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Post by dok on Aug 12, 2024 16:47:56 GMT -6
Yes, you can! In fact, this is exactly why I use the cheapest decent sub with a secondary subwoofer out, the Presonus T10. I think it's like $400. But it only works *with the method I use* because you're using it to activate the room modes. For more on that see the post/link above. The T10 probably doesn't measure great compared to something high end but that's okay in this particular application as long as we're not distorting the crap out of it. Again, you will want to measure with REQ Wizard and make sure that all results (including distortion) are within an acceptable range for you once you have things optimized. I'm reading through all the stuff on that other thread, I have a question. In your screen shot of the Room Simulator it looks like you have "crossover" unchecked. So you're disabling the crossover and letting the monitors run full frequency? Correct! Can't remember if it's Geddes or Tatz who suggests this. If nothing else it's a good starting point, your mileage may vary, and you may find that some implementation of a crossover is worth doing if you want to fine-tune results later, but I think it's worth trying at the outset so you can see how it affects your measurements.
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Post by gravesnumber9 on Aug 12, 2024 16:58:04 GMT -6
I'm reading through all the stuff on that other thread, I have a question. In your screen shot of the Room Simulator it looks like you have "crossover" unchecked. So you're disabling the crossover and letting the monitors run full frequency? Correct! Can't remember if it's Geddes or Tatz who suggests this. If nothing else it's a good starting point, your mileage may vary, and you may find that some implementation of a crossover is worth doing if you want to fine-tune results later, but I think it's worth trying at the outset so you can see how it affects your measurements. I mean that makes sense. If the idea is to excite the room to fill gaps.
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ericn
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Balance Engineer
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Post by ericn on Aug 12, 2024 19:12:58 GMT -6
Correct! Can't remember if it's Geddes or Tatz who suggests this. If nothing else it's a good starting point, your mileage may vary, and you may find that some implementation of a crossover is worth doing if you want to fine-tune results later, but I think it's worth trying at the outset so you can see how it affects your measurements. I mean that makes sense. If the idea is to excite the room to fill gaps. It’s a Theory, read the papers but unless you are a huge fan of comb filtering in your low end, no. Those pesky laws of physics have this way of showing up when you don’t them to. Anyone who has experience with large scale speaker systems will shake their head at this idea. Hear is your issue basic phase cancellation during summation. We are going to go back to physics class to diagphrams within one wavelength are going to sum together in the case of LF consider we are talking over 9 meters! So if we have 2 different subs within 32 feet of each other and less than 9 meters apart they are going to sum and all the phase level and distortion differences are going to create all kinds of comb filtering. Even the simple differences in latency because of different DSP is going to cause issues. Why do you think the original customers of ATC & Volt where live speaker manufacturers? Because when you array say 30 of the same driver you need consistency from driver to driver! Here is a very basic experiment simply compare the LF of your monitors with a bit of EQ using 2 different EQ’s say an API and an SSL as close as you can to the same EQ point. Now listen to each channel separately and then together, what you hear is comb filtering. Can you correct it with DSP? Well you can get close, but your going to need a ch of correction for each monitor ch and each sub because if you try to correct it via one simple stereo feed you won’t be able to correct the timing phase difference between the different sub. . I learned this lesson working for some regional SR companies as a kid, guys on the cheap would have some cabinets with JBL some EV and probably something like Fane they would all sound different, system symmetry just made things easier, years later I learned why. Now there is of course one exception to this rule, different models of Meyers will sum because every product Meyer has ever made has the same phase and frequency response within it’s bandwidth, even Hudsonics Aimie’s will sum with a UPA1 or M1D.
