I just wanna say thanks. This has transformed my system into something truly respectable. I multiplied the fvba filter with the inversion convolution filter I made from your other video. Not sure if that’s correct, but damn. Using equalizer apo I can quickly switch between filter and no filter and the no filter version is practically unlistenable… mind you I scratch-built some mark audio chr-120 full rangers mounted in a decware dna horn with spherical wavefront horns loading the front. Needless to say they needed some midrange attenuation. I was researching to build some notch filters when I found your channel. Thank god 😂 This is such a superior solution. I found the bass I was missing. The audio is so clear it’s almost tangeable. Imaging is like nothing I’ve ever heard! You could never get this from headphones. Thanks again. 🎉
Fvba x Inversion filter will work very similarly to the method in the video, they both allow 5dB boost but I found 1/48 auto-eq to yield just a little bit higher c80 clarity results than inversion in my tests although it could be specific to my gear/room.
My listening room is untreated so I like the idea of convolution. I was having a hard time getting the vba to move from 28 to 25 so I just left it. Bass is noticeably better. I will say that listening to hakkerskylder by heilung I could definitely hear the filter. I’m hoping it’s because I didn’t move the filter far enough. Either way, that’s one song. Also, my initial testing showed that one of my speakers ir was inverted so I switched the cables (think it’s my cheap gear). Can this be fixed in rew?
If one speaker is connected in opposite polarity relative to the other, you will experience a very audible drop in bass response. If that's not the case and you see their impulse peaks in opposite directions in REW, this could be a mic position issue and must be left alone. Rew can invert the polarity of a speaker response virtually on the charts but that's all. Check their phase responses to see in which polarity left and right phase responses are more similar to each other.
I’m gonna redo my measurements and see if there’s any difference. Sort of off topic… but Is there any way to adjust woofer Q using filters? That would really simplify subwoofer choices.
It's in fact Dr. Uli of Acourate who came up first with the VBA idea. I just developed it further: www.audiovero.de/acourateforum/viewtopic.php?p=177&hilit=VBA#p177
Quite great video!!! Let me ask two questions: - why dont you use linear phase correction? Why minimum phase over linear phase? - In my case and I suppose many other cases, I would like to have as minimum delay for convolution. Will you have any approach to load the final gilter to rew and work with number of taps to try to reduce the lengh until a few ms keeping as much as possible correction? Knowing it wont be perfect, but pretty accurate. I think doing filters with small delay is very interesting topic for a new tutorial, maybe there are some tricks to achieve best results with minimum taps/delay. Thank you for your videos and channel ;)
Minimum phase IIR filters are more efficient in the bass frequencies. The phase changes they cause are often useful (ie corrects the phase). For optimizing FIR filter number of taps see: th-cam.com/video/zmIqJ_jlZPw/w-d-xo.htmlsi=cCGBFs7WtB6rDq1u&t=1586
Hi, first of all, thank you for all the lessons on your channel. By watching your videos and experimenting a lot myself, I'm getting better at REW. However, I'm struggling with something I can't figure out. I'm trying to combine VBA with an inversion filter. I know you have a different perspective on correction of higher frequencies now, but despite all possible acoustic treatments, my room is still quite challenging, and the inversion filter works best to achieve a precise mid/high stereo image. When I combine VBA with inversion, I either get a ringing filter or the target is not accurately followed. I've tried creating both the VBA and the inversion filter from L0/R0 and then multiplying the filters. I've also tried convolving the VBA first (LVBA and RVBA) and using them as a starting point for the inversion filter. In both cases VBA and inversion work fine separately, but when combined, something goes wrong, even though I pay attention to sufficient window size and impulse alignment while making the filters. Do you know if it's theoretically possible to combine VBA and inversion? Thanks in advance! Any tips on approach / things I can try are welcome. PS: in this video you don't apply Invert Polarity after creating LPHPF. In previous VBA methods, that was necessary. Did I miss something? Regards, Hans, The Netherlands
I have tried hard but also couldn't find an optimal way to combine VBA with other filters in a meaningful manner. Physical double bass arrays also suffer from similar issues and none are perfect. It seems MIMO systems are the way to go for proper cancellation of room modes (although both ART and Trinnov keep postponing launching). PS I believe you are using the inversion method in the Mastering Digital Room Correction video. It sounds noticeably better than the method in the older REW inversion video
@@ocaudiophile wow, that is a quick response:) The vid you mention is inversion below schreuder. I my room I need the full range inversion filter, like in your video "Convolution with Inversion (no EQ filters, all FIR!)" But as far as I understand from you it is hard to combine other filters with VBA. Well, at least now I now I am not the only one who runs into troubles. But I keep trying and experimenting, maybe one day... :) Thanks again!
JBL 306p MkII + LSR310s is the computer desktop setup I am working with through this tutorial, in a medium-sized bedroom (~5m wide x 4m deep x 3m tall), not centered left-to-right. Stereo balanced output from DAC going to sub, crossover done in sub and balanced outputs to speakers. Crossover switch on sub is at 80Hz. I have three main bass peaks on both channels: both have a +20dB peak at 40Hz, +7dB peak at 58Hz, and then L has +9dB at 104Hz and R has +8dB at 139Hz. Based on the tutorial, I am working on the main 40Hz peak with VBA, and I'm getting great simulated results from it, but the "match range" instruction at 27:40 is limiting my filter's range to just 79Hz and thus not touching the L 104Hz and R 139Hz peaks. Would there be any issue using a higher "match range" upper limit to capture the 104Hz and 139Hz peaks for the filter?
@@ocaudiophile Thanks, I will give that a try today, I'm sure it will sound even better than my initial filter. As it stands, I can't believe how much better the measured step response was! Audibly the change was a little subtle (makes sense, since it was mainly working on 80Hz and below), but once I picked up on it I couldn't un-hear it in bass-heavy attacks like kick drums, guitar, etc., specifically the decay after the initial attack. Only aspect that wasn't a clear measured improvement was impulse response (first peak wasn't quite as high as unfiltered, but decay was better). I might have missed an alignment somewhere so I'm going to start from the baseline measurements and build it again. Of course measurement variation (mic placement etc.) could be at fault for that. I'm also going to change my target curve, the curve I made was much flatter than Harman in bass and sounded a little weak (I ended up adding a +3dB low shelf in my DSP app below 150Hz to get it to taste). I know you're active on ASR, which thread would be a good place to share my measurements?
