lgjr
Minor Hero
Posts: 47
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Post by lgjr on Feb 19, 2024 8:54:26 GMT -5
Thanks for the advice, but; it's a moot point now. Dropped the umik2 and the mic head came off and won't reattach. Not sure it would measure properly even if it did.
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Post by webmst007 on Feb 19, 2024 22:56:09 GMT -5
the UMIK-2 has a screw-on top mic assembly cone with a spring-loaded piston contact underneath that which pressure-connects to the top when correctly attached. Which part have you broken? Pretty sure Minidsp could replace the top cone assembly, but you would probably need a new calibration done.
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lgjr
Minor Hero
Posts: 47
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Post by lgjr on Feb 20, 2024 17:29:48 GMT -5
The top cone came off and the spring loaded piston just pushes it off. I think it's unrepairable. Don't see enough thread to catch the internal threads. Might try super glue. More likely to purchase a new one.
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Post by webmst007 on Feb 21, 2024 1:30:55 GMT -5
If the thread won't engage you could try adding plumbers tape to the thread to bulk it out or a combo of the tape and some suitable bonding agent as a mix. Or you could just buy a new one 🤪🤪
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Lsc
Emo VIPs
Posts: 3,384
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Post by Lsc on Feb 21, 2024 10:35:59 GMT -5
For whatever reason, firmware 3.2 made changed my unit to say that I have a RMC-1 and couldn’t connect to Dirac.
I had to revert back to firmware 3.1 after restoring from backup to make my unit read XMC-2 again and I can finally use Dirac.
First time since I bought the XMC2 that I’ve been frustrated. I put a service email to Emotiva in December and the communication was it was on Dirac’s side (the back and forth took 2 months for a simple data validation issue).
What’s weird is that I didn’t hear anyone else having this issue. Very weird.
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Post by marcl on Feb 21, 2024 10:49:10 GMT -5
For whatever reason, firmware 3.2 made changed my unit to say that I have a RMC-1 and couldn’t connect to Dirac. I had to revert back to firmware 3.1 after restoring from backup to make my unit read XMC-2 again and I can finally use Dirac. First time since I bought the XMC2 that I’ve been frustrated. I put a service email to Emotiva in December and the communication was it was on Dirac’s side (the back and forth took 2 months for a simple data validation issue). What’s weird is that I didn’t hear anyone else having this issue. Very weird. I would advise using the latest release which is 3.8.2. Yes that's a LONG way from 3.2. In fact I'm using the beta 3.9.1. Both work fine with my XMC-2. Be sure to always, always uninstall the old version of Dirac before installing a new version. 3.8.2www.dirac.com/live/faq/helpdesk/software-changelog/3.9.1www.dirac.com/live/faq/helpdesk/dirac-live-beta-releases/
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Post by marcl on Feb 22, 2024 8:05:03 GMT -5
Reposting ...
Listen to what he says about DLBC AND DLBM.
Notes: Start at 5:55 ..."Dirac Live corrects speakers individually" ... 9:48 with DLBC "not doing any peaks or dips ... just using all-pass filters ... just adjusting phase at different frequencies" ... 10:30 optimizing subwoofers of different models ... in Denon/Marantz implementation DLBC does all the Bass Management; all speakers have to be Small; can't send LFE to Large fronts ... 15:58 in the pairs of speakers (L/R fronts, L/R surrounds), in the crossover region, makes sure the mono material (bass) sums up in phase .... 17:54 the difference between DL Bass Control and DL Bass Management; DLBM applies the 4th order L-R crossover filters for bass management; (do all processors work this way, or just Denon/Marantz? Will Emotiva work this way?) ... 23:00 see how Dirac now shows the result of Bass Management so you can tweak the crossover and see the result ... 24:15 go from DLBM to DLBC and see how Bass Control applies all-pass filters, calculates new filters, iterates for fewest variations in bass response 31:45 benefit with single subwoofer? ... if there is a cancellation at the crossover point, DLBC can correct that 38:52 visualize how DLBC reduces seat to seat variation
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Post by PaulBe on Feb 22, 2024 10:26:53 GMT -5
Reposting ... Listen to what he says about DLBC AND DLBM. Notes: Start at 5:55 ..."Dirac Live corrects speakers individually" ... 9:48 with DLBC "not doing any peaks or dips ... just using all-pass filters ... just adjusting phase at different frequencies" ... 10:30 optimizing subwoofers of different models ... in Denon/Marantz implementation DLBC does all the Bass Management; all speakers have to be Small; can't send LFE to Large fronts ... 15:58 in the pairs of speakers (L/R fronts, L/R surrounds), in the crossover region, makes sure the mono material (bass) sums up in phase .... 17:54 the difference between DL Bass Control and DL Bass Management; DLBM applies the 4th order L-R crossover filters for bass management; (do all processors work this way, or just Denon/Marantz? Will Emotiva work this way?) ... 23:00 see how Dirac now shows the result of Bass Management so you can tweak the crossover and see the result ... 