Chris A
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Posts posted by Chris A
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Oh boy....analog out...🤢
Can you connect via a USB to a DAC?
Chris
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Since I haven't seen how you are connecting to your setup via your computer (output and input), I have to ask...
Are you using an analog output from your computer (like a soundcard) or are you using USB or HDMI?
If digital, what output driver are you using from your computer (ASIO, Java, etc.)?
Are you using a UMIK-1?
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17 hours ago, Emile said:
for unknown reasons (I had not changed ANY settings), the REW measurements now were about 20dB's louder.
So have you changed the gain of your preamp/integrated amplifier? I have a digital readout on my AVP, which I typically set to a movie--loud position (-12 dB in my setup presently), and which is part of my checklist of things every time I pull out the UMIK-1 and REW to measure again.
Also, not moving the microphone between measurements is a pretty big deal, unless you're trying to average measurements around the on-axis measurement, in which case, you'd move the microphone around by perhaps 6-12 inches or so, then average the measurements. This is much more time consuming, however, than using just the on-axis measurement.
Chris
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3 minutes ago, Edgar said:
Thank you, @Chris A, but none of this appears to address my actual concern.
Yes, I think you missed my points. Perhaps a PM/email would be a better place to discuss. (If you still want to discuss it here, I will of course.)
Chris
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35 minutes ago, ClaudeJ1 said:
Nelson Pass uses a Tannoy Co-Axial speaker to audibly differentiate the sound between amplifiers. Go figure.
Consider the Tannoy coaxial's phase response--combined with its (largely) full-range directivity. I think that's what they are listening to. Mr. Pass has talked about phase response as a factor, I'd guess that the human hearing system is extremely sensitive to amplifier--loudspeaker-direct-arrival phase response--if the phase response isn't already scrambled by one or more component in the system (including the room itself). YMMV.
An order-of-magnitude phase response plot from the 15" Tannoy HPD in an enclosure:
I find that that a dialed-in Jubilee or an off-the-shelf SH-50 will magnify phase response issues of amplifiers, recordings, and EQ (minimum phase corrections, that is). You can easily hear that phase response when you also control the early reflections in-room.
Chris
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1 hour ago, Edgar said:
...isn't the tweeter diaphragm excessively modulated by the woofer motion?...
This is a very good question, and one that I believe addresses the "fears" of all who shy away from MEH designs.
I'll have to look for the link to the reference, but I read paper that identified this and measured it, and it found that it's inaudible. That's a pretty big result/find. I can think of a couple of reasons for this:
1) The Huygens-Fresnel principle works one direction, but if you look at the reverse direction (reflections and other acoustic waves coming back into the slit/throat aperture, you'll find that the "rejection ratio" from front-to-back is extremely great. If you also look at electro-magnetic multiple-entry horns, this is also a principle that's been observed and used to advantage. (Other E/M issues exist in multiple entry horns that make them more challenging to design well.)
What happens on the back side of an acoustic horn (i.e., the higher order dynamics) tend to stay there, and the same is true for reflections and other energy coming back from the front side of the horn. I believe that this is the principle that allows horns to work so well without higher order effects completely blanketing their performance.2) There is a frequency separation of the pass bands between the throat-mounted driver and the lower frequency drivers. This isn't a trivial factor. Higher order harmonics can be excited in the horn--perhaps on the lower frequency drivers, but subharmonic frequencies getting into the throat-mounted high frequency driver are strongly rejected by the aperture of the horn at the throat and design of the higher frequency driver itself. The frequency spectrum inside the horn is also "pink", their amplitudes are being reduced by 1/f factors as the frequency rises (which is the same the principle that I use to demaster recordings).
3) Danley makes a big deal in the Synergy patent (USPTO 8284976) over the original Unity patent (USPTO 6411718) that he is crossing over below the first notch frequency of the horn (i.e., this is apparently the only difference between the two patents).
