# Reproduction of timbre

Discussion in 'Sound Science' started by old tech, Apr 23, 2019.

1. 1. Yes but we're all talking about a number of somewhat different things here and lumping them together as the same thing. For example, at the simplest level, if we have a sine wave at say 1000Hz at 80dB (peak or RMS) and change it's time, say slow it down by 1%, then it will no longer be 1000 cycles per second (cps, 1000Hz), it will be 990cps/Hz. In this example the amount of amplitude (dB) is completely unchanged, it will still be 80dB (peak or RMS) just the pitch/frequency changes. However, when were dealing with more than one instance of a signal (even the exact same signal) then we'll have a summing/cancellation effect. For example, take our sine wave again but now add an identical one that's delayed by 1 milli-sec: As it's 1000cps, that's 1 cycle in a milli-sec, therefore the second instance is 360deg out of phase, sum them together and the result will be double the amplitude (86dB). However, if the delay is half a milli-sec then the two instances will be 180deg out of phase and the result will be complete cancellation (the amplitude will be 0dB). So, depending on the time delay between the two instances we will have double the amplitude, no amplitude or anywhere in between. By the same token, if we take the same delay, it will have a different effect on a different frequency. So our 1 milli-sec delay will cause a doubling of the amplitude of two 1000Hz sine waves but will completely cancel out two 500Hz sine waves (as 1ms is half the cycle time of a 500Hz sine wave and it will therefore be exactly 180deg out of phase). Consequently, if a sound contains harmonics at 500Hz and 1000Hz and we have a second instance of that sound that's delayed by 1ms, the amplitude of those harmonics are going to change drastically (the 1000Hz harmonic will double and the 500Hz harmonic will disappear) and therefore the timbre will be different (as "timbre" is defined by the number and balance of the harmonics in a sound). This is all on paper though, in practice we won't just have two instances of the sound, we'll have two signals (stereo speakers), numerous different instruments/sounds and notes, each with numerous harmonics, numerous different room reflections and each with a different time delay. So it's all a bit of a mess and we rarely get perfect doubling or cancellation, we just get some amount of amplitude alterations of the fundamentals and harmonics. To further complicate matters, how we perceive these alterations in the amplitudes/balance of harmonics varies considerably, depending on context, expectation and other factors.

2. Jitter is an entirely different type of timing error. With jitter we're not talking about timing differences between different instances of a sound/s that result in summing/cancellation "phase" effects but if we were, then as jitter is in the pico-sec rather than milli or micro sec range, we'd be talking about significant summing/cancellation of signals with wavelengths above the giga-Hertz range! Jitter can be "thought of" similarly to quantisation errors, we effectively get a perfect signal reconstruction plus some distortion product, a "jitter spectrum" which is effectively random noise with the odd peak here and there. A decent (sub \$100) DAC/player should reduce the totality of jitter artefacts to below -120dB.
2a. It's effectively part of that "linear frequency response within +/- 0.5dB". That "+/- 0.5dB" is the total amount of frequency distortion, a combination of all the distortions in the analogue components of the DAC/player and the digital signal processing and reconstruction (which includes oversampling, reconstruction filter and jitter artefacts).

1. Yes you will, because it's likely that the recording the home audio electronics are trying to reproduce will contain timing errors in the milli-sec range and that's ignoring any reflections of the listening room/environment.
2. Yes, I agree. You seem to be talking purely about the timing errors introduced by the "home audio electronics" themselves, while I'm talking about timing errors which are going to exist anyway, regardless of the home audio electronics.

G

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2. After I experimented with the arrangement of the furniture and speakers, I used the EQ system built into my Yamaha AVR. It correctly identified a peak in the upper mids, but it didn't handle the sub well at all for some reason. It dialed the sub down to zero! I had to finesse that crossover myself. it made the rears a little too quiet too. Using the auto EQ as a default starting place, I tweaked to fix some things that the auto EQ handled ham handedly. For instance I have several listening positions in my room and I had to split differences and make compromises to make them all work equally well. The tweaks weren't huge... 3 to 5dB at the most- broad gradual adjustments, not spikes or dips.

