Chris A

I'm glad you're pleased. Chris

Here's the problem with Linux: This is a big problem as a multi-tasking OS, especially a multi-user OS. I see they still haven't done any OS development to deal with these issues. Real OSes in my terminology are either 1) RTOS's or, 2) bullet-proof multitasking/multi-user with memory protection. Linux is neither. It's still just a "me, too" PC operating system that allows the user full control of the OS and hardware support (and YOU must support it--in everything it does--even going as far as compiling and linking it). This isn't good computer science--in a nutshell. The command line syntax is basically Unix syntax (with very few differences, the last time I looked--25 years ago). I don't know what they're using for a windowing system nowadays, but there was a big push in the mid-90s to use X-Windows, which was okay. I'm sure that it's changed since then.

Reading through that BruteFIR documentation takes me back to the days of spending a great deal of time on Sun Workstations in the mid-late 1990s. I know the benefits of Linux in terms of preserving full CPU/memory horsepower of PC hardware, but you have to admit--this is a pretty deep dive for anyone but those that are very familiar with Linux and PC hardware nowadays, with significant computer engineering background, i.e., it's not anything close to "plug 'n play". The last time I was into Linux was mid-1990s...when the OS was still in its infancy. But it looks like it can fully support a 5.1 array, depending on length of the FFTs and the number of channels. Chris

Perhaps there needs to be more discussion on that subject for hi-fi audio applications, which I haven't really seen much of in the context of hi-fi audio...speaking frankly. I actually see a lot of demand materializing for flat-phase bass in loudspeakers, but no real way to get there without using thousands of taps at the resolution needed for hi-fi audio (roughly the 15 Hz-->20 kHz pass band). Chris

This is much more affordable: https://wiki.jriver.com/index.php/Convolution, especially if you already own a capable PC or laptop. This approach overlays a standard non-FIR DSP crossover with an upstream PC (as I mentioned above). Another thread that explores this subject using a miniDSP OpenDRC-DA8: https://www.audiosciencereview.com/forum/index.php?threads/multi-channel-multi-amplifier-audio-system-using-software-crossover-and-multichannel-dac.12489/page-8 It appears that the OpenDSC-DA8 doesn't have nearly enough available taps per channel. That's apparently not an issue with JRiver. Chris

Very likely. But it will probably only do this at woofer frequencies using R-51M bookshelf loudspeakers. If you buy yourself a UMIK-1 and plug it into your USB bus (after loading the shareware Room EQ Wizard), then you can see the effects of SPL on harmonic distortion and on power compression by taking in-room measurements with the microphone at 1 metre from the front of each loudspeaker under measurement (measurements taken one loudspeaker at a time). Once you plot the SPL at a lower SPL, you can measure at 0 dBFS, then move the trace downwards (using the "Controls" menu on the plot, and "trace arithmetic") to subtract the relative SPL measurement difference to match the lower SPL trace to directly read the degree of power compression that you're getting. Chris

Analog delay lines are not very "hi-fi", the last time I looked. I was aware of this, but I think my comment still stands: FIR filtering really only exists in the digital world. There are a lot of comments almost everywhere about "measures great but the results suck the life out of the music", so I think a greater knowledge of the way that the human hearing system works is probably warranted in this "new" area. I would think this should capture a lot of attention by our more serious audio enthusiasts that are just now starting out their audio journeys... SPL response flatness: I've found that a sort of "economy of filtering" works, but one with an exception: flatter SPL response. I've found that the flatter the SPL response, the better the sound. When combined with somewhat flattened phase response, and most importantly--control of early reflections in-room from the loudspeakers to the listener's ears via loudspeaker directivity and proper placement of nearfield midrange absorption--the results are startling. I'm also talking about directivity control below 500 Hz--and that really implies horn-loaded bass with directivity control down to at least the room's transition frequency (i.e., 100-200 Hz for typical home-sized listening rooms). Phase response: I've found that swings of ±90 degrees of phase vs. frequency...are not really very audible. In other words, it may be that ±90 degrees might be "good enough". This is a subject that I believe warrants further attention, because the tradeoff is time delay and added computational requirements. This also corresponds to Danley's comments about getting acoustic drivers to within a quarter wavelength at the crossover frequency. This is the principle of the multiple-entry horn (MEH), and one that I find works for subwoofer placement, too (generally at a frequency below the room's Schroeder or transition frequency). So some present "rules of thumb" that I think apply in this subject area, and other principles are just now coming into better focus. Applying IIR filters to do the heavy lifting first using the PEQ filters and time delays in typical DSP crossovers that might not have FIR filtering capabilities--or sufficient FIR filtering capacity in terms of the number of taps per channel, then applying an upstream secondary FIR filtering capability (perhaps using JRiver, etc. and rePhase) seems to be the current trend. YMMV. I do know one thing: there still is an advantage that horn-loading seems to bring to the table, and that turns into the listener being able to hear the differences between flat SPL and phase response without having to put their loudspeakers out in the middle of their listening room and thereby take a severe hit on the resulting increased bass modulation distortion that comes with this practice. You really can have the best of both worlds if using fully horn-loaded loudspeakers.: using the room corners for much better bass, and flattened SPL/phase using DSP crossovers and upstream FIR filtering to flatten phase response further. Chris

