Jump to content

All Activity

This stream auto-updates     

  1. Today
  2. Maximum horn compression ratio?

    No worries! I have considered the thickness of the main panels but I haven't yet included the bracing. That will cut down the area further, of course (or expand the box). Thanks for the tip on particle velocity. Audio folklore is that velocities should be below 10 m/s, and this gives a good rationale for this horn, I think. I had been struggling to justify all this extra carpentry versus my first design, a Stereo Integrity HST18 low-tuned vented box. Why do all the work and have a larger box if I can get the same SPL at low frequencies just by getting a bigger amp and using a conventional low-Fs, large-Xmax driver? Well, if port velocity is the measure of distortion we care about, the horn wins. I modeled both at the same RMS voltage, and at the voltage where the horn throat is hitting 10 m/s at its peaks at the horn mouth, the vented box is hitting 20 m/s at port tune at the mouth and and is over 10 m/s over most of its range. If we look at SPL, the numbers are less benign. I'm not sure what a bad value threshold is here, but we do see high pressures over more of the range at the horn throat, while the vented box is bad at resonance but much lower overall. Any ideas what SPL limit the throat should be limited to? Maybe a throat adapter could mitigate this? Yes, the fold was key to the design. I did a good bit of math to get the fold sorted. The idea was to use the expanding radius of the accordion fold to create the flare, rather than tilt the wood panels at precise angles and struggle to crumple the fold into the box. This allows a very long horn path to be crammed into a not-completely-insane box in a fairly simple-to-construct way, i.e., via a sort of matroyshka-doll structure. I'm not sure if there's a penalty to pay for the fact that the fold only really flares as it rounds the corners; I was planning on a more detailed Akabak model to capture this if it works out in Hornresp. If this thing goes anywhere, I think it needs to be called the Longhorn. No connection to Texas, but it seems pretty obvious given the length of the fold.
  3. Yesterday
  4. Maximum horn compression ratio?

    I will look at this when I get a chance. Very busy at the moment getting some more tests ready to upload. Off hand. Interesting fold. Have you considered weight of the cab and internal pressures at high output? Bracing of the panels etc? Check the particle velocities, acceleration forces and pressures at the throat and exit. They will likely be extremely high with such a high compression ratio. Also absolutely model with the Le option checked which derates motor force some.
  5. Maximum horn compression ratio?