I've had a lot of experience EQing music. It really isn't hard. It just takes a baseline to start from and analytical listening. I use specific reference recordings to balance with. For instance Reiner's Marche Slav has a descending bass pattern where each note is the same volume level. There is a Beatles song on the White Album like that too. It's really easy to spot sub crossover problems with it. I have other recordings that jump around in voices in the orchestra that are all equal levels. It helps to find gradual shifts in level across wide parts of the response. It may not be tones and SPL meters, but it can fix subtle broad things that fall between the cracks. I have recordings that are out of spec too. I put a post it on them with the corrections noted. Multichannel recordings can sometimes have level problems in individual channels. No one should be afraid to open the hood and make adjustments. The goal is for music to sound right, not to be a slave to the meter.

Everything has a resonant frequency... the lamp on the side table, the knick knacks on the shelf, even the room itself. Run a tone sweep at a healthy volume level and you'll find them all! My room has a nice big resonant frequency way down low. Thankfully, it's too low for music, but there are a couple of horror movie with a bass rumble that sets the glass in the windows and the doors all rattling. (The end of Cabin in the Woods is the worst for that.) I've gotten rid of a lot of that with weather stripping, but it still helps to put a little dip in the response there to keep the 2 by 4s in the walls from vibrating.

Yes, I play my system loud.

Last edited: Jul 25, 2019
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3. @bigshot ’s house!!!

Last edited: Jul 25, 2019
4. 1. To a certain extent that's true, particularly with stereo speakers but less so with a surround system. In practice, even if that is goal, it can only ever be partially achieved and even then, only with considerable acoustic design and treatment of the room.
1a. One of the most common errors, if not THE most common error in the audiophile world is incorrect attribution. They hear/notice something, some difference or other and then falsely attribute the cause of that difference, typically due to a lack of critical thinking and some false (often marketing driven) assumption/s. A good example of this is audiophile cables; the audiophile hears a difference and falsely attributes it to some property of the cable affecting the sound, when in fact the actual cause is their perception and there is no audible difference in the sound. The situation is far more complex with speakers because the differences between different speakers are audible, very obviously so in many cases but this fact doesn't eliminate the possibility of misattribution and arguably increases the likelihood of a partial misattribution. A dipole speaker is likely to sound at least somewhat different to a typical dynamic driver in any given room, however, even using exactly the same speaker for the centre channel as the left/right channels will still result in one being able to tell the difference between the centre channel and the left/right channels, even with a treated room and correct application of the panning law. In most home theatre scenarios a better end result is likely to be achieved using a different speaker for the centre channel, an obvious potential exception being when using an AT (acoustically transparent) projection screen.

2. The most important variable is the room acoustics and the listener/s position within that room. Both approaches can therefore work reasonably well or very poorly but generally, matching the front (3) mains would likely work better (screen allowing).
2a. Additionally, we should also consider fidelity and extremely rarely, if ever, are multichannel mixes created with mains speaker used as the surrounds. The only common exception to this is some lower budget TV (doco, etc.) which is commonly mixed on systems similar to the home/consumer 5 (identical) satellite speakers + bass managed sub systems.

1. Only partially. For example, they address the time domain in terms of the arrival time of the direct signal to the LP but even then, that's only possible to a degree as it'll be different for every LP. The freq domain is also a problem, they work by measuring the total amount of energy at the LP of the different freqs and using an EQ boost or cut to compensate for the measured peaks and troughs. The problem with this is that there is also a time domain element here too, some/many of the peaks are due to resonances which not only increases the amplitude of certain frequencies but also their duration, which further increases the total energy measured. In other words, the end result is the same amount of total energy but at any particular point in time there is likely to be too much EQ reduction (to compensate for the longer duration) and the ear/perception can (depending on context) detect this. We can clearly see this phenomena using a waterfall plot. The best solution is acoustic treatment to absorb those freqs and bring the resonance duration in line with other freqs but if this isn't possible and EQ is the only option, then the problem can't be solved, only somewhat improved according to perception.

