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My living room "make over" (aka the "surrounded by bass" project)


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#141 3ll3d00d

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Posted 06 October 2016 - 12:48 PM

For most of the measurements, there is a pretty nasty looking dip somewhere in the 600-700 Hz range, but it does shift slightly between locations.  Somewhat less obvious is that there are peaks, one at round-about 850 Hz and another broader one at 350 Hz.  Below there, the responses roll off due to a combination of baffle step effect and 110 Hz crossover.  

 

I wouldn't trust the data from a 4ms window here, at least certainly not for the 350Hz and below part and the 600Hz might be questionable as well.


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#142 SME

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Posted 08 October 2016 - 08:25 AM

I wouldn't trust the data from a 4ms window here, at least certainly not for the 350Hz and below part and the 600Hz might be questionable as well.

 

I trust the data.  The issue lies with properly interpreting it.  The 4 ms window shows me everything that happens to *every* frequency within the first 4 ms following the first arrival from an impulse fed to the system.  The crossovers add some delay, but it's mostly under 1 ms for the low frequencies here, so this is closer to 3 ms of data for the woofer frequencies.  As such, this shows me how acoustic information that arrives up to 3 ms after the low frequency part of the impulse affects the frequency response.  For frequencies below 350 Hz or so, additional window time is required for a full wavelength to be realized.  That means that the apparent direct sound may be influenced by later reflections, and these effects are not revealed here.  However, if all the diffraction occurred within the first 3 ms of the impulse response, or equivalently if the additional path lengths for sound that travels to and diffracts from cabinet edges before reaching the listener are less than about 41 inches, then these windowed responses reveal the relative contribution due to diffraction for lower frequencies too.  This is true until you approach the sub crossover 110 Hz where the delay increases again, but at that point, the later reflections I've windowed are strong enough to substantially corrupt the direct response observed at the MLP.  Even with acoustic treatment, as some point as you go lower, the room wins and you need to apply the usual methods for optimizing subs, placements, and DSP to get the response you want.

 

For the doubters, here is the exact same data except with the window extended to the full impulse response:

 

center_channel_diffraction_in-room_long_window.png

 

Please ignore the response of the subs here.  They are not optimized to play well at these measurement locations.

 

The broad ripple pattern in the windowed measurements remains in the long time measurements.  There is a lot of bass build-up at and below 250 Hz for most

locations, but this is from reflections that arrive quite a bit later than the direct sound.  I believe it's mostly side wall reflections, which arrive in the range of 10-20 ms later.  The long time response in 100-200 Hz range depends a lot on the relative time of arrival of those reflections vs. the direct sound.  Hence, measurements sharing the same color (signifying the same horizontal angle) have similar responses.



#143 3ll3d00d

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Posted 08 October 2016 - 10:03 AM

it's rather hard to read a graph with loads of lines, different scales and no legend :) 

 

are you sure about that point about usable data being found under the window limit? I've never heard anyone say such a thing. The frequency resolution is defined by the window length after all.



#144 SME

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Posted 10 October 2016 - 07:13 AM

it's rather hard to read a graph with loads of lines, different scales and no legend :)

 

are you sure about that point about usable data being found under the window limit? I've never heard anyone say such a thing. The frequency resolution is defined by the window length after all.

 

Sorry about not matching the scales between the two plots.  I didn't prepare them at the same time.  As for a legend, I don't think it would be useful for these because of how many lines are there.  ;)  I just posted the long window responses to validate what I claim is shown in the windowed responses.

 

To be honest, the only reason this validation works here is because the in-room response of my center channel is otherwise quite clean.  The diffraction appear to be the strongest acoustic effect on the center channel woofer response for the first 10 ms or so.  For the left and right, the ceiling reflection is similar in strength and alters the response quite a lot.

 

As for the question of whether data under the "window limit" is useful, the answer is yes, but it's complicated to explain the how and why.  Frequency resolution in the analysis is defined by both the window length and the window type, and you can always pad the windowed part of the IR with additional silence to get more resolution.  REW does this by defaulting to a 100 ms "left window".  The primary concern is that the window size you choose captures the effect of the acoustic phenomenon you are studying on the frequencies you are interested in.  As long as it does, then the impact of that phenomenon will be accurately exhibited in the windowed frequency response.  The big caveat is that the IR you measure is a convolution of both the acoustic response and the electromechanical response of the system.  The electromechanical response will typically delay emission of lower frequencies more than higher frequencies, so one must be aware of both the electromechanical and acoustic delays involved when deciding whether the window is big enough for the frequencies of interest.

