SME

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

65 posts in this topic

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

 

I totally disagree.  As long as we agree on the meaning of the terms, we will mostly agree on what's "thin" and what "bloated".  This is born out in studies by Harman and is also validated by the enormous success of lossy digital audio compression algorithms.

 

By employing a detailed, experimentally-verified model of human auditory perception, these algorithms are able to discard 80% or more of the information contained in a digital audio stream with almost only very minor consequences on perceived sound for a wide variety of listeners.  Where some audible artifacts are acceptable as in cellular voice service, substantially more information can be discarded.

 

In an information sense, the bandwidth of the human ear/brain system is very narrow.  One of the challenges faced by a mixer is to ensure that every element or part of the mix is balanced so that the listener actually hears it all.  But of course, this only works if the tonal balance of the production and playbacks systems are consistent.  Where there is inconsistency, relevant auditory details may be buried or lost completely.

 

The problem is that we haven't fully solved the problem of calibrating for consistency between different systems.  Only two firm standards exist for doing so.  One is the X curve standard for cinemas in which EQ is used to adjust the level of full bandwidth pink noise filtered in each 1/3rd octave RTA bin to fit the X curve target.  This standard has been proven to be seriously flawed.  The other is to listen to anechoic flat speakers (except for a UHF roll-off) at least 5 feet from nearby walls in a small to medium size room.  This isn't really a documented standard, but it is completely consistent with how most music is mixed and mastered.  And unlike X curve, Harman's research indicates that this approach yields a preferred sound for both trained and untrained listeners.  The approach doesn't fully solve the translation/consistency problem, but it works way better than X curve does.

 

That's it.  Many people will claim that a "flat frequency response" is best, but this is only close to true if the flat response was of a speaker measured in anechoic conditions.  Otherwise, it's most likely false.  And I can practically guarantee that for music that the smoothed, in-room bass response should measure at least 3-5 dB hotter than the treble because of the floor reflection.  In most situations, the difference will be even greater because of the power response difference and its impact on smoothed frequency response independent of first arrival sound.  Harman's data suggests a 10 dB gap is not unusual.  I believe the gap for my system as calibrated and with the usual boost applied is about 7-8 dB.

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I totally disagree.  As long as we agree on the meaning of the terms, we will mostly agree on what's "thin" and what "bloated".  This is born out in studies by Harman

 

 

 

I believe the gap for my system as calibrated and with the usual boost applied is about 7-8 dB.

 

Define the terms.

 

Cite the studies.

 

How do you calibrate your subwoofers to the satellites?

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My opinion is that every recording should come with a FR taken at the mix desk with mix level noted.

 

Rooms are the largest variable and should be treated accordingly.

 

Ham fisted use of parametric EQ has been the bane of audio playback.

 

Studying the distortion preferences of the general public has nothing to do with proper calibration.

 

Relying on Harmon is akin to asking Pfizer if drugs are good for your health. I read a Harmon study years back and saw this FRM graph. And when I normalized the graph to what we're used to studying... the house curve-looking graph is actually a flat response from a ported sub tuned to around 18 Hz with a typical 80 Hz LR4 LPF.

 

art-895641.jpg

 

The problem is that the sub used is covering 2 octaves of the 5-1/2 octave RB+LFE+10dB summed channel. The results are useless in my experience.

 

Most people on this forum are enthusiastic about dBSPL.

 

If you have a jazz quartet playing live in your room, you don't typically tell the upright bass player to move his instrument to a different position in the room, dampen the D string and pluck his strings a LOT harder.

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Preference-led manipulation thereafter is generally unlistenably distorted, although enjoyed, celebrated and defended regularly in these forums.

 

 

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.

 

 

Most people on this forum are enthusiastic about dBSPL.

 

If you have a jazz quartet playing live in your room, you don't typically tell the upright bass player to move his instrument to a different position in the room, dampen the D string and pluck his strings a LOT harder.

 

You frequently mention the members of this forum in conjunction with "SPL wars" and similar comments to the ones I've highlighted above, but outside of you attending a GTG of some of the members here, where 99% of the time the bass is cranked because people are drinking and just having fun, how many of our theaters have you auditioned at their "normal" playback levels (typical movie watching or 2ch listening levels)?  

 

Yes some of us occasionally have have fun with SPL, but at least in my HT, that probably only constitutes about 1% of the actual playtime.  The other 99% is spent watching movies with my wife and kids with the main volume around -15 without touching the sub trim.  Just last week we watched a movie at our normal listening levels and it wasn't enough to wake up our 9 month old from his nap.  

 

At the many GTGs I've hosted, and probably 50+ individual demos I've given to members of this board and AVS, they always start at normal listening levels with 2ch music (Livingston Taylor, Jessica Pigeon, Nils Lofgren, etc.).   For those not interested in "bass head" stuff, sometimes the subs are never even turned on and the mains are run "full range".  Typically by the end of a big GTG though most people are wanting to hear some crazy bass so I oblige, which is no different than what you did for me when you hosted a last minute GTG for me.  

 

I've also asked all those folks when listening to 2ch music or movie clips at -10db without the sub trim boosted if it seemed too heavy in the bass or out of balance and I have yet to hear a single person say the bass was too loud/heavy or out of balance compared to the mains.  

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My opinion is that every recording should come with a FR taken at the mix desk with mix level noted.

 

 

 

That would be interesting but of course it will never happen. The playback level often varies quite a bit during mixing or depending on the project to gauge how it sounds at high middle and low volume playback. Often there will be a best or "big" room / speaker setup, a smaller room / smaller speaker but still quality setup and usually a consumer grade "typical" speaker with virtually no bass and limited HF playback similar to what you'd see with a radio, tv speaker or Bluetooth dock. Clearly we're more concerned with the main system playback during mixing but there's a lot of variation there. Not to mention differences in room decay rates, tactile response and dispersion characteristics.

 

Are there any recordings that give this type of information? Perhaps some of the audiophile recordings? I like the idea of it even if it would be simplified quite a bit. I've had this urge for a long time to record a raw stereo acoustic drumset track with very high quality condenser mics and somehow calibrate the playback level of the track to reproduce it with full dynamic range. I'd probably have to include a pink noise track or something.

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That would be interesting but of course it will never happen. The playback level often varies quite a bit during mixing or depending on the project to gauge how it sounds at high middle and low volume playback. Often there will be a best or "big" room / speaker setup, a smaller room / smaller speaker but still quality setup and usually a consumer grade "typical" speaker with virtually no bass and limited HF playback similar to what you'd see with a radio, tv speaker or Bluetooth dock. Clearly we're more concerned with the main system playback during mixing but there's a lot of variation there. Not to mention differences in room decay rates, tactile response and dispersion characteristics.

 

Are there any recordings that give this type of information? Perhaps some of the audiophile recordings? I like the idea of it even if it would be simplified quite a bit. I've had this urge for a long time to record a raw stereo acoustic drumset track with very high quality condenser mics and somehow calibrate the playback level of the track to reproduce it with full dynamic range. I'd probably have to include a pink noise track or something.

 

No, I'm not aware of a single production disc that includes the mix desk FR, but some do include the mix playback level, which goes to the influence of equal loudness curves.

