SME

Time to kill the myth that "flat" bass is "correct" bass.

65 posts in this topic

I've invested a lot of time, as of late, to trying to determine the best way to calibrate an audio system.  When calibrating for in-room response, it may be readily observed that a flat frequency response does not sound so good.  The result is typically too bright indoors and lacks bass both indoors and out.  Sadly, many authorities argue for calibrating systems this way and claim that this "reference" response is the correct response and is something the listener should "get used to" unless they want to alter the response to suit their preferences.  Let me just say right here right now that this is B.S.

 

Numerous blind listening studies by Harman indicate that tonal balance preferences are actually quite universal.  Harman's work indicates that subjective sound quality impression correlates strongly to the flatness of anechoic response and the smoothness and monotonicity (greater toward the bass) of the reflected and power responses.  These preferences have also been shown to be consistent between trained and untrained listeners as well as between American and Japanese listeners,  the latter of which shatters the stereotype that Japanese listeners prefer more treble and less bass than American listeners.

 

My current calibration approach is based on the assumption that the first arrival dominates the listener's perception of tonal balance.  This is perhaps the simplistic explanation for why an anechoically flat speaker sounds best in a wide variety of listening rooms.  My reasoning is that the human hearing system analyzes the in-room sound to separate the sound of the source from the acoustical effects, where possible.  The latter point, "where possible", is key because in many situations (including bass, in general), the hearing system cannot distinguish the source independently from acoustical effects.  For practical purposes, first arrival may be approximately determined from an impulse response measurement using the FDW feature in REW.  A 1/3rd octave window works outstanding for my situation, but I haven't verified that this will work for everyone.

 

I calibrate so that my first arrival response is flat, except for adjustments at the top and bottom end.  I recently experimented with in-air high frequency absorption as a model to use for defining a family of roll-off curves, allowing adjustments to be made as needed for different content.  This experiment has been very successful, so I thought I'd try applying a similar model-based approach for determining bass lift, which is necessary for a balanced sound.

 

To understand the need for bass lift in the first arrival response, it's helpful to recall what the true reference is: an anechoically flat speaker.  Virtually all monitors used in professional audio production are designed to be as flat on-axis as possible when measured anechoically, excepting the roll-off at the top that is often added, particularly to monitors intended for near-field use.  What happens is that as frequency gets lower, reflections arrive too early to be distinguished from the first arrival sound.  Instead, these earliest reflections combine with it, contributing bass lift to the first arrival sound.  This is in addition to the bass lift observed in the full in-room response owing to both the perceived first arrival sound and the later reflections.  As a consequence, even anechoically flat speakers benefit from some bass lift when used outdoors or in very large spaces, at least if the content was mastered in a smaller room as music usually is.

 

So I'll spare the details of my analysis as it mainly involves talking about a lot of numbers, which vary quite a bit even between recommended music monitoring environments.  The typical recommendations are that the anechoically flat speakers be placed at least 5 feet from the rear or side walls, preferably in a room with a taller ceiling than the usual 8 feet that's common here in the states.  Taking into account these recommendations and assuming a modest listening distance of up to 20 feet or so, I determined that the primary contribution to bass lift arises due to the floor reflection.  The frequency region and magnitude of this lift will vary considerably depending on the height of the bass/mid driver(s) relative to the floor and the reflectivity of the surface.  By my calculations, the lift due to floor bounce falls somewhere in the range between 100-500 Hz or so.  A reasonable guess for magnitude is 3-4 dB, but it could be more in some circumstances.  I haven't nailed down any sort of "best" settings to use for the widest variety of content, and indeed, the ability to adjust the center frequency and magnitude may be necessary to achieve optimal sound with a wide range of content.  However, in my experimentation I find I get pretty good results with a +3 dB shelf centered at 130 Hz.  Any higher than that, and I get noticeably too much mud with some content.

 

Some secondary contributors to first arrival bass lift arise due to the side walls, ceiling and rear wall.  However, if the system setup follows recommended practice, these reflections won't contribute much first arrival boost until well into the sub range.  Each of these reflections will also contribute less to the first arrival response than the floor bounce will.  Right now, I'm running a +1.25 dB shelf at 52.5 Hz, and another +2 dB at 25 Hz.  Note that most monitors, even the nicer ones, don't extend below 40 Hz or so.  For content that contains substantial deep bass, subwoofers will usually be used during the monitoring process.  This presents a significant complication because the subwoofer must be some how level matched to the main monitors, and the methods of doing this vary.  However, as long as the sub is placed close to the monitors, an in-room frequency response level match ought to also provide a first arrival level match.  If the sub itself is not EQed and is anechoic flat, then some additional first arrival bass rise is warranted.  Below 20 Hz, I would say there is no clear guidance because almost any sub used for monitoring will begin to roll-off at that point, even as room gain (both first arrival and overall) begins to rise rapidly as well.  I believe in the limit of low frequency, all room gain is essentially first arrival room gain, so it probably makes most sense to just keep things flat below 20 Hz, except for maybe a gradual roll-off toward the low extreme of the system to minimize phase delay.

