filters.lib

Faust Filters library. Its official prefix is fi.

The Filters library is organized into 19 sections:

  • Basic Filters
  • Comb Filters
  • Direct-Form Digital Filter Sections
  • Direct-Form Second-Order Biquad Sections
  • Ladder/Lattice Digital Filters
  • Useful Special Cases
  • Ladder/Lattice Allpass Filters
  • Digital Filter Sections Specified as Analog Filter Sections
  • Simple Resonator Filters
  • Butterworth Lowpass/Highpass Filters
  • Special Filter-Bank Delay-Equalizing Allpass Filters
  • Elliptic (Cauer) Lowpass Filters
  • Elliptic Highpass Filters
  • Butterworth Bandpass/Bandstop Filters
  • Elliptic Bandpass Filters
  • Parametric Equalizers (Shelf, Peaking)
  • Mth-Octave Filter-Banks
  • Arbitrary-Crossover Filter-Banks and Spectrum Analyzers
  • SVF filters

Basic Filters


(fi.)zero

One zero filter. Difference equation: .

Usage

_ : zero(z) : _

Where:

  • z: location of zero along real axis in z-plane

Reference


(fi.)pole

One pole filter. Could also be called a "leaky integrator". Difference equation: .

Usage

_ : pole(p) : _

Where:

  • p: pole location = feedback coefficient

Reference


(fi.)integrator

Same as pole(1) [implemented separately for block-diagram clarity].


(fi.)dcblockerat

DC blocker with configurable break frequency. The amplitude response is substantially flat above , and sloped at about +6 dB/octave below . Derived from the analog transfer function (which can be seen as a 1st-order Butterworth highpass filter) by the low-frequency-matching bilinear transform method (i.e., the standard frequency-scaling constant 2*SR).

Usage

_ : dcblockerat(fb) : _

Where:

  • fb: "break frequency" in Hz, i.e., -3 dB gain frequency.

Reference


(fi.)dcblocker

DC blocker. Default dc blocker has -3dB point near 35 Hz (at 44.1 kHz) and high-frequency gain near 1.0025 (due to no scaling). dcblocker is as standard Faust function.

Usage

_ : dcblocker : _

(fi.)lptN

One-pole lowpass filter with arbitrary dis/charging factors set in dB and times set in seconds.

Usage

_ : lptN(N, tN) : _

Where:

  • N: is the attenuation factor in dB
  • tN: is the filter period in seconds, that is, the time for the impulse response to decay by N dB

Reference

Comb Filters


(fi.)ff_comb

Feed-Forward Comb Filter. Note that ff_comb requires integer delays (uses delay internally). ff_comb is a standard Faust function.

Usage

_ : ff_comb(maxdel,intdel,b0,bM) : _

Where:

  • maxdel: maximum delay (a power of 2)
  • intdel: current (integer) comb-filter delay between 0 and maxdel
  • del: current (float) comb-filter delay between 0 and maxdel
  • b0: gain applied to delay-line input
  • bM: gain applied to delay-line output and then summed with input

Reference


(fi.)ff_fcomb

Feed-Forward Comb Filter. Note that ff_fcomb takes floating-point delays (uses fdelay internally). ff_fcomb is a standard Faust function.

Usage

_ : ff_fcomb(maxdel,del,b0,bM) : _

Where:

  • maxdel: maximum delay (a power of 2)
  • intdel: current (integer) comb-filter delay between 0 and maxdel
  • del: current (float) comb-filter delay between 0 and maxdel
  • b0: gain applied to delay-line input
  • bM: gain applied to delay-line output and then summed with input

Reference


(fi.)ffcombfilter

Typical special case of ff_comb() where: b0 = 1.


(fi.)fb_comb

Feed-Back Comb Filter (integer delay).

Usage

_ : fb_comb(maxdel,intdel,b0,aN) : _

Where:

  • maxdel: maximum delay (a power of 2)
  • intdel: current (integer) comb-filter delay between 0 and maxdel
  • del: current (float) comb-filter delay between 0 and maxdel
  • b0: gain applied to delay-line input and forwarded to output
  • aN: minus the gain applied to delay-line output before summing with the input and feeding to the delay line

Reference


(fi.)fb_fcomb

Feed-Back Comb Filter (floating point delay).

