Automatic Room Correction has won the Subwoofer Wars
Mijostyn, you are correct that I don't have experience with the latest in room correction, but I have no reservations anyway about its use in the subwoofer region (aside from the presumably-outdated possible latency issue). My reservations about "room correction" occur further up the spectrum. And in case I haven't been clear, I have no real problem with using EQ to correct the SPEAKERS north of the bass region, but I DO have reservations about using EQ to "correct" the ROOM north of the bass region. I can explain the distinction between the two if anyone is interested. Duke |
As mentioned, a too-common post in audio fora is, "I got a sub, and it doesn’t sound good. Help me!" Unfortunately, trying to integrate a sub without measurement capability is a fool’s errand. That’s a tough thing for a newbie to accept, one who has spent a wad on a new sub, and who is reading audio fora and magazines where measurements often are held to be irrelevant. In integrating subs, many people emphasize the primacy of placement, and I agree. And it’s a LOT easier to compare placements if you measure! In some rooms, nulls can’t be avoided, but if you measure, you can place the subs so they don’t have nulls at the same frequencies as the mains. This is true whether you use DRC (digital room correction) or not. Now you have to set up good crossover frequencies and slopes. I’ve done it both ways, and I agree, it’s far easier done by using some kind of DRC. I’m currently using Anthem’s ARC, which does an outstanding job at this. I had fine results formerly from a Tact 2.2X. Finally, you have to adjust phase of the subs, and though it can be done without a microphone, it too is way easier with one. How many DRC systems do this automatically? ARC Genesis used with Anthem’s STR line does, but many do not. So yes, the DRC systems really do help in setting up subs. (And as Erik said, this is independent of the EQ features.) Even more important is the measuring mic, its software, and the patience to learn how to use them. |
I hope the OP will forgive me, but it seems this thread has already gone in bit of a different direction already. It would seem to me, despite being somewhat contrary to audiophile common sense; that flush to wall/corner mounted drivers (preferably flat, concentric and with 1st order crossovers) coupled with room correction might be most advantageous. Well, Roy Allison, Peter Snell and Klipsch certainly agreed with some of these elements in general (minus perhaps the room correction). Though the post was more about adding a sub than bass in general, your point is well taken. |
I hope the OP will forgive me, but it seems this thread has already gone in bit of a different direction already. It would seem to me, despite being somewhat contrary to audiophile common sense; that flush to wall/corner mounted drivers (preferably flat, concentric and with 1st order crossovers) coupled with room correction might be most advantageous. Such flush mounting of drivers with room correction would negate the differences and overcorrection of the direct vs the reflected sound, as the time of sounds would be nearly identical. As the room response would be in someways more predictable, some of the placement induced concerns might be built into the drivers response, further reducing the amount of room correction intrusion into the signal. |
I’m seeing the title of this thread as clickbait. Sir, do you not understand my role here?? :-) It was meant to be a discussion, in earnest, about how easy ARC systems have made subwoofer integration and, my point, that the advantages for the average audiophile outweigh the negatives. Look at how easy and how little complaints there are in Audiogon. I think that overall, ARC has proven itself quite useful and more of us should look at them as a good solution. If anything, this thread was prompted by another I started about how hotly debated subwoofer solutions, and bass in general are. ARC is the great simplifier. Best, Erik |
I'm seeing the title of this thread as clickbait. Unless we are talking about DBAs vs oldschool, there are no 'subwoofer wars'. Now one thing I'm not seeing here so far but Duke did allude to it is the Total Room Energy which dominates the room sound above 300Hz or so depending on the room. The thing is, if the speaker design is competent, the measure of this in an anechoic chamber gives a very good prediction of how the actual Total Room Energy will work out in an actual room. We all know how the off-axis response needs to smoothly drop off; the Total Room Energy is actually what dominates people's impressions of the tonality of a speaker. This is a matter of good speaker design; it appears to me that ARCs are apparently most useful in dealing with speakers with poor response. |
Configuring a sub with digital bass management is not all that hard. Choosing the right crossover points and slopes is probably the hardest part. It helps if you can see them on a screen. Then it is all trial and error. Everyone has a theory but there is no single right answer. It depends on the speakers and set up. After this the subs have to be equalized so their response is flat at the listening position and finally delays set so that the sound of the subs and satellites gets to your ear exactly at the same time in phase. This require a full 2 way digital crossover. In my system the computer measures each speaker and automatically computes correction filters and delays. They are more accurate than what you can do manually. |
