Active Crossover Benefits

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Yamaha only made a dozen or so but it seems all who bought one were famous. Vangelis as you mentioned, Keith Emerson and the guy out of ABBA had two!

Anyway one can not talk (modular) analogue synths without mentioning Vince Clarke so here goes: Vince Clarke.

obligatory link to fanfare for the common man https://www.youtube.com/watch?v=OgpnlLz7WR0

tenous link being Yamaha...

Somewhere on the web there is a link to a wonderfully insane man who shipped a GX-1 and matching speakers from Aus to UK to restore.
 
I haven't read through this entire thread, I'm just going to throw in my 2 cents. Using straight active crossovers I have never been satisfied, however, with the miniDSP and other solutions which incorporate delays and parametric EQ that can be applied to each driver, you can get the actual acoustic slopes to do what you want. It's always a process of setting the crossovers, taking a measurement, creating the EQ filters, applying them measuring again and so on. You absolutely CAN get amazing results, but it's not as simple as setting a 4th order LR at 3khz and calling it a day. I'm sure I'm echoing much of what's already been said. For high powered cheap bass the Beringer inukeDSP version is pretty impressive for the money. Something like that can really get your subwoofers to behave, it won't fix a bad room, but you can at least get the subwoofers themselves to have the proper response.

So if you want to go active make sure you have a calibrated mic, and software like REW and you'll have a good old time.
 
Have any of you tried the MiniDSP U-DAC8? Looks like it was created pretty specifically to take multi channel digital from something like JRiver and convert the individual channels to analog.

Would be really, really nice if it had volume control but I'm thinking seriously about ordering one. For $299 I'm not finding much that seems to compete with it. The main thing I'm trying to figure out is whether or not multiple Curryman DAC modules would be higher quality.

-Chris

JRiver can do (64 bit software) volume control... either trough an android or apple device or separate HT remote you can do it remotely as well.
Remotes - JRiverWiki
 
Beware the miniDSP 2x4's weak output. You need an amp with a lot of gain. The balanced versions can be run single ended and drive the amps with a lot more voltage. That helps keep the noise floor down.

Some of those JRiver systems look absolutely amazing, but you aren't going to hook up an Xbox and play a game without a huge audio lag. If you need low latency, you need to go IIR, you can always use an IIR based setup like the miniDSP and then unwrap the phase in JRiver's convolver. That's what I'm doing with my 4 way system and I find myself turning off the phase correction often because the delay is terrible. If you only care about music, or video played through JRiver then you won't have a problem.
 
It's just as easy to have a separate preset in JRiver based on IIR only for other tasks beside audio and video. It's perfectly capable of IIR only correction too. You don't have to use filters with lag for every application.
I do have separate "zones" set up for that, as I generally don't like listening to my headphones with 16 dB of boost at low frequencies 😀.
You can have a zone setup for critical listening, one for video, another for games etc...
It's way more flexible once you find your way.
 
Somewhere on the web there is a link to a wonderfully insane man who shipped a GX-1 and matching speakers from Aus to UK to restore.

Here is part one of that story: Yamaha GX1 Synthesizer, Part 1

I wouldn't call the guy insane. Apparently it was such a bargain that even with shipping costs it was dirt cheap.
The GX1 was an almost insane instrument though. I wish someone would make a version of the GX1 lead keyboard, I like the idea of creating vibrato by wiggling the key side-to-side.
May be Analogue Systems will at some point indulge us since they've got a track record regard odd analogue keyboard controllers.


PS: Gordon Reid, the guy in question, also wrote a 16 or 20 part introduction to analogue synthesis for SoundonSound which was extremely good.
 
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JRiver can do (64 bit software) volume control... either trough an android or apple device or separate HT remote you can do it remotely as well.

Well, from my point of view, using digital volume control can be not acceptable regarding soundquality (except for some fine volume tuning). I agree about the fact that well done software/hardware using high bits (48bit, 32 floating point,64bit,...) are usually not audible but the problem lies within converters. Especially with cheap ones.

