Kvalsvoll

@maxmercy There is not much correlation between velocity and spl. For the 1x config the spl looks like it follows the sideways velocity to some degree in the range where the 1x config partly fills in the cancellation dip. Velocity in 0 direction (normal to front wall) has a small dip where the cancellation occurs, but this dip in velocity does not follow the spl exactly. Especially in the 4x config the velocity is present where the spl drops off to 0. This shows that velocity and spl must be out of phase here. Just measured 3 V110 units. They are reasonably consistent, and matches the design sim quite well. Nearfield measurements at different locations in the horn mouth, and inside the horn path. I don't have any useful outdoor measurements, tried to measure the first one while moving it from the workshop, but with temp below freezing and windy conditions this was hopeless. It is also difficult to measure properly, due to the size of the radiating horn mouth - it is too long for 1m measurements, and the outdoor space is not large enough for good measurements at longer distances, may have to go up to around 4m to get correct results on the V110. Nice to see the smooth response across this wide frequency range, the rather complex internal construction with 4 damping chambers actually works.

Yes, with 4x the measurements show that most of the velocity is head-on, as in a plane wave. It is also interesting to observe that the magnitude in the 0-direction (head-on, normal to the front wall plane) is the SAME for all configurations - it does not increase when sideways and up-down velocity decreases.

Front wall has been like this since the room was built, it is 20cm Rockwool A with some slats, designed to work from around 100hz and up, which it does. There is a correlation, yes, but in this room, for this set-up, the frequency response is very similar for all those configurations. This is because the huge cancellation problem is a LENGTH reflection, it must come from the corners on the back wall. I did not believe that was the case, as there is a very large opening on the back wall, but those measurements show that a more plane wavefront from the front actually causes the cancellation dip at 60hz to be larger.

3.35m wide, around 4.6m length - before adding the 20cm front wall absorption. So this is a small and quite narrow room. It doesn't feel cramped, there are openings on back wall and right wall, and having all walls white i suppose helps. But sound doesn't care about wall color, especially the width is challenging - little room for treatment to fix it, and since it is narrow, the side wall reflections hardly do anything good. Turned out quite nice, sounds reasonably good, but this small space requires treatment to work for high quality sound reproduction. Mind that quite many will find themselves in a quite small room, or constricted to a smaller part of a larger living room, for the music and sound. It is often better to use the longer wall as front wall - turn it 90 degrees - in a small room of such proportions. Then you would sit close to the back wall, which causes its problems, but also the advantage of not having to deal with back wall reflection in the bass range. There are some pictures earlier in this thread, showing the acoustic treatment with absorption and hard back with poly diffusors.

@maxmercy, 4 units stacked (2 units stacked) near corners on front wall, so you get these systems: 1x: FL 2x: FL FR 4x: 2xFL 2xFR Let's see if it is possible to get a picture: I managed to take a full set of measurements before taking down the system. And they show that the systems have different sound field properties - but which one is better.. The 1x is all over the place, the 2x removes sideways velocity, the 4x removes both sideways and up-down. Yes, the 4x is better, but at any sane volume, it is hard to justify the double cost compared to 2 units. In a different room, or larger room, the situation may be very different. The 1x holds up surprisingly well, but it does not have enough capacity for that wall-of-sound feel that the 2x and 4x certainly has.

@Ricci, I should comment on your excellent post, just too busy with the new subwoofers. If all subwoofers were specified the way you suggest, it would be possible to see and compare models, and decide whether capacity and extension meets requirements. Set up a system with 4x V110 in Room2, to test it, pictures on facebook (kvalsvolldesign). Easy to compare going from 1 to 2 to 4 units, with dsp presets ensuring similar frequency response and level - only capacity and sound field intensity/velocity are different. Interesting.

FAQ is a good idea. Easy to add subjects. Those interested enough to consider buying use email. For questions and comments people can use the facebook page, it has both personal conversations and public comments. The blog posts has a comment option. I am now making a blog post on those specifications, basically the same text as above here, and I will add some examples and measurements to show how this works in practical situations.

