From: kong@leland.Stanford.EDU (Kong Kritayakirana)
Subject: CD Sound Quality (again?) -- Ideas
Organization: DSG, Stanford University, CA 94305, USA
Date: Mon, 28 Dec 92 08:40:12 GMT

Digital Recording: Part I: ADC
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Assuming you have good microphones and pre-mics, etc. needed to make good 
sounding recordings. You should use something like 22-bit 128x oversampling 
ADC. (Hey maybe we'll be able to get away with 20 bits but I love 
overengineering) Why?
 
Why 22-bit?
-----------
To give headroom for recording engineers so they don't have to set the 
recording level too high and risk having digital "clipping," 2 extra bits 
here provide 12dB headroom and should be pretty safe if he sets the 
recording level to about -10dB when the full orchestra blasts.
 
Extra 4 bits are used for proper dithering down to 16-bit resolution. When 
proper dithering is done, the quantization noise will become uncorrelated 
with the quantized signal and hence no digital "artifacts" of undithered 
signals (if you want to see what no dithering can cause, pick up a copy of 
Audio magazine and see the plots of undithered vs. dithered low level sine 
waves. The undithered one contains a lot of harmonic distortion. The 
dithered one has somewhat higher noise floor, but harmonic distortion is 
hardly present). 
 
So 2-MSB-bit headroom + 4-LSB-bit for dithering and we have pretty cool 
16-bit digital audio sound with real 98.1dB dynamic range.
 
Why 128x oversampling?
----------------------
 
The culprit in digital recording is the so-called "antialias filter." The 
filter supposedly filters out everything above half the sampling freq. If we 
use no oversampling during ADC, the ANALOG antialias filter must cut off 
very sharply somewhere near 22kHz. This is BAD for sharp analog filter is 
very hard to be designed and implemented w/low distortion and phase shifts.
So if we go 128x oversampling (again I LOVE overengineering) the antialias 
filter has to cut everything above 64x44.1kHz = 2.8MHz. The ANALOG antialias 
filter is a lot less likely to screw anything up in 0-22kHz range. Here's 
the picture:
                  2.8MHz "lowpass"     22-bit 128xOS
Analog Source --> Antialias Filter --> A/D converter --> Decimation/Dither
 
Aside: The antialias filter used in the diagram above can be very mild with 
       very gentle rolloff (and hence has low noise and distortion). It 
       must provide good attenuation above 2.8MHz and good pass 
       characteristics below 22kHz. The filter characteristic in 
       22kHz-2.8MHz can be anything. Who cares?
                                                         
So once we get the 128x oversampled data we can do the much less harmful 
digital filtering to filter anything above 22kHz out and use the decimation 
technique to convert it down to 44.1kHz sampling rate. Combined with the
technique suggested above to cut 22-bit data down to 16-bit, now we have 
very clean 44.1kHz 16-bit digital data ready to be printed on CD.
 
Comments? Suggestions? Any part left out? Email me.
 
Coming up next: Why oversampling during DAC (besides marketing reasons)
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Kong Kritayakirana  (kong@leland.stanford.edu)
Remotely reading and posting to rec.audio from
a nice beach somewhere in south pacific.  Damn
the technology that helps  us to get in touch.
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Newsgroups: rec.audio,rec.video