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Some audio algorithms (asymmetric waveshaping, cascaded filters, ...) can produce DC offset. This offset can accumulate and reduce the signal/noise
ratio.
So, how to fix it? The example code from Julius O. Smith's document is:
...
y(n) = x(n) - x(n-1) + R * y(n-1)
// "R" between 0.9 .. 1
// n=current (n-1)=previous in/out value
...
"R" depends on sampling rate and the low frequency point. Do not set "R" to a fixed value (e.g. 0.99) if you don't know the
sample rate. Instead set R to:
(-3dB @ 40Hz): R = 1-(250/samplerate)
(-3dB @ 30Hz): R = 1-(190/samplerate)
(-3dB @ 20Hz): R = 1-(126/samplerate)
Comments
from : andy[DOT]rossol[AT]bluewin[DOT]ch
comment : I just received a mail from a musicdsp reader:
'How to calculate "R" for a given (-3dB) low frequency point?'
R = 1 - (pi*2 * frequency /samplerate)
(pi=3.14159265358979)
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