 

For movies, the situation again varies depending on whether the mix is done at mid or far field in a dub-stage, at mid field in a smaller room (most consistent with music mastering practice), or in near field (also used for music sometimes).  As the ratio of distance vs. height increases as would be expected in a dub-stage, the magnitude of the floor bounce will increase somewhat.  Some additional floor bounce boost may be justified for dub stage mixes.  However, side-wall and ceiling reflections may arrive too late to substantially contribute to first arrival sound, so it might be better to avoid much (if any) boost in the 40-50 Hz range.  In the near-field situation the floor bounce may be diminished a bit more, and other room reflections may be a bit weaker, suggesting that less boost is appropriate overall.

 

Anyway, my work here is very preliminary.  I haven't done nearly enough listening to convince myself that my settings for bass lift here are necessarily optimal.  I am reasonably happy with the bass right now.  Something to keep in mind is that the amount of bass lift may be somewhat less important than the shape of the lift.  If the shape is wrong, it will sound unbalanced.  Unfortunately, I'd bet that a lot of people including even many on this forum don't know what balanced bass sounds like.  I admit, I'm not even certain of this, but I've learned to trust my preferences and am able to easily make adjustments that make things sound better or worse.  For example, balancing 40-50 Hz with 90-120 Hz is crucial.  Too little 40-50 Hz, and kick drum lacks weight and power.  Too much, and it sounds muddy with little punch or impact.  I'd be willing to bet that a lot of people run 40-50 Hz way too hot relative to 90-120 Hz and drown out all the punch.  Yet, as noted above, 40-50 Hz should probably still be running 1-2 dB hotter than 90-120 Hz, when measured in the first arrival.  In typical rooms, significantly more room gain may appear in the full in-room measurement than in the first arrival measurement.

 

Interestingly, the importance of first arrival may partly explain why near-field subs often sound better and feel more tactile.  When placed near-field, the ratio of direct to reflected sound may increase compared to other placements, and as long as the user calibrates in-room response to the same level and target curve, the subs will have higher first arrival SPL where direct-to-reflected sound ratio is higher.  As such, it may be possible to get good tactile punch from subs placed in other parts of the room, provided the system is calibrated in a manner more like I describe here.  Of course, the near-field placements may still offer more headroom and less interference from acoustical effects, especially where high Q room modes may be present.

 

So anyway, I can't really give a definite prescription for calibrating bass response, but I can state with moderate confidence that the notion that in-room or even first arrival bass response should be flat for a correct listening experience is total rubbish.

 

Edit: too much 40-50 Hz relative to 90-120 Hz causes bass mud and lack of punch or impact.  (These are pretty rough ranges, of course.)

Edited by SME
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Great post :)

 

 

Just to check, re: this bit:

 

 

 

Too little, and it sounds muddy with little punch or impact

 

 

Should it read:

 

 

 

Too much, and it sounds muddy with little punch or impact

 

or:

 

 

 

Too little 90-120Hz, and it sounds muddy with little punch or impact

 

?

 

 

It reads to me like it's missing something as-is!

 

 

With regards to the theory of floor bounce being the dominant boost to first arrival bass and the side walls and ceiling not adding much, would it be wrong to extrapolate that and postulate that a BossoBass-, SBA- or DBA-style array, with subs placed not just near the floor but also near the ceiling, would not just manage/remove room-nodes, but also offer a 'cleaner' in-room response due to the ceiling (if not also side-wall?) reflections arriving at the same time as the floor reflections?

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While you talk of first arrival, it's barely something that exists at the lowest frequencies in most of our listening rooms as you note at the beginning.  While related, I would say the missing component from your study/experimentation is paying attention to decay and energy vs time.  The ultimate litmus test is to create a correlation that translates the same subjective bass quantity from outdoor listening to a small listening room.

 

Floyd Toole has written a few times about work with the Synthesis systems that getting the same subjective bass from a very large, leaky space to a small, rigid room can require a lift of as much as 10dB.  We generally perceive loudness based on sound power, not just intensity, so it really is all about the intensity vs frequency vs time and finding a handy way to quantify different decay with respect to how much we hear as part of the original vs late energy.  I'm sure this requires some form of variable time window, but confidence in such a correlation still remains elusive.

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Great post :)

 

 

Just to check, re: this bit:

...

Should it read:

...

or

:...

?

 

 

It reads to me like it's missing something as-is!

