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Revel M106 Bookshelf Speaker Review

restorer-john

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Beryllium is an outlier in specific stiffness for metals, along with Boron. Most metals have the same specific stiffness, for example steel, titanium and aluminium are 3:2:1 in both density and modulus so making a titanium dome the same weight as an aluminium dome is necessary to make it as stiff (not strictly true since stiffness is geometrically proportional to thickness cubed so the aluminium will be stiffer than the titanium due to being a bit thicker.
Beryllium OTOH is a special case and its specific stiffness is much higher than other metals.
So if you want a pistonic tweeter Beryllium is the best metal since I am not sure one could make a boron dome (boron is usually built up as fibres on a thin iron wire core.

Or you can use 99.9% pure crystal ceramic alumina. :)

These tweeters are (NOS spares) for my Sony SSG-333es loudspeakers. Rare as the proverbial hen's teeth as they were only made for the one speaker and only for less than one year. Had Sony Japan find me a pair back in 1993 and have had them as spares "just in case" ever since. Ironically, it was the woofers that failed and now need new rubber surrounds...

333 tweeter (2).jpeg


Fully die-cast, spun aluminium, champagne anodized basket, finished all the way down the rim, even though they are flush rebated. Rubber trim between mesh and outer. Screw fit diffuser ring/magnet assembly and magnetically shielded (2nd magnet). Approx 1.2kg all up weight per tweeter.
333 tweeter (3).jpeg


333 tweeter (1).jpeg


They go in these:
ssg-333es.PNG
 
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Newman

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Beryllium is an outlier in specific stiffness for metals, along with Boron. ....
Beryllium OTOH is a special case and its specific stiffness is much higher than other metals.
So if you want a pistonic tweeter Beryllium is the best metal since I am not sure one could make a boron dome (boron is usually built up as fibres on a thin iron wire core.
No need to pick exotic pure metals, when ceramic coatings can be applied to form a composite sandwich with 'designed' properties for less cost.
 

restorer-john

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No need to pick exotic pure metals, when ceramic coatings can be applied to form a composite sandwich with 'designed' properties for less cost.

The thing about sandwiches is they always end up falling apart before you've finished with them.
 

Frank Dernie

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No need to pick exotic pure metals, when ceramic coatings can be applied to form a composite sandwich with 'designed' properties for less cost.
Absolutely, anodised aluminium, as used by Monitor Audio for decades, does the job very well, though I have never had enough detail to analyse the actual design, I was merely pointing out to the guy who wasn't sure about Beryllium that it has unusual properties.
My experience of its use are only owning Yamaha NS1000Ms for almost 40 years and the use in Formula 1 engines (until it was banned).
 

Chromatischism

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I would be the first to say that actual in room measurements do say a great deal. Though 1/6 octave smoothing is too much smoothing in my opinion. The most smoothing one can do would be psychoacoustic smoothing I think, otherwise stick to 1/12 or 1/24 octave.
But a mic in a room and a static graph do not say how our ear hears. There's the direct sound which hits the ear from a different angle than reflections and therefore the direct sound gets quite a different head related transfer function equalized sound than a reflection from the ceiling, floor, back wall and even side walls. On top of that modes and reverb / longer reflections arrive at a different point in time and this can psychoacoustically matter. So what your mic pics up is not the same as what your ear pics up.
Having said that, I've never heard great sound in a small room. Tried it myself once with quite a bit of treatment but couldn't get there back then (early days, was naive in many ways so not best attempt ever). Don't know if what I wrote earlier applies to a small room, quite likely other things matter more.
I thought I would get a comment about the 1/6 smoothing, as I don't normally use it, but "Psychoacoustic" smoothing was actually more smoothing than this. I know, I was surprised. It looks more like 1/4 or 1/5 if that were an option. Here is 1/24, which just shows more squiggles specific to that mic placement that we might not even hear (which is why I went with 1/6).