Usage

_ : fb_fcomb(maxdel,del,b0,aN) : _

Where:

  • maxdel: maximum delay (a power of 2)
  • intdel: current (integer) comb-filter delay between 0 and maxdel
  • del: current (float) comb-filter delay between 0 and maxdel
  • b0: gain applied to delay-line input and forwarded to output
  • aN: minus the gain applied to delay-line output before summing with the input and feeding to the delay line

Reference


(fi.)rev1

Special case of fb_comb (rev1(maxdel,N,g)). The "rev1 section" dates back to the 1960s in computer-music reverberation. See the jcrev and brassrev in reverbs.lib for usage examples.


(fi.)fbcombfilter and (fi.)ffbcombfilter

Other special cases of Feed-Back Comb Filter.

Usage

_ : fbcombfilter(maxdel,intdel,g) : _
_ : ffbcombfilter(maxdel,del,g) : _

Where:

  • maxdel: maximum delay (a power of 2)
  • intdel: current (integer) comb-filter delay between 0 and maxdel
  • del: current (float) comb-filter delay between 0 and maxdel
  • g: feedback gain

Reference


(fi.)allpass_comb

Schroeder Allpass Comb Filter. Note that:

allpass_comb(maxlen,len,aN) = ff_comb(maxlen,len,aN,1) : fb_comb(maxlen,len-1,1,aN);

which is a direct-form-1 implementation, requiring two delay lines. The implementation here is direct-form-2 requiring only one delay line.

Usage

_ : allpass_comb(maxdel,intdel,aN) : _

Where:

  • maxdel: maximum delay (a power of 2)
  • intdel: current (integer) comb-filter delay between 0 and maxdel
  • del: current (float) comb-filter delay between 0 and maxdel
  • aN: minus the feedback gain

References


(fi.)allpass_fcomb

Schroeder Allpass Comb Filter. Note that:

allpass_comb(maxlen,len,aN) = ff_comb(maxlen,len,aN,1) : fb_comb(maxlen,len-1,1,aN);

which is a direct-form-1 implementation, requiring two delay lines. The implementation here is direct-form-2 requiring only one delay line.

allpass_fcomb is a standard Faust library.

Usage

_ : allpass_comb(maxdel,intdel,aN) : _
_ : allpass_fcomb(maxdel,del,aN) : _

Where:

  • maxdel: maximum delay (a power of 2)
  • intdel: current (float) comb-filter delay between 0 and maxdel
  • del: current (float) comb-filter delay between 0 and maxdel
  • aN: minus the feedback gain

References


(fi.)rev2

Special case of allpass_comb (rev2(maxlen,len,g)). The "rev2 section" dates back to the 1960s in computer-music reverberation. See the jcrev and brassrev in reverbs.lib for usage examples.


(fi.)allpass_fcomb5 and (fi.)allpass_fcomb1a

Same as allpass_fcomb but use fdelay5 and fdelay1a internally (Interpolation helps - look at an fft of faust2octave on

`1-1' <: allpass_fcomb(1024,10.5,0.95), allpass_fcomb5(1024,10.5,0.95);`).

Direct-Form Digital Filter Sections


(fi.)iir

Nth-order Infinite-Impulse-Response (IIR) digital filter, implemented in terms of the Transfer-Function (TF) coefficients. Such filter structures are termed "direct form".

iir is a standard Faust function.

Usage

  _ : iir(bcoeffs,acoeffs) : _

Where:

  • order: filter order (int) = max(#poles,#zeros)
  • bcoeffs: (b0,b1,...,b_order) = TF numerator coefficients
  • acoeffs: (a1,...,a_order) = TF denominator coeffs (a0=1)

Reference


(fi.)fir

FIR filter (convolution of FIR filter coefficients with a signal).

Usage

_ : fir(bv) : _

fir is standard Faust function.

Where:

  • bv = b0,b1,...,bn is a parallel bank of coefficient signals.

Note

bv is processed using pattern-matching at compile time, so it must have this normal form (parallel signals).

Example

Smoothing white noise with a five-point moving average:

bv = .2,.2,.2,.2,.2;
process = noise : fir(bv);

Equivalent (note double parens):

process = noise : fir((.2,.2,.2,.2,.2));

(fi.)conv and (fi.)convN

Convolution of input signal with given coefficients.

Usage

_ : conv((k1,k2,k3,...,kN)) : _; // Argument = one signal bank
_ : convN(N,(k1,k2,k3,...)) : _; // Useful when N < count((k1,...))

(fi.)tf1, (fi.)tf2 and (fi.)tf3

tfN = N'th-order direct-form digital filter.

Usage

_ : tf1(b0,b1,a1) : _
_ : tf2(b0,b1,b2,a1,a2) : _
_ : tf3(b0,b1,b2,b3,a1,a2,a3) : _

Where:

  • a: the poles
  • b: the zeros

Reference


(fi.)notchw

Simple notch filter based on a biquad (tf2). notchw is a standard Faust function.