 

I'm aware that some mixers try their product through different systems (most don't do that as a mix tool), but they rarely adjust the final mix for those results, in my recording studio experiences. And, the mastering stage is where the final mix is actually done, where none I've ever known left their seat during the mastering process.

 

The next time you're involved in a studio, run the FR measurement yourself and note the playback level for your own reference. Like I said many years back, when the actual bass player is sitting in your sweet spot listening to a recording he played the bass on (something I did years back) and turns to you and says "THAT'S what I'm talkin' about!", how the hell do you argue with that? I could have told him that I prefer the bass bumped +10dB and I certainly have the right to do that in private listening experiences but that wouldn't have any relevance to how he told me it should sound when played back. That experience and listening to recordings on which I was the bass player opened my eyes. When you match the FR used to mix and the playback level, the result is very accurate, other parts of the discussion notwithstanding.

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Define the terms.

 

Cite the studies.

 

How do you calibrate your subwoofers to the satellites?

 

It's in the article I linked to, but maybe you disregarded it because it is associated with Harman, even though it's written by Sean Olive who you have twice quoted here.  But just to summarize:

 

    full vs. thin describes the relative level of low frequencies (bass and low mids) with "full" describing excess and "thin" describing deficiency.

    bright vs. dull describe the relative level of treble with "bright" describing excess and "dull" describing deficiency.

 

I will note that these attributes most accurately apply to particular notes or particular sound effects.  In a presentation, it's entirely possible to have thinness with some notes and fullness with others.

 

As for my sub calibration, the answer requires a bit of detailed discussion.

 

Basically, I calibrate so that the impulse response for each input channel (except LFE) measured with a -20 dBFS sine sweep and analyzed with a 1/3rd octave frequency dependent window (FDW) in REW is 80 dB SPL.  Acttualy, I bend this rule very slightly at the top end, the bottom end, and at the sub/mains XO.  At the top and bottom end, the effective LPF and HPF introduce phase shift that delays the first arrival enough that it slips past the FDW.  A similar issues happens around the sub/mains XO.  For these, I consider the response with up to 1/6th octave FDW, but note that if I use 1/6th octave FDW or higher universally I end up with subjectively imbalanced mid-range due to a very early reflection from the ceiling that my brain apparently ignores when assessing the tonal balance of the source.

 

After the calibration, I apply additional adjustments that are needed for typical content.  Pretty much everything I play gets a filter on ultra high frequencies that is consistent with the effect of in-air absorption and primarily diminishes the response at the very very top.  Almost all professional monitors have such an adjustment built-in, particularly those meant for near-field use.  The second filter is the bass boost I describe here: 3-5 dB somewhere between 100 and 500 Hz or so, again depending on content.  These two filters are sufficient to optimize the presentation of most music I own.  For movies, I also have an adjustable X-curve like roll-off of up to -3 dB/octave starting at 2 kHz.  I also occasionally use a high shelf at 2 kHz of -1 to -3 dB, which is similar to the "high trim" switch available on many monitors and is helpful at with times with both music and movies.  In the case of movies, the need for this upper mid shelf may arise due to some upper mid range energy accumulating and inflating the RTA measurements in the particular dub stage used.  With music, it may arise due to the use of the "high trim" switch during the mix or it may arise due intentional EQ of the monitoring system by the mix engineer.  Apparently many professionals believe that the response above 1 kHz should diminish by -1 dB/octave regardless of room effects and whatnot.  Needless to say, some stuff is just too thin and bright without the upper-mid shelf.  Lastly, there are occasions in which I disable the bass boost, most likely because the mixer used an in-room measurement for calibration here as well instead of relying on the informal reference of "anechoic flat speaker placed in-room at least 5 feet form adjacent walls".

 

Because cinema mixes are done in dub stages that are calibrated using in-room measurements, the bass boost may be inappropriate here as well, but it really depends.  I believe most of movie soundtrack production is done using anechoic flat monitors that may not be configured using any additional EQ.  Only the dub stage is actually calibrated to the X curve, and given growing awareness of the flaws of the X curve calibration, re-recording mixers are probably a lot less likely to make EQ adjustments once the mix gets to the dub stage.  Home mixes are also likely done in an environment without X curve calibration and may sound very good on a system that is optimized for 2 channel music playback if done correctly.  Hence, the bass boost is still often appropriate.  Since the floor bounce arrives relatively late and is somewhat diminished under near-field monitoring conditions, I tend to set the boost at around +3 or +4 dB at 100 Hz for most movies.  With most recent releases, this approach works very well, but there are still exceptions.

 

The end result of this calibration process is left and right mains that measure about 85 dBC using the -20 dBFS pink noise that's band-limited to 500-2000 Hz.  So these days, my master volume "0" is essentially compatible with the cinema reference standard.  Based on my own "first arrival" hypothesis for EQ and SPL calibration, I should be playing at MV "-4" to achieve equivalent loudness to a cinema.  And in fact, this is the level that I prefer on average after correcting for tonal balance issues with most recent film soundtracks.  Older tracks are more likely to come in at around "-6", presumably because of the early confusion about the SPL calibration standard concerning the proper meters to use and the fact that Dolby's pink noise test signal was actually -18 dBFS instead of -20 dBFS.

 

The pink noise test signal for my sub (-20 dBFS and band-limited to 40-80 Hz, IIRC) yields 85 dBC only if I have about +4.5 dB of bass boost active over my baseline calibration.  My result is probably not typical and arises because of my use of near-field MBMs.  The directivity of the MBMs in the 50-100 Hz range is effectively a lot higher than for my mains in the 500-2000 Hz region.  Below 50 Hz, directivity is fairly high too because the room response is dominated by modal behavior where global de-reverberation via EQ becomes possible.  So ironically, "boosting" by 3-5 dB gives me a sub response that's balanced and correct according to cinema standards but oftentimes sounds too hot.  I can't emphasize enough the fact that cinema calibration standards are completely broken.

 

 

My opinion is that every recording should come with a FR taken at the mix desk with mix level noted.

 

This information is mostly useless unless the full impulse response measurement data is included.  I've said many times and will say again that 1/6th octave smoothed frequency response magnitude is most useless for characterizing how a system sounds.  If the full impulse response is provided, then there is an opportunity to run the kind of analysis that's actually needed.

 

As far as what that analysis should be, I'm still working on that problem and will be probably for a long while.  The 1/3rd octave FDW worked extremely well for me in my room, but I know it is will need refinement to deal with crossover delays and what not at the very least.  It's still an open question in my mind just how much first arrival dominates perception vs. contributions from early reflections.  With good speakers like I have, flat first arrival at the MLP tends to provide very smooth response of the reflected sound as well, so I can't rule out that the reflected sound isn't at least partly important.  And this will be important for achieving accurate calibration of systems whose speakers have less-than ideal off-axis response.