    Here's the hornresp parameters, the frequency response, and a 1/4 cutaway view of the box with the accordion-fold layout I intend to use to get enough path length to get down to 10Hz. The frequency response isn't audiophile flat, but I only intend to run this up to 60 Hz, so I should avoid most of the really choppy stuff. Below that, it doesn't strike me as unreasonable, especially with some room gain and EQ shaping. Feel free to critique! This is my first horn, so if there's any obvious blunders I'm making (besides maybe a too-small compression ratio?), I'd be glad to hear about it.
  6. Thank you for your kind words. This process has been enormously challenging and time consuming, but results (so far) are immensely rewarding. I never imagined sound of this level of quality was possible, in this space or otherwise. I remember when I first moved into this house in December 2012 and set my system in the living room for the first time. The sound was so disappointing. My original plan was to try to finish part of the basement for a dedicated room, as soon as I could afford it. However, the basement options were full of ugly compromises. In one area, I would have 14 feet of width but would have to deal with ceiling obstructions with only 6 feet overhead clearance. The other part offered a full 7 feet of headroom but only 10 feet of available width. I had to come to terms with the fact that the basement options would be suboptimal regardless and decided to try to make the best of the living room space instead. Today, I could claim to have a set up that approaches "world class" performance, all while leaving the living room largely functional, albeit with lots of weird looking panels and diffusers. Thankfully, my wife has been very accommodating. Her skepticism toward acoustic treatments melted away once she heard the difference. She also happens to be quite the bass lover, lucky me! ... Some day I need to update the first posts on this thread to describe my "current" configuration. Right before @lowerFE's visit, I migrated my speaker DSP configs to use FIR filters almost entirely. I also modified the crossover to 850 Hz LR8 (acoustic). The FIR filters are much cleaner and more precise than the mess of biquads I was having to maintain. The horn/woofer crossover is also linear phase, which I opted for not so much for sound quality improvement but to eliminate group delay that confounded my tonal balance calibration method using short FDWs. The result provided a significant improvement, albeit not as dramatic as some changes in the past. Still, it was worthwhile enough for me to demo with the FIR filters, despite the fact that the bass still needed work. So @lowerFE was able to hear the speakers sounding as good as they ever have, but the bass was not as good as I think it could have been. In fact, I ended up making substantial changes on Saturday night, between his visits. On Saturday, the main/sub XO was linear phase, and I ended up redoing everything to minimum phase XOs and less aggressive shaping to reduce pre-ringing. That was kind of a hard lesson for me, which is that pre-ringing really does bad things to bass transient response and tactile sensation. The problem was most obvious to me when listening to the Danley fireworks. I could actually perceive the pressurization before the bang happened. Even with those changes, some pre-ringing persisted and is present in my current config. I don't know how perceptually important that is though. Since @lowerFE left I've EQed down the 70-100 Hz range a bit, as it was subjectively too strong, but the bigger change was to move my bass boost from being centered at 70 Hz to being centered at 155 Hz instead. I decided to try to better mimic the floor gain from a "typical floor standing speaker". I had tried bass shelves at higher frequencies like that before, but it seemed to work a lot better this time. The extra mid bass really brought more punch and overall loudness to the table. Now I'm trying to figure out how to reduce pre-ringing further while maintaining smooth frequency response, keeping excess group delay in check, ensuring coherent summing across multiple channels, and doing all of this at every seat location. It's a remarkably complicated problem, and while I have powerful DSP to attack it with, it's not at all obvious how best to apply this capability. I also have a problem of a rattling window pane (at around 60 Hz, unfortunately), so I am trying to reduce the bass build-up in that corner to keep it from rattling as much. I intend to eventually try to optimize using an automated algorithm, but automation is useless without a precisely defined objective. And in the long run, I expect I won't be able to get the results I want with the equipment I have. I still intend to replace the MBMs I have. The open question is *where* I'm going to put the new MBMs. I can put some of them behind the sofa like the old ones. I can also put some of them on top of the subs, between the subs and left/right mains. (The "pseudo-line" approach.) And I can put some up on the shelve above the TV, adjacent to the center channel. The locations behind the sofa are starting to fall out of favor with me because it's difficult to avoid pre-ringing problems. In fact, I can't really avoid pre-ringing in the dining room and kitchen areas when using behind-the-sofa MBMs without using multiple switchable DSP configs, which I'd like to avoid. So I'm curious if I can get away with MBMs on the front stage only. I think the approach has potential, given how the center channel measures. That is something I will investigate in due time. Some time, I might start a thread about bass phase response / group delay. It seems to be a substantially neglected issue with regard to system optimization and may have a strong bearing on tactile response performance. While it seems counter-intuitive that minimum phase crossovers may (often) be superior for mains/sub crossovers, minimum phase systems actually appear to have the properties we want most. We want as much energy as possible to arrive at the start of the impulse. Too much positive excess group delay, and energy does not arrive until too late to contribute to perceived impact. (Post-temporal masking effect.) But any pre-ringing has the effect of shifting the perceptual reference point, the "start of the impulse", to a place where there's very little energy at all. (Pre-temporal masking is very weak.) So what achieves these goals? For a particular magnitude response, the minimum phase response maximizes the amount of energy in the initial impulse. I suspect that this is what's needed for the best tactile "kick".
  7. Maximum horn compression ratio?

    I have, and the numbers look good with high compression ratios for sure. That’s why I’m curious if I’m missing anything. I’m more worried about what hornresp doesn’t model than what it does. For instance, it’s not going to model driver failure! It doesn’t know anything about the strength of the cone or the surround glue or whatever. On the other hand, if it does model distortion, I’d be interested to know about it. I haven’t seen that, only frequency response, max SPL, etc. Nothing about distortion.
  8. Maximum horn compression ratio?