I pretty much always take my measurements at 1 m--all of them. Sometimes I'll move the microphone back to the listening position for subwoofer EQing, but that's the only time I'll usually do that. I dial-in the subwoofer-->bass bin time delay/polarity using 1m microphone spacing, since the subs are directly behind the Jubs. There are good reasons for this. In general, don't take measurement from the listening position--because you can't isolate the non-minimum phase reflections if you do. This is actually the Achilles heel of "room correction software" why so many people have so much trouble with using those applications. I think you're getting "lost in the metaphor" here. I posted the first REW file that I could find that was calibrated at higher SPL on-axis. The measurement is a pretty old one that I generally don't refer to nowadays. For that measurement, I'm not sure that I had enough absorption on the floor to intercept the floor bounce, and I might have been dealing with connectivity issues within REW and the PC that created harmonics and other overload conditions during the sweep, etc. So again, I wouldn't read much into anything that I posted on the Jubilee at 100 dB. If you want that info, then I recommend starting a thread on that subject (i.e., not this one) and ask the question there. The tweeter won't see that power, however, but the woofer does, and the passive crossover network also sees that power. It's like driving your car by pressing down on the accelerator, then using the brake to modify the speed. That's not a very good way to do things--but most "audiophiles" think there is no other way to do it. That's the greater part of the downside of passive crossovers. They usually eat a lot of power--or they store it and try to give it back to the amplifier output at the wrong times (reactance). That's why direct coupled drivers/tri-amping or bi-amping using DSP crossovers is a far superior way to drive loudspeakers--and the ear usually concurs... Chris

Welcome to the forum.

By the way, using balanced connections (XLR, Euro/Phoenix, etc.) between the preamp and DSP crossover, and then to the amplifiers, is a pretty big deal. The Jubilee is something like 105 dB/2.83 v loudspeaker (for real), so you're going to hear the noise floors of unbalanced (RCA) connections (common mode...a.k.a., power supply noise). That's one reason why I recommend used Crown D-75A and D-45 amplifiers for those that wish to control costs initially (...or long term...) because of their balanced XLR inputs. Same thing for a preamp--you might look around for a balanced output connection preamp if you have any noise issues with the Jubilees after dialing them in. Of course, if you use S/PDIF or perhaps AES/EBU connections, then the noise issue from the preamp is completely solved. Chris