2. For this reason, I am a fan of "doing it by ear". However, the end result depends on several factors, including the critical listening abilities/perception of the person "doing it by ear". Most consumers (including audiophiles) don't have much of a clue and typically the end result is not as good as most auto-correction systems, and I've heard some examples that were truly terrible, the end result was worse than if they hadn't attempted any correction! There are some common traps that people who don't know what they're doing often fall into and doing it yourself is quite a long winded process, comparing different measurements, calculating likely EQ settings then critically listening to various test signals and reference recordings, with some blind testing thrown in for good measure. So generally I wouldn't recommend this approach to consumers/audiophiles.

G

5. It isn't that difficult to train yourself to understand how response curves work and how to manipulate them to get the results you want. Parametric equalizers are a bit non-intuitive at first, but once you get used to them, they are easy to adjust. With me the most important thing is to have a saved baseline response to compare to. Then you can easily compare to see if your adjustments have made things better or worse.

There are technical things discussed in Sound Science that are far beyond me and make my eyes glaze over. I sort of understand Gregorio's explanation of the theory behind this stuff, at least the main points. But I after all that has been addressed, there still is a bit more that can be finessed. I can clearly hear the results of my adjustments, and that is all I really need to address problems by experimenting with solutions. I guess I don't understand why it's so complicated to tweak a response curve to get it to work a little better with different listening positions, the individual character of their room, or to better suit the way a person listens to music. It definitely is something that requires critical listening and experience with identifying bands of frequencies, but that is what audiophiles aspire to do, isn't it?

Living rooms are always the wild card and they are always the biggest compromise. If you've done what you can with room treatment and auto correction, odds are you will need to do a little more EQ kludging to get it to work at its best. There are always tons of variables, making it hard to isolate it all down to one specific theory or another. You address the primary theories and try to figure out the secondary ones. But eventually the sound traveling around the room isn't so straightforward. You listen analytically, and do a little experimenting to plus it. If you can hear it, and it works, that should be fine. There is no one size fits all magic formula.

Last edited: Jul 26, 2019
6. 1. I disagree. While I agree that it's not that difficult to understand how response curves work and how to manipulate them, it is very difficult to relate that to what's actually happening and what we're actually going to hear/perceive. Our hearing/perception doesn't work the same as a response curve, depending on the characteristics of the specific sounds we're reproducing our hearing/perception can operate more like a waterfall plot. Two sounds that are identical in every respect (including level) but are of different durations can be heard/perceived as being of different loudnesses but under different circumstances (often in the case of room reflections/resonances), two sounds that measure different levels can be perceived as being the same or nearly the same level because the brain can differentiate/separate the acoustic energy of the direct sound from the acoustic energy of the reflections. In other words, different frequency regions will have significantly different RT60 times in any given (untreated) room and therefore significantly different total energy but depending on the characteristics of the sounds we're reproducing, we can perceive that as either as a difference in level (as indicated in a response curve) or as the same level just with a longer RT60 (as indicated in a waterfall plot), or as anywhere in between. Notice also that I say "reproducing sounds", this is an important distinction because many test signals do not present the circumstances under which we can differentiate the duration of acoustic energy. Pink noise is an obvious example, all we'll hear/perceive is the total energy affecting the different frequencies and therefore Pink Noise with a different "timbre".