Without being to technical a typical 24bit converter usually work on 20 at max, once you take into account that with some medium to high sensibility loudspeakers needs 18 to 35 db attenuation, your nice 24 bits converters are now working on best at 16bit more comonly at 12bit (Akai S900 sampler for everybody?! 1980's Digital sound, that's vintage! 😛 ). Well that is very simplified and probably a little bit off from numbers given but you've got it.

In my point of view i repeat you can't skip a passive attenuator (being a fixed pad, a bank of switched attenuator,...) between converters and poweramp (and fine tune volume using software/hardware volume control,it's an acceptable trade off). But that's my point of view, everybody do what he feel the most appropriate for it's own needs.
 
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. You need an amp with a lot of gain.

Udac is 2vrms maxoutput (+8dbu). Fairly common in consummer world. Usually the problem in gain staging (gain structure) is that amplifier have too much voltage gain. The only exception i know are when you use some unity gain power amp (buffer): F4 from Firstwatt for example.
 
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Here is part one of that story: Yamaha GX1 Synthesizer, Part 1

I wouldn't call the guy insane. Apparently it was such a bargain that even with shipping costs it was dirt cheap.
In his own words
[FONT=Arial,Helvetica]With 135 discrete circuit boards and what looks like 100 miles of point-to-point wiring, I wouldn't suggest that any sane person move a GX1 from the dining room to the living room[/FONT]
 
Well, from my point of view, using digital volume control can be not acceptable regarding soundquality (except for some fine volume tuning). I agree about the fact that well done software/hardware using high bits (48bit, 32 floating point,64bit,...) are usually not audible but the problem lies within converters. Especially with cheap ones.

Without being to technical a typical 24bit converter usually work on 20 at max, once you take into account that with some medium to high sensibility loudspeakers needs 18 to 35 db attenuation, your nice 24 bits converters are now working on best at 16bit more comonly at 12bit (Akai S900 sampler for everybody?! 1980's Digital sound, that's vintage! 😛 ). Well that is very simplified and probably a little bit off from numbers given but you've got it.

In my point of view i repeat you can't skip a passive attenuator (being a fixed pad, a bank of switched attenuator,...) between converters and poweramp (and fine tune volume using software/hardware volume control,it's an acceptable trade off). But that's my point of view, everybody do what he feel the most appropriate for it's own needs.

Think about it like this:
Mojave on JRiver forum said:
Here is another way to view digital volume control.
You have JRiver/HTPC, a 24-bit DAC, and an amp.

Within JRiver you can adjust the volume down and up a much as you want with the DSP.
It is always identical when truncated back to 32 bits.
Think of sliding the volume down and up in a tube that is 1000 dB long (see first picture).
You can push it all the way down and slide it all the way back up and it will still be the same.

Now you send the sound to the DAC.
With a 24-bit DAC you can attenuate by 48 dB before actually losing any audible info.
We are going to play a piece of classic music that has a range of 30-100 dB when the volume is at maximum.
This means our "tube" has been shortened from 1000 dB to 100 dB.
The music has a dynamic range of 70 dB.
You can hear the trumpets at 100 dB at 4:32.
You can hear the chimes at 30 dB at 0:10.


The DAC has a noise floor of -110 dB. It is so low you can't hear it.
When you lower the volume digitally, the noise floor stays the same.
It doesn't decrease or increase.

The noise floor of your room is 30 dB. Music played below 30 dB will be masked by room noises.

You use the digital volume control and lower the volume 20 dB and send it to the DAC.
Your entire signal has been slid down 20 dB and the chimes are now at 10 dB.
You can no longer hear the chimes due to the noise floor of the room.
The audible portion of music is still identical to when you had full volume.

You keep lowering the volume until you have attenuated by 40 dB.
Since you are only listening at 60 dB, and since the music has a dynamic range of 70 dB,
you are cutting off 10 dB at the bottom. Due to the noise floor of the room, you can only
clearly hear about 30 dB of the music.

You continue to lower the volume until you are at a max of 40 dB. You have now lost 30 dB
of dynamic range from the music and 60 dB overall.
Have you thrown away bits? Yes.
Were they audible? No, you have lowered the volume so that the lost portion wasn't audible anyway.