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How obtuse! My only point has ever been that FFT based programs can window out the later arrival times of reflections and discern them from direct (speaker to mic) impulses (ie: when performing time/phase alignments between subs and mains, etc). You’re trying to make the conversation about room/EQ measurements. Twisting in the wind (deflection)! Sorry, Erik (I’ll drop it, here). |
Room correction has not won the subwoofer wars. What it has done, is that it has subverted high quality acoustics application, room design, and system builder's base level skills, to being 'good enough for Mr Average'. Good enough for those who won't do it right, or can't do it right. It has dropped the annoyance factor for those who are not going to ever be doing it right. It is NOT correct, I repeat, NOT correct. It's akin to all the nannies and electronic additions to modern supercars, so that Mr average to poor driver can drive a $500k super or hyper car and PRETEND that they can drive. It won the popularity contest for the middle of the audiophile herd who play in the home theater space. Nothing else. Don't anyone here fool themselves into thinking it is anything else. It's not a peak, not a note on perfection, it is the opposite, it is annoyance relief, it's a comfortable hole in the ground to circle around, like circling a drain.... |
Erik, measuring and creating correction filters manually is all but impossible. Check out my system’s page and I think you will get the picture. Designing target curves is easy to do manually as I demonstrated. Measurement curves are very complex. The computer can calculate a correction filter instantly where I do not think anyone could do it that accurately in days if at all. You are talking about thousands of corners each one having to be set manually. Just doing subs is a lot easier but you miss out on a world of incredible imaging as well as other advantages by not going full range. |
Duke, you a obviously have not spent much time with one of the better digital management systems. You are making assumptions that may not be correct. I posted my system with details that help understand it. It works, Not just a little either. You can not tell there are subs in the system until a real low note comes along. The tonal balance doesn’t change with volume. Imaging is superior to anything you have ever heard because the frequency response of both sides is exactly the same. Once you use a system like this there is no going back, it is that extreme. Mike |
We got here by Duke pointing out that consumer room correction, at least currently, (to our knowledge) does not take into account reflected/directly energy when doing level balancing, and hence while it can correct for phase/amplitude, it lacks a degree of freedom in setup. Ahhh. Well, Toole dislikes them, but I think that over the last decade it's gotten better. Personally, I do all my own EQ. However!!!! Teaching a noob is a lot of work, and a lot of debate and a lot of different answers. ARC I think, over the last few years, has gotten better, and yields better than average audiophile results. Could Duke or myself do better by hand? Absolutely. |
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"Windowing" the impulse response to look at only the initial part shows us the frequency response of the direct sound with little or no contribution from the room. Yes, but this windowing feature also limits the lower boundary of the FFT. The smaller the time window, the higher the minimum frequency of the FFT, and as pointed out, get low enough and the room is unavoidable. But the entire premise of ARC is to actually correct for room behavior, so not sure how we got here. We want / need the room in measurements for ARC to work. Quasi-anechoic measurements ( i.e. simulating a speaker without a room ) is the opposite of what we are looking for here. Best, E |
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The premise/purpose of the Fourier transform is simple: Any recorded sound can be broken up into discrete sinusoidal components. For instance, a square wave, no matter how perfectly square, can be decomposed into odd-order harmonic sine waves, despite the original square wave looking anything but sinusoidal. It is not magical however, and the results will vary based on the portion of the recording analyzed. FFT is also not a substitute for all digital signal analysis. You don’t need FFT to tell you what a visual inspection of an impulse response will, such as looking for reflections and time aligning speakers. Nor do you need FFT to create an algorithm to automatically set speaker delays. https://en.wikipedia.org/wiki/Fourier_transform Where the Fast Fourier Transform really transformed acoustics was in the nearly infinite resolution. We went from band limited (octave, 1/3 octave, etc) measurements to resolution bounded only by the sample length on the low end and the Nyquist frequency on the top. Outstanding. https://en.wikipedia.org/wiki/Nyquist_frequency And of course, let’s not forget the waterfall plots. Best, E |