Usage:

_ : notchw(width,freq) : _

Where:

  • width: "notch width" in Hz (approximate)
  • freq: "notch frequency" in Hz

Reference

Direct-Form Second-Order Biquad Sections

Direct-Form Second-Order Biquad Sections

Reference


(fi.)tf21, (fi.)tf22, (fi.)tf22t and (fi.)tf21t

tfN = N'th-order direct-form digital filter where:

  • tf21 is tf2, direct-form 1
  • tf22 is tf2, direct-form 2
  • tf22t is tf2, direct-form 2 transposed
  • tf21t is tf2, direct-form 1 transposed

Usage

_ : tf21(b0,b1,b2,a1,a2) : _
_ : tf22(b0,b1,b2,a1,a2) : _
_ : tf22t(b0,b1,b2,a1,a2) : _
_ : tf21t(b0,b1,b2,a1,a2) : _

Where:

  • a: the poles
  • b: the zeros

Reference

Ladder/Lattice Digital Filters

Ladder and lattice digital filters generally have superior numerical properties relative to direct-form digital filters. They can be derived from digital waveguide filters, which gives them a physical interpretation.

Reference


(fi.)av2sv

Compute reflection coefficients sv from transfer-function denominator av.

Usage

sv = av2sv(av)

Where:

  • av: parallel signal bank a1,...,aN
  • sv: parallel signal bank s1,...,sN

where ro = ith reflection coefficient, and ai = coefficient of z^(-i) in the filter transfer-function denominator A(z).

Reference


(fi.)bvav2nuv

Compute lattice tap coefficients from transfer-function coefficients.

Usage

nuv = bvav2nuv(bv,av)

Where:

  • av: parallel signal bank a1,...,aN
  • bv: parallel signal bank b0,b1,...,aN
  • nuv: parallel signal bank nu1,...,nuN

where nui is the i'th tap coefficient, bi is the coefficient of z^(-i) in the filter numerator, ai is the coefficient of z^(-i) in the filter denominator


(fi.)iir_lat2

Two-multiply latice IIR filter of arbitrary order.

Usage

_ : iir_lat2(bv,av) : _

Where:

  • bv: zeros as a bank of parallel signals
  • av: poles as a bank of parallel signals

(fi.)allpassnt

Two-multiply lattice allpass (nested order-1 direct-form-ii allpasses).

Usage

_ : allpassnt(n,sv) : _

Where:

  • n: the order of the filter
  • sv: the reflection coefficients (-1 1)

(fi.)iir_kl

Kelly-Lochbaum ladder IIR filter of arbitrary order.

Usage

_ : iir_kl(bv,av) : _

Where:

  • bv: zeros as a bank of parallel signals
  • av: poles as a bank of parallel signals

(fi.)allpassnklt

Kelly-Lochbaum ladder allpass.

Usage:

_ : allpassklt(n,sv) : _

Where:

  • n: the order of the filter
  • sv: the reflection coefficients (-1 1)

(fi.)iir_lat1

One-multiply latice IIR filter of arbitrary order.

Usage

_ : iir_lat1(bv,av) : _

Where:

  • bv: zeros as a bank of parallel signals
  • av: poles as a bank of parallel signals

(fi.)allpassn1mt

One-multiply lattice allpass with tap lines.

Usage

_ : allpassn1mt(N,sv) : _

Where:

  • N: the order of the filter (fixed at compile time)
  • sv: the reflection coefficients (-1 1)

(fi.)iir_nl

Normalized ladder filter of arbitrary order.

Usage

_ : iir_nl(bv,av) : _

Where:

  • bv: zeros as a bank of parallel signals
  • av: poles as a bank of parallel signals

References


(fi.)allpassnnlt

Normalized ladder allpass filter of arbitrary order.

Usage:

_ : allpassnnlt(N,sv) : _

Where:

  • N: the order of the filter (fixed at compile time)
  • sv: the reflection coefficients (-1,1)

References

Useful Special Cases


(fi.)tf2np

Biquad based on a stable second-order Normalized Ladder Filter (more robust to modulation than tf2 and protected against instability).

Usage

_ : tf2np(b0,b1,b2,a1,a2) : _

Where:

  • a: the poles
  • b: the zeros

(fi.)wgr

Second-order transformer-normalized digital waveguide resonator.