Are you comparing "normalized" impulse plots? You can share mdat files here if you upload them in google drive. In ASR there's no particular threat but you can start one ;)@@MikeyAntonakakis
@@ocaudiophile Actually I just looked again, and my comment about IR was wrong - very similar between before/after but slightly higher peak after the filter, and no initial "drop" before the main peak, just like in the video. I'll start a thread on ASR and tag you, it will be a little easier than TH-cam comments :)
Very nice explanation. I have done both the v3and v3x for calculations. I think v3x is better. your eq setting makes audyssey next level. I think it is very close if not better than Dirac live. Trionove on the other hand, doesn’t suit most fans as most of us don’t have a dedicated room for movie. Living room setting is 90% of the cases I guess
Three questions please : 1. In addition to aiming to correct anomalies introduced by the room reflections, does this also correct frequency anomalies in the direct sound from the speakers, in line with the target curve? 2. Does this method also reduce the difference in arrival time of low and high frequencies, at the listening position? 3. Will the end result be accurate only at one listening position, but at more than one listening position? I must add, I have been following DRC developments for about 8 years, started to use REW, 1st with EQ Filters in EQ plugins, and later on generating IIR correction filters for use in convolution plugins, which yielded some improvements. Especially in the most recent 3 or 4 years, it appears that there has been an exponential increase in understanding, from independent sources such as this TH-cam channel. So now we have good enough tools such as microphones with calibration, at reasonable prices, and much improved versions of REW, and now "teachers" such as this TH-cam channel, the challenge is : We need some kind of roadmap, for learners, to take them step by step from elementary principles such as how to measure, e.g all the different approaches for such elementary tasks, and then build up gradually, to advanced methods, such as used in this video. I would happily pay £20 to £25, for such a book, which provided this guidance using REW. If I understood enough about this, I'd also be happy to work with anyone interested, to create such a book. Another chap MItch (AccurateSound.ca) has written such a book a few years ago, but it focused on using tools like Acourate - furthermore I discovered there was a tendency in his approach to steer the conversation towards engaging his services, and buying his convolution tools - nothing wrong with a bit of commerce - he has to eat, but that approach led me down a path where I was asked to trust one man's intuition, without really understanding what he was doing. What happens when he dies, which will eventually happen?!!!, i.e. his methods did not seem to me to be independently verifiable, cos I was forced to use proprietary tools and methods which cost a lot of money, and I would have had to invest so much in tools like Acourate, and try them out at great expense, to discover if Mitch's methods actually worked. I love videos, but thankfully TH-cam allows me to slow down the speech, cos its a bit too fast for me to comprehend otherwise. My thinking is that this level of complexity deserves capture in a book, that is revised maybe every 2 or 3 years, to keep it current. I'd happily pay £25 every 2 years for such a book. A book is also a kind of insurance. Who says TH-cam will still exist, in 5 years from now !!!????. With a good book, teaching all of these various methods, from elementary topics to advanced methods, and with good explanations, such information is then available as a keepsake. This is just my way of saying to the Obsessive Compulsive Audiophile channel producer - you clearly have a lot of ideas, but it is worth putting these down in a book, with very good explanations and justifications of why you have chosen each step, and what the various options are (at least a few), especially one which fcusses on using easly available tools like REW(and Rephase), is what most people would be highly interested in, cos it avoids leading us down a black hole path, to custom highly proprietary tools like Acourate, where one has no real clue what is going on in there. There is so much confusion in the DRC business, starting with simple things like - do I point the microphone up to the ceiling, when I take measurements, or do I point it straight ahead towards the center point between the speakers, or do I point the microphone at each speaker, and what calibration file do I use - Zero degree or 90 degree or what other kind of calibration, maybe 30 degrees. To some of you, this might be elementary, but that is the challenge with todays DRC with tools like REW and Rephase, from my perspective, there is not much guidance, or you have to crawl through so much conflicting information on the Internet, when it would have been nice to have a well curated authoritative source that says - hey this is exactly what you need to do, step by step, and this is why this is the better way to do it, and here are the other options if you want to explore. And this wonderful book I am proposing, then takes you from no knowledge to expert level, step by step. I'd rather pay £50 for such a well curated book, (based on using easily available tools like REW), that has been obviously fact checked and put into proper test, by seasoned reviewers/editors, than invest in something like Acourate, where I really am not sure of the basis of the approach used in the tool. Furthermore even if I bought Acourate, it appears I would still have to again invest in Mitch's book to learn how to use it properly. Do you see a trend, we have the tools like REW, and Acourate, but their user guides do not seem sufficient enough to take us from start to finish, so we now need well written books, to show us how to use these tools, to achieve optimal results. Please consider writing this book. Thanks.
Thanks for all the insightful comments and questions! I really appreciate your engagement. Just a quick note about some concerns around free tools and conflicting information: This channel is dedicated to free resources, so any tutorial methods use only freely available tools. While newer videos might introduce different techniques, it's all part of an evolving process in REW itself and in my methods. That journey is life long and this is in fact why most of us audiophiles are in this IMO ;) Acoustics is complex, and subjective interpretations arise due to unique listening environments and preferences. Finding optimal solutions can be tricky, which is why I started this channel - to share my accumulated knowledge from years of testing, sifting through countless forum discussions, and personal experiences. It's not exhaustive, but it's what I wish I had when I began this journey. Further Resources: REW: For the analytical minds, John's detailed REW guide offers a fantastic foundation: www.roomeqwizard.com/help.html My Basics Video: To grasp the core principles, check out this two-part series: th-cam.com/video/zmIqJ_jlZPw/w-d-xo.html Calibration Mic Direction: You're absolutely right, ideally calibration mic should be directed towards the speaker for maximum accuracy. However, with multi-channel setups, compromises are inevitable. Even with a stereo setup, one needs to compromise a good 30 degrees of accuracy to keep the mic in the middle of the two speaker. Thankfully, mic direction primarily impacts very high frequencies above 10kHz, which often benefit from being left untouched anyway.
Another great video another newb comment, but I'm dedicated to figure this out, I do everything as you do but on 18:04 when you trace arithmetic Dirac + LPHP filter I get only one convex shape at frequency that I set LP and HP filters not a wave like file like you have, needless to say that the first time I did that I got the shape like you did but during time alignment I deleted it.. not It always comes out like a all pass filter 12 db slope I don't get it....
@@ocaudiophile I'm not my rew got bugged it worked as intended on another computer, but for some freaking reason it sometimes doesnt provide the same result I'm sure I make a mistake when it doesn't get the result.... ty for your help since my bass response between first 2 peaks is not symmetrical, ie it continues to slightly dip towards peak 2 and then sharply rises, filter is not perfect but it improves something, considering my response I should have asymmetrical filter to counter the wave like you showed in your example.. I posted on avnirvana official rew section forum so you can see the examples.
OCA - Thank you for your dedication, time and shared expertise. I don't have a words to share my gratitude to everything you did so far. Thank you! Question: Once you create the filters that you present in your vidieos, where and how do you upload them to the processor, receiver or whatever the user shold use to benefit from them?