I now see this as a way to keep the lower frequency driver spectrum (from drivers closer to the horn's mouth) from affecting the higher frequencies from the throat of the horn because they are using Synergy horns at 120-130+ dB(SPL) for PA use--well above those SPLs used in home hi-fi loudspeakers.So if you're worried about this, simply cross the lower frequency drivers below the first notch frequency--but then you're in patent trouble if you do via the Synergy patent. I don't cross below the first notch frequency, and the effect of the lower frequency drivers on the compression driver is inaudible and for me not visible in electrical impedance plots.
Chris
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No offense, but I have to be honest: there is nothing about Karlson bass enclosures that interest me. Or split K-tubes. These actually have terrible acoustic performance--especially polar performance. These were inventions of the 1950s that were initially made without access to proper measurement gear that we have today. PWK tested them and handily rejected them as has been discussed on this forum many times. I agree with his assessment.
I'm also not into "cheap" or compromising bass frequency response in order to make the enclosure "small". Candidly, lower cost amplifiers can be had, but, for me, life is just too short to spend it on trying to make equipment that's too cheap produce the kind of sound that I want to listen to.
Generally, to help others understand the rationale for this thread: it was created to be about reproducing extremely high fidelity sound using emerging and modern technologies, not about rehashing yesterday's technologies because they are cheap, available, or small.
Chris
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Wow. Burning Man festivals could crop up everywhere. Kind of makes you rethink Woodstock (1969) in totally different terms. A solar panel in full sunlight could easily get close to that kind of power output to keep them recharged during the day.
Of course, there is a lot more to this than festivals and outside concerts, but the thought did initially come to mind. 😉
There are more technologies that haven't been discussed, one of them that fires the imagination: using time-domain FIR filtering to improve upon the transient response of the loudspeakers (a.k.a., Fulcrum Acoustic "Temporal Equalization"). Here's a really good video overview of that from David Gunness himself:
Here's the follow-up video that Mr. Gunness mentioned in the above video:
Chris
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I'll start by copying my thread from another that started this discussion:
1 hour ago, Chris A said:Since I was inducted into this thread involuntarily (i.e., post #6 of the previous thread that no longer exists), I think that looking forward instead of backward and discussing some of these new technologies is a lot more constructive, interesting, and friendly, unlike arguments about branding, which frankly mean extremely little to me personally---since my buying identity is not defined by brand alignment but rather by technology alignment (i.e., horn-loaded loudspeakers, etc.).
It may be that Roy can't talk about what he's looking forward to doing, so perhaps the members of this forum that don't work for, or are beholden to (by whatever services that have been provided or are hoped to provide to) KGI, can perhaps fill in the blanks. Who knows, perhaps there might be something that will help KGI's marketing arm open up their current views on market--technology subjects.
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One area that is already with us is the use of multi-amping using DSP crossovers instead of mono-amping using passive crossovers, which now very much represent the "lower performance/lowest priced" end of the consumer loudspeaker market. The evolution to DSP crossovers has already been made in the professional (commercial cinema, studio monitors, PA, etc.). KGI has made limited use of the technology, but the economics of it and the potential for performance gains is so great that it really cannot be ignored or continue to be assigned to secondary (table radio, specialty, little loudspeaker) markets.
The New Jubilee is apparently coming out with DSP crossover as standard issue. I see this as a heralding event that can easily be applied to basically all other Heritage products as add-on kits, even for very old Klipsch models. Any upgrades to drivers can be made within these "kits" along with user APIs or third party "room correction software" to dial these in after purchase. This is a pretty big deal, and I think is already beginning to affect the consumer marketplace. This is something that will see more and more penetration into the consumer marketplace as younger buyers, no longer stuck on yesterday's technologies, now represent the bulk of the buying marketplace.
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Multiple-entry horns (MEHs). This technology is now being used by KGI's competitors in the commercial marketplace, and is no longer an item that can be ignored as "not invented here" (NIH) technology. PWK himself built his first loudspeaker based on new technologies of his time, which became the original Klipschorn. This also continued with the Shorthorn, Rebel, Heresy, La Scala, Cornwall, and then Belle, which are all applications of existing technology of their time.