24:15 go from DLBM to DLBC and see how Bass Control applies all-pass filters, calculates new filters, iterates for fewest variations in bass response 31:45 benefit with single subwoofer? ... if there is a cancellation at the crossover point, DLBC can correct that 38:52 visualize how DLBC reduces seat to seat variation Good video. DLBC will have limited use in the XMC+ and RMC-1L+ processors; though an outboard DSP with allpass filters, and/or Subs with built-in variable allpass filters would do the job. The RMC1+ processor will need a Bass expansion module to make DLBC as useful as intended. Emotiva could add allpass filters to the EQ section. It's not that difficult. I have them in my DSP crossovers. The stochastic nature of the Dirac program helps me understand why people spend so much time tweeking this program. The Emotiva processor line already offers similar BM as in DLBM with 4th order L-R crossover filters for bass management. Control is more basic, but it is there. It needs a bit more refinement. I give Dirac some one handed applause for their marketing approach. Do they really have to create DLBC, a separate cost addition, just to add allpass phase control?
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Post by marcl on Feb 22, 2024 12:41:00 GMT -5
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Post by marcl on Feb 23, 2024 13:09:42 GMT -5
From Toole: "The "room correction' process is based on the belief that a room curve is the definitive statement of sound quality. Consequently, the processors perform equalization corrections, including non-minimum-phase acoustical interference irregularities, in order to hit a specific target curve. Doing so arbitrarily modifies the loudspeaker, conceivably degrading a very good one. If a high resolution "corrected" room curve looks superbly smooth, there is a possibility that something inappropriate has been done." - Sound Reproduction Third Edition, Chapter 13, page 372.
Dirac has said in multiple forums that the reason they need multiple measurements is to determine which behavior is minimum phase and which is non-minimum phase. And they do not attempt to correct non-minimum phase behavior. So what I'm saying relates directly to what Toole said. He was not talking about room correction systems that do not attempt to correct non-minimum phase behavior, presumably because at the time his only experience was with systems that attempted to correct everything.
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Post by marcl on Feb 23, 2024 13:47:28 GMT -5
Flavio works for Dirac and is active in forums. he often replies personally to support tickets. Here's an explanation of how Dirac addresses non-minimum phase aspects of room response: "Yes, we use a patented solution that uses both FIR and IIR filters. Let's consider three signals, an ideal impulse, an ideal white noise, and an ideal sine sweep. All of them have a perfectly flat spectrum, that is, they contain an equal amount of all frequencies from DC to infinity. The phase information is all the difference between these signals, and it tells us about when does each frequency arrive. Now, a speaker in a room can be measured, and showed to have some frequency response. What we try to do with Dirac live is to improve this response in frequency (to have the desired target) and in time (we want the impulse response to be as close to a perfect impulse as possible). When considering only the frequency response we can (in theory) apply any filter that have the inverse frequency response and the resulting response will be flat. This filter is often a minimum phase filter, and if the system (the speaker together with the room) was already minimum phase, the result will be a flat frequency response and the impulse response will be perfect. If the system is not minimum phase (and this is the case in our normal listening rooms) the frequency response will still be flat, but the impulse response will not be ideal, (exactly how it looks will depend on the phase). In order to get the impulse response correct for a non minimum phase system you need to use a filter that is not minimum phase. What this filter will do is shift certain frequencies in time in such a way that they all arrive at the same time, thereby achieving the desired result. You'll find all details available to public here: www.dirac.com/wp-content/uploads/2021/09/On-equalization-filters.pdfBest regards, Flavio"
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Post by marcl on Feb 23, 2024 15:14:13 GMT -5
This is just funny ... I was curious as to whether Floyd Toole had ever addressed impulse response correction as part of digital room correction. I typed "Floyd Toole impulse response correction" into the Goog ... and THIS was the first link that it returned: Funny on a couple more levels ... what this guy says makes no sense in terms of frequency response and impulse response being the same thing. And, I checked sections 7.6.4 and 7.6.5 in Toole's book and the topic is the Hass Effect (Precedence Effect) which has to do with the perception of image location as a function of time delay with two sound sources. And he didn't even watch the whole referenced video, which is an interview with Mathias Johansson from Dirac on Home Theater Geeks in 2015. He bailed at 32min of a 73 min interview. He bailed just as Johansson was describing the unique mixed phase approach that Dirac uses. twit.tv/shows/home-theater-geeks/episodes/269So Dirac corrects impulse response of each speaker. If anyone can point me to a reference where Toole discusses this topic in any context, I'd appreciate it.
