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Need Clarification about Bob Katz/Sampling Rates Effects Pedals, Units & Accessories
Old 11th September 2008
  #31
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Quote:
Originally Posted by DarkEcho View Post
So, is the analog filter built into any given interface a fixed filter? IE- super steep?

Also- the problem with the filter being so steep and causing aliasing- is that the same phenomenon that occurs when using too steep of a cutoff in a parametric EQ and getting that bulge right before the drop? (Picture posted to illustrate)

What I am getting at is, by using a more gradually sloping filter, can we minimize the effects of the aliasing imposed by the filter? (obviously we'd need to use a higher sampling rate, otherwise we'd be cutting into the audible bandwidth)

If that is the case, do our commercial converter's filters adjust their slopes when we switch sampling rates? or is it just moving the cutoff point higher and lower as we adjst the sampling rate?

If we all used 96khz, AND the analog filter started at 25khz and slowly decended into nothing by the time it got to 48khz, would we get less aliasing and all these other issues?

Well, the analog filter in many converters isn't super-steep (like JR pointed out) because of oversampling. That makes the analog filter easier to implement but then you need a digital filter to get the data at the appropriate sample rate.

The filters don't cause aliasing, they actually prevent aliasing. They can cause phase shift and amplitude ripple. That's where you might see improvement by sampling at a higher rate and moving the analog filter cutoff to a higher frequency.

Here's a simplified way to think about aliasing:

Imagine a spinning wagon wheel with spokes. Now look toward the wagon wheel with your eyes closed. Now imagine that you can only open your eyes for a brief instant every 1 second.

If the wagon wheel is rotating exactly once each second it will look to you like it's not moving at all, since you "sample" by opening your eyes and the wheel is in the same spot every time, right?

That's aliasing a signal at the sample rate to DC.

Now if the wheel was moving just a bit faster, the wheel will appear to be moving very slowly since the spoke you saw as vertical during one blink will have just slightly changed position from blink to blink.

What you can't tell is the the wheel has made more than a whole rotation between blinks, because you're not blinking fast enough to see it.

This is exactly what would happen in audio as well. A high frequency signal in the input signal would appear as a signal at much lower frequency that didn't exist in the original material at all. An audible "ghost signal" would all of a sudden be in your mix.

Now what an anti-aliasing filter would do is to be sure that the wheel doesn't rotate quickly enough so that you miss an entire rotation and you can accurately judge how fast the wheel is turning.

Now in practice there are a lot of different types of filters- Linear phase filters tend not to ring and also preserve the phase relationship between frequencies (and don't distort pulses) but they are more gradual filters than other options. They could be useful in oversampled converters possibly. There are a lot of tradeoffs between oversampling, analog filter response, jitter performance, digital filter algorithms and sample rates and a lot of ways to get good (and bad) results. There are some excellent converters out there done by guys who really know what they're doing.


Regards,

John
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