Why MEHs? Coaxial performance in a smaller package without violating Hofman's Iron Law. DSP crossovers make this an easy and low risk product development path. The performance gains available using this technology are now well known, and the time has come for introduction into the consumer market. It's been my experience that these loudspeakers generally blow away anything else that I've heard--assuming that linear phase response is retained in their design (i.e., the difference between the newer Synergy and older Unity horns). MEHs are also less expensive in terms of bought parts--only one horn is needed, and plywood works extremely well.
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Use of better and potentially lower cost high frequency drivers not requiring such high sensitivity (I.e., coupling DSP crossover/multi-amping with their employment). They can be used in two-way designs, thus eliminating a horn and extra driver used in the older three-way passive crossover designs.
It's no longer a secret that by using DSP crossovers, a whole new universe of loudspeaker configurations of extremely high performance levels are now possible that were precluded by the use of mono-amping/passive crossovers. I'm not sure why this subject is so obscure, but my experience is that the use of very low moving mass drivers (AMT-type) can now be used in fully horn-loaded loudspeakers. If you've never heard these type of drivers, you're in for a treat, because they present flat-phase response even better than "full range" two inch compression drivers costing many times more. The AMTs can now be used with DSP crossovers in horn-loaded applications, and there are many manufacturers that make these drivers nowadays.
and some follow-up posts:
1 hour ago, Edgar said:Once again, @Chris A, I find myself adding to one of your insightful posts.
First, I think that the concept of an API for EQ, room correction, etc., is genius. Those who don't want it don't have to use it. But for those of us who have the background and knowledge, it would be a delight.
Second, I'll add that high-quality, high-efficiency amplification is now available for hundreds or even tens of US dollars, instead of thousands. This has already made its mark in Pro Audio in the form of amplified loudspeakers. There's no reason that it couldn't do the same in the consumer market. To have complete control over EQ, power and overexcursion protection, thermal compensation, etc., is the objective.
Both of these also contribute to what I consider the ultimate goal: an all-digital system from source to amplification.
1 hour ago, Chris A said:One thing that I did miss that I should probably add to my last post, above: PWK was a very early adopter of stereo/multichannel technology that came out of Bell Labs studies. He basically led the industry in promoting his three-channel systems (i.e., two corner-loaded loudspeakers and a very good center). This is also highlighted in Greg's ("Edgar's") post just above in his signature line. Look at the date of that quote, and look at the date of the stereo record and tape introduction into the consumer marketplace.
PWK wasn't "behind the times", rather he led, but in another direction than the Villchur "small loudspeaker" marketplace. His company survives intact today, unlike all of those other consumer-based loudspeaker companies.
Looking forward (i.e., not really backward), some of these technologies have already been introduced into the consumer marketplace, as mentioned above.
Chris
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One area that is already with us is the use of multi-amping using DSP crossovers instead of mono-amping using passive crossovers, which now very much represent the "lower performance/lowest priced" end of the consumer loudspeaker market. The evolution to DSP crossovers has already been made in the professional (commercial cinema, studio monitors, PA, etc.). KGI has made limited use of the technology, but the economics of it and the potential for performance gains is so great that it really cannot be ignored or continue to be assigned to secondary (table radio, specialty, little loudspeaker) markets.
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18 hours ago, Madman1 said:
Should I run a new measurement and try to correct some of that every time I change a setting in crossovers?
In general, the process works like this:
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Clear all the PEQ, delay, gain, and crossover filter settings, then run a sweep (measurement) for each driver/horn, i.e., one measurement for the bass bin, one for the HF horn, etc. Do not change the position of the microphone between measurements, anywhere. Keep the microphone in the same place--don't move it--even by as little as 1/4 inch.
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Using that raw output measurements, decide where to put the crossover points, channel gains, delays, and initial PEQs (EQ filters). Set these up within the miniDSP PC application and download them to the miniDSP crossover.
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Run another sweep (measurement) using all drivers turned on for the loudspeaker, and look at the results. Adjust the settings and look at the results using REW.
- Continue the process until the desired flatness of response is achieved.
Chris
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Clear all the PEQ, delay, gain, and crossover filter settings, then run a sweep (measurement) for each driver/horn, i.e., one measurement for the bass bin, one for the HF horn, etc. Do not change the position of the microphone between measurements, anywhere. Keep the microphone in the same place--don't move it--even by as little as 1/4 inch.