 

There are other more general caveats such as the fact that you can't really isolate the low frequency effects of multiple acoustic phenomena that show up in the IR at about the same time.  So if you have a room reflection that occurs at about the same time in the IR as diffraction, you can't easily tell their effects apart.



#145 SME

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Posted 18 October 2016 - 09:47 AM

So I am very eager to start on my subs, but I got distracted by trying to get better sound out of my front stage speakers.  They already sounded great, but I got to the point where I felt that the blind, manual tweaking was no longer productive.  I decided it was time to retool.

 

I updated my measurement software and script so now it will ping all 12 amp channels (yeah, I'm up to twelve now) independently with normal sweeps as well as shorter time alignment sweeps.  I ran a set of sweeps at each seat on the sofa and quickly discovered that my speaker toe-in was in fact too steep.  I feel kind of lame for waiting so long to do it, but I went ahead and optimized the toe-in to balance high frequency response of the stereo pair across the seats.  That brought them in to an angle of about 45-50 degrees vs. 57 or so.  It was enough for the sound at the MLP to be a lot closer to on-axis, which is a good thing.  It also totally altered the tonal balance of the treble.

 

That was a bit frustrating because I still have work to do to re-factor and re-tool the analysis/simulation software I was using before to simulate the bi-quads and FIR filters I used on the old MiniDSP units.  The old software was rather slow and tedious and hasn't been upgraded to work with the libraries that the live DSP processing use.  I'm looking at updating the visualization to use a web interface.

 

In the meantime, I went ahead and downloaded the latest REW to experiment with its frequency-dependent window (FDW) capability.  I like that REW allows you to simulate filters directly against an FDWed response.  For kicks, I dropped in a 2.2 cycle window (1/3 octave) and iterated on filter adjustments until the response was flattened to +/- 0.5 dB from 200-8000 Hz.  I chose 2.2 cycles as that's long enough to capture the anechoic response of the speaker at almost all frequencies and short enough to avoid contributions from early reflections for the most part.

 

Above 8 kHz, I have roll-off on purpose as virtually every speaker rolls off above here.  (Blame problematic early audio monitors for setting a bad precedent here.)  Below 200 Hz, there is a bit more variation including a +2 dB hump at 80 Hz because of the desire to balance bass response better across seats with the subs, which I may be able to improve with better DSP and more close-placed subs.  I opted to calibrate the 2.2 cycle FDWed (approximate first arrival) response to 80 dBC.  This gives me 85 dBC on the nose with 800-2000 Hz pink noise.  With this, I expect my normal "reference" listening level to be about "-4".  I opted for 80 dBC instead of 76 dBC so that I can play DVDs and other soundtracks with the typical -4 dialnorm offset without having to put the MV above "0".

 

In any case, I've been listening with these filters and am *very pleased* with the sound.  I'm willing to bet that the filters I implemented are far from optimal as far as avoiding unnecessary resonance and so on, but system still sounds absolutely fantastic.  It sounds very balanced.  The bass is very clear and powerful but does not overwhelm or obscure any detail.  Even crappy loudness war music has plenty of kick and punch, but of course, dynamic music sounds way better.  Well-recorded vocals are extremely realistic and addicting to listen to.  I've only had one day with it and haven't had time to listen to dialog on the center, but what I've heard so far may be the best sound I've heard from my system to date.  It might be the best sound I've heard on *any* system to date.

 

I anticipate I'll get even better results when I replace the crude biquads with tightly optimized FIRs that only correct the relevant bits in the time domain as well as when I optimize directly against data from multiple measurement locations as I did before.  And of course, I have a lot of ambitious ideas about how to get better bass that I can implement completely independently of this first-arrival correction.