What are relevant and useful specifications for a subwoofer. A complete set of measurements, showing frequency response, capacity and distortion, is sufficient to tell how a subwoofer will perform. Then you can see how loud it can play at different frequencies, which is what you need to know for system design, you can see how low it can play and how loud. The graph also gives an indication of usable frequency range upwards. However, most customers don't really want to see lots of measurements, they do not understand what those graphs mean, and they acknowledge that fact. For the new Compact Horn subwoofers I did this: The output capacity number and the frequency range gives the necessary information. You want to know the output capacity to be able to dimension your bass-system, and you want to know the usable frequency range to see if it reaches low enough and covers all the range up to the desired crossover. The less tech-oriented customer still does not make much sense of the numbers, and is more likely to go by what I recommend. That's fine. The tech-experts needs to be educated on the meaning of those numbers, because they make no sense to them since they are different from what other manufacturers typically publishes. They don't recognize the meaning of Output capacity, and the frequency range is not the same as frequency response with specified tolerance limits. This is labor-intensive - requires lots of time and effort to educate and show. Perhaps these customers should be ignored - it's really a question of effort vs. value. One solution could be to make additional specifications and measurements available, so they can see exactly what the performance of the subwoofer is. The graphs still require some explanation. (The real experts usually get it, so they don't need any more education. They may ask for measurements, if they want more exact information.) Typical subwoofer specifications are useless. They say nothing about capacity, frequency range specifications are at best unreliable. One English manufacturer speccs a small egg-shaped subwoofer with two 8" drivers as "7.5Hz" - clearly very, very far off from reality. Another manufacturer makes a hairdryer with two 6" or close to that drivers, claiming "14Hz" - I have heard it, and there is no way to get anything useful out of it at that frequency, from what i heard, it struggled hard to do normal bass frequencies. Capacity is important to know because this tells how loud the subwoofer can play in the room. This is the number you use to determine how many units you need to achieve your desired spl at the listening position. Frequency range is the usable range - how low it can play at still somewhat useful output level, and how high up you can set the crossover. For a subwoofer, the frequency response is largely irrelevant, you only want to know the range, and as long as the subwoofer is designed for high sound quality the response will be smooth between lower and upper limit. If the curve is flat or tilted or in some other shape does not matter, because the in-room frequency response will be dominated by the room, and will need adjustments in dsp for optimum performance. ---------------------------- Since the real experts are on data-bass, this is the place to ask for opinions on this - how to specify subwoofer performance.

I see you are one step ahead, you already know it works. And this system should give nice performance with BEQ movies. If you rip the movies to HD you can continue to use JRiver for eq.

Yes, of course you need massive capacity and processing of individual channels. To explore the full potential. BUT. Quite many will claim that this subwoofer already is top-of-the-line, and most people does not have capacity and extension that this subwoofer can give. If BEQ is usesless on this system, then BEQ is really only for a very limited exclusive set of enthusiasts. Many of the moderately filtered movies can improve a lot with simple filters applied. And you don't need an extreme system to benefit from fixing the low end extension, it will sound better if there is extension down to below 20Hz and enough capacity to make it audible.

I suggest checking out what is possible with the eq in the SVS sub. If there are suitable parametric filters available, you can try it on some movies, to experience and then decide whether BEQ is for you. You need a parametric low-shelf filter, with adjustable gain an frequency and q. Try some of the moderate movies, Oblivion is a one that only need a moderate lift at the lowest frequencies. If you find that this improves the experience, then you can start researching what equipment to buy.

I don't know the Nanoavr products, but checking the mindsp web pages it seems the HD can do the job. Since it is digital in-out on hdmi, it will not have any negative impact on sound quality, part from the user setting up bad eq points. But check that you can actually use it, I believe it does not process TrueHD or DTS. We usually do BEQ by applying eq on the sound track in the movie file directly, which means it can be played back just like any other movie. But this requires equipment and knowledge do do it. If your bass-system has a dsp with parametric eq, you can use this to accomplish much - if not most - of the improvements. If the system has presets, you can program one or two presets with ulf bass boost, and select the most suitable one for the movie you want to watch. A simple approach, much more user-friendly. This will work very good on movies that requires modest bass eq, such as Oblivion, The later Star-Wars.