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On 9/1/2021 at 8:11 PM, Madman1 said:
Here is a pic of my graph, looks pretty bad to me but I don't know how to fix it so...
If you send me a PM, I can return my email address to you, then you can send me your .mdat measurement file exported from REW.
From that .mdat file, I can provide help, unlike screen shots. One loudspeaker per measurement (not two), microphone in front of the loudspeaker by about 39.4 inches (1 metre), centered on the higher frequency horn, with plenty of absorption material on the floor between the loudspeaker and the microphone. I use an old foam rubber mattress topper--laid out sideways, but blankets, quilts, comforters, lot of pillows, etc. will also work.
If you also send me the text file of your miniDSP settings (the one that you already have that's readable in text format), I can see what miniDSP settings you're using, and go from there.
Chris
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On 9/1/2021 at 8:11 PM, Madman1 said:
I've gotten the basic info loaded into the minidsp and have attempted to use rew....my lord that software is deep in the weeds for me. Anyway I can't get the sweep sent to the dsp because it says my biquad input file isn't formatted correctly...yea like I know wth that means.
If exporting the settings from REW, I believe you need to export in the other format provided by REW. First, you need to set the DSP crossover to "miniDSP", then "export to file" instead of "export to text" (highlighted in yellow on the right side of the figure below from REW's EQ facility). You can change the DSP crossover type in Preferences, where you only have to do it once, instead of every time you take a measurement.
Then the format of the data file from REW can be read directly into the miniDSP PC application, instead of having to transfer the settings by hand.
Chris
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You probably need to double check the connectivity you've got within the miniDSP - left input to left outputs (usually input 1 channel to output channels 1 [low frequency] and 2 [high frequency], right input to right outputs (usually input 2 to output channels 3 [low frequency] and 4 [high frequency]).
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Then make sure that the PEQ screens for each match: output channel 1 matches output channel 3, output channel 2 matches output channel 4.
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Then check to see that the input channels 1 and 2 have the same PEQ settings (there are, of course, separate input channel PEQs from the output channels).
- Then check your amplifiers for connectivity to the miniDSP and the loudspeaker drivers. If you're direct coupling the amplifier outputs to the drivers in the speakers, make sure of the connections. If you're still using some portion of existing passive crossovers, check to see if all those connections are exactly the same for the left loudspeaker as the right.
Chris
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You probably need to double check the connectivity you've got within the miniDSP - left input to left outputs (usually input 1 channel to output channels 1 [low frequency] and 2 [high frequency], right input to right outputs (usually input 2 to output channels 3 [low frequency] and 4 [high frequency]).
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Well, IIRC, tube preamplifiers don't like to drive low input impedance loads. If your tube preamplifier has an output impedance that's greater than ~500 ohms, you might have an issue.
If you have a DAC with an optical S/PDIF output, you could bypass the tube preamp and directly connect the DAC to the miniDSP--using the miniDSP as your volume control (you can order an inexpensive IR control for it), thus eliminating one ADC and DAC conversion for all digital sources.
The miniDSP 2x4 HD and 4x10 HD also have analog inputs that can be used with an analog source--like a turntable cartridge. That could use a phono preamp, etc. to do the signal conditioning (RIAA reverse-EQ curve and gain).
Chris
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43 minutes ago, jcn3 said:
as we know, it's not about how many watts, it's about the ability to provide the amount of peak amps required at a particular moment
You're right: it is amperes (current) that drives loudspeakers. Microvolts and millivolts are usually all that's needed for most music most of the time (assuming horn loading) and perhaps a volt or two in extreme peaks. Amplifiers nowadays usually can handle 2 ohms in local impedance (vs. frequency) dips without incident. It's the quality of the microamperes and milliamperes and the resulting microvolts/millivolts that are most important out of amplifiers.
