Hello Everyone! Thanks so much for watching! We had a lot of fun making this, and we hope it's useful. HUGE thank you to Warren for helping us with this, and to EchoBar for hosting us. Charles from Kali is prowling the comments section looking for questions to answer, so please don't hesitate if you have any questions.
This was a great video guys! Very unique. I was struggling with 'room' sound issues (Ribbon mic) & just reached an acceptable combination of amp, mic & mattresses for isolation.
Because I had a little 60 Hz problem in my control room, this video inspired me to test such a mini DSP myself. I ordered one the next day and measured the room with a calibrated measurement microphone. I loaded the REW data into the mini DSP and the result is impressive. The 'boom' around 60 Hz is completely gone and everything sounds much 'cleaner' now. As a compensation I had used a little LF roll-off on my EVE208 speakers in the past, but now I can use them with the LF correction at 0 dB. You probably don't want to have a look at the correction response by the DSP (!) but it is the result that counts.
Hi Warren...love your videos and I have learned so much from you about making better sounds in the studio, even after years of mixing and producing... many many thanks to you. Please read on for my brief thoughts as a studio and performance hall designer and builder with a 40 year career I loved now behind me...Fundamentally, the room should be designed to allow any speaker to perform without interference from the rooms acoustics. Best room designs will not require EQ's which have a tendency to cause other types of distortion of the sound. If a room has these "bumps" you speak of you will always be compensating a recording that will be reproduced in the real world. The best rooms are made to control the many parameters of room activity through the acoustics and this can be established by taking the time and resources to consult with an educated and Experienced designer who can demonstrate a room that itself will provide a stealthy mix environment. You need to avoid the major pitfalls of a room that soaks up bass (this causes boomy recordings) or too dead which will cause too bright and the use of too much verb etc. BOTTOM LINE in very brief terms...get the room right first, it is my belief and experience having designed hundreds of rooms (I am now retired) that great control rooms are specifically designed to let only one speaker system be the absolute standard for a truly transparent mix. Use reference monitors if you like, but to hear the truth in you recording use the speaker system the room was specifically designed for.
being a live sound engineer, I definitely get the system tuning thing. I do however wonder why they would want to do each speaker separately, and completely ignore the way they interact together in the room... it seems you'd want to do L/R separate, but then bring them together and see how that changes things.
Because you get comb filtering when you measure more than one source at a time. Check out the SMAART documentation sometime or even better, get a copy of Bob McCarthy's Sound System Design and Optimization and chew on that for a couple years.
@@jaymz168 that's just it... since they're just playing the averages here anyway, I figured you could potentially average out some of the comb filtering, but now that I think about it, that wouldn't really work with just an eq. it would have to be physical positioning to change the combing characteristics
• Thank you for mentioning this. One of the unique advantages of moving microphone method is the ability to measure interactions of multiple loudspeakers with minimal obfuscation caused by combing artifacts present in single or even multi-microphone static measurements. This is a bit of an an advanced topic, and we want to avoid some of the combing artifact objections until we can fully explain it. Expect to see this at some point in the near future.
@@kaliaudio thanks for explanation! Please do video about full system measurement, not one by one sources. And would like to know how you count in comb filtering from stereo pair to make correction curve.
Sonorworks ref is the best thing I’ve ever discovered. Our control room has a pretty good sound but ref has made that extra 15 percent of difference. Epic software. I recommended to amateurs and pros alike.
@@asemon007 each to their own mate. I’m interested to know why mixing out of the box makes any difference to ref 4? Unless you mean you are completely computerless? Just real tape? Proper old school. I suppose that would mean there’s no software that would work for you!
@@andrewstevenson3807 No no, I use Cubase and Pro Tools but a I mix in a SSL Sigma and several pieces of gear. May be I'm wrong but Sonarworks process with eq the stereo out, isnt it? I need to eq just before the monitors and not the stereo output of my interface.
@@asemon007 I’m not sure I follow you here mate. Ref 4 simply EQ’s your room, directly before the sound comes out of your monitors. It’s the very last process in your signal chain before the sound hits your ears. It shouldn’t make any difference what out board gear you use.
Eagerly waiting for the super budget option. Hopefully it will be a great way to understand more before possibly spending more. Also, I'm curious of the latency of this option for tracking compared to Sonarworks
Hey Warren. I tried calibrating my room according to this video.... I can't help but wonder if there are steps I'm missing in your video with the REW software. Maybe its because I'm using a different version of the software. I bought the sonarworks mic and I installed their trial software. Did their calibration. Its kinda fun! Then I muted that correction, to do the REW correction, and see how close it matched. Its not close. I'm going to try again with the steps in your video soon. But I'd love a more in-depth tutorial. Please!!!!
It happened the same to me, I liked a lot the idea of moving the microphone to have a quick average but when i applied the curve predicted for that measurement it did not work, it ended up with less bottom end and more highs than the predicted eq for measurements done with sweeps
Man I got sonarworks and I had to send it back. It had a lot of issues with connecting to my interface. but the eq measurement part seamed really nice. I was just gonna go the Behringer and eq wizard route but though that would make it easier. I should have known, it's just like everything else they try to idiot-proof nowadays. I did, however, point out some very important things. First off I had my speakers flip-flopped and the reason I didn't notice was that I had my mono button engaged on my monitor controller that sits behind my desk... God I'm an idiot. But it also pointed out that my control room is not as bad as I thought it is. It's very small 7'10'x 5'x11'' by 7'x10 tall. I have it pretty heavily treated, and I could tell by the test my high-end response is really good, but I just have the typical low-end boost. Aslo does not help I'm in a shed and the floor shakes even with my monitors on sand-filled stands. I guess I gotta mix really low volumes and boost the bass a bit
Very interesting method. I was familiar with the more traditional sweeping tone we usually see. This is an alternative that seems much faster, with good results. Question: It is not mentioned how to setup the level of the pink noise and the level of the microphone. I am sure it makes a difference in the response we get. How should the level on the 2i2 interface be set?
The pink noise should be set approximately at the level that you normally listen. The mic level should be set with about 20 dB of headroom. You are correct that pink noise level will change the response, but as long as you're calibrating for the amount of sound power that you're normally working with, your results should be valid
I've been using REW, a mic I calibrated, and a miniDSP for years...the price for REW and the mic is certainly right, it just takes a bit of know-how... Works about 95% + ..as good as any proprietary system costing much more, IMO...
So where is the miniDSP in the signal chain? I go optically from my interface to a D/A that feeds powered speakers. Does the main output to the interface take a loop through the miniDSP before leaving the computer? At what point are the corrective eq’s applied?
Thank you very much for the video. Very informative. Although I can't find anywhere the promised links where Warren mentions about them at and of the video. Are these videos published and available? Thanks.
Mode should be set to RTA 1/24 octave FFT length should be 16384 samples. Averages should be set to “Forever” Window should be set to Hann The boxes should all be unchecked except “Adjust RTA levels”. The rest of the default settings should be fine.