Don't read articles like this. The information presented is plain wrong and misleading. Some statements from the article: "music rarely has extremely deep, under-50Hz bass": Wrong. Most music has essential information below 50hz, and some music has content in the sub range below 20hz. In the 2-ch article I presented spectrograms taken from various music samples, which shows there is lots of low frequency information in various types of music. " most speakers with 5-inch (127mm) or larger woofers can muster 50Hz bass": No, a 5" driver can not even reproduce 200hz properly, if a realistic sound presentation is the goal. " Achieving the perfect blend isn't always possible -- subwoofer crossover tweaking isn't an exact science": Actually the integration part is science, and a manageable set of rules solves it. But you need the equipment and the knowledge to do it properly. If the sound does not improve after adding a subwoofer/bass-system, you did not do it right.

This is not true. Acoustics determines the ratio between early direct sound and later reflected sound. The ratio between early direct sound and early reflected energy defines clarity. This is the case both for reproduction and when speaking in a room, and there are established standards for this. Those standards makes is possible to predict intelligibility in class rooms and auditoriums, and adjust acoustics according to intended use to make the rooms perform well. In REW there is now a Clarity graph, which shows the performance of the measured system in regards of those parameters: "Clarity C50 The early to late energy ratio in dB, using sound energy in the first 50 ms as the 'early' part. C50 is most often used as an indicator of speech clarity." The soundtrack is of course a crucial part here, but acoustics determine how well this soundtrack is reproduced, and with several sounds going on simultaneously it will be more difficult to discern the different parts of the sound when there are more and louder late energy because this late energy will then mask parts of the transient sounds in the early arrival sound.

Too much information, but i appreciate your dedication to elaborate and explain your thoughts, and I did read it all. I have selected a small subset from your post to comment on, realizing we can not cover all aspects of sound reproduction in a few posts in this thread. Distortion and masking: Distortion components at higher frequencies are not masked by low frequency content, unless this content also has higher frequency harmonics sufficiently loud in level. Masking occurs around the fundamental tone, and when the difference in frequency is large enough there will be no masking at all. For low harmonics the masking is high, so that for 2. h the detection level is around 2%. For higher harmonics, or any other content at much higher frequency, the detection level approaches the audibility limit for a tone at that higher frequency, as if the low freq tone was not present at all. I know this because i have preformed a controlled experiment just for the purpose of investigating detection level and masking for harmonic distortion. Dialogue: Yes, some dialogue can be more difficult to understand, and whether you experience intelligibility to be sufficient or lacking will always be a subjective evaluation. And some systems will be better and some worse. The typical home system with a center speaker close to the floor and a table between this center and the listening position is not a good starting point. In such a system, it is likely that the frequency response is very compromised through the midrange, and there will be severe early reflections. This causes poor intelligibility. A simple measurement will reveal that freq resp is far from flat, and early decay is poor. On top of that, I suspect that many center speakers have problems with both on-axis frequency response and poor off-axis linearity. In many cases room acoustics is not suitable for sound reproduction because the room is not sufficiently damped, causing problems with overall decay and early reflection level. This is the reason for all those "can't hear the dialogue" comments. The only solution to this is to improve the sound system and room acoustics.

The response of the x-curve calibrated system may actually be a lot closer to neutral than the simple frequency response measurement indicates. The tonal balance depends on the direct sound and the decay, so both loudspeaker radiation pattern and room acoustics matter. Since the x-curve was made by comparing a typical cinema speaker with hf horn to a much closer typical "hifi"-speaker, the x-curve correction is supposed to fix exactly that. Though several later studies have shown the flaws of the x-curve calibration, so something obviously got lost somewhere in the process. The small-room-is-louder is another myth originated from making wrong assumptions on why the small room often sounds louder. Room size is not a property of loudness. If the decay is similar, the loudness will be the same. It is all about acoustics and speaker radiation pattern. Distortion is not a tonal issue, but radiation pattern and decay can make distortion more audible. I have at least two movies where I have different sound tracks available, where one sounds bad and the other is much better. Try to compare the first Gravity 5.1 release to the later atmos - the atmos sounds much better. And the mixers may very well be aware of issues with the sound, but for a number of reasons choose to not do anything about it. There was one movie with very bad dialogue, the noise gating was very obvious and caused the dialogue to sound distorted. I showed it to a professional sound engineer, what was his thoughts about this, could it be fixed. Yes, he could have fixed that, he could remove all the audible noise, to make the dialogue sound clean and nice. BUT: It would cost time and money. His suggestion as that there simply was not time available to fix it, since the plug-ins and method required to do it was no secret or unknown mystery to any sound engineer. For most people - even the sound enthusiasts - watching a movie is like climbing a mountain to ski it ONCE. Perfect conditions would be nice, but since you ski it only once, you take what is there, and make the best of it. If it isn't that good, you don't climb it once more to see if it got better. And you certainly can live with parts of the run being in bad condition, if other parts are excellent. We watch the movie ONCE, and if the sound was excellent, it is a plus, but if there was one scene where the dialogue sounded distorted and noisy, it doesn't destroy the film.

