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Post by Boomzilla on Feb 26, 2018 21:23:43 GMT -5
Hi leonski - Yes, the miniDSP can allow crossover customization, but... No commercial filter that I know of (even in the cheapest AVRs) is 6dB / octave. This is because of the many problems you elegantly describe. "Lower-end" gear may use second-order filters (12-dB / octave), but anything of any quality at all uses third-order (18 dB / octave) or higher. This simplifies the overlap and allows mixing and matching of disparate gear without significant issues. As I've said before, after listening to low-pass-sub and full-range-main combos, I'm a strong proponent of using low AND high pass filters to minimize overlap. The main problem in a "full-bass-management" approach is finding a high-pass filter that is inaudible over the majority of the audio range. A cheap filter (or a "pro-audio" crossover) does more harm than good. That said, there's always an exception... My Crown VFX2A, although not totally neutral, is no slouch. So is a digital or analog filter to be preferred? Provided there are no extra D-A + A-D conversions, and that the music signal remains in the digital domain all the way through, I'd think that the fully-digital filters would have the advantage. But not having heard them all, this is speculation. And in case I haven't made it clear, I really enjoyed reading your post! Cordially - Boom
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Post by leonski on Feb 27, 2018 0:42:01 GMT -5
Speakers frequently use 6 or 12 db/oct crossovers. They are easy and very known. Not to mention less $$$ to make right. Not that it matters, but my panels have a 12db octave low pass and 6db octave high pass. My preamp has been given a minor 'ding' for its ONLY 12db internal crossover provisions. Each 'order' of a crossover represents 90 degrees of phase shift. Which is why you sometimes need to wire a speaker what is technically out of phase. This is the start of the sonic holocaust which crossovers can bring.
I'll stand by the thought that crossovers introduce phase issues which can be mostly undone by going active with FIR filters. This is the start of problems with stuff like 4-way speakers.
yes, agreed. I DO use both a high pass and low pass filter for sub / main integration. I just use the pre for the high pass and the subs crossover for the low pass. I just keep the crossover values far enough apart knowing such filters are NOT brick wall and that I want the whole enchilada to sum flat thru the crossover region.
And agreed again, that IN GENERAL, better IS better. A cheap setup of filters designed for smokin' hot numbers might not sound best or be what is actually needed.
Thanks for liking my post. Keep me Honest!
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Post by ronkuper on Feb 27, 2018 1:29:37 GMT -5
Vertical biamplification alone will NOT affect IM distortion. Why? The same drivers still cover the same frequencies. Oh! So IM distortion is a driver thing, not an amp thing? I was under the impression that the article was talking about the amp's distortion in that section ("two different frequencies are being amplified simultaneously").
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Post by ronkuper on Feb 27, 2018 1:39:17 GMT -5
Agree and not forgotten! I was planning to have the active filter's rolloff point way below the respective passive XO. BTW in software, the is a "brick wall" filter, not sure about its implications though. Thanks, I'm rather familiar with MiniDSP's product line (also happens to have their UMIK-1 calibration mic in addition to my XTZ). Very good stuff and would've been relevant for me, but I'm a PC geek so part of my passion is doing anything I can on the PC with software, where also ample CPU is present. So FIR's are not an issue and I can use convolution created with REW with Roon or HQPlayer.
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Post by Boomzilla on Feb 27, 2018 3:12:40 GMT -5
Vertical biamplification alone will NOT affect IM distortion. Why? The same drivers still cover the same frequencies. Oh! So IM distortion is a driver thing, not an amp thing? I was under the impression that the article was talking about the amp's distortion in that section ("two different frequencies are being amplified simultaneously"). "Transient" IM (as first described by Mati Otala) is, indeed, an amp thing. Plain IM can happen at the speaker cone level when one frequency interferes with another.