Questions: 1. How much is the latency? 2. So this is not like Sonarworks, where you have to insert a plug-in in your daw, or systemwide bla bla? And more like let's say GLM? Bypassing the computer at the end? 3. Can we use one box with 2 sets of monitors and swap between them like normal after setting it up? 4. Why are you guys so great? (Thanks for doing this!)
1. Haven't measured for latency specifically, but it's not significant enough that we've ever noticed it. 2. Unlike Sonarworks. We're using a different approach to measuring, and a differenter approach to hardware. As you said, this lives outside the computer, as a separate part of the signal chain. 3. This is a philosophical question. If all you're doing is tuning for the room, this is a great way to go about it. If you're tuning the speakers themselves, you should use 2 boxes. ALTHOUGH you can also just make multiple presets, including output routing, in the MiniDSP, and just switch from there. You won't get instant switching though... 4. Warren must be rubbing off on us. :)
Thanks for the reply, and clearing these out! But I forget one important question that will make the buy. Any idea if the equalisation is linear phase? Like will it introduce pre-ringing? What normally happens with that type of equalisation. Thanks again.@@kaliaudio
QUESTION, the near and mid speakers position is good, but my instinct tell me that the way the far speakers and the mid ones the one on top of those that look like the focals, wont the sound from the far ones be alteres by having a speaker infront of it, wont the sound be somehow covered from the person sitting at the desk
My Equators have this built in unfortunately they've gone out of business so I can't update them but I going to use Room EQ to see if anything has change.
For Dirac live it uses this instead of moving mic as it knows what position the mic is in rather than it just knowing it’s moving about. This allows its algorithms to create complex hybrid filters
When using the MiniDSP box between your converters and the monitors, is the signal from your DAC being converted again when entering MiniDSP? Essentially, do you come out of main converters, back into another a/d, then d/a one more time? Love the video-thanks!
Hi Warren. I am impressed. However After the correction, would it be possible to make a listening test with music running through the minidsp with a bypass On/Off. To actually hear the difference With/Without the dsp ? Maybe run the same music file/ same music phrase A-B checked bypass on/off.
I just had a strange realization. I started watching this channel a few months before quarantine. And I've found Warren's videos very comforting during these stressful times. I figured it was just that he's a cheery English bloke. But the pose he makes around 11:07 made me realize that he look eerily similar to writer Sam Lake of Remedy (whose likeness was used for the original Max Payne). Given that I've replayed Max Payne & Max Payne 2 countless times, I think I understand every time I see Warren, it's like running into an old friend. Also, I think Sam Lake and Warren maybe twin that were separated at birth. ^_^
My Equators had this built in unfortunately they've gone out of business so I can't update them but I'm going to check with Room EQ and see if anything has changed
Id think that its good to treat the room after you've checked how imbalanced it is with "frequency-area imbalances", for example it might be somewhere in the lower parts or higher parts that needs treating with bass traps, absorbers, etc, but not using any "correction program in your DAW". Just my toughts to this "room problem" (not thinking about "speaker problem"). Also its better to depend on a good pair of headphones if you mix at home, like Sennheiser HD 600 is recommended, but there are many others too, and using "correction program" with those could actually be a good point, but less with speakers, imho. If the room needs correction, then correct the room, you only need to figure out what needs to be corrected, else you might continue making somewhat bad mixing decisions. Dont mix for your room, speakers or headphones, mix for best translation.
Hi and thanks! I think a absolute linear frequency response is not the target, or? I think an even drop from 100Hz to 12000Hz about 3-5dB makes more sense, or?
Great Video and very informative. But Trinnov ST2pro is all you would need. It is expensive, but it’s worth it. Warren I bet if you carry one in combination with your favorite monitors outside your environment, you would not be disappointed. Cheers sir.
Such a cool idea, but.... Reading the specs on the DSP and it runs at 48k. I realize it's self clocking and standing alone but if my system is running at 96k then I'm kind of chopping it off at the knees by putting this in line before the speakers. Very cool idea but needs to grow a bit to meet the needs of anyone working above 48k. I'll keep on using Sonarworks for now.
This is cost-effective method, and it has own drawback, but for the price 1/2 of Sonarworks it is great, and it has good benefit - it is hardware, so it doesn't load you computer. Sonarworks is addition to your computer and DAW, but this solution is more like addition to your monitors.
And of course you can easily implement it with any 5-band parametric EQ for free, for example on your master bus in DAW, as Warren says. Some interfaces has onboard DSPs with EQ, and you can use this solution inside of driver, without any additional hardware. For example RME has ability to use 3-band onboard EQ, maybe some manufacturers too.
MiniDSP do also do "beefier" versions, but they're gonna cost ya (more). At least on their own site, the Balanced 2x4 is $115, while the 2x4 HD (w/ RCA ins/outs) is $205, but does go up to 96k. But as pointed out, this *IS* the low-budget version. And if you've already splashed out on Sonarworks, this solution is probably not aimed at you anyway :D
Nice video and really instructive. I have a question ? does the sound card need to have a coaxial outputs that goes directly on the DSP ? What we can do if there is no coaxial inputs in the soundcard and speakers ? And what if speaker have a coaxial input but unbalanced ? There is some adaptor 1/4 to coaxial or something like that ? Thank you so much for this video. Really really outstanding.
Hi Warren... great topics! I've got the REW software for a couple of years now, downloading every new version appearing every once in a while. Got a miniDSP UMIK-1 calibrated mike. I used it on a fixed position, my sweet spot. I never got satisfied with the result because I never convinced myself I had to trust those curves for being the ultimate reference I needed. I recently bought the Arc System 2.5. Just about the same: no satisfaction! I don't want to get deafened by science. I know my speakers, I built my system to make music and my room ain't no lab. Those pieces of equipment are surely useful if you're designing monitors or checking modes in your room before giving it the treatment you'd like to give it... and if you need to pinpoint any specific problem in your studio. To make this stuff fully working, I think I'd need a miniDSP tuning my ears and my brain too... Earth ain't flat neither. Flat landscape is quite boring, don't you think so?
The goal of the moving mic method is to reconcile the measurements with what you hear. Our experience with stationary multiple mics, even a large number of them, was that the measurements would be completely different if you moved the array by an inch. This really doesn't reflect real-world experience. Your experience of measuring and then hearing different results is a very common one. We think that this method can get your science and your ears in much better alignment.
@@kaliaudio That's why I've been wanting to try the GLM software. I don't feel inclined to throw thousands of dollars on something I didn't get to try, in the space I'm going to use them in, is what I meant. I'm not rolling around in money, and I'm not a well known producer. :-) So this little box sounds like a solution in the meantime. Also I'm happy we got things like Kali's these days. 👍
I am using sonarworks refrence 4 with my focal alphas, is there any point of doing this befor that? Or how about doing the built in eq on the speakers befor doing any measuring with sonarworks or rew? Should I apply speaker eq to get the speakers as flat as possible befor doing any measuring?