2. They are easy to adjust but there are two problems: Firstly is the one just mentioned, comparing to a response curve, rather than to a response curve and a waterfall plot (and perception). You therefore cannot "easily compare to see if your adjustments have made things better or worse", only "better or worse" relative to a freq response curve, not to what's actually occurring or what you are hearing/perceiving. Secondly, is a trap that many fall into, applying EQ affects the signal's phase. How it affects the phase (which freq ranges and by how much) depends on the "Q" value, the amount of boost/cut and the number and location of the EQ bands engaged. Most DIYers will measure the distance between each speaker and the LP and apply an appropriate delay to each speaker to compensate and either before or after that, measure the response curve of each speaker and apply compensation EQ to each of them. However, this ignores what's going to happen in actual use, where we're using the speakers together, not just one at a time and the fact that the relative phase between the speakers is now all over the place (variable in different freq ranges) because each speaker has a different EQ applied. A simple but revealing test: With your delay and EQ compensation in place, output an equal level of pink noise to both the left and right speakers but flip the phase of one of them. An ideal setup would result in complete phase cancellation at the LP but you'll almost certainly be surprised at how little phase cancellation actually occurs. Do the same test again but with identical EQ on both the left and right channels, most likely you'll get a far greater amount of cancellation. The take-away from this is generally: Don't try to completely "fix" all the peaks/troughs with EQ, only apply it in small amounts and, only partially correct the peaks/troughs which all the speakers demonstrate, IE. The same EQ compensation is applied to all your speakers (or at least, the one compensation for your all your fronts and then another for both surrounds) thereby maintaining much better phase integrity between them. Another revealing test is to output a single channel of pink noise from say your left speaker and then pan it across to your right speaker (try it both with and without the centre channel, IE. Panning across the soundstage using the actual centre and again but using the phantom centre). Ideally, you should hear no difference in the pink noise as you pan it but in practise you'll almost certainly hear a considerable difference, both in overall level and timbre.

3. Hopefully I've explained why it's so complicated. Sure we can quite easily tweak and achieve a much better response curve but a response curve doesn't fully describe/characterise either what's actually happening or what we're likely to hear/perceive: In practice we need a response curve and a waterfall plot, we need various different listening tests and test signals (including audio recordings) and then we need to tweak the EQ to some compromise between them all and all this is just for one LP! I can hear the potential objection; that this is all overkill, going too far with fine details that few if any can hear, however this is NOT the case! Try the tests I've suggested, the audible consequences of the effects I've described can be (and commonly are!) far greater than EQ'ing various freqs/ranges by several dB.
3a. In this case, all bets are off! I'm talking about trying to achieve more accurate playback but the "way a person listens to music" almost certainly involves personal preferences, at least some of which are not related to accuracy.

G

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7. Well, maybe I took to EQing naturally like a duck to water. Dunno. It also may be that I'm EQing for a living room and you're talking about EQing a professional application. It really doesn't have to be that complicated. An equalizer made the biggest improvement in my system that I've ever had, short of buying a house with a dedicated listening room. I think everyone should EQ, not matter how far into the ozone they want to take it.

Last edited: Jul 27, 2019
8. Really bigshot, that's almost exactly the predicted response! Thanks for playing along

1. It makes no difference, the principles of EQ, phase, room acoustics and perception are the same. There are some differences in the case of theatrical mixing, as we're using mix rooms/stages many times bigger than an average living room but music recording control rooms, mastering rooms and most rooms used for mixing TV are roughly the same size as an average living room. The only other difference is that in "a professional application" it's more critical that we get it right!

2. Whether you like it or not, it is that complicated! Of course, you're entirely free to hugely oversimplify it if you want but then of course the chances of you getting it wrong are likewise also hugely increased.

4. On what basis do you advise that? If it's a basis of personal preference, then "maybe" but if it's a basis of accuracy/fidelity then "probably not" because ...
4a. Who is talking about going "far into the ozone"? Again, I'm not talking about fine details, the effects I've described can (and often do) have a much bigger impact on a system's overall performance than just EQ'ing each speaker to match a response curve. In other words, it's entirely possible that you're the one going "far into the ozone", concentrating on improving an issue of lesser importance while ignoring an issue of greater importance! Isn't this exactly the sort of stereotypical audiophile trap/fallacy that you yourself frequently argue against?