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Post by PaulBe on Feb 23, 2024 15:36:19 GMT -5
This is just funny ... I was curious as to whether Floyd Toole had ever addressed impulse response correction as part of digital room correction. I typed "Floyd Toole impulse response correction" into the Goog ... and THIS was the first link that it returned: View AttachmentFunny on a couple more levels ... what this guy says makes no sense in terms of frequency response and impulse response being the same thing. And, I checked sections 7.6.4 and 7.6.5 in Toole's book and the topic is the Hass Effect (Precedence Effect) which has to do with the perception of image location as a function of time delay with two sound sources. And he didn't even watch the whole referenced video, which is an interview with Mathias Johansson from Dirac on Home Theater Geeks in 2015. He bailed at 32min of a 73 min interview. He bailed just as Johansson was describing the unique mixed phase approach that Dirac uses. twit.tv/shows/home-theater-geeks/episodes/269So Dirac corrects impulse response of each speaker. If anyone can point me to a reference where Toole discusses this topic in any context, I'd appreciate it. My My. You and Poes are sure on a Toole bashing spree. My guess is both you will be eating crow pie. Lots of Crow Pie...
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KeithL
Administrator
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Post by KeithL on Feb 23, 2024 15:39:40 GMT -5
There's one thing that needs to at least be mentioned... It is IMPOSSIBLE to completely "neutralize" or "compensate for" room acoustics... And, likewise, it is impossible to completely compensate for how your speakers sound and how they interact with the room... If you look at ANY loudspeaker, you will measure a certain frequency response, and a certain phase response, directly in front of the speaker. More specifically even measurements taken "right in front of the speaker" will vary widely depending on exactly where you put the measurement microphone. And, if you start taking the same measurements at various angles off-axis, they will vary wildly. Now add in a room... Which has a floor and ceiling and walls, each at a different distance from the speaker, and each with different acoustic properties. In other words, each wall in your room has an absorption spectrum (equivalent to the frequency response it imparts to sound that reflects from it). And, even worse, that response may vary depending on the angle at which the sound hits that surface. (Imagine trying to calculate the frequency response and directionality of sound that comes from your tweeter, then bounces off a glass door, and diffuses after hitting a stucco wall...) This is why two different speakers, with quite similar "on axis response", may sound very different in the same room... And why the same speaker may sound different in rooms with different acoustics... And, in order to accurately characterize this, you would need to take thousands of measurements, and perform an absurd number of calculations... And those measurements would have to include separate measurements of not only every speaker but also every room surface. And, in the end, you can only make a "one dimensional correction" on the audio signal. You can increase or decrease the amplitude at a certain frequency... And you can "adjust" the phase by adding or removing delay... But you CANNOT make more of that frequency come out of the front of the tweeter while, at the same time, making less come out of the sides (because that is a physical characteristic of the speaker). And you CANNOT make the side walls reflect less at that frequency, while at the same time making the ceiling reflect more of that frequency, unless you actually change the surface of one or the other. And THIS is where the "secret sauce" of room correction comes in... Room correction starts out with relatively few measurements... And it starts out with a specified room and set of speakers... And, in