I wouldn't have both to be honest. But it may be worth fliping some EQ switches on yr speakers if there is say a general lack of bass for instance. Then Sonarworks isn't having to do large boosts down the bottom end to compensate.
@@Producelikeapro I,m well thanks man!!!, having a drumholiday (practising in my bosses drumroom daily) and searching for a permanent room for my drums serving clients (I,m building a side-business) and my complex drumbook is coming out, with an article also in a Dutch/Belgium/Luxembourg magazine: Slagwerkkrant. and ofcourse studying a lot of mixing/reading all kinds of books...Nerdmode deluxe!!! Peace, Jeroen.
Looks like a very good inexpensive way to calibrate with DSP. How about phase issues when adding so much gain? Is there any quality differences with inexpensive DSP/EQ filtering? And maybe weird to ask but why do you measure as if the head was just above the desk? Some things this would benefit from is ease of use. To many things the user needs to do. And why not add a UX specialist to the team. : ) (Software engineers and technicians are specialists in what they, do but certainly not experts on UX.)
The phase issues should be remedied by equalization. The room resonance imparts a phase anomaly. Correctly applied equalization corrects not only the frequency response, but also the phase response. That having been said, we do advise against applying massive amounts of boost as it pulls away from headroom. With regards to DSP quality, we are using a DSP with quality comparable to the monitors. For premium monitors, they might consider a higher end DSP as well. The camera angle may have created the impression we were just above the desk but the circles were centered on the ear position.
Bad call on the broad 65 Hz filter. It required a larger EQ adjustment in the 100-200 Hz range than was otherwise needed. It would have been better to just let the Auto EQ run and ignore filters below 100 Hz upon application. In reality, the response graph presented most likely does not need EQ at all. Three filters (#3, 4, 5) between 300-500 Hz (a bandwidth of only ~2/3-octave) is overkill to the max. I wish they had given us the parameters, but they are surely ultra-tight notch filters, 1/12-octave or even tighter, the type used in live sound to eliminate feedback. Keep in mind that there are only twelve notes in the musical scale, which means each one represents 1/12 octave, and you can see how potentially destructive it can be EQing with filters this tight. Filters that tight and with only a few dB in gain will probably be inaudible with program material, as I imagine an A/B listening test would show. If that ends up being the case, what’s the point of polluting your signal chain with all that extraneous processing? In addition, check the “before” and “after” response at the #5 filter. It is exactly the same for both graphs! What happened is that the #5 filter was necessary to counteract the effects of the #3 and #4 filters. This is what’s known in professional circles as “equalizing the equalizer.” It’s exceedingly poor form, to say the least. Further evidence that none of those filters were necessary to begin with. REW is a wonderful tool, and we are all eternally grateful to John Mulcahy for this gift. But it doesn’t come with training on the proper use of equalizers. Unfortunately, the easy availability of advanced measurement platforms, combined with bargain-priced yet sophisticated equalizers like those from miniDSP, has spawned a whole generation of people who essentially live and die by the graph. Not knowing any better, they’ll drown their systems in superfluous filters that accomplish nothing of audible benefit in the pursuit of “graphic nirvana.” I like what one member at the AV Nirvana Forum wrote: “I’ve had the feeling that my overzealous EQ attempts fixed the graphs but killed the sound.”
It could be to make sure the tweeter is at ear level. Placing speakers upside down or sideways usually makes a bigger difference in small rooms regarding the lower frequencies. In larger rooms not so much.
@ Charles from Kali -- I ordered the miniDSP 2X4 Balanced that you used in the video. How did you set the jumpers inside the box for input and output voltage sensitivity? I can't find a good answer anywhere on the web. Does it really matter? I'm using a UA Audio Apollo through a Dangerous Music D-Box into Presonus Eris E8XTs and a Presonus Subwoofer, if that helps.
Why it’s good to have more than one monitor and have the room EQed… I use 4”, 8” and have a PA with 2 18” subs in a fairly small room to check the low end… if I can tell it’s there on the 4” and the twin 18”s aren’t blowing me out of the room I figure I’m in the ballpark…
Thanks for the video Warren! I have have been curious of the miniDSP system for some time and it's great to see an example of it in use. My understanding was that pink noise has a drop off of so many db per octave. It seems like the objective here is to align as closely as possible to a flat line. Wouldn't it be preferred to do an overlay of the pink noise and try to align to that? (ie. pink noise graphed vs room measurements)
The settings in REW are such that pink noise in a neutral system produces a flat line. If you use the "Spectrum" setting in REW, you'd see the roll off of pink noise.
That dedicated measuring function in REW runs a frequency sweep - doesn't really work when you wave the mic around :) This way works because (and unfortunately it wasn't pointed out verbally in the video) 1) you're using pink noise, and 2) the averaging is set to "forever", which then allows you to get that "spatial average". Stuff like Sonarworks runs sweeps like REW, but also instructs you where to place the mic for each of the dozens of measurements it needs.
@@khronscave I am using Acourate from AudioVero - runs sweeps too. I believe this measuring is much faster that averaging multiple sweeps from REW, Sonarworks, ARC etc - anyhow .... interesting video/method.
How does this compare to something like IKM's ARC2? I'm using that and it makes a world of difference in the sound of my room and monitors, but it's obviously not an ideal solution.
To be honest - why didn't they use white noise wich is technically closer to sine sweeps... i did it myself with pink noise first, but if you enter this in the EQ section, you get a highly rising curve. I guess it should be white noise instead of pink noise... or my studio is completely off ;) both possible.
Adding 6db of boost to that speaker could cause problems with the speaker/amp pushing it past its limits. Wouldn't it have been better to EQ the frequency response lower to match with the lower output of the woofer? Yes you do reduce the overall SPL out of the speaker but it's better than having high distortion coming from the speakers woofer or worse the woofer hitting its Xmax and clip.
Great. Will def have to do this in my studio asap. I have got a question though. I just looked up the MiniDSP box. Is there an option for non block type input and output connections?
I noticed that he saves the files as "Right monitor" and later loads it in the DSP for output 2. That suggests that you will have to do these measurements for both speakers separately? (Which seems logical.) Also: I don't hear a word about latency, usually present when using DSP processing.
Correct. We recommend measuring one speaker at a time. Latency is an issue, but the MiniDSP is pretty good, and it's only applying 5 bands of EQ, so it's not a HUGE issue.
@@kaliaudio QUESTION: "Adding 6db of boost to that speaker could cause problems with the speaker/amp pushing it past its limits. Wouldn't it have been better to EQ the frequency response lower to match with the lower output of the woofer? Yes you do reduce the overall SPL out of the speaker but it's better than having high distortion coming from the speakers woofer or worse the woofer hitting its Xmax and clip." Thanks!
okay this was a long one but it was a great video. one question is the box they are using only for passive speakers? or can I cut a xlr cable out and match the inputs to that? im assuming its for a passive system.