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Post by Bat Lanyard on Aug 12, 2024 21:14:21 GMT -6
Alright, happy to share the method I use - I only ask that it's received in the spirit in which it was intended, which is to cheerfully outline a process to determine speaker, subwoofer, and listening position placement that is measurable and reproduce-able. I don't make any claims that it's better than any other system, only the fact that it *is* a system, compared to much of the advice you'll find online which is little better than guessing, but I've learned that especially the theory of subwoofer application used here really tends to get some people riled up. And I'm happy to answer any questions I've learned, but I really don't want to get deep into the weeds about *whether* it works - some of it is rather counterintuitive, but it always works for me and maybe it will work for you! Essentially this is a dumbed down, DIY version of Carl Tatz's Phantom Focus system - I am not affiliated with him whatsoever, but I am extremely grateful that he has posted the basics of his methodology for free on his website here: www.phantomfocus.com/Default.asp. If you read my instructions below and you want to take things to the next, actual professional level, you should 100% get in touch with him and I have no doubt it would be worth every penny. Someday when I own a permanent space that's exactly what I'm going to do. Thanks, Carl! With those disclaimers out of the way, here it is: Dok's guide to monitor placement in your HOME recording studio And, Carl makes the greatest studio chair ever.
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Post by dok on Aug 12, 2024 21:34:56 GMT -6
Well, like I say, it's important to measure your results and determine whether they work for you. It's easy enough to measure and also likely, if one is able to place everything in an ideal position with respect to the axial modes, that one would be in a better position to understand what needs to/can actually be done to improve response at the listening position. Some degree of comb filtering is going to happen any time you have more than one speaker as far as I understand it, but Geddes's protocol* actually states that the subs a) do not need to be identical, b) are not meant to be working or positioned in stereo (rather instead semi-randomly), and c) are also not meant to operate at identical output levels. It's a different way of doing things and feels very counterintuitive! And that's all I'll say by way of supporting this idea, your mileage may vary. Cheers! p.s. it's still valid to further time-align the subs in this protocol with something like MiniDSP and some people do, but that is a deeper dive for sure, and we are trying to see how far we can get *before* we start getting super granular like that. *For further reading: mehlau.net/audio/multisub_geddes/
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Post by Johnkenn on Aug 16, 2024 15:27:53 GMT -6
I just did another calibration of my room with Sonarworks...so I'm bypassing the trinnov and listening to SW, then I can flip flop and listen to the Trinnov. And it's really no comparison. The SW sounds great - but the stereo field gets wider and the bottom and top come alive. I wish I could get away with it, but it's one of those things where - don't listen to it because then you'll have to buy it.
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Post by sirthought on Oct 10, 2024 2:13:36 GMT -6
Not sure if this has been discussed on the board already... www.sonarworks.com/soundid-reference/integrationsLooks like Sonarworks has worked hard to set itself up as a big integration partner with several hardware companies. This will compete against the likes of a separate device like Trinnov or ARC or whatever else they want to sell as an added box. UAD just announced this with their newest interfaces, but it's already going with RME, Lynx, Audient ORIA, AVID MTRX, DAD 32, Merging Anubis, and DSP monitor speakers from Fluid Audio, Wayne Jones Audio, Adam Audio A Series, and Dolby Mastering Suite. This will be running the Soundreference EQ curves right on the devices' DSP, which means low latency that you can run it full time. Once you are set up you don't have to run Sonarworks system wide or have the plugin in your DAW. No more worries about forgetting to take it off the DAW mix before you print. I stopped using it a while ago just for headphones, so I'm not familiar with any updates that might impact the differences between Trinnov and Sonarworks. Just know you don't have think about different converter boxes or extra plugins.
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Post by mcirish on Oct 10, 2024 6:38:56 GMT -6
I had saw that a few months ago and looked into having it run on my Lynx Aurora (n). The problem is that it's always on, so if I switch to a different set of speakers (Dangerous Monitor ST), the wrong profile will be applied. I'm using Arc Studio now and I have it AFTER the monitor controller and right before the speakers. It is only correcting my main set of speakers. That works well for me. I still have my license for Sound ID but a friend is using it now, as I haven't touched it this year.
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Post by sirthought on Oct 10, 2024 12:14:09 GMT -6
So you are using the ARC dac instead of the Lynx or Dangerous? How are you liking that?