 

 

With regards to the theory of floor bounce being the dominant boost to first arrival bass and the side walls and ceiling not adding much, would it be wrong to extrapolate that and postulate that a BossoBass-, SBA- or DBA-style array, with subs placed not just near the floor but also near the ceiling, would not just manage/remove room-nodes, but also offer a 'cleaner' in-room response due to the ceiling (if not also side-wall?) reflections arriving at the same time as the floor reflections?

I corrected it.  I meant to say that too much 40-50 Hz, relative to 90-120 Hz causes muddiness and lack of punch or impact.

 

As for the BossoBass design or other floor-to-ceiling arrays, they substantially reduce discrete floor and ceiling reflections entirely, which is definitely beneficial in most circumstances.  Similar to (and often better than) near-field placement, these typically improve the ratio of first arrival to late arriving sound.  If you EQ in-room response to a target, then the bass will probably contribute more to tonal balance using the array if you keep the target the same.  If you opt to adjust the target in each scenario to achieve a balanced sound, then the array will likely measure lower in frequency response SPL and have lower decay time.  Either way, intelligibility should be improved.

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While you talk of first arrival, it's barely something that exists at the lowest frequencies in most of our listening rooms as you note at the beginning.  While related, I would say the missing component from your study/experimentation is paying attention to decay and energy vs time.  The ultimate litmus test is to create a correlation that translates the same subjective bass quantity from outdoor listening to a small listening room.

 

Floyd Toole has written a few times about work with the Synthesis systems that getting the same subjective bass from a very large, leaky space to a small, rigid room can require a lift of as much as 10dB.  We generally

perceive loudness based on sound power, not just intensity, so it really is all about the intensity vs frequency vs time and finding a handy way to quantify different decay with respect to how much we hear as part of the

original vs late energy.  I'm sure this requires some form of variable time window, but confidence in such a correlation still remains elusive.

 

Creating a correlation is very much what I am trying to do, and I think I may have stumbled upon a successful approach (relying on the concept of first arrival) that could potentially work in a wide variety of rooms.  Even though, I admit that I haven't tested it outside my own, and that such a correlation will be very hard to verify conclusively.  I don't have a bunch of research subjects nor multiple listening rooms to scientifically verify this, so I have to do the best I can with what I've got.

 

My hypothesis is that first arrival still exists for bass in a perceptual sense.  Because reflections that arrive early relative to the period of the wave can't be distinguished independently from the direct sound, the combined sound is perceived to be the first arrival.  At the same time, reflections that arrive later may still be distinguished from the first arrival, and the brain may treat this information differently than it treats the first arrival sound.  I believe some research suggests that the hearing system can be modelled with filters that are no broader than 1/3rd-octave, which sets an effective limit (uncertainty principle) on the temporal resolution of such filters.  Hence, 1/3rd octave Gaussian FDW (but not 1/3rd octave smoothed frequency response, which is totally different and largely meaningless as far as perception is concerned, IMO) time-frequency analysis may be a very reasonable approximation of the first stage of audio processing, after which, first arrivals are identified and processed along with information from later sound to attempt to analyze the source independently from room effects.

 

You are correct that I did not mention decay and energy vs. time, but I hypothesize that these have other, mostly orthogonal consequences for sound quality.  Depending on their level relative to the direct sound, late reflections may interfere with the brain's ability to distinguish first arrivals from the rest of the sound.  This will be dependent on content as well, with acoustic cues in the recording itself typically being the first casualties of excessive decay.

 

I assume Toole's work in which subjectively balanced bass response may differ by as much as 10 dB is considering the frequency response SPL rather than the output of the speaker or its in-room first arrival sound.  This is a crucial distinction to make.  Assuming it's frequency response SPL, this is entirely consistent with my first arrival hypothesis because differing amounts of late arriving energy in each situation contribute to higher or lower SPL, despite the same subjective balance and (hypothetically) same first arrival response.  In my original post, I'm reasoning that even the first arrival sound should have some bass lift because of boundary effects on the first arrival sound from the anechoically flat monitors used during content production.  OTOH, if they are actually boosting an anechoically flat speaker by 10 dB to get a balanced sound in-room, then perhaps I should try to track down that research to better understand their reasoning.

 

Your comment about perceiving loudness based on sound power (I think you meant "sound power over time") vs. intensity is interesting.  That's because while it's certainly true under otherwise consistent test conditions (i.e., varying the duration and/or level of a burst test signal), loudness perception also depends on other things like the overall tonal balance of the sound.  By my hypothesis, this relates back to first arrival response.  Where things become more interesting is that there is also evidence that first arrival SPL may influence apparent loudness substantially more than later reflections.  The obvious example is that theatrical reference level, calibrated using a continuous test signal, is usually too loud in smaller rooms (even those that are well-treated) and especially in near-field conditions.