@@kaliaudio oooo now i gotta make a video on doing this too XD also saw your speakers on a bunch of channels now i hear you are kicking butt. Im going to have to try some out soon!
In the DAW yes, for me, Kali are making the best sounding and most affordable speakers out there! $300 for a pair of great sounding powered speakers is truly amazing!
I would like to see the difference in response trace between a single mic on a stand in the middle of the critical listening position versus this moving average. My suspicion is that there would be no real difference.
GREAT QUESTION! Erik, who works in this room, likes to switch between his Kalis and his Focals. With the Kalis right side up on top of the Focals, the tweeters are so far apart that it's distracting. The waveguide on the Kalis is asymmetrical, so the imaging will suffer if the Kalis are on their sides. SO, in order to get the tweeters closer, we recommended that Erik flip them upside down, which preserves the imaging and achieves his goal.
@@kaliaudio Wow, thanks for the explanation! I have to say, I love my Kalis, though my purchasing experience was awful through Amazon (not your fault at all!). I have one black and one white monitor because they sent me (and I subsequently returned) 5 SPEAKERS over the past month, trying to get a single black Kali LP6. So, my studio looks... interesting with one white and one black monitor. They SOUND AMAZING!
I know there's too much low end in my room and would love to calibrate it. You should really do the inexpensive version with an sm57 and give out the microphone frequency response for people who can't calibrate for themselves... cause i mean, everyone watching this channel has a 57 and if they don't they should
@@thekramdens Fair point. Turns out i can just buy an omni capsule for my nt5 so thanks for the eureka moment there. I wonder would calibrating with a 57 be better than nothing at all though?
Behringer and Dayton Audio both sell Omni measurement mics for less than it would cost to get an omni capsule for the NT5. As handy as that would be to have anyway, it's not designed to be flat and provided with a calibration file. There's a right tool for every job. :-)
Im trying to calibrate but need more specifics of this software setup? I have behringer mic and software but what screen is constant calibration. Can someone detail please? Thanks
@@bernossi5185 it uses phoenix connectors, which you connect any balanced cable to with a little bit of slicing and splicing. Sweetwater also sells adapters if you don't want to make them yourself. For power, it uses a DC adapter. It's controlled via USB while connected to a computer, but it keeps doing its thing once disconnected.
Just to be clear...the chain is Interface > MiniDSP > Powered Speakers? What cables are used to connect these? It appears like the MiniDSP unit you have has a regular speaker cable connection. The output of my interface is either 1/4' or XLR...I'm not sure how you put the MiniDSP in the middle of this.
I´m confused, tehy are making the calibration with pink noise, I´ve always made it with a freq sweep, can someone give me some light on this? I´m not using dsp though
What i do not understand fully is that you are basically saying that it is not good to boost up around 60hz because it'd take too much headroom of the speakers and saying that using a subwoofer instead is better but then the 60hz range is boosted up with the dsp EQing software. Am I missing something? Great Video anyways!
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Do you calibrate your room? Let me know below!
Sonarworks, I was blown away with the results!
@@timjonesguitar great stuff! We love those guys!
Absolutely. I use Sonarworks and it makes all the difference
@@BengtSkogvall thanks ever so much!!
@@prodbyjharee thanks my friend!
Hello Everyone! Thanks so much for watching! We had a lot of fun making this, and we hope it's useful. HUGE thank you to Warren for helping us with this, and to EchoBar for hosting us.
Charles from Kali is prowling the comments section looking for questions to answer, so please don't hesitate if you have any questions.
So the links to the other videos aren't up yet correct?
Thank you. That is very interesting.
This was a great video guys! Very unique.
I was struggling with 'room' sound issues (Ribbon mic) & just reached an acceptable combination of amp, mic & mattresses for isolation.
What is the Calibration Mic used?
@@yungpepe Not yet, but soon. Editing as we speak.
Because I had a little 60 Hz problem in my control room, this video inspired me to test such a mini DSP myself.
I ordered one the next day and measured the room with a calibrated measurement microphone.
I loaded the REW data into the mini DSP and the result is impressive.
The 'boom' around 60 Hz is completely gone and everything sounds much 'cleaner' now.
As a compensation I had used a little LF roll-off on my EVE208 speakers in the past, but now I can use them with the LF correction at 0 dB. You probably don't want to have a look at the correction response by the DSP (!) but it is the result that counts.
Hi Warren...love your videos and I have learned so much from you about making better sounds in the studio, even after years of mixing and producing... many many thanks to you. Please read on for my brief thoughts as a studio and performance hall designer and builder with a 40 year career I loved now behind me...Fundamentally, the room should be designed to allow any speaker to perform without interference from the rooms acoustics. Best room designs will not require EQ's which have a tendency to cause other types of distortion of the sound. If a room has these "bumps" you speak of you will always be compensating a recording that will be reproduced in the real world. The best rooms are made to control the many parameters of room activity through the acoustics and this can be established by taking the time and resources to consult with an educated and Experienced designer who can demonstrate a room that itself will provide a stealthy mix environment. You need to avoid the major pitfalls of a room that soaks up bass (this causes boomy recordings) or too dead which will cause too bright and the use of too much verb etc. BOTTOM LINE in very brief terms...get the room right first, it is my belief and experience having designed hundreds of rooms (I am now retired) that great control rooms are specifically designed to let only one speaker system be the absolute standard for a truly transparent mix. Use reference monitors if you like, but to hear the truth in you recording use the speaker system the room was specifically designed for.
02:48 Marvelous!
being a live sound engineer, I definitely get the system tuning thing. I do however wonder why they would want to do each speaker separately, and completely ignore the way they interact together in the room... it seems you'd want to do L/R separate, but then bring them together and see how that changes things.
Because you get comb filtering when you measure more than one source at a time. Check out the SMAART documentation sometime or even better, get a copy of Bob McCarthy's Sound System Design and Optimization and chew on that for a couple years.
@@jaymz168 that's just it... since they're just playing the averages here anyway, I figured you could potentially average out some of the comb filtering, but now that I think about it, that wouldn't really work with just an eq. it would have to be physical positioning to change the combing characteristics
• Thank you for mentioning this. One of the unique advantages of moving microphone method is the ability to measure interactions of multiple loudspeakers with minimal obfuscation caused by combing artifacts present in single or even multi-microphone static measurements. This is a bit of an an advanced topic, and we want to avoid some of the combing artifact objections until we can fully explain it. Expect to see this at some point in the near future.
@@kaliaudio thanks for explanation! Please do video about full system measurement, not one by one sources. And would like to know how you count in comb filtering from stereo pair to make correction curve.
@@dimitrydenscikov5567 Indeed; also very interested.