S400 Pair In-Room 1-24 Smoothing.png


Noted about how our hearing is more complicated than this.

EDIT: I just conducted a test using this video. I have to conclude that there is no drop in sound levels in the region at all, so the effect measured in the spinorama is not audible on my speakers in my room. It's possible the issue is audible to others in their room. The speakers are only 1.5 feet away from each side wall so there is plenty of early reflection coming at me, which I think explains my natural downward tilted response in this room. If you look at that graph again at the Buchardt website, the side wall reflections show the same tilt so they are certainly contributing a great deal to what I'm measuring. I keep telling people to keep that in mind when looking at my graphs. Someone with a large room would measure a nearly straight line, as they would mostly be hearing direct sound and this speaker is flat on-axis.

On your last point, I think it is definitely possible to get great sound in a small room. In my opinion, it starts with eliminating SBIR, which means ignoring age-old advice to pull the speakers out into the room which will just cause you to lose 80 Hz completely to destructive interference, for example. My speakers are perceptually flat from 30-100 Hz without subs and once you achieve that, it clears up everything else, like lifting an acoustic fog. And, in a small room, the early reflections, especially the horizontal, are absolutely crucial because again, they are going to be contributing a huge amount to what you hear. No one wants to hear two different speakers playing at the same time, just a quieter version of the same speaker so it just sounds clear and neutral. Third is to not sit near the back wall. If you can achieve those three things you've gone a long way toward great sound.
 
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Chromatischism

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It is apparent that you are cooking up something very special there. I am most certainly envious. When I looked at the distortion curves for the T34A-4 my thoughts were the same as yours, but I did notice that H3 started to climb at about 1.5 kHz, however H2, H3, H4 and H5 all remain low enough for the tweeter to sound clean even if crossed over at around 500 Hz, where it would probably give a good directivity match directly to a woofer. It seems weird that it could work that way, but depending on the diameter of a woofer, somewhere around 500 Hz where wavelength is appreciably greater than the diameter of the woofer the dispersion of a woofer is about as good as it is for a tweeter.
Indeed, advancements in driver design could negate the need for waveguides, however, an argument for waveguides at this time is that they enable much less expensive drivers to perform like much more expensive ones, and the sum of the parts to enable a higher-performing speaker than it would be otherwise. You know what they say about the result being greater than the sum of the parts...
 

Chromatischism

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I don't see why 1-2kHz would be worse? Look at the equal loudness curve. There's a dip in our ears sensitivity between 1-2kHz.
And I've also tested the exact effect of crossovers on fr with simulations and actual audio. It is quite apparent that crossover between 1-2kHz is way less objectionable than the crossover between 2-5kHz. And I've shared my audio sims with various music with various people and they all agreed. Would also stand up to any blind test without any problems it's quite apparent.
You can do a very simple test of this yourself. Simply put a notch filter on your audio and put it at 1.3kHz and A/B with and without it. And put it at 3kHz and A/B with and without it. And then A/B between the notch at 1.3kHz and 3kHz. No comparison, 1.3kHz will sound way less objectionable.
Make sure you make the notch about the right width / Q though to model for instance a LR4 cancellation axis.
Considering equal loudness of the human ear and all else being equal, a speaker that measures as a straight line would sound duller in the 1-2 kHz region and brighter in the 2-5 kHz region due to our higher sensitivity there. We know this is true of the bass region and the very high frequencies. Maybe some designers know this and are okay with a small dip there at the typical 2-way crossover, for balance?

I am not saying this is the right way to do things. Technically, the place for this balance or implementation of equal loudness would be in the music or movie mixing and mastering. It would be a part of the digital track, and the speaker should just play linearly from 20-20. But I'm just thinking that maybe there is some rationale out there for it. Now if only every sound engineer could get together and agree on a target curve, we could enjoy our linear speakers...
 

