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Re: low-pass filter question
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Re: low-pass filter question


  • Subject: Re: low-pass filter question
  • From: Herbie Robinson <email@hidden>
  • Date: Tue, 17 Jan 2006 19:35:45 -0500

To throw this out there, are there any plans for the CA team to ever enhance or improve the SRC? I was always personally completely satisfied with what it is, and honestly I'd take the convenience of the CA SRC (in terms of coding) over the possible ±20dB of clarity found at <-120dB for other methods. I've never yet had a user complain about the SRC quality...

...or is there really no need when it comes to the true floating-point nature of the CA SRC? Is it possible that the "noise" in the charts comes just as much from the 24-32-24 bit conversion?

Well, let's look at the math involved. The SRC is basically a long FIR filter. Let's say 100 taps for ease of computation. The computation for each sample goes like this:


	Conversion to floating point.
	FIR Filter
	Conversion to fixed point.

The conversion of a 24 bit integer sample to floating point is lossless (even in single precision).

The FIR filter consists of 100 multiplies and 100 adds. Each one of these will accumulate a roundoff error. Using single precision SSE2 instructions, all of those roundoff errors are roughly at the 24 bit level and the roundoff errors add up -- That corresponds to distortion being in the low order 7 bits of the floating point mantissa. This can be improved upon a lot by sorting the coefficients; so, it's not as bad as it sounds, but there still has to be quite a bit of distortion build-up.

The conversion from float to fixed introduces one more roundoff error. If the conversion is dithered (it should be), then there will also be some white noise added at this stage. Note that ALL the SRCs in the tests are doing this conversion; so, the threshold in the DSP must be set higher than this level of noise (or none of the tests be showing those large areas of blue).

I may or may not be an audio purist, but at levels as low as -120dB or less, have any tests shown that that stuff is actually *audible* in the end result? Is this just grandstanding in terms of numbers and charts, or does this have real audible validity?

Surprisingly enough, artifacts in the low order 24th bit are audible, but only barely. I don't know of any formal studies, because it's difficult (i.e., expensive) to test psycho-acoustic differences at this level and the studies that get funding are along the lines of "what will some percentile 12 year old using $5 ear buds tolerate without complaining?" rather than "what can people with really good hearing detect?" or "how does the long term exposure to various kinds of truncation error affect the perception of musical content".

The only experiment I have done involved comparing two mixes I prepared myself. One mix had dither applied at the 24th bit level by the mixer and one did not. I found that I could identify a difference a significant amount of times in a blind test. I had no preference for one sounding better than the other, but I could just barely hear a difference. The material in question was recorded in a very nice acoustic setting and was mixed with no processing other than the mixer, using nothing but natural ambience and I was listening on electrostatic headphones. The methodology was ABX, with very short times between samples (to make sure the acoustic memories didn't fade). I was doing it myself; so, it couldn't be totally blind, but I put together a DAW session with the test in it and waited a day so I had hopefully forgotten what was in it.

This is a difference in only the lowest bit of a 24 bit samples and my hearing is anything but pristine (30+ years of clubbing and playing in rock and blues bands). It's been pretty easy for me to hear the difference between 16 and 24 bit material now that I know what the difference is, but it's very hard to describe it. The only thing that seems to work is visual analogies, like "veiled". Also, it seems to be more relaxing to listen to 24 bit material.

The general conjecture about this sort of thing is that the noise (or distortion or whatever you want to call it) is very un-natural. It isn't random (like white noise) and it isn't harmonic distortion. It isn't even like the Bessel function related partial series that come off drum heads. It looks like the modulation of the original signal by the sampling frequency and that aliases back into the audible range. The result is signal correlated non-harmonic content that our ears aren't used to processing. Some people are really irritated by it, but most people can't hear it until they get to compare 16 and 24 bit samples.

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