    Depends on the driver and the horn design. Driver diaphragm will limit the load it can handle, shape of the horn close to the driver affects how large pressure and flow creates distortion. Huge motor force is good in a horn, and - as you experience - you end up with larger ratios. The problem is not so much the high velocity at the throat, because it is large pressure gradients - pressure changing rapidly - together with large velocity that creates problems first. This causes flow separation and then turbulence, which means distortion, noise and efficiency lost. Too high pressure at the throat causes nonlinearities due to the nonlinear properties of air. Model it, simulate, and look at the numbers.
  9. Maximum horn compression ratio?

    Hi, I’m planning to build a tapped horn subwoofer around the BC iPal 18, which models very well in hornresp due to its extremely low q (0.14!) and high motor strength. There’s something magical about this driver; nothing else I’ve modeled manage to get as low with as reasonable a response curve in a not-too-absurdly-huge box. VERY low: I’m planning on scraping 10 Hz if I can! One problem I’ve run into, though, is that the horn compression ratio wants to be quite high, for both response curve smoothness and to keep the box size reasonable. Something in the range of 1:6 all the way up to 1:10 works best. Now this is far outside what is typically quoted for subwoofers, which are supposedly recommended to be held around 1:2, with 1:4 a typically quoted maximum. I’ve read two reasons for this maximum: avoiding excessive horn air velocity and thus distortion, and avoiding simply overstressing and blowing the woofer cone itself. However, I haven’t heard a good *quantitative* reason why the recommended ratios are chosen. If there are any. Since this is databass, I hoped I could get a data-driven discussion on the real limits of horn compression ratios for subwoofers. Is the old 1:2 ratio just based on wimpy older drivers, and is outmoded by the new generation of crazy motor force neodymium magnet woofers, or are there some very good reasons to avoid going too high? How high, exactly, could you go, before you run into problems? Is 1:6 ok? What about 1:10? I’m thinking of something like Ricci’s dual opposed 21 iPal build, which clearly ignored the typical rules of sealed box design and overwhelmed the limits of the tiny air volume with the iPals’ high motor force. If this can be done for sealed boxes, maybe something similar can be done to create mini-horn subs.
  10. Last week
  11. Had not got T5 yet on video but will eventually. People love to poop on Michael Bay and T5 has serious issues but in a hobby for A/V..... T5 brings it. Interested in the Atmos track and if it is (better be) a remarkable improvement from T4's barely-Atmos track. And yes... let's bring back the days when 20hz was cool cuz it still is, imho.
  12. Reading this (above) makes me happy. SME builds, measures, makes adjustments, attention to details that surely can not make any difference - but it does. Then he - SME - describes the amazing sound, well, what does he know that all the others don't.. And the proof is in hearing and experiencing yourself. As lowerFE did. When you focus on the parts that are important for sound quality, and fix it, you actually get results that matters.
  13. So after reading this thread over the past year and amazed and the technical depth and extreme attention to detail paid to the tuning of this system and going "man I really want to hear this!", I flew and went to check out this system. And boy what an amazing system to listen to! My mind was blown as I was amazed by one thing after the other. All the work put into getting the tonal balance of this speaker correct really paid off big time. The whole system just sounds really "correct", and the more I listen to it the more I'm amazed by it. I brought my Reference Mini's with me as a comparison, and there was a very obvious difference in sound quality. I thought my speakers sounded really great, but it sound noticeably "off" when compared to this system. The speakers had a fantastic amount of detail, and the transients are awesome! It felt like I'm listening to a pair of really good headphones (and few people realize how hard and impressive it is to achieve this), but I also get the enveloping sound that makes speaker listening so pleasurable. It's the best of both worlds. What's even more impressive is the bass. I don't think I've heard bass so tight and full sounding in a room, which is clearly due to the complex integration efforts of multiple subs and individual EQ's to get such flat bass over a large number of seats. The clarity and tightness is seriously impressive. Again, just like a headphone, and that is actually something I've never heard before from a subwoofer. It is straight up the best sounding bass I've heard in a room. Now when you also get the whole body physical sensation from bass, addictive is an understatement. One thing that is unforgettable and blew my mind is how great the speakers sound in the kitchen! I don't think SME has ever mentioned this, but it was indeed one of his goals. It was remarkable hearing a correct tonal balance with almost no treble roll off in a different room! I still can't believe this is achievable. It must be the combination of controlled directivity speakers and properly placed diffusers pulled this amazing magic trick of a feat. I've heard a lot of amazing home theaters, but this is the first time I heard imaging from surrounds. It was trippy to be able to pinpoint the location of the sound going across the rear stage. I really wish we watched an action movie and be able to so accurately track the position of the sound effects. This is even more impressive as I seem to clearly have less ability to hear imaging compared to other people. Speaking about imaging, the speakers reproduced phase manipulated music tracks far more accurately than anything I've heard so far. It must be the room treatments that are preserving the phase accuracy of the speakers. It was like "oh this is where it is supposed to sound!" I was also exposed to the dark secrets of the time domain in room correction. That was a revelation to me to be exposed to so much more information and tools to analyze room acoustics. Now it makes sense why and how the room is mucking up the sound. It's all in the time domain! Now I am able to correlate measurements and subjective judgment of how good (or bad) the room sounds. I have so much to dig and play around with now. Measurements really can tell you about how good something sounds if you look at the right things and how to interpret it properly. Thank you SME and his wife for being such amazingly gracious hosts. That was one hell of a weekend! Oh, and did I make it clear enough that your system sounds good?
  14. Replacement AVR / processor

