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Bulding the Room2 listening room


Kvalsvoll

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@maxmercy, I use a modified 10" driver, it is an old discontinued seas. 10" is a good size - good area for sensitivity, not too large so it does not change the sound field it is supposed to measure.

The 3x has less velocity in the important 30-40hz range, and a little less above up to around 120hz. Changing eq or delay on the BL back unit changes velocity response.

The 3x sounds better because it fills in a dip in the response around 60hz, phase is same, spectrogram is a little better.

The loss in velocity 30-40hz is noticeable. The increased v at ulf is more noticeable, because low freq noise stands out and becomes annoying - too much is not good. Further experiments can be to move the high-pass on the BL up in frequency, this will change the phase and perhaps make the velocity smoother - less ulf, fill in the dip 30-40hz.

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  • 1 month later...

Testing midbass horns in Room2.

Calibration is complicated, and benefits in this system and room are questionable. This is just to ensure they work, and to learn how to set up and calibrate the system, so that the customer can receive some useful guidelines.

Usable range around 45hz up to 200-300hz, but for cf above 150hz you should absolutely use stereo processing, so for a typical av-processor system with bass-management it can be used up to 150hz.

Capacity around 120dB+, depends on how much power you have.

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  • 2 weeks later...

Maximum speaker delay in processors/receivers - a critical property, which is usually not sufficiently described in the manual or product presentation.

Anyone know the limits for different types, brands? I seem to remember this issue has been up before, but oooohhh.. using the rest of the day searching will not happen, and new information and new models may be available.

 

The problem:

Getting the timing correct is crucial for high performance sound quality. For systems with front main speakers and separate bass system that means to delay the main speakers so that they sum correctly with the bass system in the frequency response AND IN TIME/PHASE.

On most processors this is done by setting a distance on the different speakers. Typically, you set the front to the measured physical distance, and end up adding several meters for the bass system (subwoofer)  - THIS WILL DELAY THE FRONT SPEAKERS.

If you know what you are doing, you set the delay using measurements, so that timing and phase gets as good as possible to achieve. If you read on-line guides and audiophile magazines and pay no attention to how things really work, you set the distance for the bass-system equal to the physical measured distance, and conclude that subwoofers always sound kind of sluggish and is best switched off for music.

Since you are smart, you want to do it the way that actually works to get better sound, and end up seeing that the distance entered can be quite huge. And in some cases it may be possible to reach limitations of the processor in use. Obviously this is a no-go limitation for a processor, so if you have an installation that you know will require large delays, you want to choose a processor that satisfies this requirement. You want to see a specification for this number.

But this number is not in the brocheur or manual, it is not in any "test" performed by on-line or paper magazines - because the don't even understand why this number is important - so the only way to know is if someone have found the data.

 

My contribution:

Denon/Marantz processors, AVR:  Max distance difference 6m / equal max delay 18ms.

Devialet amplifiers: 20ms. 

Hypex DLCP and my SA-700 amplifiers: 15ms. (Though not relevant on the sub amp, becuase it is the mains that need delay.)

Onkyo processors, AVR: ???

Some readers now realize I need those numbers for the Onkyo.

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1 hour ago, SME said:

My Denon 3313CI AVR offers up to "60 feet" distance (a bit over 50 ms, in terms of delay) and a maximum difference of "20 feet" (~17.5 ms).

Same as the Marantz and Denon I tested, it is very likely they share the same processing.

Note that it is the DIFFERENCE (17.5ms/20ft) that is interesting, as this is the number that defines how much delay is possible on the closer speaker to make it match the farther.

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  • 4 months later...
  • 1 month later...

English version of the small speaker article:

https://www.kvalsvoll.com/blog/2018/10/20/can-a-small-speaker-perform-like-a-big/

To discuss or comment, you can reply here in this thread.

Small speakers without the sound compromise have always been wanted, and there are several new speaker offerings claiming the problem is solved - small sounds as good as huge. But it doesn't. This article tries to describe some of the most important reasons for this.

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Small speakers have come a long way, especially with regard to bass performance.  I believe the Apple Homepod offers usable extension in a residential room to 35 Hz or so, especially if it's near a corner.  The servo-controlled (IIRC) internal woofer probably doesn't compress much at all unless the volume limits are being approached, which probably aren't that high.  (A lot of small speakers adjust the EQ with the volume control or may rely on some kind of bass-selective compression to allow louder sound without overloading on bass).  The Homepod won't play the lowest bass that loud, but a lot of customers are OK with that.  To be clear, I'm not endorsing Apple or the Homepod product, which I've never even listened to myself.  I just want to point out what's technically possible with current technology (and big R&D budgets).