 

I'm real stoked to try out some movie content, and I'm curious if the upper bass sounds bloated or not.  The combination of baffle edge and rear wall on the center channel causes a lot of upper bass (150-200 Hz or so) to beam diagonally in the horizontal plane.  They need a ton of boost (more than I'd rather have to run) to run close to flat on-axis with that 2.2 cycle FDW.  However, at about 6 cycles, the same response looks quite hot by several dB because of some strong early reflections.  IIRC, my steady response hits 90 dBC up there.  So this *should* give me some idea of whether 2.2 cycles or 6 cycles is better in that frequency range.  From the listening I've done so far, I'm thinking that the 2.2 cycle FDW will win, but we'll see.  If I hear too much resonance there (or if I have issues with headroom), I may need to compromise and back it down until I can improve the baffle/wall integration.



#146 SME

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Posted 19 October 2016 - 05:50 AM

So I watched a bit of film, so I could listen to the center channel.  The tonal balance sounds pretty close to right.  It's slightly bass heavy, but nowhere near as much as I'd expect looking at the 6 cycle FDW.  I can, however, hear the problems in the upper bass range.  It sounds a lot like room resonance, like in a bathroom where the acoustics may still be modal at around 200 Hz.  It is much more noticeable with speech than music and is annoying at times (to me).  It also may be increasing the apparent loudness a bit.

 

In theory, I could use more DSP to cancel that strong early reflection but in reality, this does not work well.  I'm already boosting the center by about 10 dB to get the 2.2 cycle FDW response flat.  To remove that reflection would require substantially more headroom on top of that, and I'm frankly not happy with the 10 dB boost.  It's too much.  Any distortion produced by the speaker while playing that sound will also be amplified, relatively speaking.  Furthermore, any filters that cancel that reflection at the MLP will just make things worse at off-axis seats.  Now, if I involve multiple sources in the DSP, I have a much better chance of getting a pleasing result.  With the speaker beaming diagonally, I have the extreme seats covered and could fill in the hole in the middle using close-placed MBMs.

 

But it's still far better to solve it acoustically, if at all possible.  I will have to try.

 

The other thing I did today is I relaxed my top octave high frequency roll-off by quite a bit.  I figured that now that I have the mid-range and bass very well balanced, I could maybe add back some of that top octave energy.  I chose to target the roll-off recommended by ATSC 2013 (page 34).  So far, I think it's working out very well.  A few music passages I've listened to so far sound a bit excessive up there, but it seems to work very well with most stuff I've listened to so far.  I may just have to make that roll-off curve adjustable because of the inconsistency in content.  Likewise, I could probably use something to tilt the response down slightly for some older music recordings that still seem to run a bit bright in the upper octaves in general.

 

At this point, I have to say that these are the best speakers I've heard, hands down.  By targeting flat response (except for the ATSC 2013 HF roll-off and a little bump in the sub) with the 2.2 cycle FDW I believe I have nailed the tonal balance and voicing perfectly.  I was actually very close already with my process of manual iteration, but that last little bit (literally no more than 1.5 dB change anywhere) helped everything snap into place.  Barring the minor issues discussed above, it's nigh-well perfect.

 

This accomplishment is a big deal for me!  I've been searching for this kind of sound since I started this hobby.  I obsessed over the concept of calibrating to an audio reference when I first got an AVR with Audyssey.  It's rather sad that Audyssey failed to deliver what it claimed to, but it sparked my interest and started me on the path.  It's almost 4 years later, and I think I have finally found reference.



#147 maxmercy

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Posted 08 November 2016 - 01:54 PM

I'm going to give the low-cycle FDW EQ a try to see how it affects my system.  Sounds like it worked well for you.

 

JSS



#148 SME

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Posted 08 November 2016 - 11:31 PM

I'm going to give the low-cycle FDW EQ a try to see how it affects my system.  Sounds like it worked well for you.

 

I hope it works out for you!  I'm not sure it will work for everyone.  It may just work well for me because of the exceptionally even coverage from my SEOS horn-based 2-way speakers.  My thinking was to try to capture the anechoic response and make that flat.  Along those lines, the FDW window length must be long enough to allow for crossover delay (which usually peaks at the actual XO, in terms of FDW cycle count), but not so long as to allow any close room reflections to interfere.