 

There's also some question as to how perception works in the bass region where room effects, particularly floor bounce in quality mastering studios but also other boundaries in typical home environments where speaker placements are more subject to compromise.  I've seen Harman suggest that below 300 Hz or so, perceived tonal balance is dominated by power response rather than the anechoic response of the speaker as is the case above 300 Hz.  My guess is that this is not correct, but rather appears to be true because of characteristic of the particular speaker and room used for testing.  Most speakers lose their baffle step directivity at roughly that frequency, and the wavelengths there become long enough that some early reflections (particularly the floor) become indistinguishable.  So under typical conditions, 300 Hz may be the approximate frequency where subjective tonal balance perception is no longer dominated by anechoic response, but that may be due to the particulars of speaker and room rather than any perceptual tendency.  What happens if one is using a floor-to-ceiling I.B. array speaker in which all room reflections arrive late enough to be completely distinguishable from the direct sound?  My guess is that "anechoic response" would still dominate then, even though such a system would need a 3-5 dB bass boost to sound good with most content mastered on typical floor-stander or near-field monitor speakers.

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Your 3-5dB boosted sub sounds hot because it is.

 

FR is never useless.

 

No, you misread.  The 3-5 dB boost sounds balanced with most music and many movies too.  Only with a minority of content does that boost sound excessive.

 

To repeat something I said in my previous post:  I have to have 4.5 dB boost from my baseline calibration for my subs to level match the mains with the cinema pink noise test.  So by cinema standards, I'm not running my subs hot at all!

 

FR is not completely useless, but the frequency response magnitude plots people post online mostly are useless when they are based on in-room measurements.  No one hears frequency response.  If music was nothing but unchanging sine waves, then listeners would hear the unsmoothed frequency response.  But real life content is always changing.  Even long continuous tones in music tend to flutter a bit in pitch or be subject to echo or reverb.  That's a good thing too.  Most people's unsmoothed frequency responses look like garbage.

 

To understand what happens to content other than sine-waves, you have to perform a mathematical convolution of the content with the impulse response, which can be obtained from the unsmoothed frequency and phase data.  Smoothing *does not* do this.  Smoothing does improve the appearance of the data, but most methods effectively degrades or destroys the information that's needed to do the convolution analysis described above.  Therefore, all smoothing really does is improves the visual appearance of the system's response to sine waves.  That's not very interesting.

 

Also, human hearing does not operate with regard to frequency alone.  Crucially, hearing has a time aspect to it.  So does the response of a speaker in a room.  Using the original unsmoothed frequency and phase response data, a time-frequency transform can be performed to reveal the response of the speaker and room in terms of time and frequency together.  This is entirely analogous to using Spec Lab to visualize auditory content in movies.  Without the time aspect of the in-room response, it's hard to gain insight into how your speaker and room are actually affecting what you hear.  Frequency response smoothing almost always degrades or destroys this information.

 

So to give a short answer: "when you you say frequency response, I do not think it means what you think it means."  I wish it were simpler, that we could all just calibrate our 1/6th octave smoothed in-room responses to flat, but that just isn't how things work.  On the positive side, if things did work that way, then we'd probably find listening to be very difficult and frustrating in different acoustic environments or even different places in the same room because the "frequency response" changes so dramatically.  But what happens instead is that things sound much the same in different environments, despite the dramatic changes in frequency response that result from the room.  I'm not saying that acoustics don't matter, but they don't matter as much as one might assume from looking at a smoothed frequency response measurement.  This is a crucial point to grasp.

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No, you misread.  The 3-5 dB boost sounds balanced with most music and many movies too.  Only with a minority of content does that boost sound excessive.

 

To repeat something I said in my previous post:  I have to have 4.5 dB boost from my baseline calibration for my subs to level match the mains with the cinema pink noise test.  So by cinema standards, I'm not running my subs hot at all!

 

FR is not completely useless, but the frequency response magnitude plots people post online mostly are useless when they are based on in-room measurements.  No one hears frequency response.  If music was nothing but unchanging sine waves, then listeners would hear the unsmoothed frequency response.  But real life content is always changing.  Even long continuous tones in music tend to flutter a bit in pitch or be subject to echo or reverb.  That's a good thing too.  Most people's unsmoothed frequency responses look like garbage.

 

To understand what happens to content other than sine-waves, you have to perform a mathematical convolution of the content with the impulse response, which can be obtained from the unsmoothed frequency and phase data.  Smoothing *does not* do this.  Smoothing does improve the appearance of the data, but most methods effectively degrades or destroys the information that's needed to do the convolution analysis described above.  Therefore, all smoothing really does is improves the visual appearance of the system's response to sine waves.  That's not very interesting.

 

Also, human hearing does not operate with regard to frequency alone.  Crucially, hearing has a time aspect to it.  So does the response of a speaker in a room.  Using the original unsmoothed frequency and phase response data, a time-frequency transform can be performed to reveal the response of the speaker and room in terms of time and frequency together.  This is entirely analogous to using Spec Lab to visualize auditory content in movies.  Without the time aspect of the in-room response, it's hard to gain insight into how your speaker and room are actually affecting what you hear.  Frequency response smoothing almost always degrades or destroys this information.

 

So to give a short answer: "when you you say frequency response, I do not think it means what you think it means."  I wish it were simpler, that we could all just calibrate our 1/6th octave smoothed in-room responses to flat, but that just isn't how things work.  On the positive side, if things did work that way, then we'd probably find listening to be very difficult and frustrating in different acoustic environments or even different places in the same room because the "frequency response" changes so dramatically.  But what happens instead is that things sound much the same in different environments, despite the dramatic changes in frequency response that result from the room.  I'm not saying that acoustics don't matter, but they don't matter as much as one might assume from looking at a smoothed frequency response measurement.  This is a crucial point to grasp.

 

A lot of this reminds of me of when tuxedocivic turned my world upside down a few years ago with the proper method of calibrating HF (1khz+):

 

http://data-bass.ipbhost.com/index.php?/topic/314-lukes-gjallarhornothorn-discussion/page-7#entry9037

 

The concept of how the first arrival dominates how we perceive sound is fascinating, and the proof was in the pudding.  Calibrating at the seats may have sounded semi-decent at that specific seat alone, but calibrating close mic with a gated response sounded worlds better in that primary seat and every other seat!  The new FR didn't visually look nearly as flat at the seats, but mics can't ignore the reflections that impact FR differently than the way our ears perceive them.  

 

Something else I've been confused about lately though is whether mic calibration files are valid for "close mic" HF measurements.  If you look at the calibration data and the distance they are taken from, that distance definitely impacts the amount of HF roll-off because of how the air effects the upper end response.  Anyway, that's off topic and a discussion for another thread.  

 

Quick question though. When you say you calibrate to -20 dBFS which ends up being about 85dBC, is that with an input headroom of 105db for the L/C/R and 115db for the sub?  

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A lot of this reminds of me of when tuxedocivic turned my world upside down a few years ago with the proper method of calibrating HF (1khz+):

 

http://data-bass.ipbhost.com/index.php?/topic/314-lukes-gjallarhornothorn-discussion/page-7#entry9037

 

The concept of how the first arrival dominates how we perceive sound is fascinating, and the proof was in the pudding.  Calibrating at the seats may have sounded semi-decent at that specific seat alone, but calibrating close mic with a gated response sounded worlds better in that primary seat and every other seat!  The new FR didn't visually look nearly as flat at the seats, but mics can't ignore the reflections that impact FR differently than the way our ears perceive them.