Continuing from my last post, some general thoughts. The problem with eq on playback is that there is access only to eq everything in one channel. This means that eq on the center to fix dialogue issues, such as cutting the hf, will also affect other sound effects, and that may not be desirable. Some movies sound better. But it is the odd one with the strange sound that we notice. And it gets worse with louder playback level. It is quite clear that many movies are not suitable for 0dB master for a pleasant and natural sound - especially dialogue gets way too loud. So, why not just turn it down? Turning the master down destroys dynamics and impact. At -10dB you have lost 10dB dynamics, and the experience of low frequency sound effects are compromised, tactile experience across the whole frequency range is lost. Tonal balance on dialogue is one thing. I believe distortion and noise caused by pushing the dialogue level too loud is even worse, and this is impossible to fix. You can hear this on many movies - voices are too loud, they sound hard and harsh, you can easily hear the noise when the dialogue is gated. On a decent system dialogue is easily heard and intelligible at -30dB master, on any movie. On most movies the dialogue gets louder than natural at levels beyond -10dB. If the full dynamic range was utilized, the sound would be much more pleasant and at the same time would have much more impact and realism. When the overall level is reduced, the contrast will be larger, so that transients will be perceived as more powerful, and it is not necessary to clip everything. It would sound much better.

BEQ for Valerian and the city of a thousand planets: LFE: sfm 19Hz Q=2.2 gain=+10dB LCR: sfm 22Hz Q=2.2 gain=+16dB The LCR drops off a cliff at around 45-50Hz, and trying to repair this to get it flat will only give unpredictable results, the filter suggested will only partially recover some bass below and slightly reduce the 50Hz bump. LFE turned out quite well. There is not much bass in this movie (from looking at the signals), but this filter recovers just enough to improve the experience from something that has no low end into a quite balanced, full-range sound with much more impact. The experienced difference is huge.

@SME, I tried the upper-bass/low-mid cut eq on the center now while checking the BEQ, and it worked well for this movie. BooOOomy voices are gone. I have noticed the problem before, but never thinking that this is a result of the calibration on typical monitoring systems, and that it is possible to do something about it. Could also be due to artistic choice to get fuller voices. There is no real dsp in the chain on LCR/surround in this system, but the processor allows for simple manual graphic eq. Easy to implement a crude eq on the center, to improve dialog. The problem with eq on the finished product is that everything gets the same eq, and that may not be the best solution. Doing this for the center only, fixes most of the dialog, while keeping the balance as-is on other sound effects in the other channels. Easy to see the obvious flaw - dialog in L-R and panning will be wrong. Since this will be a compromise regardless how you do it, the center only can be a simple and quick improvement on some movies.

Just checked some new movies, to verify they will play, and always curious about the sound. In one they got the low freqs right, amazing how the very last octave has so huge effect on the overall experience. But since I rarely watch movies these days, the "movie-sound" dialogue is apparent once they start talking - it booms, and the upper freq range has a very strange tonal character. So re-eq for the rest, doing not only the bass-eq, certainly makes sense. The problem is to know the eq profile. Doing this properly, for each movie, is just too much work just to watch a movie. But is it possible to do it right, so that voices have a natural tonal balance, without the excessive boom and strange nasality? Perhaps my speakers are wrong? Vocal in music does not sound like this, and a typical well-made documentary usually sounds very good. So it is definitely possible, and the problem is how the sound is made in the movies. The best solution would be if the movies were properly made - no low bass cut to adapt to bass studio and cinema speaker systems, and tonal balance that sounds natural on a reasonably flat system. The next best would be if they could provide a eq profile with the movie, so that it is possible to apply the necessary re-eq with reasonable effort.