Not in my experience. It is the most transparent process that I've heard. Of course, if you're really concerned about this you could go to the Xilica XD series. (I'm not--I've been running various DSP crossovers for 13+ years, and even have used digital-input DSP crossovers, and I can tell you that the I can't tell the difference between analog input and digital input DSP crossovers.) I'm sure that there are those here that say they can--but I'd immediately challenge them to a blind A-B test to see if they can really hear the difference. I find that the noise floor of the DSP crossover, combined with the word length/internal data rate (24 bits/96 kHz) is much more important, in my experience. (Note that I'm not talking about the el cheapo DSP crossovers from Behringer or dbx DriveRack (the "PA"), etc., or perhaps the miniDSP models--which have a higher noise floor.) Noise floor and digital resolution (i.e., 96/24) are the discriminators--in my experience. I'd stick to Xilica if going down the Jubilee path--and avoid the higher noise miniDSPs. Besides the loudspeakers themselves, the DSP crossover is perhaps the second most important piece to pay attention to. If you're going to skimp on something up-front, I'd do it with the preamp or the other upstream electronics--not the DSP crossover. But once you get to the quality level of a Xilica, you really need not spend any more money on DSP crossovers. You could go to the Xilica XD series, but then you'd have the issue of S/PDIF to AES/EBU conversion--and very little in terms of sound quality improvement that you can actually hear. I would recommend a good quality class AB or class A amplifier--and not a class D, unless you can do an A-B comparison with a good class AB or A amplifier to verify its sound quality. For instance, I don't personally recommend Hypex FusionAmps with the Jubilees (or any other Hypex class D models, for that matter). You're likely going to hear issues with dialed-in Jubilees and class D amplifiers (...and I'm actually very sorry to have to report this). In my conversations with Nelson Pass over this issue (i.e., class D), the problem appears to be centered on the amount of negative feedback that the class D amplifiers use. Chris

Usually...i.e., your present integrated amplifier would probably do the job quite well, saving a lot of money to put on loudspeakers instead. The better the loudspeakers, the better the sound. That rule of thumb doesn't usually extend to amplifiers by nearly the same scale of improvement. I'd go as far as to say that Cornwalls would sound even better (even in the size room you have)--and sometimes these are available for less than $1k USD a pair--used. A used pair of Forte I, II, or III, or Chorus I or II would also do very well--if you can find a good used pair for a reasonable price under $1K. The issue with Heresies is not so much the sound quality but the problem of putting them on the floor leaning backward (as Paul Klipsch recommended). The short height of the Heresy is problematic for many people trying to integrate them into the room without putting them on stands that deprive the woofers of room boundary loading (i.e., close proximity to the walls and/or room corners). The issue with RP-8000Fs is the crossover frequency (i.e., 1750 Hz), which means the woofers are direct-radiating from the low frequency cutoff (32 Hz) to 1750 Hz. This is the issue with the RP-8000F and all the other "tower home theater" two-way loudspeakers from Klipsch (with the exception of the Palladium series). The horn-loaded midranges of the Heresy, Cornwall, Forte and Chorus is the difference in sound quality--i.e., having controlled directivity down to 400-700 Hz is a much better proposition from a room placement perspective. You get to place the loudspeaker closer to the walls to pick up room gain when using horn-loaded midrange loudspeaker models. JMTC. Chris

This CD just came in yesterday (I wasn't familiar with the musical artist). After running "Clip Fix" and "Normalize..." within Audacity (a listing of the Audacity macro or "chain" that I used is shown just below)... ...I found that the album has about 5.5 dB of hard clipping (i.e., the album's "ReplayGain" went from -5.4 dB to +0.1 dB after declipping). The crest factor--that is, the album's "DR Database" rating--went from "10" to "15". That's a 5 dB increase in crest factor--which is a lot of improvement. Listening to the album after this fix, a lot of the artificial splattering of the transients was mostly fixed. It's a fairly pleasant CD with only a little vocal sibilance that's left on hard consonants. If anyone already owns this album and wants to hear the de-clipped version (in FLAC lossless format), you can PM me with your personal email address, and I can access you to the de-clipped version...or you can de-clip the tracks yourself using Audacity (freeware) using the little macro above. Note that the tracks will, on average, be 5.5 dB quieter (which is a welcome change in my estimation). Chris

In general, this isn't a single number, but a curve of THD vs. frequency...like the following plot: The harmonic content of the THD figure is likely dominated by second harmonic distortion (as it almost always is, which itself is almost inaudible as harmonic distortion only, but not as an indicator of modulation distortion). If the R-51M sensitivity is actually about 90 dB @ 2.83v (Klipsch stated sensitivity ratings are consistently between 3 and 6 dB overstated). Then at 12 volts input --this corresponds to about 96 dB on axis at one metre. My guess is that you're looking at between 1% and 10% THD at that voltage level, but I don't have an R-51M to test. I'd use the figure of 10% THD at 2.83v as a general rule of thumb. Almost all of the distortion will be from the small woofer in the R-51M. Chris