 

The bass may still need to be treated differently in some cases.  At some point, you go low enough in frequency that reflections and/or diffraction effects arrive too soon to clearly distinguish the first arrival from the effect of the diffraction/reflections on it.  My inclination is to then treat the response combined with these early effects as being the first arrival.  However, this may result in insufficient bass.  My thinking is that pro monitors are usually designed to be anechoically flat, and they often get used in the mixing room either as-is or with only narrower EQ corrections for room modes.  The implication of this is that the "first arrival" (including any reflections that can't be isolated) should be a bit hotter in the lower bass (say below 100 Hz) because the baffle step compensation built into the speakers effectively overcompensates once you go low enough that the strongest early reflections are constructive.

 

Of course, part of the reason I'm making this argument is that I am running my subs a bit hot with respect to the 2.2 cycle FDW, and I think it sounds better (as in closer to reference) that way.  Another possibility is that my MBMs sound better hot because they are so close to the MLP and effectively higher in directivity.  In other words, they don't put as much energy into the room for the same SPL as the speakers would if they were playing that bass instead.

 

The only real change I've made since my post above is to bump the subs up a bit higher.  I pushed the MBMs (frequencies between 50-110 Hz) up +0.5 dB, making them about +2.5 dB hot at the MLP when using the 2.2 cycle FDW.  Any more than that, and the sound is too slow and lacks punch.  The bigger change was that I bumped up the deep bass subs (frequencies under 50 Hz) by +2 dB, making them about +2 dB hot at the MLP in terms of the 2.2 cycle FDW.  I tried going up a full +2.5 dB to match the MBMs, but that extra +0.5 dB was enough to totally kill both transient detail and tactile sensation and make the bass actually sound wimpier.  (Note to readers: you read that right; excessive deep bass can kill your tactile response by masking the higher frequencies that contribute most of the sensation.)

 

After the changes, the timbre of the lowest notes is a lot more consistent with the higher notes on the same instrument, and kick drums are very satisfying.  The adjustments also tamed excess brightness, particularly in the top octave, on a lot of tracks.  (Deep bass seems to mask top octave content to quite an extent.)   Indeed, after these adjustments to the bass, I am finding the ATSC treble roll-off to be very satisfactory with a majority of recently released musical content.  These changes did not fix excess brightness for all content.  I'm fairly certain that there will always be content (particularly produced and/or mastered from the '80s and earlier) that sounds better with slight a downward slope (about -1dBish) toward the high end.  I have also noticed that quite a few releases from the mid-to-late '80s still seem too hot in the top octave.  This could be because monitors of that time were rolled off too much to try to compensate for the bad precedent set by earlier monitors.  Another possible explanation is that the excess is due to top-end roll-off in the DACs used in the monitoring signal chains of that time.  These older DACs may have rolled off at the very top because of the difficulty of creating an analog filter that cuts everything above 22.05 kHz (Nyquist frequency for CD audio) without harming response below 20 kHz.  Most of these older DACs probably had some roll-off at the very top as this was arguably a less bad compromise than the other options.  Oversampling DACs which have been in widespread use since the '90s address this problem by resampling to a much higher rate before conversion, giving the post-conversion analog filter much more room (in frequency space) to work with.

 

All of the above applies to music, and apart from the issues I've noted, music seems to be remarkably consistent as far as tonal balance is concerned.  In the absence of specific standards, precedence rules, and engineers strive to follow the precedents established by the huge body of existing recordings.  Movies are a different story, unfortunately.  Despite there being standardized calibration methods, movie sound tracks seem to vary a lot more in terms of tonal balance.  It would seem that standardization only actually helps if those standards are psycho-acoustically relevant.  And unfortunately the methods prescribed by the movie industry are a long off, even with recommended adjustments to playback level and target curve made for room size.  Indeed, standardization of this kind may have actually been counter-productive to the extent that such standards have been followed blindly.  The X curve in particular has probably contributed to a lot of movie soundtracks having too much upper mid and treble compared to typical music.  This is one area that near-field "for the home" mixes are said to improve on, but of course, the practices involved here are mostly non-standard and inconsistent.  (I say *mostly* because apparently some effort has been made, by Brian Vessa of Sony and others, to standardize home mixes, but the information on how this is done does not appear to be available to the public nor is it clear that these standards are actually being widely adopted.)