Sonorworks ref is the best thing I’ve ever discovered. Our control room has a pretty good sound but ref has made that extra 15 percent of difference. Epic software. I recommended to amateurs and pros alike.
I think a hardcore pro could afford the 2-10k it would cost for a measurement and professional-grade dsp system
It doesnt work for me. I mix out the box
@@asemon007 each to their own mate. I’m interested to know why mixing out of the box makes any difference to ref 4? Unless you mean you are completely computerless? Just real tape? Proper old school. I suppose that would mean there’s no software that would work for you!
@@andrewstevenson3807 No no, I use Cubase and Pro Tools but a I mix in a SSL Sigma and several pieces of gear. May be I'm wrong but Sonarworks process with eq the stereo out, isnt it? I need to eq just before the monitors and not the stereo output of my interface.
@@asemon007 I’m not sure I follow you here mate. Ref 4 simply EQ’s your room, directly before the sound comes out of your monitors. It’s the very last process in your signal chain before the sound hits your ears. It shouldn’t make any difference what out board gear you use.
Eagerly waiting for the super budget option. Hopefully it will be a great way to understand more before possibly spending more. Also, I'm curious of the latency of this option for tracking compared to Sonarworks
Perfect now we need a demo for speaker crossover and time alignment using miniDSP as was discussed for a subwoofer addition to the speaker system.
yes I agree
lol
Wonderful video. Opened my eyes to a whole new universe of approachable room optimization. Thanks a ton!
You're very welcome! Glad to be able to help!
Where is that $6 mic video and inexpensive miniDSP replacement that you talked about at the end of this video.?
is there a link to the budget friendly calibration video mention in this video..? i dont see it in the description..
Use what you have
Hey Warren. I tried calibrating my room according to this video.... I can't help but wonder if there are steps I'm missing in your video with the REW software. Maybe its because I'm using a different version of the software. I bought the sonarworks mic and I installed their trial software. Did their calibration. Its kinda fun! Then I muted that correction, to do the REW correction, and see how close it matched. Its not close. I'm going to try again with the steps in your video soon. But I'd love a more in-depth tutorial. Please!!!!
It happened the same to me, I liked a lot the idea of moving the microphone to have a quick average but when i applied the curve predicted for that measurement it did not work, it ended up with less bottom end and more highs than the predicted eq for measurements done with sweeps
Man I got sonarworks and I had to send it back. It had a lot of issues with connecting to my interface. but the eq measurement part seamed really nice. I was just gonna go the Behringer and eq wizard route but though that would make it easier. I should have known, it's just like everything else they try to idiot-proof nowadays. I did, however, point out some very important things. First off I had my speakers flip-flopped and the reason I didn't notice was that I had my mono button engaged on my monitor controller that sits behind my desk... God I'm an idiot. But it also pointed out that my control room is not as bad as I thought it is. It's very small 7'10'x 5'x11'' by 7'x10 tall. I have it pretty heavily treated, and I could tell by the test my high-end response is really good, but I just have the typical low-end boost. Aslo does not help I'm in a shed and the floor shakes even with my monitors on sand-filled stands. I guess I gotta mix really low volumes and boost the bass a bit
Very interesting method. I was familiar with the more traditional sweeping tone we usually see. This is an alternative that seems much faster, with good results.
Question: It is not mentioned how to setup the level of the pink noise and the level of the microphone. I am sure it makes a difference in the response we get. How should the level on the 2i2 interface be set?
The pink noise should be set approximately at the level that you normally listen. The mic level should be set with about 20 dB of headroom. You are correct that pink noise level will change the response, but as long as you're calibrating for the amount of sound power that you're normally working with, your results should be valid
Where is he link about setting up your speakers in a room, that you talked about at the end of the video?
Thanks Warren for more great information!
Thanks Darlene!
Warren,any idea when the "inexpensive calibration" video will be up?
I can't find the "most inexpensive" version with the 6$ mic... where is it?
I've been using REW, a mic I calibrated, and a miniDSP for years...the price for REW and the mic is certainly right, it just takes a bit of know-how... Works about 95% + ..as good as any proprietary system costing much more, IMO...
So where is the miniDSP in the signal chain? I go optically from my interface to a D/A that feeds powered speakers. Does the main output to the interface take a loop through the miniDSP before leaving the computer? At what point are the corrective eq’s applied?
Thank you very much for the video. Very informative. Although I can't find anywhere the promised links where Warren mentions about them at and of the video. Are these videos published and available? Thanks.
10:08 It would've been nice to mention the measurement settings as well (only here i got to see the averaging was set to "forever", for example).
Mode should be set to RTA 1/24 octave
FFT length should be 16384 samples.
Averages should be set to “Forever”
Window should be set to Hann
The boxes should all be unchecked except “Adjust RTA levels”.
The rest of the default settings should be fine.
@@kaliaudio This is what I came to the comments for! Thank you!
Questions: 1. How much is the latency? 2. So this is not like Sonarworks, where you have to insert a plug-in in your daw, or systemwide bla bla? And more like let's say GLM? Bypassing the computer at the end? 3. Can we use one box with 2 sets of monitors and swap between them like normal after setting it up? 4. Why are you guys so great? (Thanks for doing this!)
1. Haven't measured for latency specifically, but it's not significant enough that we've ever noticed it.
2. Unlike Sonarworks. We're using a different approach to measuring, and a differenter approach to hardware. As you said, this lives outside the computer, as a separate part of the signal chain.
3. This is a philosophical question. If all you're doing is tuning for the room, this is a great way to go about it. If you're tuning the speakers themselves, you should use 2 boxes. ALTHOUGH you can also just make multiple presets, including output routing, in the MiniDSP, and just switch from there. You won't get instant switching though...
4. Warren must be rubbing off on us. :)
Thanks for the reply, and clearing these out! But I forget one important question that will make the buy. Any idea if the equalisation is linear phase? Like will it introduce pre-ringing? What normally happens with that type of equalisation. Thanks again.@@kaliaudio
QUESTION, the near and mid speakers position is good, but my instinct tell me that the way the far speakers and the mid ones the one on top of those that look like the focals, wont the sound from the far ones be alteres by having a speaker infront of it, wont the sound be somehow covered from the person sitting at the desk
Thank you Charles, Erik, and Warren! I shall try and learn what I can about the software, ...sounds encouraging. 🙂
Thank YOU!
My Equators have this built in unfortunately they've gone out of business so I can't update them but I going to use Room EQ to see if anything has change.
Yeah me too. Kind of sad about that. Love coaxial speaker.
Using Dirac Live here. And it actually requires a multiple position mic measurement. 9 different readings.
For Dirac live it uses this instead of moving mic as it knows what position the mic is in rather than it just knowing it’s moving about. This allows its algorithms to create complex hybrid filters
How do you find it? Does it run system wide (all audio out) or just via a plugin on the DAW master? I use DIRAC I. My home cinema and it’s great!
I'm in outer so there's a lot to factor in. Know your Space and your mix will thank you!