Usage

_ : wgr(f,r) : _

Where:

  • f: resonance frequency (Hz)
  • r: loss factor for exponential decay (set to 1 to make a numerically stable oscillator)

References


(fi.)nlf2

Second order normalized digital waveguide resonator.

Usage

_ : nlf2(f,r) : _

Where:

  • f: resonance frequency (Hz)
  • r: loss factor for exponential decay (set to 1 to make a sinusoidal oscillator)

Reference


(fi.)apnl

Passive Nonlinear Allpass based on Pierce switching springs idea. Switch between allpass coefficient a1 and a2 at signal zero crossings.

Usage

_ : apnl(a1,a2) : _

Where:

  • a1 and a2: allpass coefficients

Reference

  • "A Passive Nonlinear Digital Filter Design ..." by John R. Pierce and Scott A. Van Duyne, JASA, vol. 101, no. 2, pp. 1120-1126, 1997

Ladder/Lattice Allpass Filters

An allpass filter has gain 1 at every frequency, but variable phase. Ladder/lattice allpass filters are specified by reflection coefficients. They are defined here as nested allpass filters, hence the names allpassn*.

References


(fi.)allpassn

Two-multiply lattice - each section is two multiply-adds.

Usage:

_ : allpassn(n,sv) : _

Where:

  • n: the order of the filter
  • sv: the reflection coefficients (-1 1)

References

  • J. O. Smith and R. Michon, "Nonlinear Allpass Ladder Filters in FAUST", in Proceedings of the 14th International Conference on Digital Audio Effects (DAFx-11), Paris, France, September 19-23, 2011.

(fi.)allpassnn

Normalized form - four multiplies and two adds per section, but coefficients can be time varying and nonlinear without "parametric amplification" (modulation of signal energy).

Usage:

_ : allpassnn(n,tv) : _

Where:

  • n: the order of the filter
  • tv: the reflection coefficients (-PI PI)

(fi.)allpasskl

Kelly-Lochbaum form - four multiplies and two adds per section, but all signals have an immediate physical interpretation as traveling pressure waves, etc.

Usage:

_ : allpassnkl(n,sv) : _

Where:

  • n: the order of the filter
  • sv: the reflection coefficients (-1 1)

(fi.)allpass1m

One-multiply form - one multiply and three adds per section. Normally the most efficient in special-purpose hardware.

Usage:

_ : allpassn1m(n,sv) : _

Where:

  • n: the order of the filter
  • sv: the reflection coefficients (-1 1)

Digital Filter Sections Specified as Analog Filter Sections


(fi.)tf2s and (fi.)tf2snp

Second-order direct-form digital filter, specified by ANALOG transfer-function polynomials B(s)/A(s), and a frequency-scaling parameter. Digitization via the bilinear transform is built in.

Usage

_ : tf2s(b2,b1,b0,a1,a0,w1) : _

Where:

        b2 s^2 + b1 s + b0
H(s) = --------------------
           s^2 + a1 s + a0

and w1 is the desired digital frequency (in radians/second) corresponding to analog frequency 1 rad/sec (i.e., s = j).

Example

A second-order ANALOG Butterworth lowpass filter, normalized to have cutoff frequency at 1 rad/sec, has transfer function

             1
H(s) = -----------------
        s^2 + a1 s + 1

where a1 = sqrt(2). Therefore, a DIGITAL Butterworth lowpass cutting off at SR/4 is specified as tf2s(0,0,1,sqrt(2),1,PI*SR/2);

Method

Bilinear transform scaled for exact mapping of w1.

Reference


(fi.)tf3slf

Analogous to tf2s above, but third order, and using the typical low-frequency-matching bilinear-transform constant 2/T ("lf" series) instead of the specific-frequency-matching value used in tf2s and tf1s. Note the lack of a "w1" argument.

Usage

_ : tf3slf(b3,b2,b1,b0,a3,a2,a1,a0) : _

(fi.)tf1s

First-order direct-form digital filter, specified by ANALOG transfer-function polynomials B(s)/A(s), and a frequency-scaling parameter.

Usage

tf1s(b1,b0,a0,w1)

Where:

   b1 s + b0

H(s) = ---------- s + a0

and w1 is the desired digital frequency (in radians/second) corresponding to analog frequency 1 rad/sec (i.e., s = j).

Example

A first-order ANALOG Butterworth lowpass filter, normalized to have cutoff frequency at 1 rad/sec, has transfer function

      1

H(s) = ------- s + 1

so b0 = a0 = 1 and b1 = 0. Therefore, a DIGITAL first-order Butterworth lowpass with gain -3dB at SR/4 is specified as

tf1s(0,1,1,PI*SR/2); // digital half-band order 1 Butterworth

Method

Bilinear transform scaled for exact mapping of w1.