Is the part you can still hear identical to when you had the volume at 100%?
Yes, it is completely identical when dithered to 32 bits or less.

Has the dynamic range of the DAC changed?
No, it has stayed at -110 dB the whole time.

Has the dynamic range of the music changed?
Yes, the dynamic range is never higher than the loudest portion and
decreases when the volume is lowered below the noise floor of the room.
 
That analysis by Mojave assumes a 24 bit DAC is linear to 24 bits. It isn't. 18, maybe 20 in reality.

If you believe that losing linearity is bad, then 4-bits of fine tuning and a range select before the pre-amp seems to be a very good compromise. Possibly not needed but keeps the worries at bay, which is worth a HUGE amount in perceived SQ.

To those lucky enough to just believe digital volume control is fine I am jealous.
 
Advantages are infinite.

It's like asking what are the advantages to fuel injection over carburetor

The advantage is, you have control over what the engine/drivers are doing.
But 80-90% of the advice given on here is for passive setups.
Also you can choose different drivers for an active setup.
Build smaller vented box with less group delay.

I use kx drivers and these are a lot different then a stand alone DSP device.
Probably not very audiopliley, but with kx drivers you have infinite options. There's others out there.

Everything I have is connected to a computer.
Passive x-overs don't make a lot of sense for me.
An active setup for me is almost free.
So a small box with nothing in it but a tweeter and woofer and has a ton of fast bass, if I want.

Also if building a 3-way, forget about passives.
I'd rather listen to nothing the rest of my life than build a passive 3-way.
Especially if I had to design the x-over.
 
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That analysis by Mojave assumes a 24 bit DAC is linear to 24 bits. It isn't. 18, maybe 20 in reality.

If you believe that losing linearity is bad, then 4-bits of fine tuning and a range select before the pre-amp seems to be a very good compromise. Possibly not needed but keeps the worries at bay, which is worth a HUGE amount in perceived SQ.

To those lucky enough to just believe digital volume control is fine I am jealous.

Adding a resistor will be much better, I'm sure 😉.
 
Adding a resistor will be much better, I'm sure

Well in fact it can be better from a snr,distortion or quantification noise point of view.

I've read the post you given and some of the claims ARE assumptions, one as Billshurv point out about the THEORICAL dynamic availlable for DAC.

As i said before everyone is own choices, mine is not to loose fidelity at DAC for many reasons, one's being that in my work in doing some mastering i've perceived some strange things when a dac is not used to his max dynamic range.

Ideally a passive pad using only (good quality, non inductive and correctly dimensionned for power rating) resistors should be harmless for your signal.

Maybe it'll add some jonhson noise (it really depend on Z of next stage and which value you choose), from my experience this is not the case if you have think your gain structure enough.

You've choosen a way of digital volume control for your system Wesayso , i respect that and i'm happy it work well for you. For my everyday work i encountered many case where the volume control could be done the same (using the master fader in many DAW or digital console), probably whith mixbuss algorythm probably even better than the one found in Jriver and most of the time an other solution is choosen and many time it's an analog one using tools like this:

http://www.cranesong.com/avocet_manual_rev8.pdf

Mastering Consoles | Crookwood

MASELEC

There is a reason if they don't use digital attenuation. But i won't argue more as it's a different field than domestic environment with different habits.

And because my choice can for some seems not being obvious nor practical for a 'home system'. Just don't think that because some ways of doing something go over what seems common sense is completly a fool choice... 😛

I'm sure you had same issues with the choice you made about your loudspeakers design, but at the end i'm sure it works great. 🙂

Maybe what i purpose is related to things i know that works and adress some real world issues too. 😀
 
I use the digital volume control in my Firewire interface.
Internally it runs at 32bit FP, in comparison to itself (ie loud or quiet) and other volume controls (ie the one on itunes, youtube etc) it doesn't appear to degrade the sound at all.
If it does I can't hear it.
 
