Same supplier, but two versions (select the size from the drop down options) 4 in stock of the first, 5 for the second listing:
1 Bit Square (Robertson) SQ 00, 0, 1, 2, 3, 4 x 25, S2 Steel, vierkantbit | eBay
1 Bit Square (Robertson) SQ 00, 0, 1, 2, 3, 4 x 50, S2 Steel, vierkantbit | eBay
1 Bit Square (Robertson) SQ 00, 0, 1, 2, 3, 4 x 25, S2 Steel, vierkantbit | eBay
1 Bit Square (Robertson) SQ 00, 0, 1, 2, 3, 4 x 50, S2 Steel, vierkantbit | eBay
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OP, you don't happen to have a little compact mixer floating around, like a Mackie or something? This could be used to provide some gain for the DAC, whose balanced headphone output should drive the mic inputs effortlessly. (You may get away with line-ins but some of those can be just as noisy as the ones on your crossover, look up the specs if in doubt. Line-in EIN of -98 dBu is not that uncommon, better ones might score around -104 dBu.)
Otherwise it takes a midrange audio interface to even make it to +18 dBu out (e.g. Focusrite Clarett 2Pre or Audient iD22, both around 500 bucks / 350€ these days), and if you want +24 dBu you'll be paying through the nose as that tends to be big studio stuff... a used Benchmark DAC might be an option.
And BTW, I fully agree with johnmath. You only need +28 dBu to drive a 93 dB / 2.83 V / m speaker to 110 dB / m anechoic, so total gain required for a +10.5 dBu output is ~17.5 dB. The XO should require about +20 dBu peak to hit a 110 dB dynamic range, so let's say 9.5 dB gain before (which may be a mic input at/near minimum gain) and 8 dB after. Comparing said 8 dB to your typical power amp gain, it should be obvious why attenuation is needed. (A line-level attenator no less. Though again, a second little mixer may be able to stand in.)
Otherwise it takes a midrange audio interface to even make it to +18 dBu out (e.g. Focusrite Clarett 2Pre or Audient iD22, both around 500 bucks / 350€ these days), and if you want +24 dBu you'll be paying through the nose as that tends to be big studio stuff... a used Benchmark DAC might be an option.
And BTW, I fully agree with johnmath. You only need +28 dBu to drive a 93 dB / 2.83 V / m speaker to 110 dB / m anechoic, so total gain required for a +10.5 dBu output is ~17.5 dB. The XO should require about +20 dBu peak to hit a 110 dB dynamic range, so let's say 9.5 dB gain before (which may be a mic input at/near minimum gain) and 8 dB after. Comparing said 8 dB to your typical power amp gain, it should be obvious why attenuation is needed. (A line-level attenator no less. Though again, a second little mixer may be able to stand in.)
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One of these interfaces might do the job between the Fiio and 10B, or something similar:
Scarlett 2i2 | Focusrite
SERIES 102i | SPECIFICATIONS | TASCAM - United States
Scarlett 2i2 | Focusrite
SERIES 102i | SPECIFICATIONS | TASCAM - United States
Sorry I haven't been active much recently, I got real-lifed and then got super ill.
I noticed right now that my DAC output wasn't being driven up to 0dBFS. Foobar has a bug in it. I use a VST called Voxengo Span to display a spectrum analyzer. Today I added another one, called Sonogram SG-1, and my output volume went way up. Foobar's VST plugin support is third-party and kind of crappy. Turns out if you have two VST plugins then the DSP chain is unity gain, but if you only have one, then there's some sort of volume reduction happening, of about 20 dB.
Neither VST plugin actually processes the audio - they all pass it on how they got it - they're merely measurement plugins.
Wow, literally every part of my signal chain went tits up! Isn't that something!
I noticed right now that my DAC output wasn't being driven up to 0dBFS. Foobar has a bug in it. I use a VST called Voxengo Span to display a spectrum analyzer. Today I added another one, called Sonogram SG-1, and my output volume went way up. Foobar's VST plugin support is third-party and kind of crappy. Turns out if you have two VST plugins then the DSP chain is unity gain, but if you only have one, then there's some sort of volume reduction happening, of about 20 dB.
Neither VST plugin actually processes the audio - they all pass it on how they got it - they're merely measurement plugins.
Wow, literally every part of my signal chain went tits up! Isn't that something!
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That is one heck of a good catch! 
The delay was/is no problem, for me at least -- just glad there was some real-life in there, that it wasn't ALL illness.
Anyway, your health is more important than any of this ego-stroking we do on here. Take your time and get fully recovered before you launch back into it all full time.
Cheers
The delay was/is no problem, for me at least -- just glad there was some real-life in there, that it wasn't ALL illness.
Anyway, your health is more important than any of this ego-stroking we do on here. Take your time and get fully recovered before you launch back into it all full time.
Cheers
That's what I'm asking myself. I might not at the moment, but ideally I would, because other noise sources will be eliminated in the room eventually.
I have some competent RMS meters coming in sometime soon. An ex Allen & Heath design engineer suggested a Levell TM3B for quick measurements. It's got much better specs than similar items from HP, Tek, Fluke, Ballantine, and with the right setup it can measure down to 1uV. And I'm getting a HP 3457A for more intense measurements. So I'll have a handle on what the actual noise and signal levels are once I have those.