Dennis Murphy

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Considering equal loudness of the human ear and all else being equal, a speaker that measures as a straight line would sound duller in the 1-2 kHz region and brighter in the 2-5 kHz region due to our higher sensitivity there. We know this is true of the bass region and the very high frequencies. Maybe some designers know this and are okay with a small dip there at the typical 2-way crossover, for balance?

I am not saying this is the right way to do things. Technically, the place for this balance or implementation of equal loudness would be in the music or movie mixing and mastering. It would be a part of the digital track, and the speaker should just play linearly from 20-20. But I'm just thinking that maybe there is some rationale out there for it. Now if only every sound engineer could get together and agree on a target curve, we could enjoy our linear speakers...
But isn't that how we hear live music? I'm not quite following why there would be any need for equalization.
 

Chromatischism

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But isn't that how we hear live music? I'm not quite following why there would be any need for equalization.
For a live music venue, levels are all over the place, so it's hard to say. However, there is indeed a need to have much higher SPL in the bass region to sound balanced, and the same would need to be applied to the studio recording. I'm not a sound engineer but they have to be doing this. Or if they aren't, we get anemic bass.
 
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JustIntonation

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Considering equal loudness of the human ear and all else being equal, a speaker that measures as a straight line would sound duller in the 1-2 kHz region and brighter in the 2-5 kHz region due to our higher sensitivity there. We know this is true of the bass region and the very high frequencies. Maybe some designers know this and are okay with a small dip there at the typical 2-way crossover, for balance?

I am not saying this is the right way to do things. Technically, the place for this balance or implementation of equal loudness would be in the music or movie mixing and mastering. It would be a part of the digital track, and the speaker should just play linearly from 20-20. But I'm just thinking that maybe there is some rationale out there for it. Now if only every sound engineer could get together and agree on a target curve, we could enjoy our linear speakers...

No that's not how it works. All natural sounds get this reduced sensitivity of the ear. If you were to EQ music to the inverse of the equal loudness curve it would sound very unnatural and bad. (though you can fool the ear a bit by playing music softly and giving it the EQ curve of a bit louder SPL, the "loudness" button on hifi amps does something in that direction).
But really, try my suggestion. Play some commercial music, or other music which fills the spectrum well, on the computer and put a notch EQ on 1.3kHz vs 3kHz and you'll hear what I mean.
I personally think there may be even more at play than I wrote before. The brain is more used to a dip between 1-2kHz and the way we hear direction switches from phase difference mode to amplitude difference mode in that region as well (look up the wiki for "directional hearing"). I don't know what it is exactly but when I put on a crossover sim in the 1.3kHz region it just sounds way less objectionable than in the 2-5kHz region, even to the extend that my ear gets used to it very quickly and when you A/B it with a flat response the flat response will sound "wrong" like it has a peak EQ boosting the 1.3kHz region for a few seconds (you get that to some extend in all regions but not in the particular way of the 1.3kHz region).
But again, run the test and you'll most likely agree (if your headphones / speakers are somewhat flat enough to do the test somewhat fair).

edit: btw the main reason designers cross higher is because of the available tweeters and the physics of tweeters in general. In order to have the response be good enough up till 20kHz the tweeter diaphragm often has to be small, which means it lacks the surface area to push enough air for a <2kHz crossover point. There are only a few tweeters which can do it without a deep waveguide / horn (btw for horn loaded compression drivers it's very common to cross <2kHz or even <1kHz because those tweeters can easily do it, but horns have a different set of problems on their own).
 
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Dennis Murphy

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For a live music venue, levels are all over the place, so it's hard to say. However, there is indeed a need to have much higher SPL in the bass region to sound balanced, and the same would need to be applied to the studio recording. I'm not a sound engineer but they have to be doing this. Or if they aren't, we get anemic bass.
I'm jumping in at the end of a thread, and there may be some context here I'm missing. Let's say the live venue is an orchestra hall, and the orchestra starts out very softly and then all instruments crescendo to a double forte. Are you saying the recording engineer then has to increase the bass level in order to capture the balance we would hear in the hall live?
 