I managed to make filters and the result is just wow... I'm speechless...Listening to Double bass on Nenad Vasalic Gavrilo's Prinzip is just unbelievable now, the precision in the bass is way better then I had before with sound ID..or with matching sound ID curve in APO witch got me better results then soundID who just pancakes the soundstage... Just to clarify this is your last video and last method? In Workshop Part II - All New FIR filters you continue to make more filters for the rest of frequency range and you make Phase Inversion filter at the end can I do that at the end as well, I suppose rest of the frequency filters are ok but can youu add Phase Inversion filter at this method? I'm sorry for bombarding you with comments on few different videos but I want to be sure I'm doing the right thing. And big thank you !
so that filter that you create from minimum phase version 1/A is added to filters from this video, or does is replace something, sorry for the confusion but was the order ?
Hello, Can I not achieve a less accurate but similar effect by placing an analog mic at the listening position and feed it to the out-of-phase rear wall sub? I'd set the rear sub volume by 30-200hz pink noise until the dips and peaks balance out the most. I can also pull that sub right behind my chair. I don't really want to ADC/DAC my records. Is that a McGyver way? :) Thank you for the great videos. Cenk
It's a bit more complicated than that: Post in thread 'Double Bass Array (DBA)' audiosciencereview.com/forum/index.php?threads/double-bass-array-dba.51919/post-2072803
@@ocaudiophile Thank you. I partially understood the post, but did not get if these "circular convolutions" are due to the filters or actual acoustic new modes due to the rear sub.
@cenkisil574 whether a sub actively sending an inverted signal or a digital filter, the residual effects are there. You need an array of subs on each wall to achieve proper flat bass response.
Awesome work you do here. Quick question, before you do the phase correction in this Video, you applied all the filters, from Dirac to VBA minimum phase plus low shelf, if I would like to apply those to a DSP, not convolution, how could I do that? Personally I would be really curious to see this, maybe even with the phase correction with APF's, would you kindly elaborate or tell us if this is possible at all, as I cant see how to apply a Dirac pulse to a DSP, or apply a delay to a Bandpass filter etc.
The saved .wav convolution files from REW at the end of the method are FIR filters and will work 'as is' with Equalizer APO, JRiver, Roon, Camilla DSP and others. If your DSP is a MiniDSP type unit, their FIR capacities are very limited usually, you can still do some of the things in the method but not all.
@@ocaudiophile Hi Serkan, Thanks for your answer. I am aware of those options, as you stated the problem is here the limited taps length in most DSP platforms. I will investigate. On another note, have you ever tried to do the same for all three axial modes, not just the length modes? Is there a way to increase the magnitude of this mode antimode VBA curve, as they seem to be really moderate? I agree with many of the comments, yu are next level smart and should work for some of the processor companies , but most likely they cant pay you enough :-) Greetings as well from Germany
@GeoffHeinzel Of course I tried axial modes and for a long time but could never get satisfactory results. You can apply a low shelf filter along with the VBA filter which increases its effect but it's been a while and I don't remember the exact details now.
Through your videos, I have noted several hardware platforms on which your filters run. I was curious if you or your viewers have ever used Camilla DSP on a Raspberry PI. I have found it to be extremely flexible and very powerful for multi-channel setups. My setup is modest ( Rpi4, Motu Ultralite mk5, Aiyma A07, 2.1) but great fun to experiment with. Your setups work wonderfully and have enabled me to create an amazing sounding yet compact setup. Thank You!
You're very welcome. I have an old i5 Intel NUC and use JRiver and Roon. I also used Equalizer APO for some time but didn't have a chance to try Camilla DSP though I know it's widely used by the audiophile community and it's MAC compatible. I think you can do even more with a Motu Ultralite. Have a look at the below discussion: The guy claims to have writteen the lowest latency multichannel (up to 256) convolution engine: www.diyaudio.com/community/threads/windows-based-asio-dsp-for-crossovers-and-eq.397971/
I have tried Camilla on Windows 10, no issues loading a convolution filter. Within reasonable filter size, should be zero issues with Raspberry either (I plan to use that exact setup). The only thing you might encounter, which may or may not matter for you, is some delay/loss of sync between audio and video if that's important - only should be an issue with non-minimum phase filters, and depends on number of taps.
Good tutorial. I will need a moment to digest the application of the concepts here. Although I'm aware of all the mathematical details of the concepts, I will need some time to understand the application in acoustics. Btw, does all this processing impose strict mic/head placement?
Thanks. Not really, the inverted impulse wavelength is about twice the room length so quite free from head movements as long as the filter is symmetric between left and right.
Hi . Greating from Malaysia. Thanks again for the wonderful video. As spoken previously in other tutorial. Is it possible you come out with a new video on calibrating Stereo system only Left and Right speakers using REW ? This are the current setup Mcintosh amp, Dynaudio standmount speakers, HifiRose streamer/DAC, Marantz CDplayer. Thank you very much.❤
This one and the workshop videos are for stereo systems. VBA cannot be applied to surround systems unless you're using a HTPC and ATmos is not availble even with that.
I've tried to adapt this to 5 full range speakers in a surround system. I use a minidsp with full range LF and LR input going to LF, LR, and LS; and a second minidsp for RF, RR, RS. Route the sub frequncies to all L speakers with FIR and time alignment, crossovers within minidsp. Similarily with right. And then time align L&R virtual sub between both sides. It definitely sounds good and i percieve directional bass. Its like a 5.7 system i guess. But is it "correct" like your videos, im not sure
@@ocaudiophile definitely a trick I agree. However the speakers only go down to 40hz so they aren't strictly sub, you still need 2 real subs to handle 20-40hz.
This can be done with many media players or convolvers most of which are free but Dolby Atmos decoding prevents communicating with a multichannel receiver from a PC so thees filters are only applicable to stereo or up to 7.1 surround.
The optimal way would be to cross the sub over with your mains at their specified -3dB low bass limit and align sub/main timing such that their group delays are level at the mid/high frequencies. After that you can measure sub/main mix as one and apply the method here as you would with just the speaker, room modes are independent of speaker placement. Port frequency will be sub's port frequency and speaker XO frequencies and phase shifts will stay the same.
Hey can you do a video on the audyssey subwoofer distance trick to help subwoofer phase at your crossing point with speakers? I always get a big dip in response around my crossover point
Thanks for all your videos, i learned so much. I started using Audiolense recently on my 2.2 system. Is there any way to add a VBA filter to the filter already generated by audiolense ? I can generate the VBA on REW and add the audiolense filters on REW too. But will this work ? Or i have to use the VBA as a prefilter on the audiolense process but i'm not sure of the interaction with one another. Many thanks.
Just import your audiolense filters in wav format to REW and you can multiply as many filters as you like and generate new combined filters (A x B trace arithmetic).
Hello OCA, After re-watching some of your current videos, the question arose as to whether aligning the pulse with the AVR distances is no longer necessary? At least you don't go into it there anymore and I could see that you created filters from measurements that weren't on time reference. Does this have to do with using “remove IR delays”? I came across this yesterday when I wanted to set the subs to the mains and was suddenly unsure which of the impulses I should now set to RF. Previously you always used the first highest 100% for this.
