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Anyone use equalizers?


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Beringer is no good quality. I have one headphone amp. But terrible. I use equalizer in media players. No need to buy hardware equalizers. Equalizer is important for headphone correction. Since headphones don't use large drivers, low frequency sound tends to weak. This can be corrected with bass boot with equalizer. Just sell! And use media player equalizer.

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I haven't used an equaliser in decades ! But, if I were to consider one, it would be the new Loki Mini from Schiit. Still on the fence whether I need one for my headphones.

Main System: NAS or QOBUZ > BlueSound Node 2i > Schiit Gungnir MultiBit > PYST XLR > Schiit Mjolnir 2 or Gilmore Lite MK2

 

Office System: iMac > Audirvana > Schiit EITR + Audiophonics LPS25 > Metrum FLINT NOS DAC (DAC TWO chips) > Schiit Magni 3+ > Aeon Flow Open

 

Loudspeaker System: NAIM Muso Gen 2

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6 hours ago, beerandmusic said:

So we buy and sell used stuff all the time, and my wife picked up a "ultra-curve pro DEQ2496" behringer equalizer at a garage sale today.

Is it worth even toying with, or should i just sell (She paid $6 and i see them sell anywhere from $100-$200).

 

If you are using computer as the source, you probably don't need DEQ2496 -- you can do all it can do in software using various convolvers. But, if you are using a CD Transport or another digital source that does not feed through the computer, DEQ2496 is an awesome piece of equipment. I used two of them to do some FR corrections built by REW, and they both worked well for many years. I still have one of them in my non-computer system. The main benefit is that PEQ can be applied completely in the digital domain.

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10 hours ago, beerandmusic said:

my wife picked up a "ultra-curve pro DEQ2496" behringer equalizer at a garage sale today.

Is it worth even toying with

 

It is ultra noisy and has 20ms of delay.

 

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I have DEQ2496. It is good product for the price. IIRC it uses the same DSP chip as Accuphase DG-28.

 

Some tips from users in Japan.

 

Use S/PDIF digital I/O (coaxial or optical) and use DEQ2496 as digital equalizer to avoid analog quality problem of built-in ADC/DAC. If it is necessary to use its analog I/O by all means, it is better to set input level and output level to consumer level (+12dBu) in the utility menu.

 

To prevent digital clip by equalizing, set Utility > Gain offset(EQ) to -15dB, then now it can adjust the gain of any frequency band up to +15dB without clipping. This is very important!

 

It has auto room equalizing function when used with ECM8000 measurement microphone (sold separately, about $60) and it is fun to use. Tip to use it effectively is, when entering target frequency response curve, it is better not to try to extend low-end frequency beyond your speaker's ability. Abandon, say, 40Hz or lower frequency. Use sound level meter to adjust its auto calibration sound (MLS white noise) level around 80dB SPL

 

About digital clip LED: The clip LED lit when input digital signal is 32767 or larger magnitude.

 

DEQ2496 has a function to control GEQ curve of left and right channel differently by setting dual mono mode but it is not good for SQ and it is better to use stereo link mode.

 

Sunday programmer since 1985

Developer of PlayPcmWin

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4 hours ago, PeterSt said:

 

It is ultra noisy and has 20ms of delay.

 

Noisy if you use analog, but completely quiet in the digital domain. 20ms of delay only makes a difference if you are playing video, and not much of a difference even then.

 

I'm curious about the aversion to proper equalization in the digital domain exhibited here. Is it really better to have an incorrect frequency response in your system? Measurably wrong, audibly wrong? Or, if using analog EQ, to introduce the significant distortions and more complexity that go along with such a device (also measurable and audible)?

 

Having done digital EQ for a long time to correct for system deficiencies, I hear nothing but improvements in my multiple speaker systems, as well as, the headphones.

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8 minutes ago, pkane2001 said:

I'm curious about the aversion to proper equalization in the digital domain exhibited here.

 

You mean from this small thread ? (nah ?).

 

I thought bass and treble knobs were old fashioned for 20 years or so. And no, no matter how easy it would be to correct for some frequency response, all it would do is kill sound.

Hey, I am 100% sure that not any analogue attenuation exists which does not kill sound (mind you, I (can) develop the things myself). That "killing" it is, once you have a fairly nice system. Only voltage controlled D/A conversion would work out (but which is far from easy to apply in good fashion, in reality).

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Phasure Mach III Audio PC with Linear PSU (manufacturer)

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4 minutes ago, PeterSt said:

 

Noisy in the digital domain just the same. But notice the lack of a preamp and the sensitivity of my speakers back at the time I tried to use it for, which was 115dB (today 118dB).