When using the MiniDSP box between your converters and the monitors, is the signal from your DAC being converted again when entering MiniDSP? Essentially, do you come out of main converters, back into another a/d, then d/a one more time? Love the video-thanks!
Hi Warren. I am impressed. However After the correction, would it be possible to make a listening test with music running through the minidsp with a bypass On/Off. To actually hear the difference With/Without the dsp ? Maybe run the same music file/ same music phrase A-B checked bypass on/off.
Hey, where is the link to adding rew's eq corrections to a "generic" eq on the master bus from your daw?
I just had a strange realization. I started watching this channel a few months before quarantine. And I've found Warren's videos very comforting during these stressful times. I figured it was just that he's a cheery English bloke. But the pose he makes around 11:07 made me realize that he look eerily similar to writer Sam Lake of Remedy (whose likeness was used for the original Max Payne). Given that I've replayed Max Payne & Max Payne 2 countless times, I think I understand every time I see Warren, it's like running into an old friend.
Also, I think Sam Lake and Warren maybe twin that were separated at birth. ^_^
My Equators had this built in unfortunately they've gone out of business so I can't update them but I'm going to check with Room EQ and see if anything has changed
Id think that its good to treat the room after you've checked how imbalanced it is with "frequency-area imbalances", for example it might be somewhere in the lower parts or higher parts that needs treating with bass traps, absorbers, etc, but not using any "correction program in your DAW".
Just my toughts to this "room problem" (not thinking about "speaker problem").
Also its better to depend on a good pair of headphones if you mix at home, like Sennheiser HD 600 is recommended, but there are many others too, and using "correction program" with those could actually be a good point, but less with speakers, imho. If the room needs correction, then correct the room, you only need to figure out what needs to be corrected, else you might continue making somewhat bad mixing decisions. Dont mix for your room, speakers or headphones, mix for best translation.
Hi and thanks! I think a absolute linear frequency response is not the target, or? I think an even drop from 100Hz to 12000Hz about 3-5dB makes more sense, or?
Yes this is a typical house curve
Great Video and very informative. But Trinnov ST2pro is all you would need. It is expensive, but it’s worth it. Warren I bet if you carry one in combination with your favorite monitors outside your environment, you would not be disappointed. Cheers sir.
Such a cool idea, but.... Reading the specs on the DSP and it runs at 48k. I realize it's self clocking and standing alone but if my system is running at 96k then I'm kind of chopping it off at the knees by putting this in line before the speakers. Very cool idea but needs to grow a bit to meet the needs of anyone working above 48k. I'll keep on using Sonarworks for now.
This is cost-effective method, and it has own drawback, but for the price 1/2 of Sonarworks it is great, and it has good benefit - it is hardware, so it doesn't load you computer. Sonarworks is addition to your computer and DAW, but this solution is more like addition to your monitors.
And of course you can easily implement it with any 5-band parametric EQ for free, for example on your master bus in DAW, as Warren says. Some interfaces has onboard DSPs with EQ, and you can use this solution inside of driver, without any additional hardware. For example RME has ability to use 3-band onboard EQ, maybe some manufacturers too.
MiniDSP do also do "beefier" versions, but they're gonna cost ya (more). At least on their own site, the Balanced 2x4 is $115, while the 2x4 HD (w/ RCA ins/outs) is $205, but does go up to 96k.
But as pointed out, this *IS* the low-budget version. And if you've already splashed out on Sonarworks, this solution is probably not aimed at you anyway :D
I use SysTune as I have this for live concert system tuning
Love how much fun Warren is having at 11:27 hahaha
This is my favorite shot! lol!
Haha thanks ever so much Roy!!
haha thanks!@@kaliaudio
Nice video and really instructive. I have a question ? does the sound card need to have a coaxial outputs that goes directly on the DSP ? What we can do if there is no coaxial inputs in the soundcard and speakers ? And what if speaker have a coaxial input but unbalanced ? There is some adaptor 1/4 to coaxial or something like that ? Thank you so much for this video. Really really outstanding.
That is a wonderful topic. Thank you very much!
Thanks ever so much Jürgen!
when can we expect the video using the home calibrated mic option?
mrbertssuitcase soon my friend!
@@Producelikeapro hossum Warren!!
@@Producelikeapro GREAT, I WAS JUST ABOUT TO ASK TOO! (shouting from Australia).
Is the video up? I cannot find it
I have been using REW for years now. Great video.
Thanks!
Wonderful to hear!
@@kaliaudio Thank YOU!
Where is the link to glenn's video?
5:06 so funny how he summed up all the info for the mere mortals like us.
Hi Warren... great topics! I've got the REW software for a couple of years now, downloading every new version appearing every once in a while. Got a miniDSP UMIK-1 calibrated mike. I used it on a fixed position, my sweet spot. I never got satisfied with the result because I never convinced myself I had to trust those curves for being the ultimate reference I needed. I recently bought the Arc System 2.5. Just about the same: no satisfaction! I don't want to get deafened by science. I know my speakers, I built my system to make music and my room ain't no lab. Those pieces of equipment are surely useful if you're designing monitors or checking modes in your room before giving it the treatment you'd like to give it... and if you need to pinpoint any specific problem in your studio. To make this stuff fully working, I think I'd need a miniDSP tuning my ears and my brain too... Earth ain't flat neither. Flat landscape is quite boring, don't you think so?
The goal of the moving mic method is to reconcile the measurements with what you hear. Our experience with stationary multiple mics, even a large number of them, was that the measurements would be completely different if you moved the array by an inch. This really doesn't reflect real-world experience.
Your experience of measuring and then hearing different results is a very common one. We think that this method can get your science and your ears in much better alignment.
Been saving up for Genelecs, but they won't lend me a set to try out so maybe this is the route to go for now.
Even on the Genelecs, calibrating your room will make a HUGE difference.
@@kaliaudio That's why I've been wanting to try the GLM software. I don't feel inclined to throw thousands of dollars on something I didn't get to try, in the space I'm going to use them in, is what I meant. I'm not rolling around in money, and I'm not a well known producer. :-) So this little box sounds like a solution in the meantime. Also I'm happy we got things like Kali's these days. 👍
I am using sonarworks refrence 4 with my focal alphas, is there any point of doing this befor that? Or how about doing the built in eq on the speakers befor doing any measuring with sonarworks or rew? Should I apply speaker eq to get the speakers as flat as possible befor doing any measuring?
Great stuff Fredrik! Thanks for sharing!
I wouldn't have both to be honest. But it may be worth fliping some EQ switches on yr speakers if there is say a general lack of bass for instance. Then Sonarworks isn't having to do large boosts down the bottom end to compensate.
Can you just apply the same EQ curve using a transparent plugin on the master bus?
You can! We'll be showing that in a VERY soon-to-come video.
@@kaliaudio What happened to these videos that were supposed to be uploaded?