My opinion is that a great speaker can be made very small, but where size ultimately comes into play is how loud the bass gets.  Furthermore, this trade-off can be partially mitigated by spending a lot more money.  This assumes active designs.  In passive speakers, good bass extension from a small space requires severe compromise, usually poor woofer efficiency.  Poor efficiency means a lot more power is being drawn from the amp and dissipated in the coils (and maybe crossover) for the crucial 100-400 Hz range, just so that the speaker can be "flat to 45 Hz".  You said the F2 extends to 60 Hz, which means it's not only using a larger woofer but one that doesn't give up as much efficiency in order to play lower than it needs to.

Another detail I think worth mentioning that I don't see in your write-up concerns speaker linear response and voicing decisions.  Passive speaker designers rarely know where the speaker will be placed relative to room boundaries, except maybe the floor in a floor-standing design.  Many designers likely voice their speakers with deficient 100-400 Hz output to try to accommodate being placed near walls or corners and also because resonances there can be quite offensive sounding.  Resonances in this "mud range" can bury/swallow voices, so many designers probably err on sacrificing punch for clarity in poor placement scenarios.

It is true that speakers with larger baffles/cabinets interact less with nearby room boundaries, often at the expense of worse baffle/cabinet diffraction effects.  The size of woofer alone (assuming <= 12") has less effect on the radiation pattern than the baffle and cabinet at these frequencies.  For wall/corner placements the benefit of increased directivity of a larger baffle and cabinet probably outweighs the increase in diffraction problems.

Most of the above discussion may be rather moot with good in-room EQ optimization.   I suspect that with good EQ optimization, differences in speaker size can be made largely irrelevant for low frequency sound quality at low levels.  On the other hand, I've noticed that speakers placed near walls and corners almost always have a suck-out somewhere in that 100-400 Hz range, which needs EQ boost to compensate for.  So at higher levels, capacity becomes an issue much sooner than without EQ.  The bottleneck region ends up being 100-400 Hz, not < 100 Hz which I find usually needs cuts rather than boosts.  So sadly, most passive consumer speaker designs sacrifice efficiency in the range that it's badly needed in return for bass extension that's not actually needed.  To be fair, they are designed to work well when placed away from walls, but who really has room / spousal approval for that kind of thing?  Not to mention how much better the sub bass is if reproduced from near a wall or corner.

So as I see it, the discussion should be more about the merits of active designs in general (including any system using powered subs) along with in-room optimized EQ and based on high efficiency, low mass, pro-style drivers.  Size does still matter, but for many listeners who only need low or moderate levels at home, a well designed active small speaker (with or without separate subs) can do the  job very well.  It's got to be active though.  Really I think it's time for passives to just die, except in custom / specialty systems where there's likely to be some kind of active processing anyway.

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Small will still not be the same, because directivity will be different. Unless directivity is designed to be similar, of course.

Active does not make any difference, and eq can not change decay profile, which is a result from room and speaker radiation. The solution lies in the physical acoustic design of the speaker.

Active is necessary to solve some issues, such as crossover at lower frequencies, and delay between drivers where this is necessary. Proper crossover between bass system and main speaker must be active, because it is too difficult and costly to make a passive network that works, and the quite large delay needed on mains can not be solved passive. Subwoofer usually requires more power, too. But then all this is not necessarily true either, because when I designed the C2 (1992) I made a passive crossover at 150hz, so it is possible, using computer simulation. But timing will be off, this ends up as a text-book 4. order crossover, with option for individual delay it is possible to design crossovers with no timing problems.

The new F105 is a small speaker with 5" lf driver and dome hf with moderate horn loading. The cabinet is a damped dipole with resistive acoustic ports. Directivity is controlled all the way down to 100hz, where it already has started to roll of. Design f range is 120hz and up. With a small subwoofer - will be half size of a V6 - that can be placed near boundaries, with dsp and eq, this gives a small system that will work well in normal rooms. The response is much smoother and more similar in different locations and rooms due to the directivity, even when this directivity control is quite small. But it doen't sound like a big speaker, regardless of sound volume. This small speaker would not benefit from active configuration, but the system as a whole is active where the crossover to the bass system needs to be done in a processor, and the bass system need dsp with eq. The next speaker will be a little larger, with 2x 5" lf drivers of a very different type. The goal for this one is to achieve good transient response - as long as you keep the volume down.

 

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  • 3 weeks later...
  • 2 months later...

There was some discussion about some subwoofers that were linked to my name, in a different thread. Please feel free to ask or comment, and you can use this thread, no need to start a new one.

If you go to my web site, you will see product presentations of the subwoofers I sell, and articles related to those subwoofers and bass in general. All presentations and product info are available in English language, most of the articles as well.

What I am going to say now, may seem a little strange:

I am not on data-bass to promote my products and services. Actually, I am not that keen to have a discussion about either products or technology, here. There are reasons for that. I want this to be my "free-space", where I can post and talk about things, and not worry about whether what I am saying is good for my own products. But feel free to ask, and I will try to answer. There is a contact-page on the web site, where you can find information on how to get in touch with my company. The company is also on facebook, where you can post public questions, or send private messages if you prefer that.