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Post by Boomzilla on Feb 27, 2018 3:14:57 GMT -5
Speakers frequently use 6 or 12 db/oct crossovers. They are easy and very known. Not to mention less $$$ to make right. Not that it matters, but my panels have a 12db octave low pass and 6db octave high pass. My preamp has been given a minor 'ding' for its ONLY 12db internal crossover provisions. Each 'order' of a crossover represents 90 degrees of phase shift. Which is why you sometimes need to wire a speaker what is technically out of phase. This is the start of the sonic holocaust which crossovers can bring. I'll stand by the thought that crossovers introduce phase issues which can be mostly undone by going active with FIR filters. This is the start of problems with stuff like 4-way speakers. yes, agreed. I DO use both a high pass and low pass filter for sub / main integration. I just use the pre for the high pass and the subs crossover for the low pass. I just keep the crossover values far enough apart knowing such filters are NOT brick wall and that I want the whole enchilada to sum flat thru the crossover region. And agreed again, that IN GENERAL, better IS better. A cheap setup of filters designed for smokin' hot numbers might not sound best or be what is actually needed. Thanks for liking my post. Keep me Honest! I think you know more about this than I do, leonski - Since we're in full agreement, we surely must BOTH be wrong! LOL
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Post by ronkuper on Feb 27, 2018 3:33:19 GMT -5
"Transient" IM (as first described by Mati Otala) is, indeed, an amp thing. Plain IM can happen at the speaker cone level when one frequency interferes with another. Makes sense. So I assume you hold the opinion that Transient IM (the amp thing) is not worth the try to reduce with linear phase FIR filters that would "brick wall" bellow/above the respected passive XOs on the speakers?
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Post by Boomzilla on Feb 27, 2018 7:03:22 GMT -5
Nah - the TIM in an amp is part of the basic design. You won't change it from external devices.
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Post by ronkuper on Feb 27, 2018 7:47:53 GMT -5
Cool!
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Post by leonski on Feb 27, 2018 15:58:08 GMT -5
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ttocs
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Post by ttocs on Feb 27, 2018 19:22:45 GMT -5
All, I'm lovin' this discussion!! Coincidentally, just today I stuck one toe into the water and ordered a active crossover with which to experiment. Baby steps.
I'm learning so much from this. Please continue! and Thank You!
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Post by leonski on Mar 4, 2018 22:17:56 GMT -5
All, I'm lovin' this discussion!! Coincidentally, just today I stuck one toe into the water and ordered a active crossover with which to experiment. Baby steps. I'm learning so much from this. Please continue! and Thank You! What kind of crossover? Of whose manufacture?
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ttocs
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Posts: 8,158
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Post by ttocs on Mar 5, 2018 10:30:17 GMT -5
All, I'm lovin' this discussion!! Coincidentally, just today I stuck one toe into the water and ordered a active crossover with which to experiment. Baby steps. I'm learning so much from this. Please continue! and Thank You! What kind of crossover? Of whose manufacture? Behringer Super-X Pro CX3400
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KeithL
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Post by KeithL on Mar 5, 2018 13:06:45 GMT -5
Just a very general consideration..... When you do what people these days call "vertical bi-amping" you feed both amplifiers feeding each speaker the same exact signal, and leave the passive crossover in place in the speaker... The main benefit there is that each of the amplifier channels is seeing a lighter load. Also, in theory, they don't interact (for example, if you drive the woofer into a non-linear portion of its operating range, that won't affect the signal going to the midrange and tweeter). With a "full real bi-amplified system" there is an active crossover before the amplifiers, and each amplifier is connected directly to its speaker driver. In general, you want to AVOID using an active crossover AND leaving the passive crossovers in the speakers. When you do that, what you have actually done is to split a single crossover into separate sections, each of which interacts with all the rest - both in terms of amplitude and phase response. In order to figure out the result, you would have to calculate the exact responses of both the active and passive portions, then add them together properly. (This is FAR more complicated then either one by itself.) Many people suggest using the same model all around... otherwise you will want to have control of the individual level of each amplifier channel. (If you're building an active crossover, adding a few level controls is trivial, and any active crossover should offer level adjustments on each output...) Hi Guys, So I've decided to check out passive* bi-amping for my next geek-out-audio mini project and wish to test some questions and assumptions with you Relevant existing gear: XPA5 (Gen 2) Quadral M50 Platinum (Bi-wiring ready, specs below ***) Current decisions and their supporting assumptions (welcome to object to any of them): Decision: Passive bi-amping - Not going to bypass the speakers' passive crossovers Supporting assumptions: - I don't have the skills to design an active crossover network that would surpass the manufacturer designed passive xo
- Passive bi-amping does offer advantages - 1. By dedicating an amp channel each for HF+MF and LF (reducing stress on the HF+MF amp)
2. By separating the electrical input (removing the external jumper) of the HF+MF XOs from the LF XOs and so preventing back current EMF from the LF woofer drivers to interfere with the HF+MF drivers - It is rather straightforward to try in my setup
The Dilemmas:- Should I configure an active (digital) low/high pass filters before the respective amp channels or give both the full signal?
