Отправлено Oval 14 февраля 2005 г. 15:56 В ответ на: Ответ: отправлено st256 14 февраля 2005 г. 13:16

You will want to read two papers:
"The ARMAdillo Coefficient Encoding Scheme for Digital Audio Filters",
Dave Rossum, IEEE ASSP Workshop on WASPAA, 1991
and
"Filter Morphing of Parametric Equalizers and Shelving Filters for Audio
Signal Processing", Yinong Ding & David Rossum, JAES vol 43, Oct 1995

Basically they take "snapshots" of normal IIRs made of second order
sections, encode the coefficients with the ARMAdillo encoding and linearly
interpolate the encoded coefficients.

To make a resonant lowpass filter, you'd design a normal IIR filter at
the following four (constant) settings:
low cutoff no resonance
high cutoff no resonance
low cutoff high resonance
high cutoff high resonance

Then you'd encode the coefficients and linearly interpolate between the
encoded coefficients (bilinear interpolation as there are two axis). For
each sample interval, you'd decode the interpolated ARMAdillo coefficients
and update your second order coefficients.

• Твоя фирма членствует в AES? — Oval   (14.02.2005 16:14, 114 байт)