@@ocaudiophile It was clear to me that the filter had to be used on the PC or MiniDSP. But that it doesn't work for surround and center, not! Can the phase method be used here? It is still unclear to me whether the Impulse is set to Ref with the distances in the AVR. Are internal delays eliminated with the IR Delays System? Can the function be used generally?
First,great channel n grt work sir...I salute you... Now I may sound dumb but pls clear what video tutorial should I follow for proper callibration of my avr n speakers....i have 7.2.4 system...thank you
Thanks again for the amazing video and method.💫 I tried it and the result is perfect👌 I would like though to discuss a difference that i tried and the outcome was very very good. Instead of calculating the target from the speakers anechoic response to create the LPF, i tried the measurement's frequency one peak below the highest peak, in your case the 44.4255 the same on HPLPF with Olives curve selected and 12db slope for better roll off. And also every target after with the same roll off. I would like to hear your experienced opinion about this. And please do try it and let me know how you feel with the result. You started this obsession 😂🤗
The goal is to remove the room from the speaker and in return, the speaker will remove itself from the room ;) Your prefered curve can coincidentally be similar to your speakers anechoic response or may be the room resonances somehow added up nicely for that curve and associated cut off filters. In general any extra bass you leave on top of speaker's capacity will be the room reflections which are by definition delayed signals and that ruins the dynamics of the bass and also it's not possible to duplicate the same response for the second speaker hence the inaccuracies in stereo imaging. Removing boomy bass is a good first step in room EQ but removing the room completely is the ultimate room EQ goal.
@@ocaudiophile Coincidentally this type of curve removed the room quite obviously and the bass response was tighter. I tried it in two different rooms measurement etc and with two different set of speaker brands and the result was similarly. So it seems to work but i have some doubts of thinking that is my perception . If there is no trouble do try it and listen on your place how you perceive it. Maybe is a matter of taste? I don't know. But the sure thing is that you have given to us Gold reproduction of Sound. I really really thank you for this!! 🙏
As long as you enjoy what you're hearing, it's the right method. But our ears are very spoiled and start to look for even better sound quite quickly (and unfortunately forever!).
@@ocaudiophile that will just lower the resolution making it even harder to read but will not change the text size on the video. If you change your monitor resolution to 1920 x 1080, Windows will automatically adjust the size of the text on your screen, making it larger and much easier for us people over 40 to read. Not trying to be a jerk, I just live your videos and would really like to be able to actually read what you are doing.
I understand. I will check using lower resolution for the next video. In the mean time, the link to the complete REW mdat file used in the video is shared in the description. It might help you.@@WillyLax24
@@ocaudiophile I watch TH-cam on my HT system usually or with a soundbar on my other TV and it’s quite distracting from your great videos. Could you isolate the mic? Yours sincerely, a two finger typer ;)
06:00 Trinnov Wave Forming is NOT generally considered to be the more advanced technology, where did you get that from? You AI gathered some text saying this, but it's unfounded. Dirac ART together with the rest of the Dirac Live suite can do what Trinnov does and more. In addition it can go beyond rectangular rooms. Other than, I appreciate your walk through in REW.
Compare waterfall graphs for both and you can easily see the major supremacy of Waveforming. How do you expect Dirac to cancel out 20Hz waves with small surround/ceiling speakers randomly placed in a room against wall to wall array of subwoofers of Trinnov? ART is only the better choice when setup costs are taken into account.
You have to compare apples to apples "... with small surround/ceiling speakers randomly placed in a room against wall to wall array of subwoofers" Can easily do Double Bass Array with ART. Just bring the equipment and ART provides tools that will enable a better result. @@ocaudiophile
and how do you know that for certain? Do you have both of the undisclosed and patented MIMO/MSMC algorithms available to you? Trinnov Waeforming isn't even publicly available yet, only installed in a couple of ultra high end home cinema systems.
You're the one that made the unsubstantiated claim that "Trinnov waveforming is generally considered to be more advanced technology..." (your claim 1) "...and it can provide better bass performance in rooms with difficult acoustics." (your claim 2) and make a blanket statement "Also correct" for both. Nothing you presented substantiates your claim. Let's start there. Then, when I point this out, instead of providing the evidence to your original claim you modify the base of comparison, comparing ART with small surround/ceiling speakers randomly placed in a room against a Trinnov wall to wall array of subwoofers, which as I pointed out is an obviously meaningless comparison, apples to oranges based on hardware differences. ART can also be set up to run a wall to wall array of subwoofers, which then should be the comparison made. ART can, as I said, do more than Trinnov based on what has been published. In addition you ask me "how do you know that for certain?" with the addition "Trinnov Waveforming isn't even publicly available yet" OK, great, then how can it be "generally considered..." [by whom? on what basis?] "...the more advanced technology" [if it's not available how can it be considered more advanced?]. I can't just sit by and take your statements as statement of facts "true" "correct" "also correct" unless you substantiate them. You haven't. @@ocaudiophile
I listened to both, one in Munich and one in Barcelona and Trinnov was on another level and most experts there had the same opinion, too. I have nothing against Dirac ART and I have no connection whatsoever with Trinnov. There's a massive price difference between the two anyway. I stand by my comments. Trinnov has recently announced a cut-down version for "normal" home cinemas that will not require subwoofer arrays, then we will be able to see who was right I guess. FWITW, Dirac Art takes 9 measurements with any USB mic that the user has avaialble, Trinnov takes 128 measurements with a custom 3D mic with 4 capsules.
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I just wanna say thanks. This has transformed my system into something truly respectable. I multiplied the fvba filter with the inversion convolution filter I made from your other video. Not sure if that’s correct, but damn. Using equalizer apo I can quickly switch between filter and no filter and the no filter version is practically unlistenable… mind you I scratch-built some mark audio chr-120 full rangers mounted in a decware dna horn with spherical wavefront horns loading the front. Needless to say they needed some midrange attenuation. I was researching to build some notch filters when I found your channel. Thank god 😂 This is such a superior solution. I found the bass I was missing. The audio is so clear it’s almost tangeable. Imaging is like nothing I’ve ever heard! You could never get this from headphones. Thanks again. 🎉
Fvba x Inversion filter will work very similarly to the method in the video, they both allow 5dB boost but I found 1/48 auto-eq to yield just a little bit higher c80 clarity results than inversion in my tests although it could be specific to my gear/room.
My listening room is untreated so I like the idea of convolution. I was having a hard time getting the vba to move from 28 to 25 so I just left it. Bass is noticeably better. I will say that listening to hakkerskylder by heilung I could definitely hear the filter. I’m hoping it’s because I didn’t move the filter far enough. Either way, that’s one song. Also, my initial testing showed that one of my speakers ir was inverted so I switched the cables (think it’s my cheap gear). Can this be fixed in rew?