Reference


(fi.)tf2sb

Bandpass mapping of tf2s: In addition to a frequency-scaling parameter w1 (set to HALF the desired passband width in rad/sec), there is a desired center-frequency parameter wc (also in rad/s). Thus, tf2sb implements a fourth-order digital bandpass filter section specified by the coefficients of a second-order analog lowpass prototype section. Such sections can be combined in series for higher orders. The order of mappings is (1) frequency scaling (to set lowpass cutoff w1), (2) bandpass mapping to wc, then (3) the bilinear transform, with the usual scale parameter 2*SR. Algebra carried out in maxima and pasted here.

Usage

_ : tf2sb(b2,b1,b0,a1,a0,w1,wc) : _

(fi.)tf1sb

First-to-second-order lowpass-to-bandpass section mapping, analogous to tf2sb above.

Usage

_ : tf1sb(b1,b0,a0,w1,wc) : _

Simple Resonator Filters


(fi.)resonlp

Simple resonant lowpass filter based on tf2s (virtual analog). resonlp is a standard Faust function.

Usage

_ : resonlp(fc,Q,gain) : _
_ : resonhp(fc,Q,gain) : _
_ : resonbp(fc,Q,gain) : _

Where:

  • fc: center frequency (Hz)
  • Q: q
  • gain: gain (0-1)

(fi.)resonhp

Simple resonant highpass filters based on tf2s (virtual analog). resonhp is a standard Faust function.

Usage

_ : resonlp(fc,Q,gain) : _
_ : resonhp(fc,Q,gain) : _
_ : resonbp(fc,Q,gain) : _

Where:

  • fc: center frequency (Hz)
  • Q: q
  • gain: gain (0-1)

(fi.)resonbp

Simple resonant bandpass filters based on tf2s (virtual analog). resonbp is a standard Faust function.

Usage

_ : resonlp(fc,Q,gain) : _
_ : resonhp(fc,Q,gain) : _
_ : resonbp(fc,Q,gain) : _

Where:

  • fc: center frequency (Hz)
  • Q: q
  • gain: gain (0-1)

Butterworth Lowpass/Highpass Filters


(fi.)lowpass

Nth-order Butterworth lowpass filter. lowpass is a standard Faust function.

Usage

_ : lowpass(N,fc) : _

Where:

  • N: filter order (number of poles) [nonnegative constant integer]
  • fc: desired cut-off frequency (-3dB frequency) in Hz

References


(fi.)highpass

Nth-order Butterworth highpass filters. highpass is a standard Faust function.

Usage

_ : highpass(N,fc) : _

Where:

  • N: filter order (number of poles) [nonnegative constant integer]
  • fc: desired cut-off frequency (-3dB frequency) in Hz

References


(fi.)lowpass0_highpass1

Special Filter-Bank Delay-Equalizing Allpass Filters

These special allpass filters are needed by filterbank et al. below. They are equivalent to (lowpass(N,fc) +|- highpass(N,fc))/2, but with canceling pole-zero pairs removed (which occurs for odd N).


(fi.)lowpass_plus|minus_highpass

Catch-all definitions for generality - even order is done: Catch-all definitions for generality - even order is done: FIXME: Rewrite the following, as for orders 3 and 5 above, to eliminate pole-zero cancellations: FIXME: Rewrite the following, as for orders 3 and 5 above, to eliminate pole-zero cancellations:

Elliptic (Cauer) Lowpass Filters

Elliptic (Cauer) Lowpass Filters

References

<http://en.wikipedia.org/wiki/Elliptic_filter * functions ncauer and ellip in Octave


(fi.)lowpass3e

Third-order Elliptic (Cauer) lowpass filter.

Usage

_ : lowpass3e(fc) : _

Where:

  • fc: -3dB frequency in Hz

Design

For spectral band-slice level display (see octave_analyzer3e):

[z,p,g] = ncauer(Rp,Rs,3);  % analog zeros, poles, and gain, where
Rp = 60  % dB ripple in stopband
Rs = 0.2 % dB ripple in passband

(fi.)lowpass6e

Sixth-order Elliptic/Cauer lowpass filter.