 

Technically speaking the mic captures almost all the information that the ear has to work with.  The information needed to isolate the speaker response from the room contribution is contained within a measurement taken at the listening position.  It just needs to be extracted with the right analysis.  Smoothing the frequency magnitude response as is typically done in measurement plots is the wrong kind of analysis.  Measuring up close is another way to get that information, but it only works as long as there are no near-field effects, which mostly limits its usefulness to high frequencies.

 

So in principle, you should be able to calibrate using measurement data at the seats.  The hard part is figuring out which analysis is most consistent with the processing that the ear and brain are doing.

 

Something else I've been confused about lately though is whether mic calibration files are valid for "close mic" HF measurements.  If you look at the calibration data and the distance they are taken from, that distance definitely impacts the amount of HF roll-off because of how the air effects the upper end response.  Anyway, that's off topic and a discussion for another thread. 

 

The simple answer is:  Yes, the mic calibration files are valid for both close and distant mic measurements.  If the calibration was done correctly, then it should be valid at all distances.  You *want* your measurements to reflect the effects of distance on the UHF response.  The roll-off of UHF is a real physical thing.

 

The more complicated answer is that almost all measurement mics are also quite directional in the UHF.  You should have calibration files for multiple mic angles, and you should use the calibration file that is consistent with the direction the mic is pointed, relative to the speaker.  When measuring up close, almost all the energy arriving at the mic is directly from the speaker, so the calibration will be completely accurate.  But when measuring at a distance, some room energy will arrive from other angles, and depending on the mic angle and calibration file used this reflected energy may be under or over emphasized in the measurement.

 

Quick question though. When you say you calibrate to -20 dBFS which ends up being about 85dBC, is that with an input headroom of 105db for the L/C/R and 115db for the sub?  

 

I don't know what you mean by input headroom.

 

I would also add that 105 dB and 115 dB are actually kind of arbitrary figures for max output from L/C/R and sub with reference level content.  Don't forget that the pink noise calibration signal is continuous and substantially energizes the room.  So it's not really accurate to say that cinema standards allow for 105 dB peaks on each channel or 115 dB peaks for LFE (not sub).  On a realistic cinema system in which a fair amount of reverb is present, digital full scale peaks may not reach those SPLs at all because the peak level measurements won't include the build-up of room energy.  Hypothetically speaking, if you created 0 dBFS RMS pink noise sample that's band-limited to 500-2000 Hz, then that pink noise would measure at 105 dB when played on each channel of a cinema system, but as you can imagine, program material looks nothing like that kind of signal.

 

Another thing is that cinema SPL calibration uses band-limited signals and is largely ignorant of what happens outside the pass band.  It is the X curve standard that prescribes that the system response be EQed to the X curve target using band-limited pink noise.  If we ignore the X curve, it's entirely possible for -20 dBFS pink noise to measure quite a bit higher or lower if it's limited to bands other than the 500-2000 Hz for the calibration.

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No, you misread.  The 3-5 dB boost sounds balanced with most music and many movies too.  Only with a minority of content does that boost sound excessive.

 

Not sure how anyone would know if content sounds bloated or whatever adjective means the sub is hot? I think that's an impossibility.

 

Also, human hearing does not operate with regard to frequency alone.  Crucially, hearing has a time aspect to it.  So does the response of a speaker in a room.  Using the original unsmoothed frequency and phase response data, a time-frequency transform can be performed to reveal the response of the speaker and room in terms of time and frequency together.  This is entirely analogous to using Spec Lab to visualize auditory content in movies.  Without the time aspect of the in-room response, it's hard to gain insight into how your speaker and room are actually affecting what you hear.  Frequency response smoothing almost always degrades or destroys this information.

 

Do you have any examples without words?

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No, you misread.  The 3-5 dB boost sounds balanced with most music and many movies too.  Only with a minority of content does that boost sound excessive.

 

Not sure how anyone would know if content sounds bloated or whatever adjective means the sub is hot? I think that's an impossibility.

 

Also, human hearing does not operate with regard to frequency alone.  Crucially, hearing has a time aspect to it.  So does the response of a speaker in a room.  Using the original unsmoothed frequency and phase response data, a time-frequency transform can be performed to reveal the response of the speaker and room in terms of time and frequency together.  This is entirely analogous to using Spec Lab to visualize auditory content in movies.  Without the time aspect of the in-room response, it's hard to gain insight into how your speaker and room are actually affecting what you hear.  Frequency response smoothing almost always degrades or destroys this information.

 

Do you have any examples without words?

 

 

How would one know if the content sounds bloated?  By using one's ears.  I realize that evaluation using one's ears is kind of heretical on a forum like this.  I can't blame anyone for being skeptical of claims that listening can be used to evaluate sound quality in any objective sense, given that so much of the audio industry is completely fraudulent and successful in so far as placebo effect influences subjective judgments.  However, until we have a complete experimentally validated model of human hearing, our ears remain the best judge of what things sound like.

 

The trouble with the whole audiophile industry is people's obsessive focus on the subtle.  "Did those new speaker cables make a difference?"  "Yes, of course.  It was subtle, but it definitely made a difference."  Uh huh.  I'm far more interested in aspects of audio that are *not subtle*.  If dialog sounds too muddy, that is not a subtle problem, especially it prevents you from understanding what is being said.  If the individual notes in a bass-line can't be discerned because of too much deep bass from the subwoofer, then that is *not subtle*.  I can demo this on my system for any interested listener who can come here to Denver.  I can shelve the bass from 50 Hz and below by +2 dB, and switch the filter in with only a fraction of a second of gap while a song is playing.  A bass line that was once snappy and a kick drum that once punched the chest turn into mush and rumble.  That's *not subtle*.  I'm convinced that everyone who heard the difference would agree that the latter sound was muddy and inferior.

 

You want some pictures?  Check out the first two frequency response plots I posted under the "Speaker / Room Calibration" section in the first post of my system/room thread.  One is processed with 1/3rd octave smoothing and the other with 1/3rd octave FDW.  The latter is a much more accurate approximation of the first arrival sound coming from my speakers and is also much closer to what I think my speakers sound like.  I don't have any good time-frequency plots to show, for a variety of reasons, chief among them is the difficulty of visualizing everything that's relevant to hearing in a single plot.  I believe this to be a problem with Spec Lab spectrograms as well.  The choice of window size always involves a compromise between time and frequency resolution.  To really understand what's going on, you need to view the data with a variety of window sizes.

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I want to just give my 2 cents on this topic, almost purely subjective without much scientific backing.

 

I work in different cities a lot. This is why I spent significant effort building a fantastic sounding small speaker that I can bring with me. Because of that, I've heard my speaker in 20+ different rooms. Here are my thoughts

 

Rooms suck ass.

 

But we all know that. The problem is, each room can sound very different even with similar frequency response at the LP. I've been in rooms that sounded absolutely terrible, and some rooms that were great! But since we are talking about the bass, I've found that some rooms are impossible to work with. I'll give a very extreme example. 

 

I once bought my speakers to a large auditorium that seated 450 people in order to get a room with a high ceiling to get high quality quasi anechoic measurements. I also brought along a small sealed 15" Martin Logan subwoofer to see how my speakers perform in a very large room if it does not have to produce bass, and to have some silly fun blasting bass :P. The bass sounded terrible. It was peaky, it had no depth, no impact, no matter where I am in the auditorium. It basically sounded like the fundamental is gone and all I'm hearing are the 2nd and 3rd harmonics.