If one speaker is connected in opposite polarity relative to the other, you will experience a very audible drop in bass response. If that's not the case and you see their impulse peaks in opposite directions in REW, this could be a mic position issue and must be left alone. Rew can invert the polarity of a speaker response virtually on the charts but that's all. Check their phase responses to see in which polarity left and right phase responses are more similar to each other.
I’m gonna redo my measurements and see if there’s any difference. Sort of off topic… but Is there any way to adjust woofer Q using filters? That would really simplify subwoofer choices.
I'm not sure how relevant is the Q of a woofer as in damping ratio to a a phase/EQ bandwidth Q.
Next level for sure!
Danke!
bitte!
You are just next level aren't you!
It's in fact Dr. Uli of Acourate who came up first with the VBA idea. I just developed it further:
www.audiovero.de/acourateforum/viewtopic.php?p=177&hilit=VBA#p177
Thanks!
🙏
Bu aldigim ilk TL cinsi tesekkur oldu :)
@@ocaudiophile :) 5-6 eurocuk
@@ocaudiophile sonra fark ettim
Always something new.
:)
👍
Quite great video!!! Let me ask two questions:
- why dont you use linear phase correction? Why minimum phase over linear phase?
- In my case and I suppose many other cases, I would like to have as minimum delay for convolution. Will you have any approach to load the final gilter to rew and work with number of taps to try to reduce the lengh until a few ms keeping as much as possible correction? Knowing it wont be perfect, but pretty accurate. I think doing filters with small delay is very interesting topic for a new tutorial, maybe there are some tricks to achieve best results with minimum taps/delay.
Thank you for your videos and channel ;)
Minimum phase IIR filters are more efficient in the bass frequencies. The phase changes they cause are often useful (ie corrects the phase).
For optimizing FIR filter number of taps see:
th-cam.com/video/zmIqJ_jlZPw/w-d-xo.htmlsi=cCGBFs7WtB6rDq1u&t=1586
Hi, first of all, thank you for all the lessons on your channel. By watching your videos and experimenting a lot myself, I'm getting better at REW. However, I'm struggling with something I can't figure out. I'm trying to combine VBA with an inversion filter. I know you have a different perspective on correction of higher frequencies now, but despite all possible acoustic treatments, my room is still quite challenging, and the inversion filter works best to achieve a precise mid/high stereo image.
When I combine VBA with inversion, I either get a ringing filter or the target is not accurately followed. I've tried creating both the VBA and the inversion filter from L0/R0 and then multiplying the filters. I've also tried convolving the VBA first (LVBA and RVBA) and using them as a starting point for the inversion filter. In both cases VBA and inversion work fine separately, but when combined, something goes wrong, even though I pay attention to sufficient window size and impulse alignment while making the filters.
Do you know if it's theoretically possible to combine VBA and inversion? Thanks in advance! Any tips on approach / things I can try are welcome.
PS: in this video you don't apply Invert Polarity after creating LPHPF. In previous VBA methods, that was necessary. Did I miss something?
Regards, Hans, The Netherlands
I have tried hard but also couldn't find an optimal way to combine VBA with other filters in a meaningful manner. Physical double bass arrays also suffer from similar issues and none are perfect. It seems MIMO systems are the way to go for proper cancellation of room modes (although both ART and Trinnov keep postponing launching). PS I believe you are using the inversion method in the Mastering Digital Room Correction video. It sounds noticeably better than the method in the older REW inversion video
@@ocaudiophile wow, that is a quick response:) The vid you mention is inversion below schreuder. I my room I need the full range inversion filter, like in your video "Convolution with Inversion (no EQ filters, all FIR!)" But as far as I understand from you it is hard to combine other filters with VBA. Well, at least now I now I am not the only one who runs into troubles. But I keep trying and experimenting, maybe one day... :)
Thanks again!
JBL 306p MkII + LSR310s is the computer desktop setup I am working with through this tutorial, in a medium-sized bedroom (~5m wide x 4m deep x 3m tall), not centered left-to-right. Stereo balanced output from DAC going to sub, crossover done in sub and balanced outputs to speakers. Crossover switch on sub is at 80Hz. I have three main bass peaks on both channels: both have a +20dB peak at 40Hz, +7dB peak at 58Hz, and then L has +9dB at 104Hz and R has +8dB at 139Hz.
Based on the tutorial, I am working on the main 40Hz peak with VBA, and I'm getting great simulated results from it, but the "match range" instruction at 27:40 is limiting my filter's range to just 79Hz and thus not touching the L 104Hz and R 139Hz peaks. Would there be any issue using a higher "match range" upper limit to capture the 104Hz and 139Hz peaks for the filter?
You can comfortably go up to 200Hz
@@ocaudiophile Thanks, I will give that a try today, I'm sure it will sound even better than my initial filter. As it stands, I can't believe how much better the measured step response was! Audibly the change was a little subtle (makes sense, since it was mainly working on 80Hz and below), but once I picked up on it I couldn't un-hear it in bass-heavy attacks like kick drums, guitar, etc., specifically the decay after the initial attack. Only aspect that wasn't a clear measured improvement was impulse response (first peak wasn't quite as high as unfiltered, but decay was better). I might have missed an alignment somewhere so I'm going to start from the baseline measurements and build it again. Of course measurement variation (mic placement etc.) could be at fault for that.
I'm also going to change my target curve, the curve I made was much flatter than Harman in bass and sounded a little weak (I ended up adding a +3dB low shelf in my DSP app below 150Hz to get it to taste). I know you're active on ASR, which thread would be a good place to share my measurements?
Are you comparing "normalized" impulse plots? You can share mdat files here if you upload them in google drive. In ASR there's no particular threat but you can start one ;)@@MikeyAntonakakis
@@ocaudiophile Actually I just looked again, and my comment about IR was wrong - very similar between before/after but slightly higher peak after the filter, and no initial "drop" before the main peak, just like in the video. I'll start a thread on ASR and tag you, it will be a little easier than TH-cam comments :)
Very nice explanation. I have done both the v3and v3x for calculations. I think v3x is better. your eq setting makes audyssey next level. I think it is very close if not better than Dirac live. Trionove on the other hand, doesn’t suit most fans as most of us don’t have a dedicated room for movie. Living room setting is 90% of the cases I guess
Three questions please :
1. In addition to aiming to correct anomalies introduced by the room reflections, does this also correct frequency anomalies in the direct sound from the speakers, in line with the target curve?
2. Does this method also reduce the difference in arrival time of low and high frequencies, at the listening position?
3. Will the end result be accurate only at one listening position, but at more than one listening position?
I must add, I have been following DRC developments for about 8 years, started to use REW, 1st with EQ Filters in EQ plugins, and later on generating IIR correction filters for use in convolution plugins, which yielded some improvements. Especially in the most recent 3 or 4 years, it appears that there has been an exponential increase in understanding, from independent sources such as this TH-cam channel. So now we have good enough tools such as microphones with calibration, at reasonable prices, and much improved versions of REW, and now "teachers" such as this TH-cam channel, the challenge is :
We need some kind of roadmap, for learners, to take them step by step from elementary principles such as how to measure, e.g all the different approaches for such elementary tasks, and then build up gradually, to advanced methods, such as used in this video.