- By giving both amps the full signal - Will it still benefit the HF+MF amp channel (less resistance from drivers?) or will it work just as hard on amplifying the full signal regardless of the drivers / passive XOs it is connected to?
- With 'full signal' to both - Does it make sense to use a shared PSU amp such as the XPA-5 or will it make it draw twice as much power (4 Amps channels getting the full fronts signal) which might degrade performance overall?
- Other drawbacks of using a shared PSU multi-channel amp vs. dedicated amps with regards to passive bi-amping?
- If using dedicated amps, should they all be the same model? Could there be amp level differences issues otherwise**?
- If using an active low/high pass filters before the amps - Should I worry about phase issues (from the active XOs) with the High-Mid section and the LF section of my speakers?
Anything else to consider? Eager to hear what you guys think on these topics. Thanks, Ron *Sources for passive bi-amping assumptions (out of many on the web) - ** Practical dilemma behind this question (5 - same model amps): Currently the XPA5 handles fronts, center and surrounds. Options are: 1. Get another 2 channel amp and use it to bi-amp the fronts with the XPA5 (One taking the mids+highs and one taking the lows) 2. Get another 2 channel amp the use it to power the surrounds and then use the freed up channels on the XPA5 to bi-amp the fronts.
*** Specs of the fronts from the manufacturer website: Nominal/music power (W) 200/300 W Frequency response (Hz) 23...50.000 Hz Crossover frequency (Hz) 280/2300 Hz Efficiency (dB/1W/1m) 90 dB Impedance (Ohm) 4...8
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Post by leonski on Mar 5, 2018 13:26:21 GMT -5
Keith, I must respectfully differ:
Each amp in a 'passive' biamp system STILL swings the entire voltage, 'loaded' or not. This really doesnt' buy you much except a 2nd amp. I'm afraid this is the sytem I use and the plan was always to go active. This might not quite apply to a current source amp, like TUBE.
Agreed about crosssover into a crossover. A nightmare and NOT to be done.
Power per 'way' in an active system depends on 2 main things. Expectations? Are you a redline type or just want background or casual listening? AND the crossover frequency. For the Redline user, who demands maximum level, than apportioning power based on crossover frequency makes a certain sense. The 50:50 power point of around 350hz is a reasonable start. Some prefer identical amps, regardless of power needed. Or in some cases, maybe amps from the same family for voicing concerns.