I don't think that any direct radiating subs will do the job (unless you have a very small listening room)...but that's me. I think you'll eventually end up looking again for something better than what you've been looking at. And I think you'll eventually kill the two subwoofers you mentioned, because they don't have enough efficiency to put out really low frequency acoustic energy below 30 Hz continuously (like your Khorns do--down to ~32 Hz in good room corners). I use two DIY TH-SPUDs (one in each corner behind each Jubilee bass bin), but most any horn-loaded sub with bass extension down to below 20 Hz will do. In my experience, direct radiating subs produce too much harmonic distortion and phase/group delay growth down low at the needed SPL that robs the performance of clean output and of good audible bass response. Richard (Coytee) owns a Danley DTS-10, one of which which would do nicely. It's big, but the shape/form factor allows you to integrate it into the room. Klipsch does produce a big horn-loaded sub (the KPT-1802-HLS), but it is so large that it would dominate everything else in the room and you'll likely have trouble getting it through any doorway. So Klipsch doesn't produce a reasonable-sized horn-loaded sub for home hi-fi duty (in my experience) and the 1802 is tuned a little too high for home theater duty (in my experience) so it has to be EQed quite heavily to reject its 40+ Hz response that it really wants to put out. There are a lot of DIY horn-loaded subs out there--notably those by Bill Fitzmaurice (you must pay for the plans, and there are builders that will build them for you if you negotiate a price--but these are not like tapped horn subs in terms of their form factor). I think there are other horn-loaded subwoofers that do just as well or better. I'd ask the members here about horn-loaded subs that are good to below 20 Hz. I also recommend crossing over from the Khorns to the sub no higher than 40 Hz. For the La Scalas, you'll have to cross at 60 Hz or above. JMTC. Chris

Sorry I missed this thread, Ron. I think that the video is sort of a "me too (but I'm not going to tell you that the technology has been around for quite a while)". I. What good is "FIR" filtering over what I currently have? FIR filtering is a way to correct not only the SPL response (a.k.a., "frequency response") of your loudspeakers in-room, but also their phase response. II. "So What?" Well, a lot...actually. Experiences using "linear phase loudspeakers" have been documented pretty widely since the early 2010s in the home hi-fi marketplace. Here is a pretty good article: http://www.linkwitzlab.com/Attributes_Of_Linear_Phase_Loudspeakers.pdf III. So What is the Difference in Flat or Linear Phase Response from What I've Got Now?" Tighter and deeper perception of bass "Phase equalization of the bass...subjectively extends the effective bass response by the order of half an octave... Wider and deeper sound stage (quite dramatic, in fact) "Without [flat phase response], the sound [is] flat and almost lifeless in comparison." Greater realism "...the initial [sound] transient and [its] relaxation time are critical for realism. Anything in a sound reproduction system which corrupts the initial transient is detrimental [to the perception of realism]." Apparent soundstage depth "This may surprise some listeners when they first hear it, since many speakers (and records) elicit only a general left-to-right spread. But "stereo", as originally conceived, implied a three-dimensional sound in which voices or instruments could be localized at different apparent distances from the listener as well as at various lateral positions. Listeners to time-aligned speakers consistently report hearing a stereo image with unusual depth." Greater Resolution "The stereo image is reproduced precisely, each voice or instrument having its proper place and width. In complex sound sources such as symphony orchestra, individual instruments can be resolved with unexpected clarity. In the old cliche, "I hear details I never knew were in the recording. " Some listeners have incorrectly attributed the improved resolution of detail to more accurate transient response, but the better definition of details is simply the result of the reduction of blending in the stereo image." Separation of ambience "With loudspeakers whose stereo image is slightly blended because of time-smear, any hall ambience or reverberation in the recording tends to become slightly mixed with the instrumental sounds, causing coloration of those sounds. Consequently, with such speakers closely microphoned recordings tend to sound better because of their distinctly defined sound. But with time-corrected loudspeakers, the ambience is resolved as a separate sound, and larger amounts of hall ambience in recordings can be enjoyed...” IV. So why haven't more home hi-fi loudspeakers incorporated FIR filtering? This is perhaps the most interesting part of this subject. Basically, I'd characterize it as anti-digital bias by what I term "mossback audiophiles"...because FIR filtering is a type of digital filtering. It really doesn't exist in the "analog" world. It's the mossback audiophiles that grew up in the 1950s-1970s that have rejected DSP loudspeaker correction...but curiously, get all of their new music digitally (whether they realize it or not). Chris