@dannad- I’ll let a copy/paste, from the information to which I earlier referred, be my last word on the subject as to whether an FFT program/algorithm can separate/discern arrival times: "Once the impulse response has been obtained, it can be analysed to calculate information about how the room behaves. The simplest analysis is the FFT, to show the frequency response between the source and mic positions. However, we have some control over it. Altering which part of the impulse response is analysed by the FFT changes what aspect of the room’s response we see. The early part of the impulse response corresponds to the direct sound from the source to the mic, the shortest path between them. Sound that has bounced off the room’s surfaces has to travel further to reach the mic, which takes longer, so the later parts of the impulse response contain the contributions of the room. "Windowing" the impulse response to look at only the initial part shows us the frequency response of the direct sound with little or no contribution from the room. A window that includes later parts of the response lets us see how the room’s contribution alters the frequency response. The ability to separate the contributions of the direct and later (reflected) sound is a key difference between the frequency response derived from an impulse response and one we would get from an RTA, for example, which can only show the total combined response of source and room." I’ll take their word for it, before yours! Apologies, to Erik |
I have ARC with my home theater system and also vastly more sophisticated and capable DEQX DSP with my two channel music system. I have never been impressed with ARC, but the DEQX is a totally different matter. To begin with performing speaker correction before performing room correction is quite important to the final result. I believe the total effect of the DEQX DSP is a big enough SQ improvement to outweigh any downside including an extra AD/DA for analog sources. Also it allows me to triamplify my my DIY fully horn loaded speakers for even more SQ improvement. |
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If you are robbing your subwoofer of power with DSP, then there must be frequency aberrations at the listening position that need to be fixed. Kind of sort of. Sorry to introduce nuance here. Not all ARC is the same, not by a long shot. Early ARC attempted a flat response and it sounds awful, so I can completely see why if you heard that you'd be of the opinion it steals power. JL Audio and Dirac are examples of ARC that no longer make these fatal flaws. I encourage you to listen to them if you can. Still, we don't have the luxury of doing everything right. We can't all have 2 subs, room treatment, and a choice of anywhere in the room to place them. In my mind, for each listener and their home there's some combination of this that will be ideal. If you can't do all, do 2 of them. If not, resort to EQ alone. For instance, I can't have 2 subs, and I can't pick a location. It's really 1 place or no sub. What I can do is add room treatment and EQ. Result is marvelous and much better than doing nothing. Despite all of this nuance, ARC is still the best fastest way for a new sub user to configure the crossover, delays and levels. Best, Erik |
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not for me. it is a band aid at best and robs your subwoofer of power at worst. I will argue that these are not either or but instead of. One really interesting thing I’m reading here is how many are only looking at ARC as EQ. ARC subwoofer integration is in addition to EQ. It sets up the crossover slopes and delays and this is where most have the biggest problems. The point of the EQ choices not necessarily being great varies with ARC vendor. Some make better choices than others. JL Audio and Dirac are two of the one's I tend to like the results of, but all will properly set the crossover settings. Optimum placement of a sub is always a good idea. |
I also should have said that how the ARC (generic) treats the rest of the bands is not what I mean to bring up, and this behavior is a lot more varied and subjective. It is in the subwoofer integration though that almost all ARC systems are better and easier than your average music-phile with nice speakers. |
I’ve been using a TacT RCS 2.2Xaaa, in my system, since the 90’s. Not much has changed, since then. FFT is called that, because it is and can discern between direct and reflected sounds. Read the ’How REW makes it’s measurements’ section (page 5), here: https://www.roomeqwizard.com/REWhelp.pdf Especially, the fifth paragraph, of that treatise |
TBC: By ARC I mean the acronym generically, not a particular brand. There are at least 4 brands of automatic room correct systems which include subwoofer integration. Adjusting the arrival time and levels of a sub to match the main speakers is trivial for these systems to do but in addition to that what they do far far better than your average audiophile is the crossover slope /phase matching and bass EQ. It is that most audiphiles have no idea what this is, that it matters, that they’ll have to learn and adjust for it that makes a new sub hit or miss. This just does not happen with most modern ARC (generic) systems. |