It's possible, though very unlikely, that you haven't introduced some significant amount of phase (or other mentioned) issues and therefore that doing it properly wouldn't make much of a difference in your particular case. If this is the case, then it's due to a lot of luck and your experience/reference of professional listening environments (studios) but of course, neither of these conditions apply to "everyone"! I therefore do not think "everyone should EQ", in many/most cases an auto-correction system would likely do a better overall job, unless the person really knows what they're doing.

G

9. The principles are the same. The purposes are different. A home doesn't require precise calibration like a recording studio does. And a home often doesn't have a physical layout that is conducive to precise calibration. In a living room, you have listening positions scattered throughout the room, not at a single spot. You have furniture in the room to make it livable. You have walls made with normal home masonry. You've got low ceilings, windows, tile floors, windows on the left- bookcases on the right, all sorts of things that you would never find in a studio. You can arrange your furniture and do room treatment to a point, but it still has to function as a living room. It isn't every going to be perfect to six decimal places.

There are a million variables involved, and all of these less than optimal aspects require balancing to take the curse off. The best way to do that is to calibrate, then live with it a little. As you do, formulate a plan for making small adjustments to find the curve that works for your particular living room. If you do it in small steps and always keep the calibration as a baseline, you won't wander astray. It'll never be perfect, because no living room is anywhere close to being textbook perfect. But it will work given the concessions to livability. Accuracy is important to a recording studio. Functionality is important to a living room.

I also strongly believe that everyone has their own tastes in how sound should be presented, and they shouldn't be discouraged from following them with their own home stereo system. They put salt and pepper on the table in restaurants for a reason. If you use a little, that isn't the same as pouring ketchup all over your meal.

I do believe that it is a very good idea to start out from as precise a calibration as you can. But there is absolutely no sin in massaging that calibration a bit to make it work better for your purposes and ears. No one should be afraid of touching an equalizer. Sure, equalizers can do a lot of horrible things if you use them wrong, but that is true of any tool. You learn how to use it, and it helps you get closer to your goal... and that goal may be a personal one, not necessarily a strictly objective one. If you start as accurate as you can, then you can make an informed choice of where you choose to deviate from strict accuracy. Your room should sound the best to you that your room possibly can. That may or may not be strictly calibrated. But always start from calibrated.

Audiophiles insist that every improvement, not matter how small is important. They have an absolutist approach that makes them follow incremental improvements beyond any sort of practical level. Sound science people can do the same thing when they hammer on accuracy too far. Accuracy is fine, but it has to be tempered with practicality for the intended purpose. Absolutism just leads down rabbit holes and becomes a baseball bat to hit other people over the head with in internet forums.

Form follows function. Theory isn't the end goal, in itself. It's just a tool to help lead to it. Living rooms are a bunch of compromises. Headphones instead of speakers are a big compromise. You just do the best with what you've got. If you can custom build a home studio that performs to spec, great! That is the ultimate. But not a lot of people have the ability to do that.

I agree that running an auto EQ is better than nothing, and I agree that people who just want plug and play and don't have an interest in learning to EQ shouldn't mess with manual EQ.

Last edited: Jul 28, 2019
10. 1. Firstly, recording studios don't have precise calibration, they have ball park calibration because even with a very large budget and control over the construction and acoustic treatment, it's still not possible to get very close to an "ideal". Virtually all studios still have freq response peaks and troughs of 3-6dB. Secondly, I'm not talking about precise calibration anyway, I'm talking about using EQ to get the best results you can, regardless of whether any other sort of treatment is available or practical.

2. No bigshot, that is NOT true! Calibrating each speaker to a FR curve might be the only "way" you are aware of, or, it might be the simplest/easiest "way" you can be bothered to use but it is NOT the best way. The best way is along the lines I've described; FR and waterfall plots, various tests for phase and other issues and an EQ setting that represents the best compromise.
2a. This is also untrue. There is a high likelihood that you WILL "wander astray" because your "way" does not consider phase issues between the speakers (and other effects) which can have a bigger affect on the performance of the system as a whole!
2b. True, it'll never be perfect but neither is any professional studio.
2c. No bigshot, you do not know if it will work unless you run tests to find out. It's entirely possible that it will NOT work, due to a number of issues you are ignoring!