Chromatischism

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I'm jumping in at the end of a thread, and there may be some context here I'm missing. Let's say the live venue is an orchestra hall, and the orchestra starts out very softly and then all instruments crescendo to a double forte. Are you saying the recording engineer then has to increase the bass level in order to capture the balance we would hear in the hall live?
No, it would have been presented at the proper perceptual level in the venue. Sorry if I wasn't clear. I meant someone doing a studio recording (especially when much of it is synthetic) would be performing that function, not someone recording a live event. Unless the capture was inadequate.

Edited my last post with the graph as I conducted a test and concluded I can't hear a dip between 2000-3000 Hz.
 
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Chromatischism

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No that's not how it works. All natural sounds get this reduced sensitivity of the ear. If you were to EQ music to the inverse of the equal loudness curve it would sound very unnatural and bad.
I mean it's totally baked into the process. You certainly wouldn't EQ music to a Fletcher-Munson curve after it's been made. I guess I'm not thinking this through and using the right terms. The creator of the mix is human and has the same ears, so they would be adjusting the balance of things as they go. The result is that the system would need to be producing higher bass levels than 3 kHz levels, for example, for perceptual equal loudness to occur to them and to us. An interesting result is our music would probably sound gawdawful to non-humans with different ears and brains. I need to stop as I'm taking this off topic :)
 

JustIntonation

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I thought I would get a comment about the 1/6 smoothing, as I don't normally use it, but "Psychoacoustic" smoothing was actually more smoothing than this. I know, I was surprised. It looks more like 1/4 or 1/5 if that were an option. Here is 1/24, which just shows more squiggles specific to that mic placement that we might not even hear (which is why I went with 1/6).

View attachment 71301

Noted about how our hearing is more complicated than this.

EDIT: I just conducted a test using this video. I have to conclude that there is no drop in sound levels in the region at all, so the effect measured in the spinorama is not audible on my speakers in my room. It's possible the issue is audible to others in their room. The speakers are only 1.5 feet away from each side wall so there is plenty of early reflection coming at me, which I think explains my natural downward tilted response in this room. If you look at that graph again at the Buchardt website, the side wall reflections show the same tilt so they are certainly contributing a great deal to what I'm measuring. I keep telling people to keep that in mind when looking at my graphs. Someone with a large room would measure a nearly straight line, as they would mostly be hearing direct sound and this speaker is flat on-axis.

On your last point, I think it is definitely possible to get great sound in a small room. In my opinion, it starts with eliminating SBIR, which means ignoring age-old advice to pull the speakers out into the room which will just cause you to lose 80 Hz completely to destructive interference, for example. My speakers are perceptually flat from 30-100 Hz without subs and once you achieve that, it clears up everything else, like lifting an acoustic fog. And, in a small room, the early reflections, especially the horizontal, are absolutely crucial because again, they are going to be contributing a huge amount to what you hear. No one wants to hear two different speakers playing at the same time, just a quieter version of the same speaker so it just sounds clear and neutral. Third is to not sit near the back wall. If you can achieve those three things you've gone a long way toward great sound.

You're half right about the phychoacoustic smoothing. I thought wrongly it was more siilar to 1/12 octave above 1kHz but it's similar to 1/6 octave above 1kHz with slightly different weighting.
Must say I've seen much worse response than your room. At what distance did you do the measurement? I assume it was fairly close?
Though I still think it's next to impossible to get a good response in especially the bass in a small room without massive treatment. The modes will be strong.

As for your test in hearing the crossover dip, that's not a good test. For sure the crossover dip will be there quite strong when you go a bit above or below the on-axis and likely with the youtube video you still won't be able to hear it in that way. Just playing a slow moving sine is a very insensitive way of listening to amplitude.
 