This was very interesting. Going to try this method soon in my attic studio.
Let us know how it goes!
Very useful and very cost effective. .nice one. .
Thanks ever so much Shaun! Agreed!
Love to see this and...The (upside down) Kali Audio lp 8 being used at Echo Bar..:)
Great stuff Jeroen! I hope you're well my friend!!
@@Producelikeapro I,m well thanks man!!!, having a drumholiday (practising in my bosses drumroom daily) and searching for a permanent room for my drums serving clients (I,m building a side-business) and my complex drumbook is coming out, with an article also in a Dutch/Belgium/Luxembourg magazine: Slagwerkkrant. and ofcourse studying a lot of mixing/reading all kinds of books...Nerdmode deluxe!!! Peace, Jeroen.
Thank you so much, Gentlemen, that was excellent, and it really helped a lot!! 👌
Thanks ever so much Frank!!
Looks like a very good inexpensive way to calibrate with DSP.
How about phase issues when adding so much gain?
Is there any quality differences with inexpensive DSP/EQ filtering?
And maybe weird to ask but why do you measure as if the head was just above the desk?
Some things this would benefit from is ease of use. To many things the user needs to do. And why not add a UX specialist to the team. : )
(Software engineers and technicians are specialists in what they, do but certainly not experts on UX.)
The phase issues should be remedied by equalization. The room resonance imparts a phase anomaly. Correctly applied equalization corrects not only the frequency response, but also the phase response. That having been said, we do advise against applying massive amounts of boost as it pulls away from headroom.
With regards to DSP quality, we are using a DSP with quality comparable to the monitors. For premium monitors, they might consider a higher end DSP as well.
The camera angle may have created the impression we were just above the desk but the circles were centered on the ear position.
@@kaliaudio Thanks for answering!
Hi Kali Audio extremely well said!!
Well, pretty similar to Reference 4. I am just concern about the DAC aspect of that cheep unit.
Reference 4 doesnt work for me. I mix out the box
Totally. And it's not only DAC, but actually AD & DA. I'd imagine some hiss is created.
Bad call on the broad 65 Hz filter. It required a larger EQ adjustment in the 100-200 Hz range than was otherwise needed. It would have been better to just let the Auto EQ run and ignore filters below 100 Hz upon application.
In reality, the response graph presented most likely does not need EQ at all. Three filters (#3, 4, 5) between 300-500 Hz (a bandwidth of only ~2/3-octave) is overkill to the max. I wish they had given us the parameters, but they are surely ultra-tight notch filters, 1/12-octave or even tighter, the type used in live sound to eliminate feedback. Keep in mind that there are only twelve notes in the musical scale, which means each one represents 1/12 octave, and you can see how potentially destructive it can be EQing with filters this tight. Filters that tight and with only a few dB in gain will probably be inaudible with program material, as I imagine an A/B listening test would show. If that ends up being the case, what’s the point of polluting your signal chain with all that extraneous processing?
In addition, check the “before” and “after” response at the #5 filter. It is exactly the same for both graphs! What happened is that the #5 filter was necessary to counteract the effects of the #3 and #4 filters. This is what’s known in professional circles as “equalizing the equalizer.” It’s exceedingly poor form, to say the least. Further evidence that none of those filters were necessary to begin with.
REW is a wonderful tool, and we are all eternally grateful to John Mulcahy for this gift. But it doesn’t come with training on the proper use of equalizers. Unfortunately, the easy availability of advanced measurement platforms, combined with bargain-priced yet sophisticated equalizers like those from miniDSP, has spawned a whole generation of people who essentially live and die by the graph. Not knowing any better, they’ll drown their systems in superfluous filters that accomplish nothing of audible benefit in the pursuit of “graphic nirvana.” I like what one member at the AV Nirvana Forum wrote: “I’ve had the feeling that my overzealous EQ attempts fixed the graphs but killed the sound.”
Great idea! Im wondering, are there any negative effects from running this box? Added noise, distortions or phase shifts?
The MiniDSP is pretty damn good. I can't say that any of us at Kali have experienced the drawbacks you listed.
@@kaliaudio cool! good to know!
Great vidéo, realy usefull ...
Thank you from France !!!
I'm very curious about the positioning of those kali monitors, what would be the reason behind it?
I would assume tweeter height, you probably don't want to tweeters at ceiling level : D
It could be to make sure the tweeter is at ear level. Placing speakers upside down or sideways usually makes a bigger difference in small rooms regarding the lower frequencies. In larger rooms not so much.
@@alone66666 That's right!
@@chrisibbetson Yes!
@@kaliaudio I have a pair of lp6 I am very pleased with them. Fantastic value for money!
How do you deal with microphonics when moving the microphone?
@Kali Audio thx for the Great information provided on this Video. Is there any solution for higher sample rates like 96k?
@ Charles from Kali -- I ordered the miniDSP 2X4 Balanced that you used in the video. How did you set the jumpers inside the box for input and output voltage sensitivity? I can't find a good answer anywhere on the web. Does it really matter? I'm using a UA Audio Apollo through a Dangerous Music D-Box into Presonus Eris E8XTs and a Presonus Subwoofer, if that helps.
Amazing!! Thank you, Warren, for the information. :)
Thanks ever so much my friend!!
Why it’s good to have more than one monitor and have the room EQed… I use 4”, 8” and have a PA with 2 18” subs in a fairly small room to check the low end… if I can tell it’s there on the 4” and the twin 18”s aren’t blowing me out of the room I figure I’m in the ballpark…
Thanks for the video Warren! I have have been curious of the miniDSP system for some time and it's great to see an example of it in use.
My understanding was that pink noise has a drop off of so many db per octave. It seems like the objective here is to align as closely as possible to a flat line.
Wouldn't it be preferred to do an overlay of the pink noise and try to align to that? (ie. pink noise graphed vs room measurements)
The settings in REW are such that pink noise in a neutral system produces a flat line. If you use the "Spectrum" setting in REW, you'd see the roll off of pink noise.
@@kaliaudio ah, ok. Thanks for explaining!
Why don´t you use the existing measuring procedure in REW ?
That dedicated measuring function in REW runs a frequency sweep - doesn't really work when you wave the mic around :)
This way works because (and unfortunately it wasn't pointed out verbally in the video) 1) you're using pink noise, and 2) the averaging is set to "forever", which then allows you to get that "spatial average".
Stuff like Sonarworks runs sweeps like REW, but also instructs you where to place the mic for each of the dozens of measurements it needs.
@@khronscave I am using Acourate from AudioVero - runs sweeps too. I believe this measuring is much faster that averaging multiple sweeps from REW, Sonarworks, ARC etc - anyhow .... interesting video/method.
how about the speaker that only have one knob volume control for both? like Eris 3.5
I am not able to upload a cal file with the frequencies of my microphone. please, how do you convert a text to cal?
This is really useful thanks. Has the 6 dollar mic and stock EQ version ever been released?