 

So I turned on miniDSP and put a huge shelf boost to crank up the deep bass. But no matter what I did, no matter how much boost I give, play with phase, or how much louder I turn up the subwoofer, there was no sense of bass below <50Hz. I measured the FR at a listening position where it sounded bad (pretty much everywhere outside of the 1 meter radius around the sub), and the subwoofer response is fairly flat, no nulls or even dips. I ran the subwoofer up to 30dB hot with the mic registering 110dB of bass, and even though the walls were very audibly rattling, it STILL sounded like there was no bass. 

 

While I haven't experienced anything to that extreme in a normal home environment, there were definitely rooms where the bass FR looks fine, but the bass sounded very different. This tells me a frequency response measurement is woefully incomplete, and there must be something else that completes the picture. 

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I don't know what you mean by input headroom.

 

dBFS = decibels relative to full scale, so when you say -20dBFS I'm wondering what you're using for scale.  Depending on which mic and preamp I'm using, -20dBFS could easily be over 140db at the seats!

 

I was curious if you were using 105db as your scale since 85dBC is about 20db below that.  

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dBFS = decibels relative to full scale, so when you say -20dBFS I'm wondering what you're using for scale.  Depending on which mic and preamp I'm using, -20dBFS could easily be over 140db at the seats!

 

I was curious if you were using 105db as your scale since 85dBC is about 20db below that.  

 

Oh!  I see what you're saying.  The -20 dBFS refers to the RMS level, relative to a full scale sine wave, of the test signal going from my PC over HMDI to the AVR.  It is definitely not the same digital scale that is captured by my measurement mic!  That would be in the neighborhood of 100 dB on my UMIK-1, I think.  It is also not the same digital scale that my Motu 16A uses.  On my 16A, I actually use different digital scales for each output channel because they go to different kinds of amps with different sensitivities.

 

Getting back to your question.  The digital scale doesn't really work like that.  If you my AVR to master volume "0" and feed it a 0 dBFS sine wave, what you measure will depend entirely on the frequency and will almost certainly not be 105 dB or even necessarily close.  There is a common misconception that if the system is calibrated right then a signal that reaches digital full-scale in the soundtrack will reach 108 dB (peak) at the seat.  Just as with sine waves, it depends entirely on the signal AND the room.  For a very transient signal, most likely you will measure quite a bit less than 108 dB at the seats because you won't have all the extra room energy that would be present with a continuous signal.  This is especially true for a bigger room with a lot of reverb.  This is a very important point, which explains a lot about why reference level typically sounds too loud in small rooms.  It's not just a psychoacoustic perception thing.  It's because the transients actually *are* higher SPL in the small room than the big room.  It's a side-effect of the calibration method.

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To be clear, my concern and comments are confined to the subwoofer (RB+LFE+10dB summed signal) and the crossover region of the subwoofer and the satellites.

 

 

How would one know if the content sounds bloated?  By using one's ears.  I realize that evaluation using one's ears is kind of heretical on a forum like this.  I can't blame anyone for being skeptical of claims that listening can be used to evaluate sound quality in any objective sense, given that so much of the audio industry is completely fraudulent and successful in so far as placebo effect influences subjective judgments.  However, until we have a complete experimentally validated model of human hearing, our ears remain the best judge of what things sound like.

 

I was afraid you'd resort to hearing as the arbiter of success after calibration... I have 60-odd discs you're invited to "listen" to and grade the muddiness or the thinness thereof. I've conducted these sorts of casual listening tests myself as well as exploring the directional cognizance claims by many that supposedly correlate to crossover point. I've asked elderly, young, clueless, musicians, producers, male, female, friend and foe. Listeners are generally clueless and so extremely biased (or in serious stages of hearing loss) as to border on the comical (not that there's anything wrong with that <disclaimer for the feelings of the less well-adjusted readers>).

 

Hacksaw Ridge is a good example. If your system is sharply rolled off <20 Hz (as yours is), the presentation will be noticeably lacking, and the calibration will be a fruitless exercise. Even if your system is full bandwidth to 1 Hz and flawlessly calibrated, there is no, zero, minus infinity chance you can judge the systems accuracy with your ears without a reference. As I've said far too many times in various forums, applying a steep HPF at 20 Hz is not a trivial tweak to content bandwidth and certainl;y affects playback accuracy and perceptions.

 

The trouble with the whole audiophile industry is people's obsessive focus on the subtle.  "Did those new speaker cables make a difference?"  "Yes, of course.  It was subtle, but it definitely made a difference."  Uh huh.  I'm far more interested in aspects of audio that are *not subtle*.  If dialog sounds too muddy, that is not a subtle problem, especially it prevents you from understanding what is being said.  If the individual notes in a bass-line can't be discerned because of too much deep bass from the subwoofer, then that is *not subtle*.  I can demo this on my system for any interested listener who can come here to Denver.  I can shelve the bass from 50 Hz and below by +2 dB, and switch the filter in with only a fraction of a second of gap while a song is playing.  A bass line that was once snappy and a kick drum that once punched the chest turn into mush and rumble.  That's *not subtle*.  I'm convinced that everyone who heard the difference would agree that the latter sound was muddy and inferior.

 

You want some pictures?  Check out the first two frequency response plots I posted under the "Speaker / Room Calibration" section in the first post of my system/room thread.  One is processed with 1/3rd octave smoothing and the other with 1/3rd octave FDW.  The latter is a much more accurate approximation of the first arrival sound coming from my speakers and is also much closer to what I think my speakers sound like.  I don't have any good time-frequency plots to show, for a variety of reasons, chief among them is the difficulty of visualizing everything that's relevant to hearing in a single plot.  I believe this to be a problem with Spec Lab spectrograms as well.  The choice of window size always involves a compromise between time and frequency resolution.  To really understand what's going on, you need to view the data with a variety of window sizes.

 

This is what I suspected is the final answer... frequency response. It is indeed what we hear, irrespective of the particular curve you or anyone else might prefer and regardless of the method used to arrive at it. It remains one of the best tools for calibration.

 

 

A few of us have discussed and explored windowing in years past. I was once told that I need to use 'X' window for better accuracy of measurement, for whatever 'scientific' reason, which I forget now. Smoothing can take you all the way to anechoic response, as in a close mic with 1/2 octave smoothing (which is always what I've used). Shown below id the no-smoothing vs 1/2 octave smoothing close mic of my subwoofer system:

 

iI3MJ4h.jpg

 

This (comparing the close mic with the seats mic versions) is basically what you're doing with smoothing, but with your own choice of compromise:

 

oquLJdj.jpg

 

My response graph was a result of the suggestion by AVS member <forgot his SN> that I have bad response measurements because i use the wrong window. I used my usual window and then used the one he suggested. That result also has the smoothed close mic version laid over it.

 

Then there are your graphs, smoothed and windowed, with scale normalized.

 

Of course, I do not "hear" the 6 Hz resonance exhibited by my floor system, but I think it a bit absurd to suggest its influence should be noted by the close mic response or any in-between resulting FR due to smoothing and/or window views.

 

Curious; are you suggesting that the peak your 1st FR shows at 110 Hz isn't what I would hear at the mic position?