I would happily pay £20 to £25, for such a book, which provided this guidance using REW. If I understood enough about this, I'd also be happy to work with anyone interested, to create such a book. Another chap MItch (AccurateSound.ca) has written such a book a few years ago, but it focused on using tools like Acourate - furthermore I discovered there was a tendency in his approach to steer the conversation towards engaging his services, and buying his convolution tools - nothing wrong with a bit of commerce - he has to eat, but that approach led me down a path where I was asked to trust one man's intuition, without really understanding what he was doing. What happens when he dies, which will eventually happen?!!!, i.e. his methods did not seem to me to be independently verifiable, cos I was forced to use proprietary tools and methods which cost a lot of money, and I would have had to invest so much in tools like Acourate, and try them out at great expense, to discover if Mitch's methods actually worked.
I love videos, but thankfully TH-cam allows me to slow down the speech, cos its a bit too fast for me to comprehend otherwise.
My thinking is that this level of complexity deserves capture in a book, that is revised maybe every 2 or 3 years, to keep it current. I'd happily pay £25 every 2 years for such a book. A book is also a kind of insurance. Who says TH-cam will still exist, in 5 years from now !!!????. With a good book, teaching all of these various methods, from elementary topics to advanced methods, and with good explanations, such information is then available as a keepsake. This is just my way of saying to the Obsessive Compulsive Audiophile channel producer - you clearly have a lot of ideas, but it is worth putting these down in a book, with very good explanations and justifications of why you have chosen each step, and what the various options are (at least a few), especially one which fcusses on using easly available tools like REW(and Rephase), is what most people would be highly interested in, cos it avoids leading us down a black hole path, to custom highly proprietary tools like Acourate, where one has no real clue what is going on in there.
There is so much confusion in the DRC business, starting with simple things like - do I point the microphone up to the ceiling, when I take measurements, or do I point it straight ahead towards the center point between the speakers, or do I point the microphone at each speaker, and what calibration file do I use - Zero degree or 90 degree or what other kind of calibration, maybe 30 degrees. To some of you, this might be elementary, but that is the challenge with todays DRC with tools like REW and Rephase, from my perspective, there is not much guidance, or you have to crawl through so much conflicting information on the Internet, when it would have been nice to have a well curated authoritative source that says - hey this is exactly what you need to do, step by step, and this is why this is the better way to do it, and here are the other options if you want to explore. And this wonderful book I am proposing, then takes you from no knowledge to expert level, step by step.
I'd rather pay £50 for such a well curated book, (based on using easily available tools like REW), that has been obviously fact checked and put into proper test, by seasoned reviewers/editors, than invest in something like Acourate, where I really am not sure of the basis of the approach used in the tool. Furthermore even if I bought Acourate, it appears I would still have to again invest in Mitch's book to learn how to use it properly.
Do you see a trend, we have the tools like REW, and Acourate, but their user guides do not seem sufficient enough to take us from start to finish, so we now need well written books, to show us how to use these tools, to achieve optimal results.
Please consider writing this book. Thanks.
Thanks for all the insightful comments and questions! I really appreciate your engagement.
Just a quick note about some concerns around free tools and conflicting information:
This channel is dedicated to free resources, so any tutorial methods use only freely available tools. While newer videos might introduce different techniques, it's all part of an evolving process in REW itself and in my methods. That journey is life long and this is in fact why most of us audiophiles are in this IMO ;)
Acoustics is complex, and subjective interpretations arise due to unique listening environments and preferences. Finding optimal solutions can be tricky, which is why I started this channel - to share my accumulated knowledge from years of testing, sifting through countless forum discussions, and personal experiences. It's not exhaustive, but it's what I wish I had when I began this journey.
Further Resources:
REW: For the analytical minds, John's detailed REW guide offers a fantastic foundation: www.roomeqwizard.com/help.html
My Basics Video: To grasp the core principles, check out this two-part series: th-cam.com/video/zmIqJ_jlZPw/w-d-xo.html
Calibration Mic Direction:
You're absolutely right, ideally calibration mic should be directed towards the speaker for maximum accuracy. However, with multi-channel setups, compromises are inevitable. Even with a stereo setup, one needs to compromise a good 30 degrees of accuracy to keep the mic in the middle of the two speaker. Thankfully, mic direction primarily impacts very high frequencies above 10kHz, which often benefit from being left untouched anyway.
Another great video another newb comment, but I'm dedicated to figure this out, I do everything as you do but on 18:04 when you trace arithmetic Dirac + LPHP filter I get only one convex shape at frequency that I set LP and HP filters not a wave like file like you have, needless to say that the first time I did that I got the shape like you did but during time alignment I deleted it.. not It always comes out like a all pass filter 12 db slope I don't get it....
You may be forgetting to time shift the lpf filter with "offset t=0" before A+B adding it with Dirac pulse.
@@ocaudiophile I'm not my rew got bugged it worked as intended on another computer, but for some freaking reason it sometimes doesnt provide the same result I'm sure I make a mistake when it doesn't get the result.... ty for your help since my bass response between first 2 peaks is not symmetrical, ie it continues to slightly dip towards peak 2 and then sharply rises, filter is not perfect but it improves something, considering my response I should have asymmetrical filter to counter the wave like you showed in your example.. I posted on avnirvana official rew section forum so you can see the examples.
OCA - Thank you for your dedication, time and shared expertise. I don't have a words to share my gratitude to everything you did so far. Thank you!
Question:
Once you create the filters that you present in your vidieos, where and how do you upload them to the processor, receiver or whatever the user shold use to benefit from them?
You can use free Equalizer APO for PC or Camilla DSP for Mac. Paid options are many including River, Roon Hang Loose etc.
@@ocaudiophile But I am running my movies via media player box, that is connected to server and to receiver. How about in this case?
All you need is a CPU somewhere in the chain to process the filters.
I managed to make filters and the result is just wow... I'm speechless...Listening to Double bass on Nenad Vasalic Gavrilo's Prinzip is just unbelievable now, the precision in the bass is way better then I had before with sound ID..or with matching sound ID curve in APO witch got me better results then soundID who just pancakes the soundstage... Just to clarify this is your last video and last method? In Workshop Part II - All New FIR filters you continue to make more filters for the rest of frequency range and you make Phase Inversion filter at the end can I do that at the end as well, I suppose rest of the frequency filters are ok but can youu add Phase Inversion filter at this method? I'm sorry for bombarding you with comments on few different videos but I want to be sure I'm doing the right thing. And big thank you !