 

Perhaps the more crucial thing I failed to mention is that above the frequency at which a nearby boundary combines constructively with the direct sound as part of the first arrival, there is a region where the combination is destructive and causes a broad dip in the first arrival response.  For floor bounce, this dip can be quite substantial, and it's probably there in almost every mastering room that doesn't use EQ of some sort to try to correct it.  My guess is that most experienced engineers are acquainted with this phenomenon and take care not to boost too much to compensate there, but I honestly don't know.  It's also possible that some kind of psychoacoustic compensation occurs where the brain has enough information at higher frequencies to identify the reflection as distinct.  However, I am doubtful this is the case because it contradicts my own experience.

 

To give an example:  My center channel currently exhibits a significant on-axis direct sound hole around 190 Hz because it is placed flat against the wall without significant absorption on either side.  At the same time, significantly more sound at 190 Hz is radiated off-axis and eventually reflects from the side-walls to the on-axis listening position.  (I hope to fix it eventually by effectively extending the baffle, but that's another story.)  I did an experiment in which I boosted the dip to make the first arrival sound flat at the on-axis position.  I have a lot of headroom, so was able to get away with it for the most part.  The measured frequency response SPL in that region was up to 100 dB with a -20 dBFS sine sweep and a calibration to theatrical reference level, but the tonal balance actually sounded pretty normal at that seat position, albeit with a strange side-effect.  The side effect is that any content centered there tended to induce a lot of strange tactile sensations as it resonated with my skin and even my clothes.  I could also hear and even feel the longer decay time of that sound, but this was perceived to be independent from the perceived tonal balance of the source.  Of course, I opted to abandon this approach because the first arrival SPL was even more excessive at the off-axis seats, which totally ruined dialog intelligibility and caused excessive boom there.

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Creating a correlation is very much what I am trying to do, and I think I may have stumbled upon a successful approach (relying on the concept of first arrival) that could potentially work in a wide variety of rooms.  Even though, I admit that I haven't tested it outside my own, and that such a correlation will be very hard to verify conclusively.  I don't have a bunch of research subjects nor multiple listening rooms to scientifically verify this, so I have to do the best I can with what I've got.

 

 

 

I'll volunteer.  I can make measurements (with and without FDW), send them to you, and see what you come up with for EQ solutions, and I will give you feedback.

 

We can make this single-blinded if you want, where you send me MiniDSP 10x10Hd Configs, and I'll compare configs by ear them without knowing what EQ/bass lift amount is there.

 

I can also do post-EQ measurements after subjective impressions have been tabulated.

 

 

JSS

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I'll volunteer.  I can make measurements (with and without FDW), send them to you, and see what you come up with for EQ solutions, and I will give you feedback.

 

We can make this single-blinded if you want, where you send me MiniDSP 10x10Hd Configs, and I'll compare configs by ear them without knowing what EQ/bass lift amount is there.

 

I can also do post-EQ measurements after subjective impressions have been tabulated.

 

I'm interested in taking you up on your offer.  But I'll need more info and we may run into some issues.  One of those is crossovers, where phase difference come into play requiring measurements with a consistent, reliable timing reference.

 

For starters, have you tried the 1/3rd-octave FDW approach I've suggested for calibration at all yet?  If not, we should start there and calibrate your mid-range and treble before worrying much about bass lifts.   Indeed, I wouldn't trust any kind of subjective evaluation of bass performance without having mids and treble balanced.  Too much treble in the wrong places can easily ruin bass, perceptually speaking.

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I used 1/3 FDW, but used it inappropriately, using it far to low in frequency.  I need to try an amalgam of long-window EQ for bass to tame resonances, and FDW above 150-250Hz or so, or above where I no longer have a minimum phase response.

 

I use LR24 crossovers, within the AVR.  My AVR provides an LR24 lowpass, and only a BW12 highpass, assuming you have sealed speakers and that you are using the -3dB point as the crossover point.  I cross higher than my LCRS -3dB point, and add a BW12 via MiniDSP to the AVR's highpass to get an LR24.

 

Measurement would be tough, as I measure baselines from my soundcard directly to the amps powering the speakers.  Running through HDMI and Umik to see the crossover's effects, there is no timing reference.

 

JSS

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I used 1/3 FDW, but used it inappropriately, using it far to low in frequency.  I need to try an amalgam of long-window EQ for bass to tame resonances, and FDW above 150-250Hz or so, or above where I no longer have a minimum phase response.

 

I use LR24 crossovers, within the AVR.  My AVR provides an LR24 lowpass, and only a BW12 highpass, assuming you have sealed speakers and that you are using the -3dB point as the crossover point.  I cross higher than my LCRS -3dB point, and add a BW12 via MiniDSP to the AVR's highpass to get an LR24.

 

Measurement would be tough, as I measure baselines from my soundcard directly to the amps powering the speakers.  Running through HDMI and Umik to see the crossover's effects, there is no timing reference.