JustIntonation

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I mean it's totally baked into the process. You certainly wouldn't EQ music to a Fletcher-Munson curve after it's been made. I guess I'm not thinking this through and using the right terms. The creator of the mix is human and has the same ears, so they would be adjusting the balance of things as they go. The result is that the system would need to be producing higher bass levels than 3 kHz levels, for example, for perceptual equal loudness to occur to them and to us. An interesting result is our music would probably sound gawdawful to non-humans with different ears and brains. I need to stop as I'm taking this off topic :)
Music tends to follow pink noise in general, not an inverse equal loudness curve. Though there are some specific curves which deviate a bit from pink noise for for instance commercial hits.

But yes, I took it too far and too long off-topic too :) Will stop now.
 

Chromatischism

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You're half right about the phychoacoustic smoothing. I thought wrongly it was more siilar to 1/12 octave above 1kHz but it's similar to 1/6 octave above 1kHz with slightly different weighting.
Must say I've seen much worse response than your room. At what distance did you do the measurement? I assume it was fairly close?
Though I still think it's next to impossible to get a good response in especially the bass in a small room without massive treatment. The modes will be strong.

As for your test in hearing the crossover dip, that's not a good test. For sure the crossover dip will be there quite strong when you go a bit above or below the on-axis and likely with the youtube video you still won't be able to hear it in that way. Just playing a slow moving sine is a very insensitive way of listening to amplitude.
I listen at 9 feet. Measurement was taken from normal head position at the seat.

Do you have a better test for hearing a crossover dip?
 

JustIntonation

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I listen at 9 feet. Measurement was taken from normal head position at the seat.

Do you have a better test for hearing a crossover dip?
Ah then your waveguide is working well >3kHz or you have some treatment :)
But no I don't have a good test for listening to crossover effects of your own speakers in your own room.
 

Newman

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You're half right about the phychoacoustic smoothing. I thought wrongly it was more similar to 1/12 octave above 1kHz but it's similar to 1/6 octave above 1kHz with slightly different weighting.
From REW Help: "Psychoacoustic smoothing uses 1/3 octave below 100Hz, 1/6 octave above 1 kHz and varies from 1/3 octave to 1/6 octave between 100 Hz and 1 kHz. It also applies more weighting to peaks by using a cubic mean (cube root of the average of the cubed values) to produce a plot that more closely corresponds to the perceived frequency response."
cheers
 

richard12511

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Can you explain your math here? Lets assume an 8 ohms speaker.....any less than that will require more power.

88dB at 1m would probably be around 80 dB at 12 feet in room. We need to add 25 dB to hit reference level.
2w = 83
4w=86
8w=89
16w=92
32w=95
64w=98
128w=101
256w=104. Still not quite there....We are a massive distortion/clipping/compressoin generator before this point.

Wait, we forgot to factor in the 3-6 db of eq used, at minimum, below 300Hz or so. This takes us from 104 back to 98...and thats not happening since we are trying to dump 256 watts in the speaker.

For four ohms, double all the above power requirements. This is why -10 MV runs people out of the room with low sensitive speakers...."It's too loud!"..yes...distortion/compression/clipping is very offensive.

I'm not sure how much louder than the M105s these can get, but I can say for sure that the M105s start sounding way too loud around 95db in my main room(~12ft). 95db is about all I can stand with them, and honestly they start sounding kinda cringe a few db below that. As is the case with the vast majority of hifi speakers, they definitely can't reach reference levels. Each speaker needs to be able to hit 105db on its own at 12ft. Taking into account the headroom eaten up by EQ, and the fact that 105db is not a strict limit(different movies can have peaks anywhere from 100-110db), I'd say mid 90s(at the very least) sensitivity is what you should be looking for if you want movie theater dynamics. Power Sound Audio makes some relatively affordable loudspeakers that are capable of doing it. I've never heard PSA speakers, though, so I can't really comment on sound quality.
 