fact, it doesn't even ask you to move your speakers, or to take measurements with your speakers at different locations, or to take measurements of your walls and ceiling. And, on top of all that, it also takes into account both what it can hope to fix and what it expects to be most audible... And, from all that, it determines USEFUL corrections that will actually, with luck, remove or minimize most or all of the imperfections that are likely to be AUDIBLE... That's the reason why getting it right is so difficult... And it's why there are so many different solutions to what at first seems to be the same equation... And it's also why some solutions work better than others in different situations... The reason I gave this such a build-up is this... That's ALSO why it's important to get things like your room, and your speakers, and the location of your speakers, as good as you can FIRST. And even to pay attention to things like your listening position... and the toe-in and distance from the wall of your speakers, and both "intentional" and "incidental" room acoustics. It's not even theoretically possible to have a room correction system that can "fix any room and system so its perfect"... So it's really important to give your room correction solution the best possible point to start from... And, like it or not, you should never assume that some wondrous software is going to save you the trouble of getting the other stuff as close to right as you can. You're always going to end up with a better final result if you start from a better starting point. From Toole: "The "room correction' process is based on the belief that a room curve is the definitive statement of sound quality. Consequently, the processors perform equalization corrections, including non-minimum-phase acoustical interference irregularities, in order to hit a specific target curve. Doing so arbitrarily modifies the loudspeaker, conceivably degrading a very good one. If a high resolution "corrected" room curve looks superbly smooth, there is a possibility that something inappropriate has been done." - Sound Reproduction Third Edition, Chapter 13, page 372. Dirac has said in multiple forums that the reason they need multiple measurements is to determine which behavior is minimum phase and which is non-minimum phase. And they do not attempt to correct non-minimum phase behavior. So what I'm saying relates directly to what Toole said. He was not talking about room correction systems that do not attempt to correct non-minimum phase behavior, presumably because at the time his only experience was with systems that attempted to correct everything.
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Post by marcl on Feb 23, 2024 15:48:07 GMT -5
This is just funny ... I was curious as to whether Floyd Toole had ever addressed impulse response correction as part of digital room correction. I typed "Floyd Toole impulse response correction" into the Goog ... and THIS was the first link that it returned: View AttachmentFunny on a couple more levels ... what this guy says makes no sense in terms of frequency response and impulse response being the same thing. And, I checked sections 7.6.4 and 7.6.5 in Toole's book and the topic is the Hass Effect (Precedence Effect) which has to do with the perception of image location as a function of time delay with two sound sources. And he didn't even watch the whole referenced video, which is an interview with Mathias Johansson from Dirac on Home Theater Geeks in 2015. He bailed at 32min of a 73 min interview. He bailed just as Johansson was describing the unique mixed phase approach that Dirac uses. twit.tv/shows/home-theater-geeks/episodes/269So Dirac corrects impulse response of each speaker. If anyone can point me to a reference where Toole discusses this topic in any context, I'd appreciate it. My My. You and Poes are sure on a Toole bashing spree. My guess is both you will be eating crow pie. Lots of Crow Pie... Specifically what did I say to "bash" Toole? Quote me, please ...