 

JSS

 

Did you listen with the filters anyway?  How did it sound?  How about just the mids and highs?  Anyway, it should suffice to treat any resonances (acoustic or speaker) with higher Q filters first and then use the lower Q filters to shape the 1/3rd octave FDW response to the intended target for the full range of frequencies, bass included.

 

If the raw responses of speakers and subs share a timing reference, then I can correctly sum them to evaluate the combined response.  Using REW's interactive editor, I don't see any obvious way to do this though.  I would have to do this with my own code, which I haven't yet re-factored from the proof-of-concept I used to EQ my system back when I was running the OpenDRC-ANs.  That said, I plan to revamp that code very soon, so I can integrate all my new subs.

 

Do you have a .mdat for the raw LCR?  I don't see an equipment list for you.  Things I would be interested to know: speaker makes/models or DIY specs, sub crossover frequency, amp power, room dimensions (or rough description if irregular like mine), seat distance from speakers, etc.  Do you have any measurements that involve more than one location?  Multiple locations are the surest way to distinguish resonances from ephemeral peaks and dips that shift with location.  IMO, only the former should be EQed in most circumstances.  Otherwise, if you know approximately where the modal region starts in your room, that would also be helpful.

 

Send over the data you have, and I'd be happy to take a stab at doing an "optimized" HD 10x10 profile.  Maybe it'll sound bad, and we'll get a good laugh about it later.  Or it'll sound good, and you can be the first member of my new cult of audiophilia.  ;)

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Since I rearranged some things in the room, I do not have a current .mdat, but that is easy to get, my last two .mdats are 8 locations for each speaker (3 very near MLP, rest on other seating locations.  I use:

 

LCRS - 7 'Big Malcolms' - SEOS12/DNA350 with 4 Aura NS6s, crossover by bwaslo.

Subs - 8 Dayton RefHF 15s in 3 cuft sealed.

 

I cross at 100Hz to keep the NS6s under Xmax with significant power.

 

Each LCRS has 1 channel of an A500 amp.

 

The subs share 2 EP4ks.

 

I use a MiniDSP 10x10 for LCRS and Sub processing.

 

The AVR is a Denon DN500-AV.

 

 

JSS

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Sorry, but there's no myth here. Surely, no thinking person would say that these two exact same subwoofers with different signal shaping upstream placed differently (from a GTG experience) would sound similar or accurate?

 

art-716025.jpg

 

I prefer Olive's opinions:

 

 

Accuracy is Not Applicable to Most Recordings Made Today


Most recordings made today are not intended to sound like the live performance. Anyone who heard Taylor Swift's live performance with Stevie Nicks at the 2010 Grammy Awards understands why. (Note: you can relive the magical moment on Youtube. Warning: this may be offensive for the musically-inclined). About 90% of commercial recordings are studio creations consisting of a series of overdubs, processed with auto-tuning, equalization, dynamic compression, and reverb sampled from an alien nation. For these recordings, there is no equivalent live performance to which the recording/reproduction can be compared for accuracy. The only reference is what the artist heard over the loudspeakers in the recording control room. If the important performance aspects of the playback system through which the art (the music and recording) was created can be reproduced in the home, then the consumer will hear an accurate reproduction of the music, as the artist intended. It is possible to achieve this if we adopt a science in the service of art philosophy towards audio recording and reproduction.

The bolded part is by me as it sums my consistent opinion on the subject over the years in one sentence.

Ever since the invention of bass and treble "controls" the distortion of recorded source through playback devices was off to the races. That grotesque distortion (typically +10dB <100 Hz and many times up to +20dB manipulation of the signal) has become indelibly etched into generations of brains.

The nearly unanimous misapplication by enthusiasts (and many so-called professionals) of the Equal Loudness Curves has served to justify this learned preference for distortion.

 

The point here is that subjective preference is irrelevant. Utterly and completely irrelevant. Unless, of course, you're using it to sell audio hardware or to justify your systems lack of accurate playback capability.

 

Who here can explain what the sound effects in Hacksaw Ridge should sound like or review a systems calibration as to the accurate playback of those effects by employing subjective listening preference as the measurement tool?

 

Flat level calibration and response magnitude, through placement and phase/trim adjustments is the best method to approach accuracy. Preference-led manipulation thereafter is generally unlistenably distorted, although enjoyed, celebrated and defended regularly in these forums.

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The point here is that subjective preference is irrelevant. Utterly and completely irrelevant.

 

This statement sums you up perfectly, and is probably why you find so much contention on countless boards.  

 

In this hobby, subjective preference is everything.  Neither you nor anyone else can tell someone what they should like in their system.

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This statement sums you up perfectly, and is probably why you find so much contention on countless boards.