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KeithL
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Post by KeithL on Mar 5, 2018 14:51:52 GMT -5
1) Even though each amplifier swings the same voltage load, the current load on each is greatly reduced over a certain frequency range. How much this matters will, of course, depend on the amplifier itself. I do agree that the benefits of doing this are relatively minimal. 1a) There is also a "special case" situation involved with overdriving a woofer. If you overdrive a woofer into excessive excursion, you may in fact drive the voice coil to move outside the linear portion of the motor structure's magnetic field some of the time. As a result, you end up getting abrupt jumps in the impedance of the driver as seen by the amplifier, which CAN modulate the output of the amplifier (as the load impedance jumps abruptly). If you're using a single amplifier, and you drive the woofer into clipping, this modulation CAN produce audible distortion in the other parts of the audio spectrum that same amplifier is powering. You clip the woofer, which causes modulation of the output of the amplifier, which, in turn, causes audible distortion in the midrange and tweeter. Obviously, if you cause the amplifier to clip due to insufficient voltage, it's going to clip no matter the load. However, you can isolate the midrange and tweeter from any modulation that may be caused by the woofer if you use separate amplifier channels for each. Therefore, even a "passively bi-amped" system may be more tolerant of overloads, and sound less obviously bad when a small amount of overdrive occurs. 2) Tube amps are NOT "current source amplifiers". Virtually all audio tube amplifiers are simply "a voltage source amplifier with a relatively high output impedance". (While most solid state audio amps are "a voltage source with a very low output impedqance.) This makes them a very poor approximation of "a load invariant voltage source".... but still far from a pure current source. (It is fair to claim that a tube amplifier is "more like a current source than most solid state amplifiers.) Keith, I must respectfully differ: Each amp in a 'passive' biamp system STILL swings the entire voltage, 'loaded' or not. This really doesnt' buy you much except a 2nd amp. I'm afraid this is the sytem I use and the plan was always to go active. This might not quite apply to a current source amp, like TUBE. Agreed about crosssover into a crossover. A nightmare and NOT to be done. Power per 'way' in an active system depends on 2 main things. Expectations? Are you a redline type or just want background or casual listening? AND the crossover frequency. For the Redline user, who demands maximum level, than apportioning power based on crossover frequency makes a certain sense. The 50:50 power point of around 350hz is a reasonable start. Some prefer identical amps, regardless of power needed. Or in some cases, maybe amps from the same family for voicing concerns.
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Post by leonski on Mar 5, 2018 20:38:58 GMT -5
right you are. no pure 'voltage' or 'current' source amps, or so few as to not really matter. Carver experimented with a series resistor for use with certain loudspeakers and Pass has an article about this same subject...
Redline users will ALWAY be able to find a way to break something. I had some AWFUL Fisher Speakers once. Still can't figure out just Why I bought them. but when I finally destroyed the midrange? i unplugged 'em inside and replaced with a 10ohm 5 or 10 watt resistor and they sounded BETTER. Junk.
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KeithL
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Post by KeithL on Mar 6, 2018 10:10:31 GMT -5
I had one of the Carver amps with the resistor..... It was a 1 Ohm series resistor, which raised the output impedance to 1 Ohm, lowering the damping factor to 8 (with an 8 Ohm speaker). That put the output impedance and damping right on a level with many tube amps - which was the "real idea". (And, no, it sounded much better on every speaker I had at the time when connected properly. - without the resistor.) The IDEA of a current source amplifier is interesting. Since the magnetic motor in most speakers delivers a motive force that's proportional to the CURRENT passing through it, it seems like you would get a more linear response that way. In practice, however, that doesn't usually work out very well. With a typical driver, at the resonance frequency, you get both a big peak in impedance, and a BIG peak in efficiency - mostly due to mechanical issues. With a voltage drive amplifier, the impedance peak causes the speaker to draw less current at resonance, which tends to partially cancel out the mechanical efficiency peak. With a current drive amplifier, the exact opposite happens, so the drive remains very consistent, which makes the effects of the mechanical efficiency peak even worse. In theory, there might conceivably be other benefits, at frequencies far away from the resonance frequency. However, in practice, when you get away from there, the load becomes more purely resistive (or, at least, non-resonant), and it doesn't make much difference either way. I've always thought a true current-drive headphone amp might be.... interesting. (It's actually very easy to design one... I've just never gotten around to it.) right you are. no pure 'voltage' or 'current' source amps, or so few as to not really matter. Carver experimented with a series resistor for use with certain loudspeakers and Pass has an article about this same subject... Redline users will ALWAY be able to find a way to break something. I had some AWFUL Fisher Speakers once. Still can't figure out just Why I bought them. but when I finally destroyed the midrange? i unplugged 'em inside and replaced with a 10ohm 5 or 10 watt resistor and they sounded BETTER. Junk.