 

I haven't watched a whole lot of movies since I found my reference response for music, so I don't have an especially wide sample of content with which to formulate an opinion.  With that said, movies do seems to generally run brighter than music.  Even the "made for the home" mixes seem like they might sound a little better with the -1 dB slope I mention above for older music.  My guess is that some of the studios monitor the tracks this way on purpose under the assumption that it better reflects what speakers at home are doing.  I would guess that they are correct that most home speakers have the slight downward slope because this does sound better with older recordings and because excessive brightness is usually more offensive to the listener than the opposite.  However, I'd argue that they should still monitor home mixes flat (albeit with a roll-off at the very top similar to ATSC) to sound best on today's state-of-the-art monitors and to be consistent with modern music releases.

 

To deal with inconsistent tonal balance in movie releases, I plan to eventually implement a few controls into my DSP system to allow fine adjustments to the upper mid, treble, and top-end response.  While such controls will be nice to have for older music, it appears they will be critical for getting the best sound with a variety of movie soundtracks.  It's important to keep in mind that even 0.5 dB too much treble can add a few dB of loudness to a track, so if (room appropriate) reference level is too loud, there's a good chance it's just because the balance is too bright.

 

And just to reiterate, the sound I'm hearing on my system now (for music especially) is just stunning.  The bass is strong and tactile but remains in balance with the mids or treble.  Cymbals and hats are incredibly clean, detailed, and realistic (subject to the quality of the recording, of course), yet there's no harshness or fatigue even at high listening levels.  I get decent tactile bass slam even from loudness war music, but of course, dynamic music is way better.  I have some good dynamic live music recordings where the sensations from the kick drum are very strong, like something blowing up inside my chest.



#149 3ll3d00d

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Posted 09 November 2016 - 01:25 PM

I hope it works out for you!  I'm not sure it will work for everyone.  It may just work well for me because of the exceptionally even coverage from my SEOS horn-based 2-way speakers.  My thinking was to try to capture the anechoic response and make that flat.  Along those lines, the FDW window length must be long enough to allow for crossover delay (which usually peaks at the actual XO, in terms of FDW cycle count), but not so long as to allow any close room reflections to interfere.

 

I don't really follow that, a 2.2 cycle FDW is much smoother than an anechoic response.



#150 SME

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Posted 10 November 2016 - 07:38 AM

I don't really follow that, a 2.2 cycle FDW is much smoother than an anechoic response.

 

No, not exactly, but I believe it's a useful approximation in my case.  If my XOs were linear phase (maybe some day soon they will be), I would have tried an even shorter window.  Other sound systems may need different strategies depending on XOs and other sources of delay in response.  And to repeat what I said earlier, this strategy may not work nearly as well for speakers that are messier off-axis than those based on the SEOS horns.  Tonal balance may be substantially affected by later arriving energy, which is influenced by off-axis response.



#151 SME

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Posted 23 January 2017 - 09:27 AM

So lately, I've probably been enjoying my existing system too much and not spending enough time working on my subs.  I've continued to make very minor tweaks to the system.  The biggest change was to back off on the boost to 190 Hz on the center channel and instead make it ~flat "on-average" across the seats.  This is mostly because I often sit off-axis, and it didn't sound so good in those other spots.  The excess 190 Hz tended to overwhelm and mask out a lot of detail in voices whose fundamentals hit that spot.  It is an acceptable compromise until I extend the baffle.  The center still sounds very good, and it's nice to not have the weird feeling (literally) that arose from the build-up of later arriving energy at that frequency.

 

Otherwise, I boosted my sub bass response a bit, giving it a bit of an upward tilt.  I think the "first arrival determines tonal balance" concept doesn't apply in the same way to bass because room gain tends to naturally boost even first arrival sub bass in most situations.  An anechoically flat speaker will typically run hot in the bass without EQ adjustment, and that's often just the way things are monitored.  Instead, I've opted to adjust things by ear, and a few extra dB with some slope up toward the bottom seems to sound better.  There is a very clear point where too much low end washes out much of the transient detail.  Kick drum might go boom but not really thump, and bass instrument can sometimes go from being punchy and well-defined to indiscernible sludge.  If I raise the response to that threshold and then back-off about 0.5 dB, repeating this process for different regions of the response, I get a very solid, satisfying thump with no overhang from most kick drum.