 

20ms of delay is crucial once you refuse to let lose such sh*tty devices on your mid-high frequencies, and thus apply it to bass only (which is in my view perfectly allowed with 96KHz sampling). The kind of luck is that the noise is not much audible in 200Hz limited drivers. But also notice that you can't get away with digital input in such a case. So you'd need analogue.

I also recall that this Behringer can't do anything under 20Hz. So for example, might you want a graduate roll off under e.g. 24Hz, then this is hard-cut at 20Hz.

 

Behringer is cheap and is for cheap purposes. That it is used in the Pro world is, well, because it is the Pro world (which is all but audiophile).

 

End of rant. Haha.

 

Peter, I measured mine down to about 120dB, I found no noise attributable to DEQ2496 when using it in the digital domain.

 

But, yes, if you use the DEQ for a cross-over, obviously a delay will be a bad thing. And I agree, its analog stage isn't the best.

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5 minutes ago, PeterSt said:

 

You mean from this small thread ? (nah ?).

 

I thought bass and treble knobs were old fashioned for 20 years or so. And no, no matter how easy it would be to correct for some frequency response, all it would do is kill sound.

Hey, I am 100% sure that not any analogue attenuation exists which does not kill sound (mind you, I (can) develop the things myself). That "killing" it is, once you have a fairly nice system. Only voltage controlled D/A conversion would work out (but which is far from easy to apply in good fashion, in reality).

 

Sure, this small thread just repeated the question in my mind. Obviously the question didn't arise from just reading the posts here :)

 

It takes some learning and effort to do digital EQ properly. But it produces amazing results when done well. Much greater difference than, for example, a different USB cable ;)

 

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49 minutes ago, pkane2001 said:

It takes some learning and effort to do digital EQ properly. But it produces amazing results when done well. Much greater difference than, for example, a different USB cable ;)

 

 Haha.

 

OK, in my previous post I wanted to say the opposite but I let it be. Here goes :

 

Those who know Phasure know that we (or I) attack everything and all by one means only : make it better (in short : improve). So I told it many times more often : I have a pile of PEQ's in the garage because I thought I needed them. And I did for 4 years or so (in the very same room they are now removed from, for 10 years already). So what happened : I improved. I am still doing that today and all I really do is making playback software better (less OS influence and what-not shyt), update the D/A converter (the one you can buy), improving the grounding on the mains supply, and for example, make a better USB cable (blurp).

And with each improvement the need for "equalizing" disappears more.

 

A few of you (CA readers) visited us, and the same gag applies always : go walk around this room and try to detect frequency anomalies. Might one of you read this (15 or so from the US) then confirm. And I really use no-thing for  attacking frequency anomalies explicitly. I don't have room treatment. I play close to 100dBSPL at times, but commonly 90dBSPL. I don't use treble etc. controls. I also don't use analog attenuators, and there it starts (would I do so, then you'd walk out screaming). And I also don't use ringing filters ... (which is prerequisite #1).

 

Now you know. :P

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59 minutes ago, pkane2001 said:

But it produces amazing results when done well.

 

I can not say that this can not help as well. But the problem with that is : you'd have all right for one listening position only.

And I continuously walk about. I don't even have a listening chair. Really not ! (we have a lot of bar stools though).

 

A maybe known means : Dirac live. It does what it is supposed to do. And the sound from is is crazily bad.

Bet that no-one recognizes this ?

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Phasure Mach III Audio PC with Linear PSU (manufacturer)

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34 minutes ago, PeterSt said:

 

 Haha.

 

OK, in my previous post I wanted to say the opposite but I let it be. Here goes :

 

Those who know Phasure know that we (or I) attack everything and all by one means only : make it better (in short : improve). So I told it many times more often : I have a pile of PEQ's in the garage because I thought I needed them. And I did for 4 years or so (in the very same room they are now removed from, for 10 years already). So what happened : I improved. I am still doing that today and all I really do is making playback software better (less OS influence and what-not shyt), update the D/A converter (the one you can buy), improving the grounding on the mains supply, and for example, make a better USB cable (blurp).

And with each improvement the need for "equalizing" disappears more.

 

A few of you (CA readers) visited us, and the same gag applies always : go walk around this room and try to detect frequency anomalies. Might one of you read this (15 or so from the US) then confirm. And I really use no-thing for  attacking frequency anomalies explicitly. I don't have room treatment. I play close to 100dBSPL at times, but commonly 90dBSPL. I don't use treble etc. controls. I also don't use analog attenuators, and there it starts (would I do so, then you'd walk out screaming). And I also don't use ringing filters ... (which is prerequisite #1).

 

Now you know. :P

 

OK, so you're saying that if you have all perfectly designed components working perfectly together, you don't need to EQ. I guess I must agree to that, otherwise the components wouldn't be perfect, right? Those of us who live in the real world, with imperfect systems, EQ can certainly help. Of course, it's easy enough to measure if your system has a perfect frequency response. And, yes, minimum phase filters are a good thing, although the improvement in my experience is not huge.