3. That's the point I'm trying to make bigshot, getting the best accuracy you can using EQ, which does not affect the functionality of a living room.
4. I believe that too. The problem is that what you're actually doing (and advising) is quite different from what you say you believe! You are NOT "starting out from as precise a calibration as you can", you are starting out with a calibration of each of your speakers relative to a FR plot, which is not precise relative to what's actually occurring, it completely fails to consider the phase interaction between the speakers (and other effects). Your starting point is therefore an imprecise calibration, possibly a hugely imprecise calibration and certainly NOT "as precise a calibration as you can"!

5. Agreed, everyone is free to change their system to satisfy their preferences. But if you're talking about massaging a "calibration that's as precise as you can [get]" then that calibration has to actually be as precise as you can get, not some screwed up calibration because you didn't know about or couldn't be bothered to do the required tests.

6. That's exactly my point, the fact that you're effectively advocating the wrong use of equalisers! ... Equalisers always "do a lot of horrible things" however you use them, they always affect phase and different EQ settings affect phase differently. Calibrating/EQ'ing each speaker to a FR plot will result in a different EQ setting for each speaker and therefore a different phase relationship between each speaker. The right way to use EQ in this scenario is therefore not to calibrate/EQ each speaker to an FR plot but use the same EQ setting for all of them (except the sub of course). That way you're affecting the phase of each speaker's output exactly the same and maintaining the phase relationship between them. And again, we're not talking about some small, inaudible or nearly inaudible theoretical effect here. It might be relatively small/inaudible in a few specific cases but it's just as likely to be relatively huge, in most cases it makes a very significant difference and is clearly isolated/revealed by the tests I suggested.

7. Then why don't you? Why do you advise a "way" that is not doing "the best with what you've got" and in some cases may actually be worse than not doing anything?!

Bigshot, you've done something with your system that you perceive as the best you can get (given your restrictions) or at least somewhat close to it but you haven't tested it properly to actually find out, you then advise everyone else to do the same, regardless of the fact that your advice ignores and/or contradicts some highly relevant science/facts. Upon being challenged/refuted, you just rephrase and defend the same assertions, multiplying the contradictions. Isn't this pretty much exactly what so many audiophiles do and what you spend a considerable amount of your time arguing against?

G

Last edited: Jul 29, 2019
11. You're talking about how studios do it. I'm talking about how regular people do it... using normal consumer electronics, like an AVR with Audessey and five band parametric equalizers on each channel. Normal people with furniture in their living rooms and multiple seating positions. Normal people who want to optimize the sound of their home audio system using EQ. I keep repeating what I'm talking about and you keep answering as if I'm talking about something different. The reason we keep talking past each other is that you don't see the parameters I'm speaking about. You are redefining it to your own parameters and my comments don't fit that context.

Imagine you're someone with a typical living room. Your wife limits what you can do with it. You don't have access to anything except the Audessey measuring mike and your AVR. This describes 90% or more of the people who might be interested in this subject. Now read my comments with that situation in mind. I think that might make it clearer to you what I'm saying. Compromises and circumstances dictate the degree of precision you can achieve. But even with compromises and circumstances, EQ can improve the sound of a home system. Just because you can't measure everything to the nth degree, it doesn't mean that you haven't made a significant improvement. There are techniques for doing that besides measuring and making charts. They may not be as precise, but they do work.

Last edited: Jul 29, 2019

12. If the goal is accuracy of reproduction and you don’t have anything other than Audyssey and it’s associated mic, most people should stop there and not try to make manual EQ adjustments without the ability to properly measure the impact.

If the goal is personal preference, have at it. But don’t refer to the result as an accurate/improved calibration.

