Dunno that its quite so absolute. Years ago I regularly put together systems using JBL and Altec horns, including large bass bins. They speakers all sounded different, different versions of diaphragm surround design sounded different, how the systems were configured made them sound different, if the bass bins were fitted with wings they sounded different, etc. More recently I had some JBL M2 speakers here on long term loan, so I know what those sound like. The other speakers here are relatively new large panel Sound Lab ESL, with stereo REL subs. A fellow forum member visited here a couple times and described the system here verses other systems he is familiar with: https://www.diyaudio.com/community/threads/dac-recommendation.376015/post-7560022
Here is a brief quote:
...The first thing I noticed was that the electrostats could play loud with little to no distortion. Much more so than my horn speakers, which can go plenty loud. I had heard electrostats before and was always disappointed that I couldn't turn it up without hitting the limit. On these, the limit is high enough that you can enjoy realistic sound levels and stunningly low distortion...
This low distortion from the speakers resulted in a lot of transparency and ability to hear down into the preceding components.
I noticed that you quoted less than one sentence of mine. I was talking about electronics performance, not loudspeaker (i.e., as a result of the much higher efficiencies involved). Is that what you just commented on: audible electronics performance at very low signal levels?
I think last part of the quote I just finished up editing makes clear that it does include hearing down into the electronics at pretty much all signal levels.
Anyway, in case that wasn't clear yes, its possible to hear very low level detail although of course it takes more power than horns would for the same SPL.
Anyway, in case that wasn't clear yes, its possible to hear very low level detail although of course it takes more power than horns would for the same SPL.
Yeah, i dunno either....about fully horn-loaded speakers needing different/better electronics, ones that can run very cleanly at low signal levels.
Made the point earlier, but it seems to be worth repeating......ime/imo, proper gain structure rules.
Keep line level signal reasonably hot all the way to the amp. Either have a good way of attenuating the post DAC analog signal into the amp, if such attenuation capability isn't built into the amp, or choose an amp with lower gain.
Theoretically, best gain structure is when everything in the chain clips at the same time.
Also think gain structure is an issue for not just horn-loaded speakers, but any speaker using a compression driver or other high-efficiency device.
So I'd say electronics' concerns are about the same for many direct radiator speakers, just like for fully horn -loaded.
Made the point earlier, but it seems to be worth repeating......ime/imo, proper gain structure rules.
Keep line level signal reasonably hot all the way to the amp. Either have a good way of attenuating the post DAC analog signal into the amp, if such attenuation capability isn't built into the amp, or choose an amp with lower gain.
Theoretically, best gain structure is when everything in the chain clips at the same time.
Also think gain structure is an issue for not just horn-loaded speakers, but any speaker using a compression driver or other high-efficiency device.
So I'd say electronics' concerns are about the same for many direct radiator speakers, just like for fully horn -loaded.
I may be remembering wrong, but wouldn’t an appropriately large capacitor protect against out of band bass frequencies to the mid or tweeter without introducing a phase shift in the pass-band ?
After reading through a bunch of this stuff I’m definitely going to give it a shot on my LS soon.
I know that the reactance on the amplifier output terminals will increase significantly if you do that. I'm not a big fan of that approach, to be honest. If you've got the kind of amplifier that can often fail with DC on the output terminals (i.e., tube-type amplifiers), then the capacitor-blocking output is okay.
One thing that I think is difficult to convey in a way that will stick--is that the compression drivers that I use are many times more robust than the typical K-77 tweeter of the PWK era (i.e., the only tweeter that met PWK's audible performance requirements of that day/age and that had high enough sensitivity). I think that experience is very much present in a lot of Klipsch owner's minds that have blown K-77 diaphragms in the past.
I can relate a story that might help to alleviate some of that simmering fear: the first guy that I helped to dial-in his loudspeaker remotely (a K-402-MEH in the same setup as his 1st-Gen Jubilees) also chose to tri-amp one or more of his Belles, and he kept popping K-77 diaphragms using a Xilica XP series. I proposed to set a fast limiter on the Xilica to a good setting for him to prevent that from occurring. He said that all the problems with blown K-77 diaphragms completely ceased after that, and he could not detect any difference in the audible output of the Belle.
A success story, if I might say so myself. Consider this approach if you have any fears of blowing HF driver diaphragms. There is no effect on the output audio fidelity if the limiter is set properly. If you need to set a limiter for your system, let me know in a PM (i.e., a "conversation" on this forum).
As far as the out-of-band bass frequencies affecting HF compression driver output at crazy-high drive levels (i.e., above 110 dB continuous), first I'd say that you need to factor in the response of your compression driver at bass frequencies before worrying. Second, setting another attenuating PEQ or shelf filter in the compression driver's stop band to drive the relative signal level below -40 dB is pretty easy to do. If you've run out of PEQs in the output channels of your compression driver, simply copy one of the compression driver output channel PEQs that isn't close to the crossover point(s) into the input channel PEQs to free up an extra PEQ in the output channel, then add a big attenuating PEQ or shelf filter deep into the stop band to assuage your fears.