 

Where I've struggled more is with taming the high end.  Running the top end flat just doesn't work for my ears and for most others.  In fact, it can be very irritating with a lot of content that I think most would regard as being well recorded.  What I'm finding is that the shape of the roll-off is to an extent more important than the amount of roll-off involved.  Very minute changes to the shape can have a big impact on the sound.  At the same time, content is quite variable in overall top octave strength, so I feel that having an adjustable UHF control will be advantages. 

 

I recently decided to experiment with a hypothesis that the best roll-off shape will almost approximate the roll-off caused by dissipation of sound as it moves through the air.  I was surprised to see that dissipation is actually pretty significant even over short distances for the highest frequencies.  After reviewing data of UHF roll-off vs. distance at various humidity levels, I settled on simulating distance roll-off via a single biquad centered at 12750-13000 Hz (higher for greater roll-off amounts) with a Q of 0.5 and gain ranging from 0 to -10 or so, approximately equivalent to roll-offs due to distances ranging from 0 to 20 meters or so.  With most content, I seem to do well by about -3 dB or ~6 meters of distance and occasionally opt to push it out to -6 or -10 dB for ~12 meter or ~20 meters of distance.  The consequence of adjusting the roll-off while maintaining the constraint of physical realism is is remarkably subtle.  It kind of acts like the audible equivalent of a sharpness control.  Too much, and the transients have a bit too much bite.  Too little, and there is a loss of fine detail as well as crispness.  These changes are very subtle, even compared to say boosting 15-20 kHz by 1 dB *without* altering the frequencies below and thus ending up with a physically unrealistic curve.

 

Along these lines, it's very fascinating to me that UHF and very low frequencies can mask one another substantially.  For a few hours, I accidentally had 15-20 kHz and up running about 1 dB hotter than I meant to run it, and it caused much of the sub bass to sound very distant and lose almost all of its weight and feeling.  But it only happened on certain soundtracks, those with full extension.  I was really weirded out until I recognized that it was the UHF that was masking the bass and causing it to diminish.  I tried pushing the subs up 3 to 6 dB, and it did nothing for its subjective loudness or power and merely muddied the sound.  Without fixing the UHF, there was no helping the bass.  Bringing the UHF to proper balance totally fixed the problem completely to my great relief.  I have also observed masking in the other direction.  When I had some harshness in the UHF, boosting around 45 Hz or so seemed to quell it considerably, even when I couldn't conciously hear any content down there.  Psychoacoustics are very weird.

 

Another thing I encountered when playing with UHF response and ending up too hot in parts was a sort of unrealistic hyper-detailed sound.  I haven't seen a Darbee before, but I imagine it to be the audio equivalent of turning the Darbee effect up way too high.  Upon first impression, the sound is incredibly enhanced, with far more richness and detail than is heard normally, but only upon more careful disciplined listening does it become clear that the sound is completely inaccurate and unrealistic.

 

At this point, I am declaring this experiment a success.  The sound seems to have improved by another notch.  Cymbals and hats sound absolutely great.  Indeed, I'm hearing cymbals emerge in background content of soundtracks where I never heard them before.  And there's even less harshness than there was before.  For the occasional soundtrack that still "bites" a little, I can easily dial-down the UHF without losing detail or weirding out the tonal balance.



#152 Ricci

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Posted 23 January 2017 - 05:03 PM

I tend to agree with much of the above. I tried a ruler flat on axis response outdoors and at the LP in smaller rooms a few times and neither sounds good to me. The top end is subjectively too much for my tastes. Sounds unnatural and the low end sounds anemic. This was especially true for the LF outdoors. I assume it is from the lack of direct vibrational queues, or other types of coupling with the body directly, that are much more prevalent in a vehicle or small room, that might help fill in the LF experience. Overall I prefer a bit of a downward tilt from the bass to the treble range, with the bass becoming a bit more aggressively boosted below 100hz. Most of my friends who are not very technically savvy when it comes to acoustics or speakers seem to agree with that type of general shape as well. Maybe it's technically a coloration preference but whatever. At the end of the day it is all about enjoying your music or movies isn't it?