The obscenely large "horn" pictured in that other thread has a story behind it. The driver is a 24", so that thing is quite a bit larger that what it seems like from a quick look at the picture. The origins of this design was that the builder was curious about whether a compact-horn using some 24" drivers he already had, was possible. And it is possible, but even with the chosen tuning it gets very large, and the performance per size-unit is not particularly good. The driver is simply too large. It is a tuning with very large rear chamber and short horn channel, closer to a ported pox with huge port, than a real horn. This design performs a little better than 2x V110 - but those V110 would be half the size of one of these overly huge cabinets. The sound quality, however, should be quite good, as there are no resonances and very smooth response in the intended pass-band, it also has a very low cut-off. He has built several of those now, and it would surprise most of you what they have replaced, how sound quality improved, and I have never seen any complaints or concerns about capacity. Which does not surprise me at all, and I have not even heard them.

As for the question - why? They are too large, similar performance can be had in half the size, on paper it may seem like even much smaller than that using regular "mickey-mouse" subwoofers could do the job. So, why on earth choose something like this, even if the price-per-size-unit is extremely good due to cabinet made in construction-chipboard. My take is, that the 'I have a 24"' is the major factor, and combine that with the knowledge that when it comes to bass, the real thing requires something very different from what you find in the typical shop, price is low, and suddenly you find yourself looking at ways to make room for a couple of those small houses inside you listening room.

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  • 1 month later...

New articles in the blog-section, published quite recently:

The F105 loudspeaker - review-style article on this small loudspeaker with some quite interesting technical solutions:

https://www.kvalsvoll.com/blog/2019/04/23/the-f105-loudspeaker/

(Use google translate for the text that did not make it through the author's translation service.)

Bass and sound quality - 4 real world examples:

https://www.kvalsvoll.com/blog/2019/03/10/bass-og-lydkvalitet-4-eksempel-fra-den-virkelige-verden/

(Use google translate, it continues to improve, and has now reached a performance level sufficient to make most of my articles understandable in English.)

Looking at what is happening around on the net and otherwise, it is quite apparent that audio has died and become sort "Reign of the nonsense". There is no interest for technology, little innovation in new products, what is left of audio press are rendered totally irrelevant, lots of nonsense products gets whatever remains of attention. And there is simply no advancement in knowledge and technology - you find yourself just repeating and trying to explain things that by now should be known ("it is known"), which of course hinders further development up to the next level because you are stuck with this debunking of myths that should not be alive. Most of the really serious enthusiasts now build their own custom systems, which is good, but also not so good for audio as a business, because there simply is no market.

Read the articles. Is this interesting? Is it relevant? Is it entertaining? Does it make you inspired to learn more about audio, put on some music and listen, find out what you can do with your own system?

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Re the holes on the side, translate says they are "acoustically filtered", does it mean some damping material behind the holes? I looked into doing this once but the amount of trial and error (and/or lack of modelling tools) put me off. One thing I do remember though is that any such speaker tended to be pretty deep in order to get enough delay on the side output to affect directivity in a desirable way (iirc). These are v shallow speakers though by the looks of it. Are you using them for this purpose or something else?

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7 hours ago, 3ll3d00d said:

Re the holes on the side, translate says they are "acoustically filtered", does it mean some damping material behind the holes? I looked into doing this once but the amount of trial and error (and/or lack of modelling tools) put me off. One thing I do remember though is that any such speaker tended to be pretty deep in order to get enough delay on the side output to affect directivity in a desirable way (iirc). These are v shallow speakers though by the looks of it. Are you using them for this purpose or something else?

Yes, something inside the cabinet behind the ports to control acoustic resistance. This is modeled and simulated, at least to some extent. Dimensions and placement of ports, cabinet volume, is nice to get reasonably correct. The damping will always need some experimenting, because it is not straight forward to get good models of the acoustic properties of the damping material. This is not a problem, it does not take much time to get it right.

The cabinet does not need to be very deep, but the location of the ports, baffle width, speaker driver size all matter for resulting response.

I plan to develop more of this type speakers, with damped ports to control radiation. The F105 is small, with low-cost drivers, I designed them to test the concept. And it works.

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  • 2 months later...

Audibility of group delay at low frequencies:

https://www.kvalsvoll.com/…/audibility-of-group-delay-at-l…/

It is then confirmed and proven that group delay at low frequencies is audible. 

This means timing - GD, phase - matter for sound in the bass range, it is not sufficient to tune towards a flat frequency response alone.

(Yes, we knew, told you so..  But, now it is actually proven with evidence in a replicable,  described, controlled experiement.)