    @SME , this discourse kind of crossed the OT border somewhere, and should probably be moved to another thread.. I was going to say you are wrong, but I actually have to say I mostly agree. When I got back into audio some years ago, I found that people could not set up the system properly to get a good integration, and this severely compromises sound quality, because the timing in the most important frequency range goes bad. But it is not extremely difficult to get it a reasonably good result, at least so it sounds better than main speakers alone. But it requires much more than the casual buyer can do, because you need to measure. When you have the measurement capability, it is possible to achieve predictable results using a manageable set of rules. Especially for 2-ch, the systems often end up with a very low crossover, because that sounds better. And does sound better, if you are not able to set the delay on the mains properly.
  15. Replacement AVR / processor

    @Kvalsvoll, I was talking about 2 channel systems with full-range speakers without subwoofers. They don't generally need delay because the low frequency drivers are co-located with the rest of the drivers. This includes speakers using either passive crossovers to the LF section or active crossovers (with built-in amplification). Some speaker / amp combos can extend as low as most subs go. Either way, the integration is relatively trivial. It could be said that using subwoofers, in separate room locations, solves one problem but creates another. It (partly) solves the problem of poor in-room response in the sub frequencies at the locations that are otherwise ideal for the speaker. It creates the new problem of integrating the speaker and sub response, which typically affects a region of frequencies that is crucial for reproduction of bass in music, around 60-120 Hz. I'd argue that, for the vast majority of music, accurate reproduction of those frequencies is far more important than the extra octave or two of extension that people using subs chase after. Nevertheless, I'd bet that the vast majority of systems using subs have serious frequency response problems in that range because this integration is not at all trivial. I'm not saying that using separate subs is inferior to using standalone speakers. A system that uses separate subs will absolutely out-perform a system using standalone speakers, *if* they they are configured optimally. Rather, achieving the optimal configuration for a system with subs is not at all trivial. It's hard enough to do with a single sub, and enormously more complicated with multiples. Most consumers don't have the knowledge or equipment to achieve even half-decent results, and even the more advanced consumers struggle to get "good" results. Count me among them. I have more DSP capability than just about anyone on these forums along with subwoofers-only response that looks almost "picture perfect" , yet I'm still trying to find the best strategy for integrating my subs + MBMs with my mains.
  16. Replacement AVR / processor

