What about EQ??? (Please don't ignore me... :( )
Nov 25, 2001 at 8:53 PM Post #46 of 57
hmm... the image is k1kfr... what could that stand for?
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K1000, perhaps? that's impressive.
 
Nov 25, 2001 at 11:37 PM Post #49 of 57
BTW, you can hook up the best transducers in the world without a DSP, such as playing back reel-to-reel tape. And it'll sound a hell of a lot better than a rock hooked up to an SACD player with the ultimate DSP EQ system in existance.
 
Nov 25, 2001 at 11:50 PM Post #50 of 57
Quote:

Originally posted by kwkarth
...EQ when performed properly in the digital domain can be pretty cool. When done in the digital domain, you don't introduce the phase anomalies that you would otherwise incur...


You don't? I thought you did. Isn't "ringing" a phase anomaly?
 
Nov 26, 2001 at 12:51 AM Post #51 of 57
"You don't? I thought you did. Isn't "ringing" a phase anomaly?"

1st of all, I did say properly,
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and secondly, no, I would not refer to ringing as a phase problem.

Ringing is basically a form of harmonic distortion and can be caused by a poorly designed analog brick wall filter, a poorly designed analog transformer, gross impedance mismatch, etc... Depending upon the timing of the overshoot, reflection, & resonance, it can manifest itself as a phase anomaly, but more often you see it as a sub harmonic or harmonic (sideband) of some fundamental. TIM (Transient Intermodulation Distortion) is actually an indicator of this phenominae.
 
Nov 26, 2001 at 2:05 AM Post #52 of 57
Quote:

Let me ask you this...lets record the response of a Sennheiser driven by a Max. Lets hook up a TA, and record the response. Lets compensate the difference by EQ. Wow you don't need Sennheisers + MAX you can just get Sennheisers + TA + digitally EQ'd CDRs! Maybe I'll fudge up some graphs so some people will believe it too.


I think I did mention that you can't formulate the distortion caused by random noise? (like what the TA would introduce?)
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I would admit that I read the graphs a bit off...
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But I guess I'd just have to wait and see if I can audition a Compensator. Hey, it's all about the sound right?? So go read a review or something
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BUT...

Quote:

Lol actually the website says that the second graph is of a speaker...yet if you LOOK at the title you can see that the measurement is of a signal with a voltage measurement instead of SPL! Jeez try to drum up some good looking graphs and they forget to white-out the titles LOL.



graph.php

Would you care to look at the left axis? It's also measured in volts
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Nov 26, 2001 at 2:31 AM Post #53 of 57
hey joe, where are you goin' with that gun in your hand...
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(that's a Hendrix quote for all you philistines
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)

Anyway, there are still a lot of things EQing can't do that I enjoy from headphones... ie, I can't ever give Senns headshaking bass because the drivers are so far away. But whatever floats your boat. It's useless to argue that EQing things by ear makes something more accurate. More pleasing maybe, but not more accurate... but that's already been established... i'm rambling...
 
Nov 26, 2001 at 3:20 AM Post #54 of 57
But what do the headroom graphs have to do with the graphs I am talking about? The headroom graphs have some value because they demonstrate some consistency. When we compare some headphones we hope that there is enough consistency to show some information.

But I am confused on why the site you link has a Sensitivity Bode plot being compared to a Function Bode plot. I am also confused on why they neglect to mention what exactly they are measuring besides an "audiophile speaker".

Also if you are a mathematician that you should notice that the time interval of the "before" impulse graph is about 2 msecs, while the "after" impulse graph is about 4 msecs. Using my l33t math skills, the "after" impulse graph will produce an impulse that looks twice as short. Oh and the amplitude difference is something altogether LOL...it is of MAGNITUDES on a different scale it is almost ridiculous that they are even being compared. Hell we don't even know if they are using the same speakers.

Graphs only MEAN something if there is good measurement and comparison methodology. If you really want I can take some napkins and crayons and draw some graphs on the spot too...but it isn't going to mean that much. Showing before and after graphs of different parameters of unknown variables and constants is pointless.

Just because someone spits out specs, graphs, and numbers does not mean it is absolute truth. Graphs and specs are attempted measurements of the real-world. The real-world is what matters. You can make graphs or specs look better or worse to give people a distorted picture of their real-world performance. Also my bet is it is the marketing department that had a good hand in writing the "faq" and not engineering. That isn't to say they might not have a good thing going on in terms of engineering, but they have flaky marketing common to a lot of "audiophile" marketing.
 
Nov 26, 2001 at 5:21 AM Post #55 of 57
Quote:

Originally posted by kwkarth
Ringing is basically a form of harmonic distortion and can be caused by a poorly designed analog brick wall filter...


I thought all digital filters did it, it's just brick wall filters did it the worst, and therefore audibly. But at the level that Joe is talking about, it's only phase type stuff, because it only traverses the immediately neighboring sample slices...
 
Nov 26, 2001 at 5:50 AM Post #56 of 57
Ya lost me Dusty. Huh?
 
Nov 27, 2001 at 3:53 AM Post #57 of 57
Quote:

Originally posted by kwkarth
Huh?


Alright, sorry for the delay, and lack of subsequent quality in my response...I started doing online research, and I just don't have time right now, so I'll go from memory. Please feel free to correct me on technical aspects:

I didn't think it was possible to create a digital filter that performed frequency domain transformations without getting some amount of "ring".

No single sample contains frequency information. In order to obtain enough information to convert a signal into the frequency domain, you have to look at the samples on either side at the timeslice for which you are calculating the frequency content.

Hence, there will be "spillover" -- the adjacent time slices will get a little bit of frequency information from the adjoining samples without actually containing that frequency information. Another way of saying it will "ring" -- the frequency information will "smear" slightly, in the time domain.

Of course, this information could be based on technology which is simply old, but I didn't think they found a way to get rid of ring completely (...yet). Hence the reason some of the highest-end and most expensive DAC's don't have digital filters at all.

NP: Radiohead's Amnesiac
 

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