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Post by PaulBe on Feb 23, 2024 16:11:30 GMT -5
Flavio works for Dirac and is active in forums. he often replies personally to support tickets. Here's an explanation of how Dirac addresses non-minimum phase aspects of room response: "Yes, we use a patented solution that uses both FIR and IIR filters. Let's consider three signals, an ideal impulse, an ideal white noise, and an ideal sine sweep. All of them have a perfectly flat spectrum, that is, they contain an equal amount of all frequencies from DC to infinity. The phase information is all the difference between these signals, and it tells us about when does each frequency arrive. Now, a speaker in a room can be measured, and showed to have some frequency response. What we try to do with Dirac live is to improve this response in frequency (to have the desired target) and in time (we want the impulse response to be as close to a perfect impulse as possible). When considering only the frequency response we can (in theory) apply any filter that have the inverse frequency response and the resulting response will be flat. This filter is often a minimum phase filter, and if the system (the speaker together with the room) was already minimum phase, the result will be a flat frequency response and the impulse response will be perfect. If the system is not minimum phase (and this is the case in our normal listening rooms) the frequency response will still be flat, but the impulse response will not be ideal, (exactly how it looks will depend on the phase). In order to get the impulse response correct for a non minimum phase system you need to use a filter that is not minimum phase. What this filter will do is shift certain frequencies in time in such a way that they all arrive at the same time, thereby achieving the desired result. You'll find all details available to public here: www.dirac.com/wp-content/uploads/2021/09/On-equalization-filters.pdfBest regards, Flavio" Impulse response is the inverse of frequency response. One can be derived from the other. If on axis frequency response is good, impulse response is good. Frequency response is correctable with minimum phase filters. What Flavio is describing is exactly what Toole said about “including non-minimum-phase acoustical interference irregularities”. You stated in your post above – the post where you copied my words and Toole’s quote from my post on the G4P board - “Dirac has said in multiple forums that the reason they need multiple measurements is to determine which behavior is minimum phase and which is non-minimum phase. And they do not attempt to correct non-minimum phase behavior.” Obviously, Dirac DOES “attempt to correct non-minimum phase behavior”, with “non-minimum phase correction”. You can’t talk out of both sides of your mouth at the same time. It is true that flat frequency response can have a non-flat phase response, and this is true in all speaker systems – some more, some less. It’s a good thing that our ears don’t require both, or audio communication in any form would be very difficult outside an anechoic chamber. Paraphrasing Toole – our ears don’t respond to waveforms. Very few speaker systems reproduce full band sound within one phase rotation of the original source – which always makes me chuckle when we talk about accuracy. Many good multi-way systems have up to three phase rotations over the full audio band. Our ears also tend to not be concerned about floor bounce, walls in a hall, or objects in a room that produce diffractive non-minimum phase sound. These aberrations need to be really big to create issues. Great Halls have their own sound signatures with TONS of diffractive non-minimum phase sounds. Think what happens when you calibrate your room, and then invite friends over for a movie. How much of your calibration is useful when you now have 3-8 more people in the same room then when you did the calibration? Imagine all the "non-minimum-phase acoustical interference irregularities" sitting around you at a live concert. This all brings me back to Toole’s research conclusion – frequency response is the most important parameter… We might want to differentiate between power and direct frequency response. Reflectivity will matter, but, a system with controlled directivity in a hard surfaced room can have the same frequency response as a wide dispersion system in a room with treatments… Even have the same level of reflected sound, but, the sound signatures of the respective rooms will be different. Imagine all the reflected audio in a room occupied with dipoles. Concerning Flavio: Impulse is corrected by correcting frequency response. Frequency response can be good with non-ideal impulse response – Hmmm. “In order to get the impulse response correct for a non-minimum phase system you need to use a filter that is not minimum phase.” Shifting “certain frequencies in time in such a way that they all arrive at the same time” is adjusting group delay. PEQ affects group delay. Crossover slope affects group delay. The low frequency group delay difference between a ported box and a closed box is large. How does Dirac correct phase of different speakers in a system, with different crossovers and different phase responses? If you make the porridge coming from each speaker taste the same, is consistency achieved? Give me a room that responds as if the MLP is closer than ‘critical distance’ and let the rest lay. As long as direct sound hits my ears at least one wavelength And enough milliseconds before the reflected sound, And with enough attenuation of the reflected sound, I’m good. If Dirac is doing something like this, good for them and us. My 'rule of thumb' remains at 4x Schroeder Frequency as max frequency for useful correction - about 340Hz for me. I'll try DL with the G4P, and set the curtains, IF both are stable. IF they are not stable, I'll close the curtain on Dirac. I look forward to more Dirac Wisdom…
