would lead to extra boost at 9 kHz in the correction filter set. My target curve has enough loss at that frequency that the brightness is kept in control"
> 
> If you want to be more precise you may use an utility by Robert Cohen to convert your UMIK-1 0° calibration file to a 90° cal file, it is accurate and you will find it here:
> http://www.minidsp.com/forum/umik-questions/10088-translate-0-degree-calibration-to-90-degree
> 
> To use that "micdelta" utility with Win7...
> 
> - The 0° calibration file of your UMIK-1 should be copied inside the micdelta folder together with the other files
> 
> - Press and hold Shift and right click on the micdelta folder to open the command prompt at that location and click on Open Command Window Here
> 
> - Enter the following string replacing umik1_serXXXXXX.txt with the name of your calibration file:
> micdelta.exe umik1_avg_0.txt umik1_avg_90.txt umik1_serXXXXXX.txt umik1_serXXXXXX_90.txt
> 
> - Enter
> 
> DONE
> Flavio


Flavio,

That is super helpful. Thanks for the great tip.


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