If your DAC has a local buffer on the USB input channel, then I'm not sure that your remaining issue is the USB connection, but perhaps the analog transfer function out of the DAC. There are USB to AES3 and I2S converters that have their own clocks internally and use those clocks to get rid of USB bus jitter. I'd look for something in that area (if it were me). I know you seem really invested in the USB bus only, JC, but the other approaches will provide equal or better results. About two years ago, I was looking for a USB (from a laptop) to AES/EBU converter with an internal clock. I found some, but beware: the device drivers to recognize these converters on a USB bus weren't up to snuff (mainland China products). Chris http://archimago.blogspot.com/2018/08/demo-musings-lets-listen-to-some-jitter.html http://nwavguy.blogspot.com/2011/02/jitter-does-it-matter.html https://www.stereophile.com/content/2020-jitter-measurements

This subject comes up often. The last time I remember was October 2020. Here was my response at that time: The above tracks that you see listed are "demastered", i.e., most of the creative "mastering EQ" and the remaining specular noise (50/100/150 Hz for European recordings, 60/120/180 Hz for North American-produced tracks), and plenty of instances where HVAC specular noise is pretty bad (usually one or more of the following frequencies: 17, 19, 23, 27, 29, 31, 39, 41 Hz) has been removed or re-EQed. If you're interested in hearing these tracks in FLAC format (lossless), just shoot me a PM with an email address (the email address is necessary for Google Drive authorization where the tracks reside). In general, I select tracks based on the issues that come up via measurement. I find it's much better to test them first, then listen to the loudspeakers using specific tracks to hear the issues identified via those measurements. This typically takes a lot less time to find the issues than trying to use one or two "go-to tracks", and trying to discern what the issues might be. A lot of the time, the problems that arise are very difficult to hear at first, but over time, begin to stick out like a sore thumb (at least, that's been my experience). Chris

I recently came across this Klipsch Blog article written by Trey Cannon with the following quote: I guess that I touched on some of the "why" of PWK's response in my comments on passive crossover design, above. I find that there is a general tendency to oversimplify the loudspeaker design/modification process--sometimes on the order of being 10x or more too simple. PWK was clearly trying to convey what would happen without a proper approach and follow-through if trying to modify one of his designs. At the time (early 2000s), the DIY tools to do the job were still not really available at or near DIY levels (and costs). It's only been in the past 10 or so years that the tools and the tutorials have appeared to do a credible job on DIY loudspeakers--especially horn loaded designs. But the available tools need to be used correctly to achieve good results. Chris

Iain is correct: this is why I don't recommend DIY loudspeakers using passive crossovers--because those doing them usually don't do the measurements, then design the full network with balancing notch filters (using something like LspCAD or VitiuxCAD) and also ensuring that the minimum input impedance doesn't go below 3 to 4 ohms, etc. You also have to watch the phase alignment of the tweeter to midrange, and midrange to bass bin to ensure that there aren't dropouts or other severe lobing issues in the SPL response at the crossovers due to phase mismatches between the drivers brought on by the relative mounting distances (fore-and-aft, and vertically to each other within 1/4 inch for the tweeter and midrange and 1-2 inches for the midrange to bass bin), and the phase growth due to the electrical filters themselves (generally, 90 degrees of phase lag per order of the filters is induced on the lower frequency drivers). All of the above issues are moot if using REW/UMIK-1 and a good DSP crossover, which can be dialed-in in minutes--instead of days or weeks of passive crossover design--and have outstanding resulting sound quality, to boot. And now the cost of the DSP crossovers and multi-amping are really not worth wringing one's hands over trying to use passive crossovers nowadays. If you were going to produce many of the same loudspeaker configurations for a lot of people that can't do their own measurements and dialing-in in-room, then passive crossovers might make sense. But most try to oversimplify the passive crossover design and implementation task and wind up unsatisfied., etc., eventually leading to getting rid of everything. I'm sure that Roy discussed some of these issues in his class, as well as other issues that I haven't identified, above. Copying Klipsch passive crossover designs just isn't good enough in my experience since almost no one is using the exact same drivers and box and driver mounting geometries as Klipsch. Chris