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KeithL
Administrator
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Post by KeithL on Feb 23, 2024 16:15:47 GMT -5
I'm not going within a mile of the actual math... because it's way over my head... But a lot of this confusion has to do with what you meant by "impulse response" and "frequency response"... I'll try not to get any of this too wrong... In mathematical terms an impulse is a pulse of infinite amplitude and infinitely short duration (an "infinitely loud infinitely short click"). And, if I make that click in a room, and record the result, I can know EVERYTHING about the system. And, yes, that includes things like the frequency response, and the phase response, and how ANY sound played through that system will come out. And, if you can correct your system so that its response to that impulse is perfect, then, by definition, all of that other stuff will also be perfect... end... mic drop... we're done. Of course, since "infinitely loud infinitely short pulses" are somewhat impractical we're stuck with approximations... HOWEVER, go look up, "convolvers"... That's where you go to Winchester Cathedral, play a loud click, record the result with a really good microphone, and save a really good copy of the result. You can then analyze that recording and have a COMPLETE description of "what Winchester Cathedral sounds like". And, once you take that "impulse file", you can use it to "process" a recording of yourself playing a guitar in an anechoic chamber so it sounds EXACTLY like you were playing in Winchester Cathedral. And, yes, that software does exist... And, yes, it does work... to varying degrees... (You can even get free plugins, and sample impulse files, to play with yourself.) So, logically, if you have "a perfect impulse file"... (which you can make by calculation)... And you do all the math properly... Then theoretically you could use that file to "make perfect corrections".... (You could measure your room, subtract that from what you have, end up with a perfect "anechoic recording", and then add in the sound of any room you liked.) Of course there are lots of reasons why nobody has actually managed to do that so far... - for one thing an "audio track" is "2D" (so your result would only be perfect at exactly one point) - for another multiple channels make it all a lot more complicated - for another it only works perfectly for linear systems - (and the list of "catches" sort of goes on and on Now, remember where Toole mentioned "the precedence effect"...? That's how those plugins manage to actually deliver a somewhat convincing result... It's a fancy way of saying that "as long as you get the one or two things that our brains lock onto about right we tend to ignore all the small things that aren't right"... If you want a great example just take a look at a balance control... or a "pan pot". If someone is standing on a stage talking while walking back and forth from left to right... They get louder and softer in either your right or left ear... And the loudness of the reflections from the left and right walls that you hear in each ear also vary in loudness... And the various delays of both their voice and those reflections that you hear in each ear also vary... BUT... If I have a monaural recording of a singer, I can "move her from left to right" by simply adjusting JUST the relative levels in each channel. (If you look at a schematic you'll see that the "pan-pot" on an analog mixer JUST changes the amplitudes of the channels.) However, to we who are listening, it sounds like she's moving back and forth, and we fail to notice that the delays are wrong most of the time. This works because, thanks to that precedence effect, your brain decides where she is based mostly on the relative levels, and IGNORES the fact that all of the delays aren't actually right. (A very few people might notice that she sounds "more natural" if we also adjust the delays, and the levels of the echoes, and the delays of the echoes... but it works OK even if we don't bother.) That's a big art of "the secret sauce of room correction".... Deciding what CAN be fixed... and what NEEDS to be fixed... and successfully fixing what needs to be fixed. I should note that, when we talk about "impulse response", we're usually really talking about... "making relatively short pulses that look about right". And we do this by... "making sure that all of the main frequency components arrive at about the right relative times". (Which is an approximation of all of that fancy math that's good enough to work for us mere humans...) This is just funny ... I was curious as to whether Floyd Toole had ever addressed impulse response correction as part of digital room correction. I typed "Floyd Toole impulse response correction" into the Goog ... and THIS was the first link that it returned: View AttachmentFunny on a couple more levels ... what this guy says makes no sense in terms of frequency response and impulse response being the same thing. And, I checked sections 7.6.4 and 7.6.5 in Toole's book and the topic is the Hass Effect (Precedence Effect) which has to do with the perception of image location as a function of time delay with two sound sources. And he didn't even watch the whole referenced video, which is an interview with Mathias Johansson from Dirac on Home Theater Geeks in 2015. He bailed at 32min of a 73 min interview. He bailed just as Johansson was describing the unique mixed phase approach that Dirac uses. twit.tv/shows/home-theater-geeks/episodes/269So Dirac corrects impulse response of each speaker. If anyone can point me to a reference where Toole discusses this topic in any context, I'd appreciate it.