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To be clear, my concern and comments are confined to the subwoofer (RB+LFE+10dB summed signal) and the crossover region of the subwoofer and the satellites.

 

Why?  I would regard the bass range to formally start at around 200-250 Hz.  Actual bass instruments and sound effects tend to have harmonics well above there that affect the perception of the overall sound.  Even tactile sensation is affected by higher frequencies, up to 500 Hz in my experience.  If you don't care about the rest of the spectrum, why not just low pass everything at 120 Hz?

 

I was afraid you'd resort to hearing as the arbiter of success after calibration... I have 60-odd discs you're invited to "listen" to and grade the muddiness or the thinness thereof. I've conducted these sorts of casual listening tests myself as well as exploring the directional cognizance claims by many that supposedly correlate to crossover point. I've asked elderly, young, clueless, musicians, producers, male, female, friend and foe. Listeners are generally clueless and so extremely biased (or in serious stages of hearing loss) as to border on the comical (not that there's anything wrong with that <disclaimer for the feelings of the less well-adjusted readers>).

 

Grading the quality of bass on discs is not my focus.  My focus is to optimize the tonal balance of my playback system to maximize enjoyment of those discs.  This requires matching the tonal characteristics of my system to those used in the mastering studio or the mix room, if no mastering was done.

 

Even though I talk of a "bass" boost, the transition point of this shelf falls in the range of 100-500 Hz.  For what it's worth, the effect of changing this transition point on the response in the sub range is actually very minor.  Changes below 100 Hz are actually very minor.  My choice for bass boost transition point and gain for any particular content is guided more by the subjective balance of lower mids with higher harmonics.

 

The transition frequency and gain I use most often with music, kind of my default, is about +3.5 dB @ 120 Hz.  This works with a large variety of music I listen to, particularly a lot of electronic music, much of which was probably mixed on near-field monitors and never even mastered.  Content that is released by major labels is more likely to have been mastered, probably with floor standing speakers that exhibit a somewhat greater boost with a higher transition frequency.

 

If there are vocals in the track, a clear give-away is that the female voices sound thin even though the male voices sound full.  Likewise, the piano may sound tinny with higher notes but not lower notes.  Or maybe the cello sounds good but violin that sounds harsh as it ascends to higher notes.  On the flip side, if I set the transition frequency for the bass boost too high, then I might notice clear and detailed female vocals but muddiness in the male vocals.

 

Hacksaw Ridge is a good example. If your system is sharply rolled off <20 Hz (as yours is), the presentation will be noticeably lacking, and the calibration will be a fruitless exercise. Even if your system is full bandwidth to 1 Hz and flawlessly calibrated, there is no, zero, minus infinity chance you can judge the systems accuracy with your ears without a reference. As I've said far too many times in various forums, applying a steep HPF at 20 Hz is not a trivial tweak to content bandwidth and certainl;y affects playback accuracy and perceptions.

 

Umm, I thought we were talking about listening tests with music?  And anyway, if your goal is to hear what the director or artist intended, you'd best throw away that stuff under 20 Hz.  We both know probably 99% of pro system used for mastering and re-recording mixing roll-off at 20 Hz and above.  Probably 99.9% roll-off at or above 15 Hz, which is where my system actually rolls off, by the way.

 

This is what I suspected is the final answer... frequency response. It is indeed what we hear, irrespective of the particular curve you or anyone else might prefer and regardless of the method used to arrive at it. It remains one of the best tools for calibration.

 

Of the two plots of mine that I had you look at, neither is my frequency response.  One is frequency response magnitude smoothed with 1/3rd octave resolution.  The other is frequency response with 1/3rd octave FDW.  For various reasons, neither of them reveals my extension below 20 Hz.  If you insist on calling one of these "frequency response", then which one?  Which one should be used for calibration?

 

A few of us have discussed and explored windowing in years past. I was once told that I need to use 'X' window for better accuracy of measurement, for whatever 'scientific' reason, which I forget now. Smoothing can take you all the way to anechoic response, as in a close mic with 1/2 octave smoothing (which is always what I've used). Shown below id the no-smoothing vs 1/2 octave smoothing close mic of my subwoofer system:

 

Windowing is a complex subject.  There are many different kinds and sizes of windows that may be used, and of course, frequency-dependent windowing opens up an entire new range of possibilities.  Each of them may be used to explore the data in different ways, to create a multitude of different SPL vs. frequency curves, whose meaning must be interpreted in the context of the windowing and smoothing that is used.

 

You cannot smooth "all the way to anechoic response".  You can use windowing to recover the anechoic response for mid and high frequencies where the direct sound is readily distinguished from the reflected sound.  For bass, you can't recover the anechoic sound at all.  Nor will close mic measurements give you the anechoic sound in the bass.  Such measurements will be affected both by near-field effects and by room effects, where they are strong enough to overwhelm the direct sound even at the close distance.  There is a reason Josh Ricci does his measurements outdoors.

 

At the same time, my interest is to recover the anechoic sound only for those frequencies in which it can be distinguished from room reflections.  For bass frequencies, I care about the first arrival sound that is perceived by the listener.  And I believe that first arrival sound exhibits a bass boost when an anechoic flat speaker is used in a typical mixing or mastering environment.

 

Of course, I do not "hear" the 6 Hz resonance exhibited by my floor system, but I think it a bit absurd to suggest its influence should be noted by the close mic response or any in-between resulting FR due to smoothing and/or window views.

 

Curious; are you suggesting that the peak your 1st FR shows at 110 Hz isn't what I would hear at the mic position?

 

I'm not sure what you're saying here, but resonances are a different story.  They are typically audible well beyond their influence on first arrival sound.  But I consider fixing resonances in the response to be a separate activity, one that improves sound quality, but is not directly relevant to tonal balance calibration.

 

My center channel response has inconsistent coverage in the bass range because of its placement up against the wall.  I am planning to address that problem using either thick absorption or an extended baffle wall, but until then, the response shown reflects a compromise to minimize deviation across multiple seats.  At the MLP, center channel output is weak in the broad region around ~180 Hz and strong in the broad region around ~90 Hz.  These features are visible in the second plot, and while they constitute deviations of only +/- 2 dB, they are in fact quite audible.

 

In contrast, the peak at 110 Hz in the second plot results from a combination of the strong direct sound output around 90 Hz and constructive interference from a series of reflections that arrive in the range of 10-20 ms after the initial impulse.  These reflections blend together in the bass frequencies, but the entire group can be readily distinguished from the direct sound.  By my hypothesis, the series of peaks and dips that arise from the interference of the reflections with the direct sound are not likely to be heard as such.  Instead, the brain is likely to hear the direct sound and reflections as separate events, each with their own tonal composition.  The brain may use information from the later arriving reflected sound to revise its assessment of the tonal qualities of the direct sound, but the overall effect of that later arriving sound on perception will likely be greatly diminished compared to the direct sound.

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Okey Dokey, we're in unrecoverable mode now...

 

I've played electric bass for some 50 years now. Not sure if you're that confused or yanking my chain? I'm keenly aware of its range and possible harmonics that may or may not survive production methods. You can't write music without knowing the ranges of the instruments involved.