    @SME, the typical 2-ch system actually lack basic and necessary functionality, such as delay on main speakers. This is the case for most 2-ch preamplifiers, including those with dedicated subwoofer outputs, even the digital ones. The most sophisticated may have some sort of low pass filter on the sub output, usually fixed slope and cut off frequency. And if the system has a AV-processor/AVR, they usually bypass all processing for 2-ch listening, in "direct" or "pure" mode. This efficiently disables all calibration settings for bass system integration - no delay, no filter on the mains. Alternatively, a dedicated 2-ch preamp is installed - all necessary functionality is lost. So, it is no wonder the typical 2-ch system sounds better with subwoofers disabled. Even if they know how to set this up properly, it is not possible because the playback chain lacks necessary functionality. Of relevance for this thread: Note that the amplifier test was done using a quite ordinary AVR as processor/DAC for playback, and no one has been able to detect any audible difference from the original to the sample that was passed through the playback-recorder loop 4x times. A reasonably good AVR, used correctly, does not have any "sound" at all, it is completely transparent. All this is caused by bad advice given from manufacturers and dealers who want to sell more equipment and does not understand how to set up a sound system properly.
  17. The Low Frequency Content Thread (films, games, music, etc)

    Is no one going to talk about the audio in Transformers "The Last Knight" whenever Megatron is on camera? The audio has been lightly brushed on here at Data-Bass but I don't recall anyone going into detail. I SpecLab'd a few scenes last night and when Megatron lands in the salt flats the hottest spot is centered at 20 Hz. Sure it's no 7 Hz WOTW, but the sound in the movie makes it worth watching and also using for demo's. I haven't seen much talk about it so I'll go out on a limb and say that I love the proper audio in this movie.
  18. Replacement AVR / processor

    I will never need to turn my MV up that high, I have downstream processing ( DCX 2496's ) before pro amps. I don't have horn loaded subwoofers at the moment, if I did, that may be a consideration, but it also depends on the latency of your display device when watching a movie, when you start adding tens of feet or more delay to all other channels to match up to a horn subwoofer. Thanks for the insight.
  19. Interested in doing this mod to my inuke6000dsp. Has the Op’s Amp remained reliable since the mod?
  20. Replacement AVR / processor

    Yes, I did generalize for a tapped horn, and now that I think about it, I might not be correct even for that case. Several 10s of milliseconds sounds very high for "room acoustics" effects. A full cycle at 60 Hz is 17 ms. If you are delaying more than that (in addition to distance and "internal" effects), then you are probably adding unnecessary group delay, which likely impacts transient response sound quality and slam. FWIW, I've been studying this problem quite intently lately, trying to improve integration between my speakers and subs. Unlike most people, I have practically unlimited DSP resources to throw at the problem, where the only real practical limit is latency. I would say that the processing capabilities built into AVRs and most processors are woefully inadequate for achieving an optimal outcome. The"THX "LR4 sub/sat crossover" is largely fantasy that rarely occurs in real world conditions. The best that most people can do is a brute force evaluation of different XO frequencies and sub delays, where typically response is only optimized on one channel and at one seat. Yet even this effort requires more sophistication than most users are capable of. (Readers of DataBass and some of those who read AVSForum are obvious exceptions.) No wonder a lot of people prefer bass from 2 channel full-range speakers vs. subs. While the in-room "placement" of the LF drivers in such speakers is non-optimal, the XO is (ideally) optimal for that placement. I've noticed that good anechoic-flat full-range speakers, when pulled far enough from walls, can deliver impressive slam; whereas many sub systems including many with big horns or many large drivers struggle in this respect. My recent experience suggests that phase (or rather group delay) effects are more important than most people realize. And it's not what people think. I.e., a ported sub isn't necessarily sloppier than a sealed sub, though that obviously depends on the competence of design. Such effects are largely minimum phase. (A good thing.) Rather, it is the excess group delay, which arises from crossovers and distance differences that appears to be important. Pre-ringing in particular seems to really kill tactile slam, and it should be noted that FIR filters are not the only way to introduce pre-ringing into a system. Pre-ringing can arise merely from placements and/or delay settings. Any situation in which sound from a sub may reach the listener before sound from a speaker potentially involves pre-ringing. Rooms with rear subs are likely to exhibit pre-ringing for rows behind the one used for calibration. What's not at all clear is where the perceptual thresholds lie for hearing and feeling of pre-ringing effects. Anyway, I still have a lot of work to do here, and at some point, I may try to do some more formal testing of excess group delay effects, including pre-ringing, as this information would be very useful for optimizing sub systems for multi-listener environments.
  21. Replacement AVR / processor

