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Kvalsvoll last won the day on December 8 2017

Kvalsvoll had the most liked content!

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  1. X-curve compensation re-EQ

    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.
  2. X-curve compensation re-EQ

    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.
  3. The Bass EQ for Movies Thread

    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.
  4. X-curve compensation re-EQ

    @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.
  5. X-curve compensation re-EQ

    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.
  6. Bulding the Room2 listening room

    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.
  7. Bulding the Room2 listening room

    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.
  8. Bulding the Room2 listening room

    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.
  9. Bulding the Room2 listening room

    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.
  10. Bulding the Room2 listening room

    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.
  11. Bulding the Room2 listening room

    This is how it looks up front in Room2 now. Had the photographer visiting some days ago, to take photos of the new subwoofers, and we snapped this picture. It is interesting to notice what is in the media console - rather, what is not. There are only two things left - an amplifier, and the retired cd-player. The cd-player is connected, but never used. This is what has been removed: - Preamplifier - Exotic class-A output stage (25W, lots of heat, no global feedback yet still distortion below noise floor, exotic output devices, single stage input and voltage gain). - A NAS with separate dedicated wireless router. - SOTA CD-player can also be considered removed, as it is never used. This is a picture of todays dilemma in state-of-the-art audio. The very nice, expensive items of yesterday is rendered obsolete. The new configuration destroys the traditional old-style hifi-setup on all parameters except for two - price and looks. It is not very expensive, and it does not look very expensive and sophisticated. The sound is much better, convenience of operation is much better, it takes up a lot less space, it has functionality such as dsp in the amplifier that makes it possible to integrate the bass-system properly. The current system consist of a laptop computer connected to the local network streaming from the server or the net, a cheap amplifier with dsp, dsp amplifiers for the bass system. This system is guaranteed sonically transparent up to the terminals on the main speakers.
  12. Bass system integration

    Yes, timing is the key. Good idea to use lower cut-off for rear subs in a multisub system, it is not easy to achieve a very good result with sub units both up front and back.
  13. Bass system integration

    Integration basics: We start with the simple basics that everyone on this forum already knows. And then we will see that what should be quite simple and straight forward, just is not so, at least not always. Assuming the subwoofers are in place, and any eq and other settings on the bass system is completed. If it isn't perfect - well, it never is - then that's what we have. The end goal is to end up with a system with a smooth frequency response matching our chosen target and very good transient behavior. Transients means timing - we need to get the time domain right, it is not enough to just eq or whatever to get a perfect frequency response. To achieve this, we can adjust the level on the bass system to match the mains, we set delay on the mains so that the bass system and main speakers sum up correctly in the time domain, and we set a low pass filter on the bass and high pass filter on the mains - this is the crossover. The typical processor allows tuning of these parameters. Level: This is where the typical buyer fails first - level is set too high, often 20dB higher on the bass system, because the subwoofer was expensive and must be heard. The frequency response measurement effectively reveals this problem, simply adjust the level so that the frequency response matches your target as close as possible. Now if both mains and bass system are smooth/flat, this is quite easy, but can be more tricky to find the best compromise if the responses are very nonlinear. Level can usually also be adjusted on the bass system amplifier, so you don't need a typical processor for this. Delay on mains: This is the subwoofer distance setting. This should not be set to the distance to the subwoofers, it must be set to match the total delay in the bass system, which includes dsp, additional filtering, acoustic delay in subwoofers, acoustic delay due to room interaction. It will usually be a lot more than the physical distance. One method to get it right is to look at the frequency response around the crossover - the correct setting is when the bass system and mains sum up to a reasonably flat level with no dips. The problem is that there are an endless number of distances that will look right in the crossover region, because if the delay is exactly one period wrong, it will still sum up. Fortunately the frequencies involved are quite low, so the delay for one period at crossover is long, and we can see this on the time step response or in the spectrogram. Tune it approximately first, using spectrogram and time step, then fine tune using frequency response. Have an expensive audiophile preamp with no delay for mains? Well, why do you have that, sell it and get gear that is possible to set up properly to get good sound. Crossover: Choice of crossover frequency depends on main speaker response and capacity, bass system response and capabilites in the upper bass range, location of subwoofer units. Generally, a higher cf gives better results, 120hz is often a good starting point to try. If your bass system has asymmetrically located units, a lower cf is needed to avoid localization, try 80hz, in bad situations as low as 60hz can be necessary. It is also possible to run the mains with no high pass, running full range. If the main speakers have extension and capacity at low frequencies, they will now act as part of the bass system, hopefully resulting in a smoother total response. All this is quite easy to do right if the bass system and mains have flat and perfect responses. In real world scenarios, this is not the case, and decisions has to be made to choose the best compromises. It is quite clear that the set up an calibration of the bass system before integration is also a part of the integration process, and then this is no longer easy to do just by following a few rules.
  14. Bass system integration

    How to integrate the bass-system/subwoofers with the main speakers, so that the end result actually sounds better. This involves setting crossover frequency and delay and level, so that the bass system and main speakers integrate seamlessly. Get those 3 steps right, and the result will likely have significantly improved sound quality compared to switching off the subwoofers. Additional steps includes dsp processing - like eq - on bass system, dsp/eq on main speakers, placement of main speakers and bass system, room acoustics, placement of listening position. A complex process with so many variables. Kind of nirvana for the hobby enthusiast looking for problems to spend time on, a real pain for someone who just bought this awesome bass system and now want to hear some music and watch a movie. Integrating the bass system is like engineering a speaker system in your room, which requires knowledge in electroacoustics and equipment to do acoustic measurements. So , what do you do if you have neither? You use a set of rules to make the necessary settings. Perhaps helped by automated set-up functionality, such as audyssey. The result is what one can classify as "variable success" - in some cases, it ends up with something that sounds reasonably good, in other cases it is so flawed that the sound is better when subwoofers are disabled. Even for experts, a set of rules makes it possible to do a system in a reasonable time frame, and also give more predictable results. So, which set of rules applies. How do we set this up, simple and easy, with predictable results. Share your way of doing the integration, ask questions. Let us kill the myth that subwoofers must be switched off for music. This thread hopefully will provide useful information on how to integrate a bass system, and when we enter the more advanced parts we may be able to further improve the methods we use.
  15. JTR Noesis 215RT Measurements Discussion

    Jeff is right, if you design a full-range speaker with anechoic response flat down to 20hz and below, you will experience a huge low frequency boost. All commercial speakers for home use are designed with some roll-off in the low end, and they all end up with a more or less balanced level down to where the speaker finally rolls off. Most full-range systems are set up with a separate bass system / subwoofers, and the bass system is level matched and usually equalized. What you see in the measurements when adjusting the eq is of course the in-room response. Now one can argue that the radiated sound energy falls at lower frequencies, but if you eq for flat energy level - or have a flat speaker - the sound pressure will be too loud at low frequencies. Very, very few speakers are true full-range. This one is the rare exception. Some very small speakers with dsp claim to have response down to 20hz or 14hz or whatever, but that is pure nonsense, because they do not have the output capacity required for those low frequencies to even be noticed. But this speaker has, confirmed by the measurements.