 

In this hobby, subjective preference is everything. Neither you nor anyone else can tell someone what they should like in their system.

Funny, I was going to say something very similar.

 

How I choose to setup my system to my preferences is the ONLY thing that is relevant. I prefer a lot of bass and mid bass. But, not so much to where is sounds bloated or muddy. However, I think a flat system sounds awful. I find a middle ground between the two. I suspect I would walk out of your room due to your preferences. Does that make yours wrong? Certainly not. It's YOUR preference; I just happen to prefer something different. Isn't that what this hobby is all about? Sharing our own preferences without someone coming in and shitting on them?

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Funny, I was going to say something very similar.

 

How I choose to setup my system to my preferences is the ONLY thing that is relevant. I prefer a lot of bass and mid bass. But, not so much to where is sounds bloated or muddy. However, I think a flat system sounds awful. I find a middle ground between the two. I suspect I would walk out of your room due to your preferences. Does that make yours wrong? Certainly not. It's YOUR preference; I just happen to prefer something different. Isn't that what this hobby is all about? Sharing our own preferences without someone coming in and shitting on them?

 

I assume this is directed toward Dave and not me?  ;)  :D

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Hacksaw Ridge: Digits vs Mic'd

 

3fb9ef3011464e1ad17055a0468bd2ed.gif

 

Doesn't sound thin (or any of the silly adjectives used to describe accurate playback in these forums) at all. The Oscar winning re-recording mixer saw to that.

 

I'm pretty sure SME is talking about calibration methods here and not trying to argue that anarchy is the preferred course because otherwise sensibilities might be triggered.

 

A flat response is the reference in-room. It's just common sense. What one does to the response after that is his or her choice... of course, but not relevant to calibration.

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This statement sums you up perfectly, and is probably why you find so much contention on countless boards.  

 

In this hobby, subjective preference is everything.  Neither you nor anyone else can tell someone what they should like in their system.

This post sums you up perfectly and is probably why you end up apologizing in PM regularly.

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Funny, I was going to say something very similar.

 

How I choose to setup my system to my preferences is the ONLY thing that is relevant. I prefer a lot of bass and mid bass. But, not so much to where is sounds bloated or muddy. However, I think a flat system sounds awful. I find a middle ground between the two. I suspect I would walk out of your room due to your preferences. Does that make yours wrong? Certainly not. It's YOUR preference; I just happen to prefer something different. Isn't that what this hobby is all about? Sharing our own preferences without someone coming in and shitting on them?

 

 

Hi Rowan,

 

Can you post any data that helps explain your preference vs flat response? Trying to stay on topic here with calibration methods that explain preferences. For example, I've found that what most people think is a flat calibration is usually far from it.

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This post sums you up perfectly and is probably why you end up apologizing in PM regularly.

 

I stand by my statement and I certainly hope it does sum me up well.  You meant it as an insult but it's a compliment.  

 

I try to be at peace with people and if that means apologizing for saying things in a way I regret later then yes, I'll apologize to them.  You dogging me for apologizing for such things is shameful.

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I assume this is directed toward Dave and not me?  ;)  :D

Um, yep. ;)

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Hi Rowan,

 

Can you post any data that helps explain your preference vs flat response? Trying to stay on topic here with calibration methods that explain preferences. For example, I've found that what most people think is a flat calibration is usually far from it.

 

Skipping the other silliness for the moment, In the rooms you have calibrated, has every room of differing size, construction, and acoustics subjectively sounded to have the same spectral balance just by adjusting for a flat magnitude response as measured by a sine-sweep?  How about measured by RTA?  If the answer is yes they sound the same beyond loudspeaker/subwoofer deficiencies, then there's not much to discuss.  I don't know anyone with experience calibrating more than a dozen rooms who would answer yes to that question, and that gives us something to discuss.

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It's been a while since I updated, but I've been experimenting with variations of bass boost with music and have studied the situation with movie soundtracks a bit more closely.

 

I'll discuss movies first because there's not much to talk about here.  The short answer is that cinema calibration standards are fundamentally broken, so you can't calibrate for a reference response.  The standards aren't just broken for home playback.  They are broken in cinemas and dub stages too.  Recent studies have proven that the subjective responses of various cinemas and dub stages are all over the map.  As a consequence, theatrical mixes are all over the map themselves.  There's no fix for this other than to re-EQ for each feature, and because we don't have data on the dub stages used for each production, the re-EQ can only be optimized by ear.  In principle, home mixes have the potential to be better, but I'm sure a lot have been ruined  by using the same broken calibration method, albeit without the X curve treble roll-off.  Either way, they are broken.  As of 2014, the industry is aware of the problems with cinema calibration standards and are working, albeit slowly, to some kind of resolution.  Until then, the only option is re-EQ.  I've been doing this myself a lot lately.