    No, it is not that simple. The delay of the horn itself depends on the tuning and type of horn. Front loaded, back loaded, size of compression chamber, tuning. And delay is frequency dependent, though less so for a horn than a typical ported box. Then room acoustics come to play, as for any type of subwoofer, adding delay that is also frequency dependent , and can be several 10's of milliseconds. So the actual delay you need on the L R depends on the crossover frequency, the horns in use, and room acoustics and placement. (And this is before entering the more advanced configurations, which can include multiple subwoofers set up to solve problems that typically occurs in small, closed spaces).
  22. Replacement AVR / processor

    I keep my L/C/R trims at -9 on the Marantz and the SW out at -12. Since i keep my amp gains at max for the subs, the input on the MiniDSP for the sub channel is at -25, so anytime I want to bump the bass trim I adjust it there. I also don't use Audyssey.
  23. Replacement AVR / processor

    I do, but I don't count. IIRC, my channel trims are +3 for the mains, which allows me to run at up to +5 MV without clipping for those odd tracks that need higher than theatrical reference level playback. This minimizes between it and my DSP. I still have the sub out set to -12, even though I technically could do -7 while retaining the same headroom, and that's because my bass management is done downstream in my DSP so the "sub out" only has to pass LFE. If I need more gain in the system, I can boost directly in my DSP, which can easily clip any of my amps. That what I usually do for those occasions in which I want to "boost the sub". That's pretty rare for me as well. I demoed some dub step at ref level with a +/- 6 dB tilt for LowerFE when he was visiting. It was funny because at first he was surprised to see my cones moving with music that was "not so loud". Then he loaded up the SPL meter in REW on his laptop, which showed continuous output in the one-teens with occasional peaks clipping the UMIK (> 120 dB). I gotcha. I feel like I've asked this question before. (head smack) So the delay is 1/4 length at tune? So 12.5 ms for a 20 Hz horn? I can see where that'd be a problem.
  24. Replacement AVR / processor

    You have to add the horn path length to the distance of the sub from the main listening position in addition to other delay caused by the LPF or DSP.
  25. Replacement AVR / processor

    No way. My house would explode. Lol, not really but...
  26. Replacement AVR / processor

    Thanks for the tip, I assumed they were all limited to around 20ms. This can be a problem with horns, I have often reached this delay limit when experimenting with different calibrations. I have only measured Denon/Marantz units, sub output is fine for -12dB trim and 0dB master, anything higher and it will clip.
  27. Replacement AVR / processor

    I'm still using the old Onkyo 886 myself. Probably get the Denon DN-700AV when I do change unless something better comes along. Related to the clipping with WCS signals... Most of us with big, capable systems, by default have HE / sensitivity. I've done a lot of gain investigation with my current components. My REF playback is at -7 on the master with my FL and FR channels at -15 channel trim. This is largely set by the high sensitivity of the CF-4 speakers and the extra hot input sensitivity of the Emotiva XPA-5 amp. The sub channel runs about 3dB hot at -7 trim with my K20 input sensitivities at minimum. The Onkyo will spit out over 12v through the SW jack without clipping but I do not know what point it would occur internally with a WCS. Back when I did all of that testing I did not have an easily accessed WCS signal. K20 clipping occurs well below 12 volts input. I could always increase the input gain on the amps if needed but I don't run REF +15 on the subs virtually ever. Is anybody actually running their processors at (0) for assumed REF with the channel trims near 0 or higher? You really shouldn't need to with high capability speakers but this makes me curious.
  1. Load more activity