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Post by leonski on Mar 6, 2018 13:33:53 GMT -5
I can't say what speakers you tried with the resistor equpped Carver (Sunfire series?). The difference might be that you used all more 'modern' speaker designs. The big break between small / low sensitivity and the old chool 'big box' very high sensitivity was maybe when AR started with their original acoustic suspension deisgns. I always wanted a pair of AR3a.
Because of all the things noted, above, the old school speakers responded better to an amp with high output impedance. I wish I could quickly find the article, but Pass did experiments with VERY old school single driver speakers and series resistance.
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KeithL
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Post by KeithL on Mar 6, 2018 15:00:33 GMT -5
Actually, the speakers I had at the time were two different Vandersteens (a pair of Vandersteen 2's and a pair of Vandersteen 4's), several different M&K's, and a pair of KEF RDM's (I forget if they were RDM-1's or RDM-2's). There were a few others, but I only ever tried that resistor once each on a few speakers.... and it always resulted in bass that was, to some degree, poorly controlled. And, yes, they were all relatively modern designs. (The amp I had with the resistor was the original "Sunfire Load Invariant Power Amplifier" - which was actually a pretty decent amp in its day.) And, yes, you're quite right. Speakers designed back when tubes were current technology had to be designed to work well with an amplifier that didn't provide much damping. Because of this, they relied less on electrical damping, and more on mechanical damping... Those older speakers tended to be more efficient, but less tightly controlled..... Note that the low damping factor common to virtually all tube amps was never a design CHOICE. From early times, most amplifiers were designed with the highest damping factor that could practically be achieved. Certain limitations in how tube power amps are designed actually LIMIT the damping factor you can achieve in standard designs; if you push them beyond that limit they buzz like a giant door buzzer. (Damping factor is related to feedback; the phase shift present in all output transformers causes stability problems if you apply more than a tiny bit of feedback.) In comparison to older speakers, most modern speakers have heavier cones, much more powerful motor structures, and proportionally less mechanical damping. Modern super-powerful magnets have helped to make the more-powerful motor assemblies practical. (Even those enormous magnets you see on some old speakers were pretty wimpy compared to modern ones.... Neodymium is something like 20x as powerful as AlNiCo). Modern amplifiers, with high damping factors, are better able to utilize those more powerful motors to maintain tighter control over the speakers... and that's what everybody expects to happen. From the theoretical side of things this is the right way to do things. You want the speaker to FOLLOW THE SIGNAL..... However... Put an amplifier with a low damping factor on a modern speaker and the woofer tends to flop around because the amplifier doesn't control it well at all. (As a result, you'll usually get more bass, but it will tend to be "wooly" or "tubby", because the amplifier fails to control the woofer, so it keeps bouncing around after the signal stops.) Put a modern amplifier on an older speaker, and the modern amplifier controls it TOO well.... and the result is often not enough bass. (The older speaker sort of expects the amplifier to "let it do its own thing" and counts on that extra bass contribution to come up even; without it the bass often sounds thin.) I can't say what speakers you tried with the resistor equpped Carver (Sunfire series?). The difference might be that you used all more 'modern' speaker designs. The big break between small / low sensitivity and the old chool 'big box' very high sensitivity was maybe when AR started with their original acoustic suspension deisgns. I always wanted a pair of AR3a. Because of all the things noted, above, the old school speakers responded better to an amp with high output impedance. I wish I could quickly find the article, but Pass did experiments with VERY old school single driver speakers and series resistance.
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