Usage

_ : lowpass6e(fc) : _

Where:

  • fc: -3dB frequency in Hz

Design

For spectral band-slice level display (see octave_analyzer6e):

[z,p,g] = ncauer(Rp,Rs,6);  % analog zeros, poles, and gain, where
 Rp = 80  % dB ripple in stopband
 Rs = 0.2 % dB ripple in passband

Elliptic Highpass Filters


(fi.)highpass3e

Third-order Elliptic (Cauer) highpass filter. Inversion of lowpass3e wrt unit circle in s plane (s <- 1/s)

Usage

_ : highpass3e(fc) : _

Where:

  • fc: -3dB frequency in Hz

(fi.)highpass6e

Sixth-order Elliptic/Cauer highpass filter. Inversion of lowpass3e wrt unit circle in s plane (s <- 1/s)

Usage

_ : highpass6e(fc) : _

Where:

  • fc: -3dB frequency in Hz

Butterworth Bandpass/Bandstop Filters


(fi.)bandpass

Order 2*Nh Butterworth bandpass filter made using the transformation s <- s + wc^2/s on lowpass(Nh), where wc is the desired bandpass center frequency. The lowpass(Nh) cutoff w1 is half the desired bandpass width. bandpass is a standard Faust function.

Usage

_ : bandpass(Nh,fl,fu) : _

Where:

  • Nh: HALF the desired bandpass order (which is therefore even)
  • fl: lower -3dB frequency in Hz
  • fu: upper -3dB frequency in Hz Thus, the passband width is fu-fl, and its center frequency is (fl+fu)/2.

Reference


(fi.)bandstop

Order 2*Nh Butterworth bandstop filter made using the transformation s <- s + wc^2/s on highpass(Nh), where wc is the desired bandpass center frequency. The highpass(Nh) cutoff w1 is half the desired bandpass width. bandstop is a standard Faust function.

Usage

_ : bandstop(Nh,fl,fu) : _

Where:

  • Nh: HALF the desired bandstop order (which is therefore even)
  • fl: lower -3dB frequency in Hz
  • fu: upper -3dB frequency in Hz Thus, the passband (stopband) width is fu-fl, and its center frequency is (fl+fu)/2.

Reference

Elliptic Bandpass Filters


(fi.)bandpass6e

Order 12 elliptic bandpass filter analogous to bandpass(6).


(fi.)bandpass12e

Order 24 elliptic bandpass filter analogous to bandpass(6).


(fi.)pospass

Positive-Pass Filter (single-side-band filter)

Usage

_ : pospass(N,fc) : _,_

where

  • N: filter order (Butterworth bandpass for positive frequencies).
  • fc: lower bandpass cutoff frequency in Hz.
  • Highpass cutoff frequency at ma.SR/2 - fc Hz.

Example

  • See dm.pospass_demo
  • Look at frequency response:

Method

A filter passing only positive frequencies can be made from a half-band lowpass by modulating it up to the positive-frequency range. Equivalently, down-modulate the input signal using a complex sinusoid at -SR/4 Hz, lowpass it with a half-band filter, and modulate back up by SR/4 Hz. In Faust/math notation: pospass(N) =

An approximation to the Hilbert transform is given by the imaginary output signal:

hilbert(N) = pospass(N) : !,*(2);

References

Parametric Equalizers (Shelf, Peaking)

Parametric Equalizers (Shelf, Peaking).

References


(fi.)low_shelf

First-order "low shelf" filter (gain boost|cut between dc and some frequency) low_shelf is a standard Faust function.

Usage

_ : lowshelf(N,L0,fx) : _
_ : low_shelf(L0,fx) : _ // default case (order 3)
_ : lowshelf_other_freq(N,L0,fx) : _

Where: * N: filter order 1, 3, 5, ... (odd only). (default should be 3) * L0: desired level (dB) between dc and fx (boost L0>0 or cut L0<0) * fx: -3dB frequency of lowpass band (L0>0) or upper band (L0<0) (see "SHELF SHAPE" below).

The gain at SR/2 is constrained to be 1. The generalization to arbitrary odd orders is based on the well known fact that odd-order Butterworth band-splits are allpass-complementary (see filterbank documentation below for references).

Shelf Shape

The magnitude frequency response is approximately piecewise-linear on a log-log plot ("BODE PLOT"). The Bode "stick diagram" approximation L(lf) is easy to state in dB versus dB-frequency lf = dB(f):

  • L0 > 0:
    • L(lf) = L0, f between 0 and fx = 1st corner frequency;
    • L(lf) = L0 - N * (lf - lfx), f between fx and f2 = 2nd corner frequency;
    • L(lf) = 0, lf > lf2.
    • lf2 = lfx + L0/N = dB-frequency at which level gets back to 0 dB.
  • L0 < 0:
    • L(lf) = L0, f between 0 and f1 = 1st corner frequency;
    • L(lf) = - N * (lfx - lf), f between f1 and lfx = 2nd corner frequency;
    • L(lf) = 0, lf > lfx.
    • lf1 = lfx + L0/N = dB-frequency at which level goes up from L0.