#153 MemX

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Posted 23 January 2017 - 05:21 PM

Good posts :)

 

I smiled at the first part, though - it does seem as though sometimes the tweaking time overtakes the actual time spent just playing content for fun!

 

I tend to like the presentation of mine, but I'm just running Audyssey for processing.  Perhaps one gets to like what one is used to until one is presented with something that is better?

 

I do take my hat off to everyone who spends so much time improving their system - I've still not got round to putting the MiniDSP (which I bought about 18 months ago) into the system, and I really must do it soon before I have to move out and back into the missus' dad's house so we can save for a bigger place...



#154 SME

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Posted 26 January 2017 - 12:35 PM

I added a discussion along with some in-room response measurements of my system, as it was last time I measured, to the first post of this thread.  (Scroll down to the second section.)

 

I tend to agree with much of the above. I tried a ruler flat on axis response outdoors and at the LP in smaller rooms a few times and neither sounds good to me. The top end is subjectively too much for my tastes. Sounds unnatural and the low end sounds anemic. This was especially true for the LF outdoors. I assume it is from the lack of direct vibrational queues, or other types of coupling with the body directly, that are much more prevalent in a vehicle or small room, that might help fill in the LF experience. Overall I prefer a bit of a downward tilt from the bass to the treble range, with the bass becoming a bit more aggressively boosted below 100hz. Most of my friends who are not very technically savvy when it comes to acoustics or speakers seem to agree with that type of general shape as well. Maybe it's technically a coloration preference but whatever. At the end of the day it is all about enjoying your music or movies isn't it?

 

As per the discussion in my first post, there's a world of difference between flat *first arrival* frequency response and flat frequency response in a typical listening room.  I'm fairly certain that this is why a frequency response with significant slope sounds better indoors.  As for outdoors, I would expect a flatter looking response to sound better, albeit with some high frequency adjustments and some bass boost.  I already discussed the reasoning for high frequency adjustments in my first post.  I did not discuss the bass boost however.

 

I believe I've stated this elsewhere before, but I believe bass boost may actually be more correct.  The reason is that a typical monitor with flat anechoic response will exhibit room gain in a typical listening space, even for the so-called first arrival.  In the early days of audio, this room gain was probably never compensated for in the monitoring systems.  Instead, the room size and speaker placements in the mastering studio were chosen to be approximately representative of listener's homes.  Even today, room gain is likely not compensated for directly.  Instead, a lot of mixers will adjust the bass response of the monitoring system by ear until it sounds good, using trusted recordings for guidance.  In other words, bass boost is established by precedence, just like high frequency adjustments are.

 

When you say that you and most of your friends prefer the response a certain way, it's probably not because you prefer a colored sound.  Sure, there are people that prefer to play their subwoofers as loud as possible so they can experience their eyes wobbling and struggle to breathe.  That's a special case, where extreme bass at the expense of the rest of the sound is enjoyed as sort of a sport.  But when it comes to fully appreciating real world music and movie content, outside of a few dedicated "bass" genres, I believe preference is a lot less subjective than we are commonly led to believe.  The sound that you and your friends prefer to listen to may be a lot closer to what the mix and mastering engineers heard than you would think.

 

 

Good posts :)

 

I smiled at the first part, though - it does seem as though sometimes the tweaking time overtakes the actual time spent just playing content for fun!

 

I tend to like the presentation of mine, but I'm just running Audyssey for processing.  Perhaps one gets to like what one is used to until one is presented with something that is better?

 

I do take my hat off to everyone who spends so much time improving their system - I've still not got round to putting the MiniDSP (which I bought about 18 months ago) into the system, and I really must do it soon before I have to move out and back into the missus' dad's house so we can save for a bigger place...

 

I've put an enormous amount of time into tweaking, and I believe it's really paying off for me now.

 

I liked Audyssey MultEQ XT while I used it, but I now consider it to be seriously flawed.  Unless one's room is acoustically dead, it yields a very top heavy tonal balance.  Admittedly, I didn't realize this is for at least a couple years.  It certainly improves the sound in some aspects, but these improvements come at great cost with respect to other aspects.  Once I got the Pro kit with the adjustable target curve, I figured out how wrong flat was, but unfortunately, the Pro kit was too limited (+/-3 dB max difference in the target curve) and too tedious to use for me to find an optimum target curve.