I should probably first put some resistive pads in the cables from the crossovers to the amplifiers. It would be nice to build a remote controlled attenuation box for putting between the 10Bs and the amplifiers. This could then also house a preamp for best dynamic range.
What's the best way to build a resistive pad? This article talks about some circuits, but doesn't really mention which one is better to use. Also it talks about all of them in terms of when they're being used with a microphone, not with a hi fi interconnection.
Uneeda Audio - Build your own attenuator pads
That is one heck of a good catch!
The delay was/is no problem, for me at least -- just glad there was some real-life in there, that it wasn't ALL illness.
Anyway, your health is more important than any of this ego-stroking we do on here. Take your time and get fully recovered before you launch back into it all full time.
Cheers
Thanks. This thread is one of the things that keep me going through all this stuff, so I'm glad it's still kicking. Unfortunately the real-life stuff isn't all that pleasant either, the owner of my house sold it and is trying to kick me out. Which isn't that great given that I'm still trying to recover from a very serious case of covid I got in march last year and had to go to the ER yesterday. But tinkering with that stuff and making it golden is perfect for me. It's funny, literally ever part of my set up was misconfigured at some point. I should write a booklet about this or something, because there's simply nothing out there to prepare you for all the gotchas. Here are some things I didn't know about that tricked me:
- I inspected the PMC-provided banana plug to speakon cables and as it turned out, some of the banana plugs disassembled themselves. Caught it before plugging in.
- there was a lot of hum happening. This was a pin 1 problem. I had to cut off pin 1 on all the XLR cables I bought. Oh, and one of the plugs wouldn't fit. On further inspection it turned out it was squashed in production. Good thing I bought spares.
- the 10B has switches to turn off each of the three bands. However, the one for high tones goes the other way. For no reason. I was wondering why the speakers sound crap for a couple days before I figured it out through trial and error.
- the speakers were light on bass. Turns out I had all the power amps set to 1V sensitivity except the bass amps which were set to 2V sensitivity.
- I randomly ran a stereo test on youtube and one of the sides was clearly missing high frequencies - but when HF was coming out, it wasn't coming out of either side, really. I was scared I had burnt out one of the tweeters, or bought a dud. After almost a full day of seeking the fault it turned out that the HF amp was set to bridged mode. The switch is very light and I probably touched it while plugging cables in. I should 3D-print a cover for those.
- turns out a piece of antique furniture is resonating for almost a full second after the bass is done playing at higher volumes. I was wondering why the bass wasn't that great. I have some remodeling to do. Might put my clothes rack in there, I think those should be fine.
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Have a read of the following information on pads. To be truly balanced you need to use 1% resistors or better or match them on test.
Uneeda Audio - Build your own attenuator pads
Uneeda Audio - Build your own attenuator pads
We posted the same link! I don't know which pad to use, though. U? Balanced T? Balanced Bridged T? Do they differ in performance at all?
The only comment the website provides is that Balanced Bridged T is neat if you can find the right kind of potentiometer. But I'll be using them fixed, and I'd like to be able to fit them in the XLR shell, but I can solder SMD in the air easily.
I did realize you need to match the resistors really well. Holding out for my HP meter.
Oops, my mistake for inattentiveness! For a specific use like amplifier input attenuation the pad does not have to be reversible. It's easy to fit three resistors (or five) into an XLR connector, however since XLRs are sexed and not reversible there's not really a reason to make a H pad, just use a U pad. You just need to make sure that the pad's input impedance is a suitable load for the source, and that it's output impedance is suitable for the load i.e. the amplifier input impedance.
I haven't made pads for 30 years or so and don't remember what values we used to use. As a starting point, multiply the resistances for the microphone example by 4; that will give a 600Ω source for the amplifier and a ~5kΩ load for the 10B. These values could be doubled, tripled or quadrupled to taste. I'd probably double them myself. (Hopefully someone will jump in to correct me if I have made a silly mistake.)
I haven't made pads for 30 years or so and don't remember what values we used to use. As a starting point, multiply the resistances for the microphone example by 4; that will give a 600Ω source for the amplifier and a ~5kΩ load for the 10B. These values could be doubled, tripled or quadrupled to taste. I'd probably double them myself. (Hopefully someone will jump in to correct me if I have made a silly mistake.)
My only caveat would be: Avoid the temptation to build the pad into what would otherwise be a 'standard cable'. I used to do it routinely in pursuit of *connector minimization*, believing that it was improving reliability. Almost every single one of those cables came to bite me in the ar$$ years later. I'd have long since forgotten, the marking had worn off, and it'd get placed in service and have me pulling out hair trying to track down the signal loss! (OK, I admit -- my approach to troubleshooting isn't always the most efficient! 
)
Cheers
)Cheers
My only caveat would be: Avoid the temptation to build the pad into what would otherwise be a 'standard cable'. I used to do it routinely in pursuit of *connector minimization*, believing that it was improving reliability. Almost every single one of those cables came to bite me in the ar$$ years later. I'd have long since forgotten, the marking had worn off, and it'd get placed in service and have me pulling out hair trying to track down the signal loss! (OK, I admit -- my approach to troubleshooting isn't always the most efficient!
Cheers
Fair warning. Thanks
That depends on the resistor matching, the closer the better. I think 0.1% will get you 60dB. CMRR is not important here. Agree that best to make these into an adaptor plug, not a cable.