Chris
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Unfortunately I'm currently in the same boat as Gigantic as I'm using the same Dayton DSP-408 to run things at the moment.
It was a good deal at less than $100 for an open box but it obviously has a number of limitations.
The PEQ and high/low shelf settings only have a range of plus or minus 12dB.
I might try setting the shelf filters where they need to be and then for the mid and tweeter I could try the regular high-pass crossover filter set maybe an octave lower and see how it measures and what the phase response looks like.
I'm currently running some Selenium D220TI for my tweeters so they are much more robust than the old K-77's which I think I sold off over 15 years ago.
-Josh
It was a good deal at less than $100 for an open box but it obviously has a number of limitations.
The PEQ and high/low shelf settings only have a range of plus or minus 12dB.
I might try setting the shelf filters where they need to be and then for the mid and tweeter I could try the regular high-pass crossover filter set maybe an octave lower and see how it measures and what the phase response looks like.
I'm currently running some Selenium D220TI for my tweeters so they are much more robust than the old K-77's which I think I sold off over 15 years ago.
-Josh
That's a significant limitation for my use. I ran into that with the ElectroVoice Dx38. I would factor that in if anyone else is thinking about using the same model number. It's about as limiting as only having 6 PEQs per output channel (like the Dx38). I regularly use big attenuating PEQs in my dial-ins via email when I'm helping others--notably using either Xilica XP/XD series or miniDSP (2x4 HD or Flex).
In those cases, a big cap in series with the compression driver may be your only approach if trying out zeroth order crossover filters (i.e., PEQs/shelves only to limit phase growth), and you expect to use your setup at extremely high output levels (i.e., above the levels that I would use my setup).
Yes, I'd always encourage others to look at the natural response of the drivers themselves before panicking about out-of-band modulation distortion.
One thing that I can say is this: you can hear modulation distortion, and if you find you're experiencing this when you really crank your volume control, you know exactly where to go and what to do. It doesn't hurt the driver, per se, but only sounds bad (if excessive).
Another way to look at it: if you're hearing modulation distortion while using while using zeroth order crossover filters in your setup, then this is an indication that, in advancing years, you may want to turn it down a bit...today... 😂 Sort of an automatic "turn it down" indicator. 😁
I do know that at least one guy that I helped in the past started talking about how it "sounded bad" when he cranked it up after I'd taken some effort to get his phase response flattened. When I made some inquiries as to how loud he was playing his basement system, I found out that he was running out of amplifier power (dual McIntoshes)- using Jubilees. My guess is that he was listening at near 120 dB (1m) in his basement. I believe that he could probably file for disability now for hearing loss. The human hearing system simply can't take that level of SPL without permanent damage occurring almost instantaneously--permanently. I know my own father had permanent hearing loss, but in his case, it was jet engine noise (he was the lead stability and control engineer on the XF8U3 project in the late 1950s--"the best fighter that the Navy ever cancelled"...in lieu of the F4 Phantom II. I think that phrase was correct...the following from Wikipedia):
Chris
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spalmgre, a quick reply before I have watched the UMIC phase correction video, I swapped from a UMIC usb to a Behringer ECM8000 XLR mic connected as shown in post #6 in this thread https://www.audiosciencereview.com/...e-more-measurement-program.20062/#post-670479
Open Sound Meter is also a useful tool to use.
Open Sound Meter is also a useful tool to use.
The correction file phase correction can probably be done to any microphone that has a correction file.
Thank you the Motu + measure microphone is the solution I have searched for. Using acoustic timing reference from a speaker is a hassle.
It can probably be done with several different USB DACs but I have not had the time to also solve this challenge.
I have been thinking of buying a Motu but I have some questions.
1) can you direct an analogue input stereo signal from Motu into a USB computer with a convolver software like Hang Loose Convolver. And then back 8 channels to Motu through the USB interface. This way I would not need to have any other line connection input to the computer. Motu would be the preamp with input and output connections. Streaming of course from the computer so that there would not be a back-and-forth loop.
2) How easy is it to make a decent interface with Motu? Switching between signal sources and using the volume control.
This would preferably be done with an iPad app.
3) If you are running the CamillaDSP on the computer can you also switch input channels on the Motu with the Camilla interface? Or are you running two separate interface apps.
Thank you the Motu + measure microphone is the solution I have searched for. Using acoustic timing reference from a speaker is a hassle.
It can probably be done with several different USB DACs but I have not had the time to also solve this challenge.
I have been thinking of buying a Motu but I have some questions.