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Post by ttocs on Feb 23, 2024 16:20:08 GMT -5
So Dirac corrects impulse response of each speaker. If anyone can point me to a reference where Toole discusses this topic in any context, I'd appreciate it. I'd like to know this also! I always love reading the findings from Toole.
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Post by marcl on Feb 25, 2024 9:17:20 GMT -5
lrobertson lcseminole ttocs I think it was just you guys commenting on this. Tag others if I forgot somebody. Magic Beans
Okay I looked into this but honestly I'm not going to go back 18 months and watch every video that Joe and Channa made to try to piece together what Joe is trying to accomplish and what (if any) scientific basis he has for his approach to Magic Beans (because I'll bet my magic beans he doesn't). But I think I have enough info to make some comments: What do you do with Magic Beans?- Take near-field measurements (within a range similar to speaker baffle width), and MLP measurements for each speaker using pink noise, and moving the mic around in a pattern at the measurement position. No multiple listening position measurements, just MLP.
- Magic Beans uses this data to (somehow) come up with a response for each speaker.
- You pick a target curve, and Magic Beans generates the coefficients for PEQ filters to correct the response, and they can be imported into various processors (Emotiva included). Similar to how using REW EQ automatically generates filters and gives you coefficients that can be imported.
- You import the PEQ coefficients to your processor.
- If you have the Audyssey version that they reference running in your processor, you can import the data into Audyssey Otherwise you use the PEQ function of your processor.
- You also import the target curve from Magic Beans. You can import this target curve to any system, including Dirac.
What DON'T you do with Magic Beans?- You don't use the filters with Dirac in any way. There is no way to import filter coefficients if you're using Dirac.
- This means, you can't use Dirac for impulse response correction and time alignment ... and Magic Beans for filter coefficients.
- You have to do time alignment some other way, such as manually entering distances or using the automated tool in Audyssey or in some processors.
What is the point of using Magic Beans?Well ... uh ... I mean ... there is no scientific precedent or logic to using near-field measurements for room correction. Taking measurements at multiple listening positions and applying some algorithm to optimize response makes sense. But taking measurements at ONE listening position and combining it somehow with near-field measurements makes no sense. If a speaker had an inherent flaw that revealed itself in a near-field measurement, you could argue that it could be corrected with EQ, and in fact you could even legitimately correct a dip in the bass response if there was one. But it really doesn't matter because it's the listening position response that matters. So ... the only point of using Magic Beans would be if - for whatever personal reason - someone does not LIKE what Dirac does, or does not LIKE what Audyssey does ... then this would be a way to play with an alternate app until they get what they want. p.s. In at least one video Joe talks about impulse response and time alignment and trying to get similar results to Dirac or whatever using REW. Note the following: - Time alignment and impulse response correction are two different things. Dirac does both. There is no way to do impulse response correction without some sophisticated algorithm and filter implementation similar to Dirac. Maybe Trinnov does it, but I know of no other way.
- Yes, you can determine time alignment using the impulse response measurements in REW. These can be converted to distance and used to manually enter distances to a processor. But if you use Dirac, you can't enter distances manually with Emotiva processors.
- In one video Channa talks about how, after doing a Dirac calibration, he used the time alignment files in Spatial Audio Toolkit and (amazed look on his face) Dirac had aligned everything so perfectly that he could not hear the slightest discrepancy in the alignment.
p.p.s. Further evidence that the vast majority of people - even those making YouTube videos - ignore the impulse response correction that Dirac does.
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Post by marcl on Feb 25, 2024 10:17:28 GMT -5
Okay, first ... note that I had omitted something from my previous post so I corrected it. Magic Beans determines a target curve in addition to filter coefficients. You import the target curve to your system as well as the filter coefficients. But if you're using Dirac, you can only import the target curve ... just like you would import a Harman curve or whatever.
Now ... the crux of the biscuit ...
I found this video where Joe is doing a demo for some folks, and most of it ends up being a dialogue with Justin Zazzi, Senior Acoustic Research Scientist/Engineer at Harman X.
Justin does what a respectful scientist/engineer would do. He asks questions, paraphrases, offers analogies ... to try to understand what Joe is saying.
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