Try a UMIK-1 and REW. It's a lot easier and much more effective than the analyzer that you mentioned. It's actually amazing what can be measured nowadays, and will enable you to proceed directly to the core issue(s). It's definitely worth the price of a UMIK-1. The microphone is self calibrating after you download the serial-number-unique calibration file from the linked page that I posted. The rest is very straightforward--and gives you a lot more than 1/3 octave SPL response. Chris

IIRC (and according to Roy) the K-691 is a B&C DE75 driver with a modified phase plug, ostensibly to get higher response out of them(?). The original DE75 (with I believe was another modified phase plug design) is the K-69. We heard this driver in Hope in 2009 with Roy-by accident. It was superior in the 10-20 kHz band to its successor, the K-69-A. The later K-69-A was a P.Audio BM-D750 Series I driver (identifiable by the case design), and was a lower cost driver, IIRC. I think it has a modified phase plug, too. The provenance of the K-1132 and K-1133 drivers was never really explained to me. Roy indicated that they were Klipsch produced. I suspect something like an EV driver or similar was the real starting point, but no one has ever identified where the basic 1132 and 1133 design(s) came from. Chris

I'd use the 4-8 ohm setting. The 2-3 ohm setting is nice to have (i.e., someone worked harder to provide that capability, which says that your amplifier is pretty stable, or conversely, it has a larger output transformer that can take output loads down to 2 ohms). All the DSP crossovers that I've personally used have separately input and output channel gains. This is in the digital domain, so all that means is the input and output channel data words are separately able to multiply by a channel gain. Try it. If the channels are noisy or sensitive to adjustments, then you probably need to change the gain structure. I'm a big fan of trying different configurations and listening to the results. I've found a lot of advice that's "set in stone", really isn't when I try different approaches. Of course, it does help to think in engineering terms--what's really occurring internally--to keep analog noise, digital quantization errors, and overall adjustability within bounds and find a good set of operating points. You can experiment, but remember to keep things closer to nominal values (it's like living a good life: "moderation in all things"). It usually takes some experimenting to achieve this, no matter how many calculations are made. Chris

Only if your amplifiers are unreliable, i.e., if they have failed in the past in short circuit fashion or power supply A/C is no longer rectified to DC. It's the direct coupling of the amplifiers to drivers which increases the fidelity of the setup, eliminating the added reactance of passive crossover circuitry that your amplifiers have to deal with. In order to lower the output impedance of the amplifier, I'd use the 4 ohm settings--all the time. Only if using very high output impedance amplifiers (i.e., transconductance amplifiers) would a recommend using the higher output impedance settings. "Noise" is a good term. Sometimes called the "noise floor" of the amplifier. Usually, turning down the amplifier gain is the first thing to try. I'd recommend reading up on "gain staging" or "gain structure" to help you understand the concepts used: https://www.minidsp.com/applications/dsp-basics/gain-structure-101 EDIT: By the way the following text in that miniDSP link, above, isn't something that I recommend actually. It's better to set the gain of the DSP to the maximum that the drivers can take (voltage input or integrated power over time) or the maximum that you ever want to hear in order to protect your drivers from overload (the most common reason why driver diaphragms are destroyed): Chris

This sounds like you didn't really dial them in well. Did you ever take in-room acoustic measurements and dial-in the EQ properly on them? I believe that everything that you said changes once you do that (below crossed to a K-500 horn in a Belle, but even better using a K-510 horn with 2" compression driver and CP25) : Chris