Thanks. The last one (which needs to be also replaced in fact due to some improvements in REW itself) is "Mastering Digital Room Correction".
so that filter that you create from minimum phase version 1/A is added to filters from this video, or does is replace something, sorry for the confusion but was the order ?
@@boogiexx You can multiply (A x B trace arithmetic) as many filters as you like to generate a final combined FIR filter.
Hello, Can I not achieve a less accurate but similar effect by placing an analog mic at the listening position and feed it to the out-of-phase rear wall sub? I'd set the rear sub volume by 30-200hz pink noise until the dips and peaks balance out the most. I can also pull that sub right behind my chair. I don't really want to ADC/DAC my records. Is that a McGyver way? :) Thank you for the great videos. Cenk
It's a bit more complicated than that:
Post in thread 'Double Bass Array (DBA)' audiosciencereview.com/forum/index.php?threads/double-bass-array-dba.51919/post-2072803
@@ocaudiophile Thank you. I partially understood the post, but did not get if these "circular convolutions" are due to the filters or actual acoustic new modes due to the rear sub.
@cenkisil574 whether a sub actively sending an inverted signal or a digital filter, the residual effects are there. You need an array of subs on each wall to achieve proper flat bass response.
@@ocaudiophile understood. Thank you
Awesome work you do here. Quick question, before you do the phase correction in this Video, you applied all the filters, from Dirac to VBA minimum phase plus low shelf, if I would like to apply those to a DSP, not convolution, how could I do that? Personally I would be really curious to see this, maybe even with the phase correction with APF's, would you kindly elaborate or tell us if this is possible at all, as I cant see how to apply a Dirac pulse to a DSP, or apply a delay to a Bandpass filter etc.
And on top, can this be done for multi sub systems in home theaters, and how?
The saved .wav convolution files from REW at the end of the method are FIR filters and will work 'as is' with Equalizer APO, JRiver, Roon, Camilla DSP and others. If your DSP is a MiniDSP type unit, their FIR capacities are very limited usually, you can still do some of the things in the method but not all.
@@ocaudiophile Hi Serkan, Thanks for your answer. I am aware of those options, as you stated the problem is here the limited taps length in most DSP platforms. I will investigate. On another note, have you ever tried to do the same for all three axial modes, not just the length modes? Is there a way to increase the magnitude of this mode antimode VBA curve, as they seem to be really moderate? I agree with many of the comments, yu are next level smart and should work for some of the processor companies , but most likely they cant pay you enough :-) Greetings as well from Germany
@GeoffHeinzel Of course I tried axial modes and for a long time but could never get satisfactory results. You can apply a low shelf filter along with the VBA filter which increases its effect but it's been a while and I don't remember the exact details now.
Through your videos, I have noted several hardware platforms on which your filters run. I was curious if you or your viewers have ever used Camilla DSP on a Raspberry PI. I have found it to be extremely flexible and very powerful for multi-channel setups. My setup is modest ( Rpi4, Motu Ultralite mk5, Aiyma A07, 2.1) but great fun to experiment with. Your setups work wonderfully and have enabled me to create an amazing sounding yet compact setup. Thank You!
You're very welcome.
I have an old i5 Intel NUC and use JRiver and Roon. I also used Equalizer APO for some time but didn't have a chance to try Camilla DSP though I know it's widely used by the audiophile community and it's MAC compatible.
I think you can do even more with a Motu Ultralite. Have a look at the below discussion: The guy claims to have writteen the lowest latency multichannel (up to 256) convolution engine:
www.diyaudio.com/community/threads/windows-based-asio-dsp-for-crossovers-and-eq.397971/
I have tried Camilla on Windows 10, no issues loading a convolution filter. Within reasonable filter size, should be zero issues with Raspberry either (I plan to use that exact setup). The only thing you might encounter, which may or may not matter for you, is some delay/loss of sync between audio and video if that's important - only should be an issue with non-minimum phase filters, and depends on number of taps.
Good tutorial. I will need a moment to digest the application of the concepts here. Although I'm aware of all the mathematical details of the concepts, I will need some time to understand the application in acoustics. Btw, does all this processing impose strict mic/head placement?
Thanks.
Not really, the inverted impulse wavelength is about twice the room length so quite free from head movements as long as the filter is symmetric between left and right.
Hi, Many thanks for the new version of the VBA. Why don´t you invert the vba filter anymore ? Thanks T
Because I'm dividing it rather than multiply and inversion is achieved in that operation automatically.
Hi . Greating from Malaysia. Thanks again for the wonderful video. As spoken previously in other tutorial. Is it possible you come out with a new video on calibrating Stereo system only Left and Right speakers using REW ?
This are the current setup Mcintosh amp,
Dynaudio standmount speakers,
HifiRose streamer/DAC,
Marantz CDplayer. Thank you very much.❤
This one and the workshop videos are for stereo systems. VBA cannot be applied to surround systems unless you're using a HTPC and ATmos is not availble even with that.
I've tried to adapt this to 5 full range speakers in a surround system.
I use a minidsp with full range LF and LR input going to LF, LR, and LS; and a second minidsp for RF, RR, RS.
Route the sub frequncies to all L speakers with FIR and time alignment, crossovers within minidsp. Similarily with right. And then time align L&R virtual sub between both sides.
It definitely sounds good and i percieve directional bass. Its like a 5.7 system i guess. But is it "correct" like your videos, im not sure
It surely is a very uniform bass with 7 woofers. Nice trick!
@@ocaudiophile definitely a trick I agree. However the speakers only go down to 40hz so they aren't strictly sub, you still need 2 real subs to handle 20-40hz.
This is done with Rhoon right?
How would you put Rhoon between you media and your AVR? (for movies)
This can be done with many media players or convolvers most of which are free but Dolby Atmos decoding prevents communicating with a multichannel receiver from a PC so thees filters are only applicable to stereo or up to 7.1 surround.
Amazing work. how do you add a subwoofer to the main speakers with this method?
The optimal way would be to cross the sub over with your mains at their specified -3dB low bass limit and align sub/main timing such that their group delays are level at the mid/high frequencies. After that you can measure sub/main mix as one and apply the method here as you would with just the speaker, room modes are independent of speaker placement. Port frequency will be sub's port frequency and speaker XO frequencies and phase shifts will stay the same.
Hey can you do a video on the audyssey subwoofer distance trick to help subwoofer phase at your crossing point with speakers? I always get a big dip in response around my crossover point
Align the tails of sub and speaker responses in Overlays/GD graph and your XO point dips should greatly improve.
Thanks for all your videos, i learned so much.
I started using Audiolense recently on my 2.2 system. Is there any way to add a VBA filter to the filter already generated by audiolense ?
I can generate the VBA on REW and add the audiolense filters on REW too. But will this work ? Or i have to use the VBA as a prefilter on the audiolense process but i'm not sure of the interaction with one another.
Many thanks.
Just import your audiolense filters in wav format to REW and you can multiply as many filters as you like and generate new combined filters (A x B trace arithmetic).