Dannad wrote: " Does ARC do this? I don’t know, but it is certainly possible. " I would be surprised if automated room correction algorithms are that sophisticated. But perhaps they are... I must admit that my knowledge of the topic is apparently outdated. " If we are talking low bass, then the latency can be minimal." I recall seeing where latency on the order of 16 or 18 milliseconds was a bragging point for a subwoofer-specific DSP system. That corresponds a one wavelength delay at ballpark 60 Hz, relative to the mains (and more than one wavelength north of 60 Hz). Maybe that 16 milliseconds figure was on the high side relative to what’s being done today? "you either have timing information that is created by two speakers... On the other hand, a lot of spatial cues are volume based... " My understanding is that localization cues (sound image locations including soundstage width) are primarily conveyed by the first-arrival sound, influenced somewhat by early lateral reflections. And that good ambience (soundstage depth, envelopment, spaciousness, immersion, "you are there") without degradation of clarity calls for a very clean first-arrival sound, followed my minimal early reflections, which in turn are followed by a fair amount of relatively late spectrally-correct reflections, which hopefully then decay neither too quickly nor too slowly fairly uniformly across the spectrum. (These aren't the only things that matter, but are among them.) Duke |
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Dannad wrote: " ARC can detect accurately speaker distance based on first time of arrival and based on timing could differentiate between direct and reflected." It depends on the wavelengths and reflection path lengths. In order to get good direct-sound data at a given frequency the time window has to be open and reflection-free for at least one wavelength, and possibly more. If the wavelengths are long relative to the reflection path lengths, the reflections start to arrive before the first wavelength is even finished. Now factor in the time-delay-induced phase rotation of that arriving reflection and we have bad data, because the microphone can ONLY sense the combined net pressure at its location - it CANNOT tell how long a path some or all of that pressure (or lack thereof) travelled before arriving. I use time-gated measurements regularly, and in a "normal" room there will be a lower limit, typically in the several hundred hertz region, below which we simply cannot get good first-arrival-sound data. We can get good in-room data all the way down, but we can usually only get good first-arrival sound data down to several hundred hertz. " However, from a perception standpoint, correcting on total volume is not a bad idea as you will have both speaker irregularities and room peaks and valleys that are not in the recording so removing them, on balance is better." Imo the first-arrival sound is what matters most north of the Schroeder frequency region (yeah I know that term is debatable within the context of a small room). If the off-axis response (which dominates the "room response") has not been addressed by the loudspeaker design, then address it with acoustic treatment if we can. And if we can’t THEN imo it might well make sense to compromise the first-arrival sound for the sake of the overall tonal balance, but I’m not sure I’d trust an algorithm to make those decisions - I’d want to make the adjustments by hand. Imo what we definitely DO NOT want is that a DIFFERENT "room EQ" be applied to the left and right channels (north of the Schroeder frequency), as despite the term what we’re REALLY doing is EQing the first-arrival sound, and we don’t want to introduce a channel mis-match into the all-important first-arrival sound. (Just to be clear imo the in-room sound matters a LOT, but the most effective place to address it is at the loudspeaker design stage, followed by room treatments.) " I can’t see latency being an issue as it would be consistent across channels by design (or intentionally different)." If we run everything through the DSP unit, that is true. If we only run the subwoofer’s signal through the DSP unit, then latency could be an issue. And not everyone wants to run the signal going to their mains through a DSP unit. So, imo, whether or not latency is an issue "depends on the specifics". Duke |
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Imo room correction works great in the subwoofer region, but not so much further up the spectrum. This is because the microphone picks up the reflections along with the direct sound, and does not differentiate between them. That’s fine at very low frequencies where the ear’s time-domain resolution is poor, so the ear is likewise not differentiating between the direct and reflected sound. But further up the spectrum the ear differentiates between the direct sound and the reflections, perceiving and processing them differently. And in general it would be a mistake to "correct" the direct sound based on the contribution of the reflections. I’m not saying that improvement via EQ isn’t possible further up the spectrum, but imo it should be done by hand, not "automatically" based on what an ungated microphone captures in-room. Latency might be an issue with some digital room correction systems, I suppose it depends on the specifics. Duke |