 

I've heard that XT32 uses a completely different algorithm, but I'd hazard a guess that the result is still very top heavy.  Audyssey makes many claims that their system leads to a "reference" response, implying that a flat frequency response (with a tiny bit of top end roll-off) allows one to hear what the mix/mastering engineers heard in their studios.  Audyssey also claims that if one does not like the flat sound, then he/she has a preference that is inconsistent with the reference response.  These claims are totally wrong, and I'm not sure if anyone at Audyssey actually knows any better.  It's really quite absurd, because despite all of the psycho-acoustic research that supposedly goes into their technology, their approach appears to be based on a flawed understanding of how hearing works and a flawed understanding of how mixes are produced.

 

Definitely get that MiniDSP unit up and running.  Are you going to use it for your mains or just your subs?  Either way, I suggest using the MiniDSP exclusively for EQ and turning off Audyssey.  If your speakers are decent, they should have a much nicer tonal balance than what Audyssey gives you, and if you have EQ capability for them with the MiniDSP, then you can fine tune them as needed.



#155 SME

SME

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Posted 23 February 2017 - 09:56 AM

I recently migrated bass management from my AVR to my custom DSP.  This will allow me to do BEQ correctly and to do more powerful room EQ when I get there.

 

I've mentioned in the past that I found calibrating "first arrival" flat sounds best, with some exceptions.  One of those exceptions is the apparent need for a shelf of -1 dB centered in the upper mids for some music.  However, I'm trying a different approach now using a bass boost instead.  The idea is that an anechoic flat speaker has a bass boost from the floor reflection that shows up somewhere in the low mids or upper bass, depending on the design of the speaker and listener distance.  The boost is roughly in the range of 3.5-4.5 dB or so, and I make the center frequency of the boost adjustable to account for different mastering environments.  I'm still doing a lot of experimentation and listening, but moving the center frequency of the boost a bit higher seems to work well (as in sounds good) in lieu of the upper mid shelf.  It's not perfect, but I don't know if it ever will be.  A real floor bounce also has a dip in first arrival sound that appears above where the boost sets in.  Should I simulate the response dip too?  I'd rather not, especially being that its location, shape and depth will vary a lot more in different environments than the boost will.

 

So I'm pretty happy with how my music sounds.  However, I've been recently reminded that cinemas and dub stages are usually calibrated so that in-room power response follows the X curve.  Because of this, some movie content needs the X curve applied to the highs to sound right.  Some also seems to need an extra bit of upper mid cut, perhaps because some room reverb may be present there.  It depends on the room.  Making matters worse is the low frequencies.  In a dub stage with a lot of bass build-up the mains may be calibrated with a lot of attenuation down there in order to keep the power response flat.  As such, the bass boost I use is likely inappropriate for a lot of films.  At least its presence is less likely to be offensive than the lack of X curve where it's needed, but the bass boost may cause some boominess in some of the voices.   To make matters weirder, I actually need that bass boost in my calibration in order for my near-field subs to give me the same SPL as my mains via pink noise.  So is my bass boosted or not?

 

Needless to say, the cinema standard is a disaster for good sound.  The existence of home mixes just throws another wrench into the works.  Do they calibrate to a target curve or do they let the monitors run as they are designed (i.e. flat anechoic)?  Unfortunately, the one example of a home mix I'm aware of is very irritating without a -3 dB high shelf in the upper mids, and the voices are boomy unless I disable my bass boost.  I have no idea how they calibrated their system, but clearly it produced a mix with a very different tonal balance than most music.

 

It's now abundantly clear to me that the only way to get high end sound with a wide variety of movies is to re-EQ them as needed, and not just to restore filtered bass.  Right now, I must make the EQ changes manually, which is disruptive when trying to watch movies, but in the long run, I plan to implement a handful of enlightened tone controls that I can adjust remotely while a movie.  I could use this for music too, where I'd like to be able to adjust bass boost center frequency and UHF/distance roll-off amount more easily as well.






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