I use the digital volume control in my Firewire interface.
Internally it runs at 32bit FP, in comparison to itself (ie loud or quiet) and other volume controls (ie the one on itunes, youtube etc) it doesn't appear to degrade the sound at all.
If it does I can't hear it.

In the Lake this is same (i can't remember if it's 48bit or 32bit buss i'll have to check the manual) and i repeat this is not the culprit (for me fine volume tuning is transparent this way (let's talk about 12db attenuation approx at max).

It's all about the converters and the way DACS reconstruct analog signal: they re is some quality loss either you tend to restrict the dynamic range by not using the most bits offered or by trying to run them to much close to the limits ( near 0dbfs) as it is possible for some peaks in INTERSAMPLE to overload the analog parts.

Is this hearable or not depends on many factors which are out of control in domestic environment. In professional studios with acoustic treatments and great monitoring i've heard both artifacts and theyre effects.

I'm not saying i give a correct way of doing things or whatever, just that for what i've heard and to stay 'high fidelity' i've made choice about this particular issue. I know some thinks it is idiot choice, but hey, if Cranesong, Manley or Maselec sell they're units with high price tags and some pro buys (while they already have ready made solution in the hardware they already own) maybe there is a reason after all.

But by all means don't think i'm giving lessons to anyone, that's not my point. I just answer to question asked with my own point of view.
 
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when you have an active crossover you do need to think a bit harder about gain structure tho. Some times it falls into place, sometimes it doesn't. Likewise sometimes you can change the gain of your power amps to maximise things and sometimes you can't.

what is important is that there is choice and the ability to have open adult discussion about it.
 
its very simple, anyone doubting that digital volume control degrade the sound, just try it.
in my experience, the sound quality is dramatically affected using internal jriver volume control.
get a good headphone amp with passive VC, enable and disable the jrive internal VC. the difference is very huge to me.
 
krivium;4611215[I said:
]-snip-[/I] You've choosen a way of digital volume control for your system Wesayso , i respect that and i'm happy it work well for you. For my everyday work i encountered many case where the volume control could be done the same (using the master fader in many DAW or digital console), probably whith mixbuss algorythm probably even better than the one found in Jriver and most of the time an other solution is choosen and many time it's an analog one using tools like this: -snip-

Well I do completely understand where you're coming from. But I can live perfectly happy because I do maximize my gain structure for the DAC feeding my amps. Set that as a base level I'd like to listen at. (I can still turn up the amps if I have to as I leave a little room there).
Most of the listening is done at that level. So I only attenuate on requests (lol) and for movies set to a pleasing sound level which usually only requires only a small change. But due to living with a multi channel soundcard and a DAC fed from that card it would be hard to do it any other way.
JRiver has a well thought out volume control and you'd be hard pressed to hear the differences. The ideal staging and imaging usually is maximized at the normal listening level anyway. As all my files are analyzed in JRiver and automatically gain controlled to peak levels I don't need to touch the volume very often.
I do understand the commotion but I optimize gain structure for my preferred listening level. So I do get to hear the maximum my DAC has to offer. If I ever need more, just turn the amps up that last bit, left on the table.
 
Well I do completely understand where you're coming from. But I can live perfectly happy because I do maximize my gain structure for the DAC feeding my amps. Set that as a base level I'd like to listen at. (I can still turn up the amps if I have to as I leave a little room there).
Most of the listening is done at that level. So I only attenuate on requests (lol) and for movies set to a pleasing sound level which usually only requires only a small change. But due to living with a multi channel soundcard and a DAC fed from that card it would be hard to do it any other way.
JRiver has a well thought out volume control and you'd be hard pressed to hear the differences. The ideal staging and imaging usually is maximized at the normal listening level anyway. As all my files are analyzed in JRiver and automatically gain controlled to peak levels I don't need to touch the volume very often.
I do understand the commotion but I optimize gain structure for my preferred listening level. So I do get to hear the maximum my DAC has to offer. If I ever need more, just turn the amps up that last bit, left on the table.

no matter the gain on the jriver volume control, I hear obvious degradation. best bypass it and use a good pre amp. try it for yourself and youll know. to me its very obvious.
 
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