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Post by mcirish on Oct 10, 2024 12:21:06 GMT -6
The Arc is not messing with the sound even though it's an extra AD/DA conversion. I really didn't notice a sonic change, other than the room correction. I'm very happy with it at this point. Sonarworks crashed Nuendo too often but I used it till Arc Studio came out. Trinnov looks great but latency would be an issue for me. The price is out of my range currently too.
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Post by sirthought on Oct 10, 2024 12:25:01 GMT -6
With the new option Sonarworks wouldn't run a plugin in the DAW. So hopefully no crashes.
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Post by craigmorris74 on Oct 10, 2024 12:53:55 GMT -6
I wish Sonarworks or ARC would do time domain correction too.
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Post by sirthought on Oct 10, 2024 13:38:28 GMT -6
I wish Sonarworks or ARC would do time domain correction too. It's not available yet, but the way UAD are talking about it with the new Apollo stuff, it sounds like Sonarworks will. I'm not aware if this is a change with all of these various DSP integrations or just the one with UAD. UAD is implementing a new bass management feature that will filter certain frequencies going to your satellites vs a sub, and then they say the Sonarworks will handle time alignment.
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Post by phantom on Oct 10, 2024 14:42:30 GMT -6
I wish Sonarworks or ARC would do time domain correction too. It's not available yet, but the way UAD are talking about it with the new Apollo stuff, it sounds like Sonarworks will. I'm not aware if this is a change with all of these various DSP integrations or just the one with UAD. UAD is implementing a new bass management feature that will filter certain frequencies going to your satellites vs a sub, and then they say the Sonarworks will handle time alignment. I don't know if this will change anything. Sonarworks and UA might be referencing the usual method of correction from Sonarworks as "time alignment". Because what I understood is that it's the same Sonarworks correction, even the same mic and the usual calibration file, that you upload on UA Console, with a new plugin or whatever there. What is confirmed to be different is that the eq correction will be made using UAD filters, not the usual ones from SoundID. I believe that this will help with the sound - I have never liked SoundID correction.
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Post by Johnkenn on Oct 10, 2024 14:51:02 GMT -6
Yeah, don’t think it’s doing any phase correction.
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Post by Johnkenn on Oct 10, 2024 14:51:45 GMT -6
Now that I’m thinking about it - maybe phase correction is the wrong term.
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ericn
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Balance Engineer
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Post by ericn on Oct 11, 2024 11:10:48 GMT -6
Now that I’m thinking about it - maybe phase correction is the wrong term. The problem, with phase correction is it is very DSP intensive lots of little delays with a band delay.
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Post by Tbone81 on Oct 11, 2024 11:23:29 GMT -6
Not sure if this has been discussed on the board already... www.sonarworks.com/soundid-reference/integrationsLooks like Sonarworks has worked hard to set itself up as a big integration partner with several hardware companies. This will compete against the likes of a separate device like Trinnov or ARC or whatever else they want to sell as an added box. UAD just announced this with their newest interfaces, but it's already going with RME, Lynx, Audient ORIA, AVID MTRX, DAD 32, Merging Anubis, and DSP monitor speakers from Fluid Audio, Wayne Jones Audio, Adam Audio A Series, and Dolby Mastering Suite. This will be running the Soundreference EQ curves right on the devices' DSP, which means low latency that you can run it full time. Once you are set up you don't have to run Sonarworks system wide or have the plugin in your DAW. No more worries about forgetting to take it off the DAW mix before you print. I stopped using it a while ago just for headphones, so I'm not familiar with any updates that might impact the differences between Trinnov and Sonarworks. Just know you don't have think about different converter boxes or extra plugins. The hardware integration is one of the main reasons I went with SW over the others. Trinnov is so expensive, probably worth it, but damn my wallet won’t let me do it. Where as I’m looking forward to more and more monitors and interfaces integrating SW directly. It should open up a lot more options, especially for folks who use multiple sets of monitors (which is most of us I’d guess).
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