At moderate loudness levels, capacity will not be a factor. But when you turn it up, there may be very significant differences between something that overloads and the one with headroom to spare. I don't have any cd-speakers here now, but I do have 2 systems where one has the F1 with limited hf capacity in the horn-loaded ribbons, the other has the F2 with horn-load large AMT. The AMT wins, and the difference is obvious when you have the opportunity to compare instantly - same music, same volume, very different resolution and easy in the upper octaves. Something like Jøkleba with the quite loud trumpet really brings those differences - at +6dB, the F1 struggles, while the F2 sound exactly the same regardless of volume. And there may be compression on transients long before it starts to sound really bad. I believe this affects realism. The peak transient level can be very high. I measured this once, on music, and if I remember correctly the transient level was around -6dB at high frequencies, while the rms level in the same frequency range was in the -20dB range. Radiation pattern differences are also very important. A directive horn will throw more sound energy towards the back of the room, while the typical dome will spread the sound more closer to the speaker. This causes significant differences in how the total sound appears at the listening position. In the decay plot they can look quite similar, though the horn will tend to fall off sharper in early decay, and have more late decay.

I try to tell they have to find out for themselves. Listen and experience. It all starts with curiosity and an open mind, if they don't want to learn and are not ready to accept that their current beliefs can be wrong, it is hopeless and a waste of time.

I think they certainly would hear a difference, but I also think they would fail to tell which is which. The room would also be a major factor here - the horn system would interact different with the room, creating a different sound-stage. And what is a horn speaker - could be anything from something with a hf horn and trad low frequency drivers, or a large system with horn all over. And horns can have very different radiation patterns.

Just to continue.. But horns are actually back in fashion - among some people. Not only for home theater, but for dedicated 2-channel. Those systems are typically diy, with large horns, often front-loaded bass horns with directivity control from around 100hz and up. Some of them are now trying SEOS horns. We also have commercial horn speakers, like Avantgarde. Still, it seems like there are two sorts of people - those who like horns, and those who do not.

I have never been a dealer or manufacturer of audio equipment, the professional side of this is new with the company Kvålsvoll Design. But I was one of the last people to convert from vinyl, I have had speakers of very different types, though all of them have been designed and built by me. In the mid-80ies I was part working in a local audio shop, so I have quite good experience with commercially available speakers too. But back then, it was not common to find 400 liter ported cabinets loaded with 15-inch woofers with high Bl, in any shop. This was also the time when the Apogees came, and that was something that actually did sound different and on some parameters quite an improvement. If you look in the designs section on my we page, there is a short note about the planar speakers I made late 80ies, with some pictures from a newspaper article. Those could never play loud, but they had some qualities that I suspect my current design never will be able to match. Trying to explain this (about the electronics) to the typical hard-core audiophile is pointless, you will never break through. For those who are not that much emotionally connected to the tech side there is hope, if you get them in to the room to listen. Most of this is actually quite simple. If you hear a difference in a dac, but this difference disappear if you do not know what you listen to, the only logical explanation is that this experienced difference is created in your brain, and has nothing to do with sound. For rational people capable of some very simple logical reasoning, this is possible to understand. But all electronics must be good enough. This does not necessarily cost much money, and as an electronics designer this is obvious to me, the parts to make an amplifier circuit does not cost a lot of money, and there are no mysterious phenomena unknown to science, that strangely only affect audio signals. The cheap amplifier in Room2 has more power, less noise, inaudible distortion - as long as you don't push the output stage beyond limits. Sound quality improve because the noise that was audible on the audiophile preamp is now gone, and there is more power available before clipping. This goes well only because the F2 speakers have decent efficiency, they are true 8 ohm - not "8 ohm dipping down to 3 ohm - and they are placed in a small room. But adding a couple of decent output stages does not need to cost so much either, like I had to in the Moderate Cinema, because the F1s kept on killing the Marantz unsufficiently dimensioned output stages. When we get into functionality, the new cheap amp kills the old on all aspects. I have already mentioned the dsp functionality for delay and crossover to the bass system. Then we have the built-in dac - no need for a separate box, and it has hdmi input for best possible connection to the computer. and then there is the calibrated master volume. No need for this on music, some will say. I say I love it, I always know how loud I have my volume turned up, on any system, because they are all calibrated to the same level. Not to mention when you want to measure something - you always know the volume is correct and repeatable. Then we have the speakers and the room. Not so easy. But solving and leaving those other issues that proved insignificant, at least leave all our time and resources and effort available to improve and solve what matters. And they say "high-end is dying".. Yes, I certainly hope so, to be replaced by good sound instead.