 

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1 minute ago, pkane2001 said:

OK, so you're saying that if you have all perfectly designed components working perfectly together, you don't need to EQ.

 

That is the very simplified version. Think of the importance of this (try to envision it) :

 

2 minutes ago, pkane2001 said:

And, yes, minimum phase filters are a good thing, although the improvement in my experience is not huge.

 

 

I think I said "not ringing". So also not post (like Minimum Phase does). Now we have a "pinpointed" transient which is not "blurred" (sorry) by ringing and which is theoretically super lousy in the frequency domain (say that a sharp frequency for sine wave now is distorted by other components). This you can attach to a certain extent with an interpolating filter and now yo have what I use.

There is now no buzzing and standing waves problem, or at least not emphasized with "ringing" as such which in the end mixes discrete frequencies themselves.

What the mentioned improvements do, is make the discrete frequencies even more that because of less distortion. But do notice that this is difficult to measure as such, because you can only measure music (it is the mix between time (impulse) and frequency response). But you can hear it all the more better.

 

10 minutes ago, pkane2001 said:

Of course, it's easy enough to measure if your system has a perfect frequency response

 

Maybe. But I would not know how to.

Of course, a 1000 people are now going to tell how to, but what you'd measure mostly is room response which is outside the importance of what you want to know - and that is how your ears/brain respond. I won't say "this can't be measured !" but I do say that it is beyond of what you like to know. To understand better : do you really need to position yourself in a certain corner or square inch if you want to understand what someone in the room is telling you ? I know, this is half-flawed, but still something to think about.

I "live" by improving and listening for standing waves as a measure. Are they there ? then something *will* be wrong. But this is easy talking because I could get rid of it all (in say 6-7 years of time). Can't I measure SPL differences for a frequency (or more of them) in all places in the room ? I sure can ! and see, this is where we go wrong. I would correct it, while I can't even hear it as anomaly.

Approach this with room correction software and then for sure it will be wrong everywhere (except for one position of you apply all right) and the cause was not attacked.

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15 minutes ago, wgscott said:

Huh?

 

Yes. Because of the processing.

Not really a wonder for someone like me who tries to get rid of the processing of one superfluous CPU cycle because that is audible just the same (OK, not one, but ...).

I start out with no ringing and I'm left with ringing from all angles. It is a matter what you are used to (yup).

 

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2 minutes ago, wgscott said:

I just filter it though a Belkin USB cable and it takes care of any potential problem.  But it set me back $8.

 

That could be $6 too much. But they are not easy to find (in a working fashion).

But the message is clear (and in case you were not serious, I would be, with the same message).

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3 hours ago, PeterSt said:

 

Noisy in the digital domain just the same. But notice the lack of a preamp and the sensitivity of my speakers back at the time I tried to use it for, which was 115dB (today 118dB).

 

20ms of delay is crucial once you refuse to let lose such sh*tty devices on your mid-high frequencies, and thus apply it to bass only (which is in my view perfectly allowed with 96KHz sampling). The kind of luck is that the noise is not much audible in 200Hz limited drivers. But also notice that you can't get away with digital input in such a case. So you'd need analogue.

I also recall that this Behringer can't do anything under 20Hz. So for example, might you want a graduate roll off under e.g. 24Hz, then this is hard-cut at 20Hz.

 

Behringer is cheap and is for cheap purposes. That it is used in the Pro world is, well, because it is the Pro world (which is all but audiophile).

 

End of rant. Haha.

 

No one in pro audio uses Behringer.

 

I have seen riders that said a band will not play if Behringer gear is used.

[br]

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Just now, pkane2001 said:

So you have no filters in the signal path to eliminate all ringing? How did you achieve this?

 

 I have in one place : the loudspeaker crossovers/filters. This is minimum phase and quite unavoidable.

 

Otherwise all DC coupled (no filtering capacitors) and a digital interpolating filter upsampling to 705600 / 768000. This does not ring even one sample.

 

Something quite else is that I feel that this one occasion where thee filter post-rings (the LS) could be OK because it does not oscillate with any other ringing at different frequency. But this is highly speculated (a very wild guess). I add this, because all other ringing is highly disturbing (in my room, to my ears etc.) while the speakers ring too (well, they have to) and it is OK by all means.

Lush^3-e      Lush^2      Blaxius^2.5      Ethernet^3     HDMI^2     XLR^2

XXHighEnd (developer)

Phasure NOS1 24/768 Async USB DAC (manufacturer)

Phasure Mach III Audio PC with Linear PSU (manufacturer)

Orelino & Orelo MKII Speakers (designer/supplier)

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