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KaiserSoze

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Yes, that's the idea :) The baffle will be 29cm wide + the 12.5cm of the roundover on each side so in total 54cm wide. It will indeed reduce the first peak which sits on the top of the baffle step (though not completely eliminate it, it will be very smooth and insignificant, and pretty much no diffraction above that at all).

I realized afterward that I muddled that. Manifestly, making the baffle wide moves the 1st peak of the diffraction ripple lower in frequency. As such the larger the radius of the round-off relative to the width the baffle the better the effect will be, the ideal being a cylinder with just enough flat region at the face to mount the drivers. For a driver mounted equidistant from both edges the 1st peak will occur at wavelength matching the baffle width. For a driver mount 2/3 of the way from the further edge, that distance will account for 180 degrees (the other 180 degrees is due to the 180 degree phase shift in the "soft" reflection), which means that 2/3 of the baffle width will equal 1/2 wavelength, and that the wavelength will equal 4/3 of the baffle width. For full width equal to 54 cm, if the affected driver is mounted at the midpoint the 1st peak will occur at 635 Hz. If the driver is mounted 2/3 of the way from the further edge (so that the harmonically related peaks occurring higher in frequency than the 1st peak will line up with the dips, i.e., with the frequencies where the wave coming off the edge is out of phase with the direct wave), the 1st peak will occur at 476 Hz. Given the low crossover point you indicated, the affected driver will be the midrange. Since the round-off will just barely be effective at these frequencies, the implication is that the midrange and tweeter should be placed off-center, such that the distance from one edge is twice greater than the distance from the other edge.

I've done this simple analysis a number of times before, and it always comes out this way. This was Linkwitz's point about trying to using rounded edges to overcome the diffraction ripple. In order for it to work, the enclosure pretty much needs to be a cylinder, i.e., the round-off radius has to be not much less than half the baffle width. When this is the case the diffraction ripple is avoided but of course the overall step remains. And to analyze the effect of the step in any meaningful way you have to make assumptions about the speaker placement, especially the distance from the baffle to the wall behind the speaker. The reflection from the wall is sort of the inverse of the baffle step with diffraction ripple. At very low frequency the wall reflection is in phase with the wave coming directly from the driver (which is why it doesn't make sense to try for 6 dB of baffle step correction, contrary to what a lot of hobbyists seem to believe). As the wavelength shortens the reflection from the wall becomes increasingly out of phase with the wave coming directly from the driver, until eventually it is 180 degrees out of phase and a dip occurs in the sound we hear.

An obvious question is whether it is possible to force the frequency where this dip occurs to line up with the first peak in the diffraction ripple. It turns out that the only solution to the equation you end up with is for the baffle to be mounted flush with the wall. This is an interesting exercise for the purpose of demonstrating that this is the only solution. The dip will occur when the total distance travelled by the wave, from the center of the driver to the edge of the baffle, then to the wall, then to the listener, is greater by one-half wavelength than the direct distance from the driver to the listener. For a driver mounted equidistant from both edges, the wavelength for the dip will be equal to the baffle width plus 4x the distance from the baffle edge to the wall. The wavelength of the first peak in the diffraction ripple will be equal to the baffle width, for a driver mounted equidistant from both edges. Thus, the dip will always occur at wavelength greater than the wavelength for the first peak in the diffraction ripple, the difference between the two wavelengths being equal to four times the distance from the baffle edge to the wall. The more closely the speaker is placed to the wall behind it, the more closely the dip will line up with the first peak in the diffraction ripple. If the speaker is mounted flush with the wall there is no baffle step and no diffraction ripple. Of course there is a popular notion that a speaker is best placed out away from the wall, but notwithstanding the popularity of this notion, I have personally not ever encountered an explanation that makes sense. If in fact a speaker sounds better when placed further from the wall, the reason is most likely that the speaker has a poor off-axis response which is rendered less obvious by moving the speaker out away from the wall.
 
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