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Not only improperly calibrated subwoofers, but crossovers can also degrade the phase/GD:

X-over.thumb.png.f8a2234b5fde7a300f1391748393d8ee.png

It is a generic drum sound (top) with 80Hz 1st order BW, LR2, and LR4 crossovers applied.  For best fidelity/coherence, FIR can be used to fix the phase/GD problems that a subwoofer crossover imparts to the signal.  

To tell the truth, if you get this far and have solved all of the other problems with bass and sub integration in small rooms that you are correcting the sub crossover, you probably already have a well optimized, terrific system.

JSS

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@maxmercy, yes, this is true. But it is actually possible to remove all gd/phase from the crossover itself, by adjusting slopes and delay. This is also possible to do higher up in frequency, in even in passive crossovers, but then you don't have the option of adjustable delay, so you depend more on simulations to get it right.

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10 minutes ago, maxmercy said:

How is this done?  In AVRs, the subwoofer crossover is not adjustable for slope in most cases.

JSS

Delay is adjustable, and the rest is fixed in the dsp on the bass-system. For a text-book filter there is a delay, and if you adjust delay on mains to remove this, it wqill not sum correctly. But if the slope on the bass-system is different from the text-book, it is possible to end up with something that sums correctly and has zero delay. you end up with flat phase and flat GD.

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Thanks for creating and posting these samples.  Can you explain a bit more about the reasoning that went into creating the particular filter response shape?  Is this supposed to be group delay related to some kind of filter (e.g. crossover), an in-room acoustic effect, or something else?  FWIW, I definitely notice a difference between the 100 ms and original with a clear reduction of impact.  Between the 20 ms and original, I think I notice a very slight reduction in impact, but I'd want to do blinded A/B/X to be confident.

Now having said this, there is a big caveat with your study because there's a big difference between GD applied electronically and GD that arises from acoustic effects.  That's why I asked my first question above.  Time and time again, engineers try to treat room acoustics as a "linear transform along a wire", when this is not the case at all.  The ears and body are capable of sampling pressure at multiple locations (the ears and tactile), and the brain is very well adapted to parsing the content of the source  (both time and frequency aspects!) from what could be a very messy sound-field with dramatic local variations in measured frequency response and group delay.  So in general, electronic changes may be far far more audible than FR and GD features of similar magnitude that appear in in-room measurements.

Another potential caveat here.  You indicate that the frequency response of your filters "is reasonably flat, considered below threshold for audibility".  I can't comment with certainty in your specific case, but in general, I would not be surprised if the frequency response changes you show were well above the audibility threshold on a system with strong accurate bass.    This alone could have substantially affected the amount of perceived impact.  Again, there is a big difference between filters applied electronically and influence of acoustics on measured sound vs. perception.  Depending on the circumstances, I believe the brain can pick out excruciatingly small changes, likely below 0.01 dB for bass.  These can be perceived most readily on transients.

Regardless of the audibility of your filters, this experiment says little about the audibility of characteristics arising from room acoustics or whether it's necessary to "correct" the group delay deviations seen in in-room measurements.  I can't emphasize enough how important it is to keep this distinguish in mind clearly when optimizing response.

Also a comment about the sample material.  The kick seemed a bit soft, diffuse, and fluttery to begin with, and it didn't really sound consistent between beats.  Audacity spec analysis suggests that the kick has some high Q ringing at various frequencies, which also does not appear to be consistent between different beats.  Differences in group delay might be a lot more apparent on tighter transients that don't ring so much.

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8 hours ago, maxmercy said:

Not only improperly calibrated subwoofers, but crossovers can also degrade the phase/GD:

===cut===

It is a generic drum sound (top) with 80Hz 1st order BW, LR2, and LR4 crossovers applied.  For best fidelity/coherence, FIR can be used to fix the phase/GD problems that a subwoofer crossover imparts to the signal.  

To tell the truth, if you get this far and have solved all of the other problems with bass and sub integration in small rooms that you are correcting the sub crossover, you probably already have a well optimized, terrific system.

I believe these group delay shifts are pretty modest.  IIRC, an ideal LR4 at 80 Hz has excess delay in the 5-10 ms range.  I used to think that getting more energy in the first arrival is important for the most intense slam, but I now believe that the experience of tactile slam is actually a very high level perception.

My thinking is that the brain samples information from both ears and vibro-tactile receptors and then applies sophisticated "room correction" processing to this information in order to reconstruct an accurate estimate of the actual time-frequency aspects of the original sound.  In my own room, as I tighten up the bass response accuracy, the sound and tactile sensation I perceive become remarkably more uniform throughout the room, despite what the localized measurements suggest.  If accurate enough, one can get an extremely tight "thump inside the body" effect that is more like what one experiences with a powerful system outdoors.

I'll hopefully be soon exploring this in other rooms and am particularly interested to see what I can do in tiny pathological rooms.

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