 

TvCl522.gif

 

Actual bass instruments have fundamentals in the bass range. Their harmonics are played back by satellites and are far easier to manipulate to taste. My basses are all 4 string and cover low E (41.2 Hz) to 2 octaves higher E on the G string (123.5). A 100 Hz crossover point takes the subwoofer to 200 Hz, where it is down -30dB with LR4 LPF. Keyboard and 5-7 string basses go lower. Synth can generate fundamental to single digits.

 

Transients (an impulse is one) have content to DC, regardless of the fundamental, unless they are purposely filtered. I'm not aware of anyone who has successfully generated, played back and measured transients cleanly and accurately.

 

If you intend to only discuss your front 3 satellites, that would be a useful disclaimer. Of course, this isn't the case according to what you've posted thus far, so not sure what the LPF @ 120 Hz comment is supposed to mean..

 

My close mic measurements have turned out to be more accurate that Ilkka's ground plane measurements of the same Tumult driver in slightly different Vb Josh chooses to measure ground plane because the battery of tests he uses that were devised by others requires it. My close mic showed a perfect 12 dB/octave roll off whereas Ilk's showed a 10dB/octave roll off (which appeared in nearly all of his GP measurements of an unfiltered sealed subwoofer as unexplained phenomena in our private conversations about the subject).

 

art-598742.jpg

 

The bottom line is that the close mic method can be extremely accurate and has been used my many noted engineers over the decades.

 

I realize that both of your frequency response magnitude graphs are smoothed. The question remains.. do you think I would not hear that peak because it isn't comprised of only the anechoic response of the loudspeaker or that it's possible to distinguish that distortion by your brain separating the anechoic (direct radiated) sound from the latent release of the stored energy sound? If not, how does smoothing help with the dominant distortion in your system at the mic?

 

And, let's just say that I don't believe a human can even acknowledge 10-20 ms of delay much less distinguish the first and second sounds. Curious to know if you've ever employed a processor with delay and increased the delay until it is audible to you? And, that sort of experiment is with only a single sound and repeat, but it might help prioritize.

 

You can use any means of manipulating the FR at the primary listening position that you feel distinguishes itself from the multiple theories on the subject best to your logical thinking. The best smoothing method is that of treating the room to the degree of dead you prefer. I remember the avalanche of posts when Audyssey first hit the streets that exclaimed and extolled the religious experience of a difference once Audyssey was run. Thank goodness, like most theroies, the crowds have calmed down and the manufacturers have made improvements. None of that will help you with the subwoofer which dominates the distortion of the input signal at the PLP. That's why my focus is there and not with the short wave frequencies of the audible bandwidth.

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Okey Dokey, we're in unrecoverable mode now...

 

I've played electric bass for some 50 years now. Not sure if you're that confused or yanking my chain? I'm keenly aware of its range and possible harmonics that may or may not survive production methods. You can't write music without knowing the ranges of the instruments involved.

 

[---cut---]

 

Actual bass instruments have fundamentals in the bass range. Their harmonics are played back by satellites and are far easier to manipulate to taste. My basses are all 4 string and cover low E (41.2 Hz) to 2 octaves higher E on the G string (123.5). A 100 Hz crossover point takes the subwoofer to 200 Hz, where it is down -30dB with LR4 LPF. Keyboard and 5-7 string basses go lower. Synth can generate fundamental to single digits.

 

You totally missed the point.  The point is that the bass instrument *usually* has strong harmonic content, which has a big impact on the sound.  Do you ever just plug your electric bass directly into a subwoofer when you play?

 

Transients (an impulse is one) have content to DC, regardless of the fundamental, unless they are purposely filtered. I'm not aware of anyone who has successfully generated, played back and measured transients cleanly and accurately.

 

All realistic transients are rolled off somewhere at the bottom or they wouldn't be transient.  An impulse is a theoretical construct that does not exist in real life in an unfiltered form.  An unfiltered impulse also has content to an infinitely high frequency, which is impossible in reality.

 

Anyway, I don't know why you are bringing this up other than to try to turn this conversation into *yet another debate about ULF*.  No thanks.

 

If you intend to only discuss your front 3 satellites, that would be a useful disclaimer. Of course, this isn't the case according to what you've posted thus far, so not sure what the LPF @ 120 Hz comment is supposed to mean..

 

You told me: "My concern and comments are confined to the subwoofer (RB+LFE+10dB summed signal) and the crossover region of the subwoofer and the satellites."  I pointed out that any discussion about calibration and bass balance has to include the mains well above the crossover frequency.  If you don't believe me, then try running an LPF at 120 Hz, or just don't bother to turn on the amps for your mains.  Great bass, huh?  Yeah.  I thought so.

 

My close mic measurements have turned out to be more accurate that Ilkka's ground plane measurements of the same Tumult driver in slightly different Vb Josh chooses to measure ground plane because the battery of tests he uses that were devised by others requires it. My close mic showed a perfect 12 dB/octave roll off whereas Ilk's showed a 10dB/octave roll off (which appeared in nearly all of his GP measurements of an unfiltered sealed subwoofer as unexplained phenomena in our private conversations about the subject).

 

[---cut---]

 

The bottom line is that the close mic method can be extremely accurate and has been used my many noted engineers over the decades.

 

Haven't we discussed this before?  Close mic measurements have uses, and can approximate the shape of the ground plane curve, but they don't replace ground plane measurements.  Even if you avoid the caveats of near-field effects (you certainly won't with your Raptor's) and interference from strong room modes, they don't give you absolute sensitivity needed to compare performance.

 

I realize that both of your frequency response magnitude graphs are smoothed. The question remains.. do you think I would not hear that peak because it isn't comprised of only the anechoic response of the loudspeaker or that it's possible to distinguish that distortion by your brain separating the anechoic (direct radiated) sound from the latent release of the stored energy sound? If not, how does smoothing help with the dominant distortion in your system at the mic?

 

No, I don't think you are really going to hear a peak at 120 Hz.  You are more likely to hear the presence of the reflections that contribute to the peak there as well as the peaks and dips across the 100-400 Hz bandwidth of the reflections.

 
The part of the spectrum that sounds less than great to my ears is at 190 Hz where the direct sound is weak and the reflected sound is stronger than the direct sound.  This can make voices sound a bit weird.  Solving this problem requires me to improve the acoustic integration ff the speaker with the wall, something I plan to do fairly soon.  Other than that, the group of reflections is unusually strong because of symmetry.  Both the center channel and MLP are centered in the room, so the reflections from each sidewall arrive at the same time within that range.  This is probably a common problem in home theaters.

 

 And, let's just say that I don't believe a human can even acknowledge 10-20 ms of delay much less distinguish the first and second sounds. Curious to know if you've ever employed a processor with delay and increased the delay until it is audible to you? And, that sort of experiment is with only a single sound and repeat, but it might help prioritize.

 

The ear and brain are absolutely able to discern audible events separated by 10-20 ms, but they won't necessarily be perceived as discrete echos.  The separate arrivals of the direct sound and the group of reflections are likely to be fused into a single perception of the sound of the speaker being heard in a room like the one you are sitting in.

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You totally missed the point.  The point is that the bass instrument *usually* has strong harmonic content, which has a big impact on the sound.  Do you ever just plug your electric bass directly into a subwoofer when you play?