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13. The goal is the best sounding music in your living room from whatever listening position. The best sound is always the goal isn't it? A studio needs accuracy because they need consistency from session to session. You might start a mix in New York and want to finish it in LA. It has to be calibrated. A home doesn't serve that function. It just needs the best sound. Absolute accuracy isn't really a practical goal in a home situation. It's difficult enough in the controlled environment of a studio. Too many variables and compromises to make. You do the best you can at achieving accuracy, then you see if you can make it a little bit better.

In my case, if I had followed the settings of my Yamaha auto EQ, my sub would be dialed down to zero, and the rear channel would be very weak sounding. Should I have just stuck with that because it's "accurate"? Nope. I trust my ears for what accurate sounds like too. I can use them to EQ. I used the parts of the auto EQ that worked and revised the ones that didn't. An ideal response curve isn't a hard and fast fixed thing. It's a range, and your particular circumstances and ears have an impact on what the ideal curve is. Calibration is just the starting point, not the destination. There are too many variables in a normal living room to depend just on that.

If you do want to stick to accuracy, then you need to remove the compromises. Have just one listening position. Remove a lot of your furniture. Apply acoustic treatment everywhere you can. Create a room with the proper angles, proportions and building materials. If you are able to create something resembling a sound studio, then accuracy will work fine. But if your living room isn't like that, you need to accommodate the compromises. One size does not fit all in that case.

I see people all the time posting pictures of their listening situations. One guy had a big plate glass window on the right and bookcases covering the left wall. His fireplace wasn't centered in the room, so the wife insisted that he put the speakers evenly on each side of the fireplace shoved against the wall. One was in the corner, the other was around the middle of the room. His listening position was smack dab in the middle of the room. He was asking for advice, and the advice was pretty obvious. But it was equally obvious that he wouldn't be able to follow all of it. How he would make the changes he could make was just as important as how he would deal with the changes he couldn't make. You do as much as you can with the room acoustics, then you calibrate, then you adjust the calibration a bit to make the compromises work a little better.

Life is a compromise. How you deal with the compromises matters. It isn't good to just follow the rules and then just ignore the rules you can't follow. You have to find a way to massage that stuff into working as best as it can. Is that perfect in an absolute sense? No. Is it important from a practical standpoint? Yes.

Last edited: Jul 29, 2019

14. Agree that having it sound the best is the goal, but disagree that using your ears to get there is actually going to give you that result. You keep ignoring issues other than FR.

A specific question on your approach - without measurements, how do you identify, let alone correct for phase cancellation around the crossover point of your sub and the other 5 speakers (or in my case, multiple subs)? That’s likely a bigger issue in most HT setups than the sub’s in room FR. In my experience, it’s extraordinarily difficult to well by ear and something many baseline AVR EQ solutions get wrong if they attempt to address it at all, I’m choosing that example because it’s not limited by mixed room concerns, has large impact, shows up in FR, and is usually easy to remediate using measured response and the settings commonly available in a consumer AVR.

15. The sub is the one thing that I couldn't follow the AVR's auto EQ on. I have one sub placed in the optimal spot for it very close to the mains. It is a really good one and fills the room easily by itself. It has its own auto EQ and I ran that when I first got it. It doesn't have a display, so I have no idea what it did. So maybe even though the AVR failed at that, the sub corrected for it itself. I don't know. The phase on the sub is set to normal and the crossover is at 80Hz, and I've checked the crossover with tones to make sure it is balanced handing off to the mains. I think trying to do that with multiple subs in different positions and a lower crossover might have given me a lot more problems. The AVR has the distances of the speakers set. Phase is a little more fluid in a multichannel system. I use a stereo to 5.1 DSP that I'm sure is altering phase a lot. Not sure exactly how it works. It just sounds really good so I use it.

Maybe I lucked into a good spot. Or maybe keeping it simple handled it. Or perhaps the sub calibration did something. Not sure,