@@ocaudiophile Thanks i will try this
Hello OCA,
After re-watching some of your current videos, the question arose as to whether aligning the pulse with the AVR distances is no longer necessary?
At least you don't go into it there anymore and I could see that you created filters from measurements that weren't on time reference.
Does this have to do with using “remove IR delays”?
I came across this yesterday when I wanted to set the subs to the mains and was suddenly unsure which of the impulses I should now set to RF. Previously you always used the first highest 100% for this.
This method is not applicable to surround system and AVRs. It's for stereo setups with a convolution filter run from a PC of some sort.
@@ocaudiophile
It was clear to me that the filter had to be used on the PC or MiniDSP.
But that it doesn't work for surround and center, not!
Can the phase method be used here?
It is still unclear to me whether the Impulse is set to Ref with the distances in the AVR.
Are internal delays eliminated with the IR Delays System? Can the function be used generally?
It can be used with any device which can process FIR filters!
First,great channel n grt work sir...I salute you...
Now I may sound dumb but pls clear what video tutorial should I follow for proper callibration of my avr n speakers....i have 7.2.4 system...thank you
Audyssey ART
Thank you
Thanks again for the amazing video and method.💫
I tried it and the result is perfect👌
I would like though to discuss a difference that i tried and the outcome was very very good.
Instead of calculating the target from the speakers anechoic response to create the LPF, i tried the measurement's frequency one peak below the highest peak, in your case the 44.4255 the same on HPLPF with Olives curve selected and 12db slope for better roll off. And also every target after with the same roll off.
I would like to hear your experienced opinion about this. And please do try it and let me know how you feel with the result. You started this obsession 😂🤗
The goal is to remove the room from the speaker and in return, the speaker will remove itself from the room ;)
Your prefered curve can coincidentally be similar to your speakers anechoic response or may be the room resonances somehow added up nicely for that curve and associated cut off filters. In general any extra bass you leave on top of speaker's capacity will be the room reflections which are by definition delayed signals and that ruins the dynamics of the bass and also it's not possible to duplicate the same response for the second speaker hence the inaccuracies in stereo imaging.
Removing boomy bass is a good first step in room EQ but removing the room completely is the ultimate room EQ goal.
@@ocaudiophile Coincidentally this type of curve removed the room quite obviously and the bass response was tighter. I tried it in two different rooms measurement etc and with two different set of speaker brands and the result was similarly. So it seems to work but i have some doubts of thinking that is my perception .
If there is no trouble do try it and listen on your place how you perceive it.
Maybe is a matter of taste? I don't know.
But the sure thing is that you have given to us Gold reproduction of Sound.
I really really thank you for this!! 🙏
As long as you enjoy what you're hearing, it's the right method. But our ears are very spoiled and start to look for even better sound quite quickly (and unfortunately forever!).
Love your videos, just struggle to be able to read the text on your screen, which makes is so small even on my 27” monitor.
I understand but 2560x1440 at 60Hz is the best I can do with screen recording in 16:9 format
@@ocaudiophile are you not able to do 1920 X 1080? Would make the text bigger.
You can adjust to a lower res in YT settings.@@WillyLax24
@@ocaudiophile that will just lower the resolution making it even harder to read but will not change the text size on the video. If you change your monitor resolution to 1920 x 1080, Windows will automatically adjust the size of the text on your screen, making it larger and much easier for us people over 40 to read. Not trying to be a jerk, I just live your videos and would really like to be able to actually read what you are doing.
I understand. I will check using lower resolution for the next video. In the mean time, the link to the complete REW mdat file used in the video is shared in the description. It might help you.@@WillyLax24
Would anyone care to share their results after following this guide?
I have shared a link to the actual REW mdat file used in the video in the description. It also contains measured results after calibration.
I can FEEL every bang of the keys on your keyboard 😅 it’s too loud man
The mic is on the same desk, key clicks are amplified and besides I am a 2 finger typer :)
@@ocaudiophile I watch TH-cam on my HT system usually or with a soundbar on my other TV and it’s quite distracting from your great videos. Could you isolate the mic? Yours sincerely, a two finger typer ;)
noted!@@jimmykaka89
06:00 Trinnov Wave Forming is NOT generally considered to be the more advanced technology, where did you get that from? You AI gathered some text saying this, but it's unfounded. Dirac ART together with the rest of the Dirac Live suite can do what Trinnov does and more. In addition it can go beyond rectangular rooms. Other than, I appreciate your walk through in REW.
Compare waterfall graphs for both and you can easily see the major supremacy of Waveforming. How do you expect Dirac to cancel out 20Hz waves with small surround/ceiling speakers randomly placed in a room against wall to wall array of subwoofers of Trinnov? ART is only the better choice when setup costs are taken into account.
You have to compare apples to apples "... with small surround/ceiling speakers randomly placed in a room against wall to wall array of subwoofers" Can easily do Double Bass Array with ART. Just bring the equipment and ART provides tools that will enable a better result. @@ocaudiophile
and how do you know that for certain? Do you have both of the undisclosed and patented MIMO/MSMC algorithms available to you? Trinnov Waeforming isn't even publicly available yet, only installed in a couple of ultra high end home cinema systems.
You're the one that made the unsubstantiated claim that "Trinnov waveforming is generally considered to be more advanced technology..." (your claim 1) "...and it can provide better bass performance in rooms with difficult acoustics." (your claim 2) and make a blanket statement "Also correct" for both. Nothing you presented substantiates your claim. Let's start there. Then, when I point this out, instead of providing the evidence to your original claim you modify the base of comparison, comparing ART with small surround/ceiling speakers randomly placed in a room against a Trinnov wall to wall array of subwoofers, which as I pointed out is an obviously meaningless comparison, apples to oranges based on hardware differences. ART can also be set up to run a wall to wall array of subwoofers, which then should be the comparison made. ART can, as I said, do more than Trinnov based on what has been published. In addition you ask me "how do you know that for certain?" with the addition "Trinnov Waveforming isn't even publicly available yet" OK, great, then how can it be "generally considered..." [by whom? on what basis?] "...the more advanced technology" [if it's not available how can it be considered more advanced?]. I can't just sit by and take your statements as statement of facts "true" "correct" "also correct" unless you substantiate them. You haven't. @@ocaudiophile
I listened to both, one in Munich and one in Barcelona and Trinnov was on another level and most experts there had the same opinion, too. I have nothing against Dirac ART and I have no connection whatsoever with Trinnov. There's a massive price difference between the two anyway. I stand by my comments.
Trinnov has recently announced a cut-down version for "normal" home cinemas that will not require subwoofer arrays, then we will be able to see who was right I guess.
FWITW, Dirac Art takes 9 measurements with any USB mic that the user has avaialble, Trinnov takes 128 measurements with a custom 3D mic with 4 capsules.
Can you please not use chat gpt
Why is that?
OCA ….. OMG !