 

This is waaaaay off topic from the OP. I'm not interested in the rabbit hole of educating a non-musician as to what my bass sounds like or can sound like when played through my HT subwoofer system. The reasons should go without saying but if they don't in this case c'est la vie.

 

All realistic transients are rolled off somewhere at the bottom or they wouldn't be transient.  An impulse is a theoretical construct that does not exist in real life in an unfiltered form.  An unfiltered impulse also has content to an infinitely high frequency, which is impossible in reality.

 

Anyway, I don't know why you are bringing this up other than to try to turn this conversation into *yet another debate about ULF*.  No thanks.

 

Yet another debate about ULF? Wow. How about yet another SME lecture point about what calibration philosophy is bullshit?

 

You told me: "My concern and comments are confined to the subwoofer (RB+LFE+10dB summed signal) and the crossover region of the subwoofer and the satellites."  I pointed out that any discussion about calibration and bass balance has to include the mains well above the crossover frequency.  If you don't believe me, then try running an LPF at 120 Hz, or just don't bother to turn on the amps for your mains.  Great bass, huh?  Yeah.  I thought so.

 

Yes, I've analyzed my many systems over the decades with and without the satellites dating back to the 70s. Analysis of subs only is useful to hear the effect of crossover point and slope selection and extraneous vibration noise. Oh, and yes, of course the subs only listening test is crucial, but I'm not the one here who claims his pschoacoustic abilities are a primary tool in calibration... you are.

 

Haven't we discussed this before?  Close mic measurements have uses, and can approximate the shape of the ground plane curve, but they don't replace ground plane measurements.  Even if you avoid the caveats of near-field effects (you certainly won't with your Raptor's) and interference from strong room modes, they don't give you absolute sensitivity needed to compare performance.

 

Ground plane measurements can approximate close mic... yes. Sometimes ground plane measurements contain non-trivial errors, as shown in my previous post. How many times that fact is discussed isn't relevant. Comparing performance is also not relevant to the discussion.

 

No, I don't think you are really going to hear a peak at 120 Hz.  You are more likely to hear the presence of the reflections that contribute to the peak there as well as the peaks and dips across the 100-400 Hz bandwidth of the reflections.

 

Reflections are responsible for the peak at 110 Hz? You mean reflections that create a standing wave or RT60 reflections? And, you don't hear the +7dB peak, but you hear the individual reflections that produce it? I'm not misreading, I just don't get what you're trying to say. Please provide some... any... evidence/data when you make proclamations. It makes for a much less frustrating discussion.

 
The part of the spectrum that sounds less than great to my ears is at 190 Hz where the direct sound is weak and the reflected sound is stronger than the direct sound.  This can make voices sound a bit weird.  Solving this problem requires me to improve the acoustic integration ff the speaker with the wall, something I plan to do fairly soon.  Other than that, the group of reflections is unusually strong because of symmetry.  Both the center channel and MLP are centered in the room, so the reflections from each sidewall arrive at the same time within that range.  This is probably a common problem in home theaters.
 
Agreed that room treatment is the fix for how it sounds and has something to do with calibration but this is straying from your OP and falling into the nuance category you've mentioned to be unimportant.

 

The ear and brain are absolutely able to discern audible events separated by 10-20 ms, but they won't necessarily be perceived as discrete echos.  The separate arrivals of the direct sound and the group of reflections are likely to be fused into a single perception of the sound of the speaker being heard in a room like the one you are sitting in.

 

You're obviously confusing echolocation with detection of latency. Test yourself and get back to me.

 

I responded to your thread based on your OP in which you declared that it's sad that any authority would suggest the best calibration from a recorded source playback system is a flat response. You declared it, right here and now, to be bullshit with no disclaimer. You support that absurd declaration by mentioning "numerous blind listening studies conducted by Harman" though you don't define numerous nor do you cite any of them.

 

I've said about a thousand or more times; the equal loudness curves are built into all commercially available recorded material. I have been a participant in enough of those sessions and processes over the past half century, beginning at age 13YO, to assure you that no producer has ever mixed the content flat assuming that SME or anyone else will re-mix the product, post production, using a one size fits all calibration adjustment. For various reasons, that material may end up anywhere on the quality scale which exposes the flaw in such an approach.

 

Think about it. A producer is deduced to have radically different hearing than some random collection of listeners to the point of requiring post production production to get the mix right. And, THAT conclusion isn't bullshit?

 

For the record, I don't have any music program that requires a +5dB boost in the bass region to correct for a thin sound. That doesn't mean there is no such program, it just means that i don't keep poorly produced source in my collection.

 

If one falls prey to Harman (or any other of the thousands of 'listening studies') conclusions and calibrates to some distorted bias, then every recording played back on that system will show that bias. This is proven on the production side. Mix on a system other than flat and get a result that has too much bass or too little bass, which is what happens in reality. The proper method is indeed to calibrate flat and "season to taste' after that on a disc-to-disc basis. What others preferred during a listening test positively changes in those test subjects over time with the evolution of hardware and software, room construction differences, age and preference adjustment. Why you or anyone would think that he/she should conform to such a metric is beyond me, but calling a flat calibration bullshit is just not the way to be taken seriously.

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Here is a link to a Harman study about target curves.Hopefully it will work.

Thanks for the link.

 

I've read this and the auto EQ vs the "target FR" articles in the past. I compared (one of many that have changed with remodels and changes in hardware over the past 15 years of measuring in-room) my response at the LP to the results of the auto EQ article.

 

jrASsYb.jpg

 

1) One guy preferred the low end bumped +15dB (IIRC).

2) IMO, 1dB/octave, which amounts to (+/-5dB) is tantamount to a flat response without EQ post smoothing, which I don't use.

3) My high end is bumped because I've lost a chunk of hearing capability up there, but I don't find a more rolled off high end objectionable.

4) I listen at significantly higher playback levels, which alters perception. Any test should use only program where the mix level is known and matched.

5) A noted caveat in the auto EQ article is: "Program material is a nuisance variable".

6) The results of all of Harman's studies leave out the fact that they reveal a simple conclusion; a) either the producers of the selected program source had seriously flawed  monitoring hardware or b ) they really sucked at mix production.

7) Playback systems that roll off sharply below 50 Hz shouldn't even be a part of a listening session of any kind, IMO.

8) All of the hardware used is a Harman product.

9) Sine sweep FRM graphs are always the metric used.

 

Reminds one of "listening tests" over the past century:

 

N77xIJA.jpg

 

Was Edison serious, or was he selling his invention? Note that some of the "listeners" have the blindfolds covering their ears!! Also note that they are all old guys whose hearing might be dubious.

 

Acoustic Research conducted many "listening tests" comparing "Live vs Playback": WUT??

 

PF18eIy.jpg

 

If I post the Ground Plane measurements results of the Raptor, someone please predict what will happen before the virtual ink is dried?

 

This "test" cracks me up. It seems the principals got the wrong memo on which of the human senses were being tested?

 

anxyeCK.png

 

Preferences don't mean much and the industry should get that and move on. Nothing against any preferred curve, just that proclaiming one a "target" over others is just silly. Members are appreciated for their posting efforts, especially guys like SME who have interesting stuff to post. But, edicts such as "your method is BS because Harman said so" should expect some blowback.

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