See lowshelf_other_freq.


(fi.)high_shelf

First-order "high shelf" filter (gain boost|cut above some frequency). high_shelf is a standard Faust function.

Usage

_ : highshelf(N,Lpi,fx) : _
_ : high_shelf(L0,fx) : _ // default case (order 3)
_ : highshelf_other_freq(N,Lpi,fx) : _

Where:

  • N: filter order 1, 3, 5, ... (odd only).
  • Lpi: desired level (dB) between fx and SR/2 (boost Lpi>0 or cut Lpi<0)
  • fx: -3dB frequency of highpass band (L0>0) or lower band (L0<0) (Use highshelf_other_freq() below to find the other one.)

The gain at dc is constrained to be 1. See lowshelf documentation above for more details on shelf shape.


(fi.)peak_eq

Second order "peaking equalizer" section (gain boost or cut near some frequency) Also called a "parametric equalizer" section. peak_eq is a standard Faust function.

Usage

_ : peak_eq(Lfx,fx,B) : _;

Where:

  • Lfx: level (dB) at fx (boost Lfx>0 or cut Lfx<0)
  • fx: peak frequency (Hz)
  • B: bandwidth (B) of peak in Hz

(fi.)peak_eq_cq

Constant-Q second order peaking equalizer section.

Usage

_ : peak_eq_cq(Lfx,fx,Q) : _;

Where:

  • Lfx: level (dB) at fx
  • fx: boost or cut frequency (Hz)
  • Q: "Quality factor" = fx/B where B = bandwidth of peak in Hz

(fi.)peak_eq_rm

Regalia-Mitra second order peaking equalizer section.

Usage

_ : peak_eq_rm(Lfx,fx,tanPiBT) : _;

Where:

  • Lfx: level (dB) at fx
  • fx: boost or cut frequency (Hz)
  • tanPiBT: tan(PI*B/SR), where B = -3dB bandwidth (Hz) when 10^(Lfx/20) = 0 ~ PI*B/SR for narrow bandwidths B

Reference

P.A. Regalia, S.K. Mitra, and P.P. Vaidyanathan, "The Digital All-Pass Filter: A Versatile Signal Processing Building Block" Proceedings of the IEEE, 76(1):19-37, Jan. 1988. (See pp. 29-30.)


(fi.)spectral_tilt

Spectral tilt filter, providing an arbitrary spectral rolloff factor alpha in (-1,1), where -1 corresponds to one pole (-6 dB per octave), and +1 corresponds to one zero (+6 dB per octave). In other words, alpha is the slope of the ln magnitude versus ln frequency. For a "pinking filter" (e.g., to generate 1/f noise from white noise), set alpha to -1/2.

Usage

_ : spectral_tilt(N,f0,bw,alpha) : _

Where:

  • N: desired integer filter order (fixed at compile time)
  • f0: lower frequency limit for desired roll-off band > 0
  • bw: bandwidth of desired roll-off band
  • alpha: slope of roll-off desired in nepers per neper, between -1 and 1 (ln mag / ln radian freq)

Examples

See spectral_tilt_demo.

Reference

J.O. Smith and H.F. Smith, "Closed Form Fractional Integration and Differentiation via Real Exponentially Spaced Pole-Zero Pairs", arXiv.org publication arXiv:1606.06154 [cs.CE], June 7, 2016, http://arxiv.org/abs/1606.06154


(fi.)levelfilter

Dynamic level lowpass filter. levelfilter is a standard Faust function.

Usage

_ : levelfilter(L,freq) : _

Where:

  • L: desired level (in dB) at Nyquist limit (SR/2), e.g., -60
  • freq: corner frequency (-3dB point) usually set to fundamental freq
  • N: Number of filters in series where L = L/N

Reference


(fi.)levelfilterN

Dynamic level lowpass filter.