1) can you direct an analogue input stereo signal from Motu into a USB computer with a convolver software like Hang Loose Convolver. And then back 8 channels to Motu through the USB interface. This way I would not need to have any other line connection input to the computer. Motu would be the preamp with input and output connections. Streaming of course from the computer so that there would not be a back-and-forth loop.
2) How easy is it to make a decent interface with Motu? Switching between signal sources and using the volume control.
This would preferably be done with an iPad app.
3) If you are running the CamillaDSP on the computer can you also switch input channels on the Motu with the Camilla interface? Or are you running two separate interface apps.
Nope, how can I possibly reject the experience you had. I appreciate you reporting it.
However, I do reject that your experience applies universally to others, even in similar gear/setup situations.
ALL of our experiences, mine certainly, and yours certainly too...are just that....what we we each individually experienced.
They all need a YMMV....(your mileage may vary disclaimer)....
spalmgre, your questions on the Motu UltraLite Mk5:
1. I had not heard of the Hang Loose Convolver before your post, maybe the vendor will be able to tell you if their software works with USB Audio Interfaces. With REW connected to the Motu M4 for measurement, the M4 shows up as both an input device for the microphone and an output device for the speaker.
2. Look through this post
https://audiosciencereview.com/forum/index.php?threads/rpi4-camilladsp-tutorial.29656/post-1038294
It is part of Michael's tutorial, I followed it and now use a FLIRC and an old Squeezebox remote to switch sources and adjust volume. I linked to a picture in an earlier post. I also use the volume display (but in a cheap black plastic 1 unit half width rack case)
3. Switching channels - when I am testing and only want one speaker (in a tri-amped speaker) then I use the CamillaDSP GUI, click on the Mixer tab and mute the channels that are not being measured.
1. I had not heard of the Hang Loose Convolver before your post, maybe the vendor will be able to tell you if their software works with USB Audio Interfaces. With REW connected to the Motu M4 for measurement, the M4 shows up as both an input device for the microphone and an output device for the speaker.
2. Look through this post
https://audiosciencereview.com/forum/index.php?threads/rpi4-camilladsp-tutorial.29656/post-1038294
It is part of Michael's tutorial, I followed it and now use a FLIRC and an old Squeezebox remote to switch sources and adjust volume. I linked to a picture in an earlier post. I also use the volume display (but in a cheap black plastic 1 unit half width rack case)
3. Switching channels - when I am testing and only want one speaker (in a tri-amped speaker) then I use the CamillaDSP GUI, click on the Mixer tab and mute the channels that are not being measured.
I made some changes yesterday.That's a significant limitation for my use. I ran into that with the ElectroVoice Dx38. I would factor that in if anyone else is thinking about using the same model number. It's about as limiting as only having 6 PEQs per output channel (like the Dx38). I regularly use big attenuating PEQs in my dial-ins via email when I'm helping others--notably using either Xilica XP/XD series or miniDSP (2x4 HD or Flex).
In those cases, a big cap in series with the compression driver may be your only approach if trying out zeroth order crossover filters (i.e., PEQs/shelves only to limit phase growth), and you expect to use your setup at extremely high output levels (i.e., above the levels that I would use my setup).
Yes, I'd always encourage others to look at the natural response of the drivers themselves before panicking about out-of-band modulation distortion.
One thing that I can say is this: you can hear modulation distortion, and if you find you're experiencing this when you really crank your volume control, you know exactly where to go and what to do. It doesn't hurt the driver, per se, but only sounds bad (if excessive).
Another way to look at it: if you're hearing modulation distortion while using while using zeroth order crossover filters in your setup, then this is an indication that, in advancing years, you may want to turn it down a bit...today... 😂 Sort of an automatic "turn it down" indicator. 😁
I do know that at least one guy that I helped in the past started talking about how it "sounded bad" when he cranked it up after I'd taken some effort to get his phase response flattened. When I made some inquiries as to how loud he was playing his basement system, I found out that he was running out of amplifier power (dual McIntoshes)- using Jubilees. My guess is that he was listening at near 120 dB (1m) in his basement. I believe that he could probably file for disability now for hearing loss. The human hearing system simply can't take that level of SPL without permanent damage occurring almost instantaneously--permanently. I know my own father had permanent hearing loss, but in his case, it was jet engine noise (he was the lead stability and control engineer on the XF8U3 project in the late 1950s--"the best fighter that the Navy ever cancelled"...in lieu of the F4 Phantom II. I think that phrase was correct...the following from Wikipedia):
Chris
My ears tell me I’m on the right track.
I need to move some some stuff out of my living room to get some better measurements to see what’s actually going on.
I recently picked up a used Yamaha 2060 for just under $500...same model you suggested.
The Yamaha SP2060 runs at 96 kHz. It also has a higher noise floor level (using balanced analog inputs) than the Xilica XP series. You need to take care in setting up your gain structure. I haven't tested the digital input for noise, however. Other than the noise floor, it's a very hi-fi crossover.
Chris
Chris