How does this compare to something like IKM's ARC2? I'm using that and it makes a world of difference in the sound of my room and monitors, but it's obviously not an ideal solution.
To be honest - why didn't they use white noise wich is technically closer to sine sweeps... i did it myself with pink noise first, but if you enter this in the EQ section, you get a highly rising curve. I guess it should be white noise instead of pink noise... or my studio is completely off ;) both possible.
Adding 6db of boost to that speaker could cause problems with the speaker/amp pushing it past its limits. Wouldn't it have been better to EQ the frequency response lower to match with the lower output of the woofer? Yes you do reduce the overall SPL out of the speaker but it's better than having high distortion coming from the speakers woofer or worse the woofer hitting its Xmax and clip.
I follow you on TikTok! Found your TH-cam here organically!
Great. Will def have to do this in my studio asap. I have got a question though. I just looked up the MiniDSP box. Is there an option for non block type input and output connections?
:( Unfortunately no.
@@kaliaudio seems odd to not have that.
I noticed that he saves the files as "Right monitor" and later loads it in the DSP for output 2.
That suggests that you will have to do these measurements for both speakers separately?
(Which seems logical.)
Also: I don't hear a word about latency, usually present when using DSP processing.
Correct. We recommend measuring one speaker at a time. Latency is an issue, but the MiniDSP is pretty good, and it's only applying 5 bands of EQ, so it's not a HUGE issue.
@@kaliaudio QUESTION: "Adding 6db of boost to that speaker could cause problems with the speaker/amp pushing it past its limits. Wouldn't it have been better to EQ the frequency response lower to match with the lower output of the woofer? Yes you do reduce the overall SPL out of the speaker but it's better than having high distortion coming from the speakers woofer or worse the woofer hitting its Xmax and clip." Thanks!
@@mauiztic either way you're taking away headroom from the system. Both approaches have the same limits.
okay this was a long one but it was a great video. one question is the box they are using only for passive speakers? or can I cut a xlr cable out and match the inputs to that? im assuming its for a passive system.
This is a powered system. Yes, cut the cables.
@@kaliaudio oooo now i gotta make a video on doing this too XD also saw your speakers on a bunch of channels now i hear you are kicking butt. Im going to have to try some out soon!
In the DAW yes, for me, Kali are making the best sounding and most affordable speakers out there! $300 for a pair of great sounding powered speakers is truly amazing!
I would like to see the difference in response trace between a single mic on a stand in the middle of the critical listening position versus this moving average. My suspicion is that there would be no real difference.
Whats the difference with this calibration and the calibration of a software like Sonarworks Reference 4?
will this cause latency? is it usable when you play instruments and monitoring at the same time?
Cant buy MiniDSP balanced anymore due to parts shortage, any options?
isnt it simply easier to use Sonarworks or ARC since they do the same thing?
Great vid as always Warren. I'm very interested in trying this out. But, it's killing me and I have to ask, why are his Kali's upside down?!
GREAT QUESTION! Erik, who works in this room, likes to switch between his Kalis and his Focals. With the Kalis right side up on top of the Focals, the tweeters are so far apart that it's distracting. The waveguide on the Kalis is asymmetrical, so the imaging will suffer if the Kalis are on their sides. SO, in order to get the tweeters closer, we recommended that Erik flip them upside down, which preserves the imaging and achieves his goal.
@@kaliaudio Wow, thanks for the explanation!
I have to say, I love my Kalis, though my purchasing experience was awful through Amazon (not your fault at all!). I have one black and one white monitor because they sent me (and I subsequently returned) 5 SPEAKERS over the past month, trying to get a single black Kali LP6.
So, my studio looks... interesting with one white and one black monitor. They SOUND AMAZING!
Where can I buy this box, mic and software?
I know there's too much low end in my room and would love to calibrate it. You should really do the inexpensive version with an sm57 and give out the microphone frequency response for people who can't calibrate for themselves... cause i mean, everyone watching this channel has a 57 and if they don't they should
You really need a omni condenser mic. A 57 isn't sensitive enough and it's quite directional
@@thekramdens Fair point. Turns out i can just buy an omni capsule for my nt5 so thanks for the eureka moment there. I wonder would calibrating with a 57 be better than nothing at all though?
@@johndcoffee632 I'd avoid it. Your ears and a 57 hear things very differently.
Behringer and Dayton Audio both sell Omni measurement mics for less than it would cost to get an omni capsule for the NT5. As handy as that would be to have anyway, it's not designed to be flat and provided with a calibration file.
There's a right tool for every job. :-)
Thanks for the advice guys, much appreciated
Im trying to calibrate but need more specifics of this software setup?
I have behringer mic and software but what screen is constant calibration. Can someone detail please? Thanks
So good
Harmony Studios Thanks ever so much!!
Thanks!
Thank you so much, great!
Great!!
Thanks ever so much!
is there any way of using this DSP with active monitors? or am i just blind because it seems like its designed for passive ones...
Hi Bernossi the monitors here are active!
@@Producelikeapro ok i think im getting something wrong here because the mini dsp youre using does not seem to use XLR or a 6,35mm jack.
@@Producelikeapro so how do i hook this up with the xlr outs of my subwoofer/monitor chain.
@@bernossi5185 if you check the accessories page you can find Phoenix to XLR cables for this
@@bernossi5185 it uses phoenix connectors, which you connect any balanced cable to with a little bit of slicing and splicing. Sweetwater also sells adapters if you don't want to make them yourself. For power, it uses a DC adapter. It's controlled via USB while connected to a computer, but it keeps doing its thing once disconnected.
This video convinced me that just paying for sound ID is worth it.
Just to be clear...the chain is Interface > MiniDSP > Powered Speakers?
What cables are used to connect these? It appears like the MiniDSP unit you have has a regular speaker cable connection. The output of my interface is either 1/4' or XLR...I'm not sure how you put the MiniDSP in the middle of this.
Phoenix style connectors. You'll have to get XLR to Phoenix adapters. They aren't too expensive.
Am I right suspecting you were using a Behringer MIc ?
Who said that? Could be any mic!
Dayton Audio has calibrated omnis too. :-)
@@nickwallette6201 Just because the calibration file started with " Beh"
guys I can't find those 3 links :( am I missing something?
13:35 Where are those "4-inch monitors"? :D
"Full Range." But you're right; it totally sounds like "4 Inch"
Thank you so much for the information, super amazing
You’re very welcome!
what mic is being used?
Ultra interesting Episode i loved it ! :)
I´m confused, tehy are making the calibration with pink noise, I´ve always made it with a freq sweep, can someone give me some light on this? I´m not using dsp though
Awesome mate
What i do not understand fully is that you are basically saying that it is not good to boost up around 60hz because it'd take too much headroom of the speakers and saying that using a subwoofer instead is better but then the 60hz range is boosted up with the dsp EQing software. Am I missing something? Great Video anyways!