 

With that said, let's talk about music again.  My earlier point about "flat in-room bass" being a myth is absolutely relevant with music where the reference is an anechoic flat speaker that extends down to at least 40 Hz or so.  Below there, subwoofers are typically in use, and a common practice is to level match the subs with the mains at the crossover point.  This reference has essentially been proven by Harman's research.  However, it is an incomplete reference.  It is a much better reference than flat in-room response, but room effects still alter subjective response in the bass region, at frequencies up to a few hundred Hz or so.

 

The best we can do when calibrating a system for music is to anticipate the consequences of the room on an anechoic flat speaker response in a typical mastering studio.  In the studios, the existence of a bass lift of 3-5 dB is practically universal.  It arises because the woofers are close enough to the floor to make the direct sound and floor reflection indistinguishable.  Even if the room was otherwise acoustically dead, that reflection would still boost the bass.  The transition frequency depends on the particular speaker design and on listening distance.  An engineer working on a particular system will tend to cut frequencies below that transition frequency, wherever it is.  Therefore, in order to hear the music the way the artist intended, we need a matching bass lift in our own response.

 

If we run high quality passive speakers (i.e., anechoic flat) without EQ, we will get the bass lift naturally, but it may not occur at the same frequency that it did in the mastering studio for whatever content we are listening to.  So as a consequence, some recordings will sound very good (similar transition frequency and gain), and others will seem to have flaw such as muddiness (if our transition frequency or gain are higher than in the studio) or thinness or harshness (if our transition frequency or gain are lower than in the studio).

 

For several weeks now, I have experimented with using an adjustable bass lift of 3-5 dB with an adjustable transition frequency to allow me to optimize playback for a wider variety of content.  My experience is that adjusting the transition frequency for different music allows me to hear the best sound with a variety of material.  Whereas in the recent past, I tended to use a -1 to -3 dB high shelf centered around 2 kHz to tame thinness and harshness with a lot of music, I am finding that upward adjustment of the bass boost transition frequency works better most of the time.  (I still use the 2 kHz shelf with movies from time to time.)

 

To re-iterate, the typical transition frequency ranges from 100-500 Hz with gain ranging from 3-5 dB.  Typically higher gains are appropriate where the turnover frequency is higher.

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Funny, I was going to say something very similar.

 

How I choose to setup my system to my preferences is the ONLY thing that is relevant. I prefer a lot of bass and mid bass. But, not so much to where is sounds bloated or muddy. However, I think a flat system sounds awful. I find a middle ground between the two. I suspect I would walk out of your room due to your preferences. Does that make yours wrong? Certainly not. It's YOUR preference; I just happen to prefer something different. Isn't that what this hobby is all about? Sharing our own preferences without someone coming in and shitting on them?

 

Exactly.  If your sound is bloated or muddy, then there's probably too much bass or too much low frequency energy somewhere. Otherwise, more bass is usually a good thing.  Strangely enough, almost all listeners would agree with this point.  Furthermore, the thresholds for muddiness and bloatedness tend to be quite consistent between listeners.  So the fact that flat sounds bad to you should tell you that flat is not correct.  For music, this is absolutely true for in-room response as well as outdoor response.  A boost of 3-5 dB is likely best for outdoor response.  The boost for indoor response may be higher because your in-room response may include a lot of reflected energy that is not necessarily heard as being part of the speaker innate voice.

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Skipping the other silliness for the moment, In the rooms you have calibrated, has every room of differing size, construction, and acoustics subjectively sounded to have the same spectral balance just by adjusting for a flat magnitude response as measured by a sine-sweep?  How about measured by RTA?  If the answer is yes they sound the same beyond loudspeaker/subwoofer deficiencies, then there's not much to discuss.  I don't know anyone with experience calibrating more than a dozen rooms who would answer yes to that question, and that gives us something to discuss.

 

There's an even bigger issue lurking in this discussion and in many others about frequency response flatness.  A sine sweep measurement yields an impulse response, which can be converted using the Discrete Fourier Transform to a frequency response consisting of magnitude and phase vs. frequency.  That data contains an enormous amount of information, yet it looks like a total mess and bears little direct relevance to what the system actually sounds like.

 

So the usual approach taken is to apply some kind of smoothing.  When people say they have a flat in-room frequency response, they almost always refer to the frequency response data after some kind of smoothing has been applied.  However, there is more than one way to smooth the data, even after selecting a bandwidth (e.g. 1/3rd octave) for the smoothing filter.  Smoothing is effectively the application of a low pass filter to frequency data, and obviously there are countless possibilities for such filters.  I'm not aware of any standard that specifies which kind of smoothing filter should be used for audio data.  Believe me, I tried to track down such a standard.  I can't even find documentation detailing the filters used in programs like REW beyond some ambiguous description like "moving average filter".