Usage

_ : levelfilterN(N,freq,L) : _

Where:

  • L: desired level (in dB) at Nyquist limit (SR/2), e.g., -60
  • freq: corner frequency (-3dB point) usually set to fundamental freq
  • N: Number of filters in series where L = L/N

Reference

Mth-Octave Filter-Banks

Mth-octave filter-banks split the input signal into a bank of parallel signals, one for each spectral band. They are related to the Mth-Octave Spectrum-Analyzers in analysis.lib. The documentation of this library contains more details about the implementation. The parameters are:

  • M: number of band-slices per octave (>1)
  • N: total number of bands (>2)
  • ftop: upper bandlimit of the Mth-octave bands (<SR/2)

In addition to the Mth-octave output signals, there is a highpass signal containing frequencies from ftop to SR/2, and a "dc band" lowpass signal containing frequencies from 0 (dc) up to the start of the Mth-octave bands. Thus, the N output signals are

highpass(ftop), MthOctaveBands(M,N-2,ftop), dcBand(ftop*2^(-M*(N-1)))

A Filter-Bank is defined here as a signal bandsplitter having the property that summing its output signals gives an allpass-filtered version of the filter-bank input signal. A more conventional term for this is an "allpass-complementary filter bank". If the allpass filter is a pure delay (and possible scaling), the filter bank is said to be a "perfect-reconstruction filter bank" (see Vaidyanathan-1993 cited below for details). A "graphic equalizer", in which band signals are scaled by gains and summed, should be based on a filter bank.

The filter-banks below are implemented as Butterworth or Elliptic spectrum-analyzers followed by delay equalizers that make them allpass-complementary.

Increasing Channel Isolation

Go to higher filter orders - see Regalia et al. or Vaidyanathan (cited below) regarding the construction of more aggressive recursive filter-banks using elliptic or Chebyshev prototype filters.

References

  • "Tree-structured complementary filter banks using all-pass sections", Regalia et al., IEEE Trans. Circuits & Systems, CAS-34:1470-1484, Dec. 1987
  • "Multirate Systems and Filter Banks", P. Vaidyanathan, Prentice-Hall, 1993
  • Elementary filter theory: https://ccrma.stanford.edu/~jos/filters/

(fi.)mth_octave_filterbank[n]

Allpass-complementary filter banks based on Butterworth band-splitting. For Butterworth band-splits, the needed delay equalizer is easily found.

Usage

_ : mth_octave_filterbank(O,M,ftop,N) : par(i,N,_); // Oth-order
_ : mth_octave_filterbank_alt(O,M,ftop,N) : par(i,N,_); // dc-inverted version

Also for convenience:

_ : mth_octave_filterbank3(M,ftop,N) : par(i,N,_); // 3rd-order Butterworth
_ : mth_octave_filterbank5(M,ftop,N) : par(i,N,_); // 5th-order Butterworth
mth_octave_filterbank_default = mth_octave_filterbank5;

Where:

  • O: order of filter used to split each frequency band into two
  • M: number of band-slices per octave
  • ftop: highest band-split crossover frequency (e.g., 20 kHz)
  • N: total number of bands (including dc and Nyquist)

Arbitrary-Crossover Filter-Banks and Spectrum Analyzers

These are similar to the Mth-octave analyzers above, except that the band-split frequencies are passed explicitly as arguments.


(fi.)filterbank

Filter bank. filterbank is a standard Faust function.

Usage

_ : filterbank (O,freqs) : par(i,N,_); // Butterworth band-splits

Where:

  • O: band-split filter order (ODD integer required for filterbank[i])
  • freqs: (fc1,fc2,...,fcNs) [in numerically ascending order], where Ns=N-1 is the number of octave band-splits (total number of bands N=Ns+1).

If frequencies are listed explicitly as arguments, enclose them in parens:

_ : filterbank(3,(fc1,fc2)) : _,_,_

(fi.)filterbanki

Inverted-dc filter bank.

Usage

_ : filterbanki(O,freqs) : par(i,N,_); // Inverted-dc version

Where:

  • O: band-split filter order (ODD integer required for filterbank[i])
  • freqs: (fc1,fc2,...,fcNs) [in numerically ascending order], where Ns=N-1 is the number of octave band-splits (total number of bands N=Ns+1).

If frequencies are listed explicitly as arguments, enclose them in parens:

_ : filterbanki(3,(fc1,fc2)) : _,_,_

SVF Filters

References

Solving the continuous SVF equations using trapezoidal integration * https://cytomic.com/files/dsp/SvfLinearTrapOptimised2.pdf


(fi.)svf

An environment with lp, bp, hp, notch, peak, ap, bell, ls, hs SVF based filters. All filters have freq and Q parameters, the bell, ls, hs ones also have a gain third parameter.

Usage

_ : svf.xx(freq, Q, [gain]) : _

Where:

  • freq: cut frequency
  • Q: quality factor
  • [gain]: gain in dB