If you really are concerned about hearing bass out of your loudspeakers/room, I recommend bi-amping or tri-amping and a good DSP crossover, instead of trying to mono-amp using passive crossovers. Then you can get direct coupling to the woofers and avoid the problems of back-EMF generated by moving mass and electromagnetic issues on the higher frequency drivers. Multi-amping also avoids all the "padding" (added resistance and reactance) that occurs in passive crossovers in order to balance the driver outputs--which saps the amplifier's overhead capability.
I've found that it's the room acoustics and the loudspeakers themselves that almost exclusively determine sound quality--not electronics. That's the focus of this thread. It's actually a trap that many fall into to place too much money on electronics and far too little on loudspeakers and rooms. If sound quality is your measure of merit (and not the "look" of the other gear), spend money in the preference order: room acoustics, loudspeakers, source music (recording) quality. Lastly, electronics.
Typically, audiophiles get that formula quite backwards. Just look at the audio forum where the word "science" is in the name. I've just described the most fundamental issue there.
Chris
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1 hour ago, NBPK402 said:
...just that your ears are more accustomed to a speaker that is not as accurate IMO.
Exactly. Being used to hearing the real thing (acoustically) typically results in more expensive loudspeakers and better rooms acoustically to reproduce the sound of the real performance, in my experience. For example, I'm used to live classical music performance, especially classical organ and large ensembles (wind symphony and orchestral), so the price goes up to do a credible job reproducing this type of music, I've found.
The upside is that the setup optimized for acoustic music can then play anything quite well (in the case of horn-loaded loudspeaker setups), especially if you reduce the damage done to your CD music albums, i.e., fixing the extreme clipping and perhaps compression used to make CDs sound louder over the past 30 years (since 1991) and largely undoing the mastering EQ on CDs produced before then.
There are some apparent exceptions to this rule, i.e., large monetary outlays for your chosen music genres to reproduce well, but those are typically associated with large amounts of low bass and/or generally extreme SPL without audible modulation distortion, the latter types of which I don't recommend if maintaining whatever hearing acuity you still possess has any value for you.
Chris
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What I'm saying is that the people doing it may or may not be well trained, since there are no apparent schools to train those guys (and it takes a while and some interest on the part of the observer to develop an ear, in my experience), the apprenticeship approach may be more of "the blind leading the blind" locally than having true expertise in getting the most out of the room and loudspeakers, etc.
I can say that having a more-than-entry-level understanding of the acoustics and the applicable physics/psychophysics has been more than a little help in understanding and identifying issues and solutions encountered in the process. Hiring a young guy right out of high school (i.e., grade 12 in the US) typically doesn't result in that kind of understanding of the technical portions of the problem.
Chris
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In passing , note I had three things to dial-in my setup over those 12 years:
1) The exact position of the loudspeakers in the room corners (including the TH subwoofers co-located, using my ears and REW measurements),
2) Acoustic absorption, bass trapping, and diffusion treatments (using my ears and REW measurements), and
3) DSP crossover dial-in of crossover filters, channel delays, and EQ (using REW measurements and my ears).
The thing that took all the time I mentioned was really understanding what each part could influence/control, what choices there were in the resulting sound quality, and how to actually do each part, separately and in combination. I've written about those experiences so that others don't have to start from scratch--like I did. Doing it again for other locations/setups would take perhaps a day or two (not 12 years), assuming all the needed items were on hand.
All the available books and articles on the subject were not worth much, I found. Toole's book didn't provide any help except for understanding some of the effects of phase response. Mostly, it was through first-hand experimentation that I learned about how to use each part.
I actually find it hard to believe that those people that do custom dial-ins as a business (along with selling the hardware to customers) don't really publish anything of much use in terms of what they're doing--as if it's some sort of trade secret. I'm also not sure that those custom dial-ins are really getting anything close to the best performance, and certainly not with horn-loaded loudspeakers having DSP crossovers (since those doing the installs usually handle direct radiating loudspeakers, only). There are apparently no schools to train those that do in-room dial-ins, and the variation in quality of those doing the tasks must be very great, indeed.
Chris
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Using REW to Find Parametric Equalizer (PEQ) settings
in Technical/Restorations
Posted
Do you have an HDMI out from your computer? That is the preferred method if using an Oppo player.