 

The only real standard for smoothing data is the cinema X curve, which relies on measurement with a 1/3rd octave RTA using full-band pink noise.  This data is binned rather than being smoothed, and the averaging is totally different from the averaging that's applied to smooth frequency data.  In any case, it is a broken standard that produces widely variable subjective results on different systems and in different rooms.  It is believed, with good reason, that the variance is due to the fact that RTA measurements fail to distinguish between the sound from the speaker versus the sound contributed by the room.  Almost all frequency response smoothing methods suffer the same deficiency.

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Skipping the other silliness for the moment, In the rooms you have calibrated, has every room of differing size, construction, and acoustics subjectively sounded to have the same spectral balance just by adjusting for a flat magnitude response as measured by a sine-sweep?  How about measured by RTA?  If the answer is yes they sound the same beyond loudspeaker/subwoofer deficiencies, then there's not much to discuss.  I don't know anyone with experience calibrating more than a dozen rooms who would answer yes to that question, and that gives us something to discuss.

 

Of course not. Since the bandwidth includes quite a bit of infrasonic content, and boundary rigidity and transmission losses are not trivial, the reaction of the environment  can completely change the presentation.

 

I think the point of the OP is calibration method and the results therefrom.

 

Back to Olive quote, with a different emphasis this time:

 

 

Accuracy is Not Applicable to Most Recordings Made Today
Most recordings made today are not intended to sound like the live performance. Anyone who heard Taylor Swift's live performance with Stevie Nicks at the 2010 Grammy Awards understands why. (Note: you can relive the magical moment on Youtube. Warning: this may be offensive for the musically-inclined). About 90% of commercial recordings are studio creations consisting of a series of overdubs, processed with auto-tuning, equalization, dynamic compression, and reverb sampled from an alien nation. For these recordings, there is no equivalent live performance to which the recording/reproduction can be compared for accuracy. The only reference is what the artist heard over the loudspeakers in the recording control room. If the important performance aspects of the playback system through which the art (the music and recording) was created can be reproduced in the home, then the consumer will hear an accurate reproduction of the music, as the artist intended. It is possible to achieve this if we adopt a science in the service of art philosophy towards audio recording and reproduction.

 

The experiments I've looked into, since both of my sons are musicians who both record and mix their own audio creations and who both mix in the nearfield with a subwoofer, is along the lines of what Olive says in the quoted article.

 

If my sons mix a snippet of sound, music or otherwise, and accompany that recording with a FR at their ears and recording level, it is notable that when you match that FR and level in the playback environment, the experience is far more accurate than the randomness of what you find in modern recordings.

 

That would be the reference from which the "silliness" can feel free to roam, whether it be the spawn of any of the thousands of listening experiments done in the 20th century or the car sub enthusiasts-turned-home theater buffs.

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Exactly.  If your sound is bloated or muddy, then there's probably too much bass or too much low frequency energy somewhere. Otherwise, more bass is usually a good thing.  Strangely enough, almost all listeners would agree with this point.  Furthermore, the thresholds for muddiness and bloatedness tend to be quite consistent between listeners.  So the fact that flat sounds bad to you should tell you that flat is not correct.  For music, this is absolutely true for in-room response as well as outdoor response.  A boost of 3-5 dB is likely best for outdoor response.  The boost for indoor response may be higher because your in-room response may include a lot of reflected energy that is not necessarily heard as being part of the speaker innate voice.

Exactly the problem, IMO. One man's "bloated" is another man's "thin". They are both meaningless terms to the reader.

 

Mark mentioned decay earlier. As a matter of physics, generally speaking, the lower the frequency, the longer the decay, the more weight in the presentation. If a recording is brick wall filtered at 30 Hz, no amount of SW trim boost will add the weight that will be perceived when the same content is unfiltered.

 

Subjective silliness aside (I'll use that word as no one has built a cross on which to hang Seaton for using it) "I like the filtered version better" "I like the unfiltered version better", Tastes Great", "Less Filling"... irrelevant.

 

Mark then brings the differences in rooms into the mix. I've studied the FRM measurements of 15 forum members who can be relied upon to produce fairly accurate data. I've compared those to the known ground plane measurements of their subwoofers by Josh and Ilkka. I've found the room gain curve, which is derived from those differences and then averaged, to be very similar. FRM (frequency response magnitude) is predictable and has little to do with room size. I've argued this before in what the kids today call "contention", but the results are what they are, silliness notwithstanding.

 

Rooms differ in how the boundaries are constructed and with what materials they are decorated. This has been known since the beginning of audio hardware and is discussed in detail in several books in my library, circa 1945-1960.

 

I think you are correct (as Olive and others have suggested) that calibration begins with standardization on the production side. Absent that, FRM and level via the placement/phase method can be a practical one and an effective one.

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