vaeffects.lib
A library of virtual analog filter effects. Its official prefix is ve
.
References
Moog Filters
(ve.)moog_vcf
Moog "Voltage Controlled Filter" (VCF) in "analog" form. Moog VCF
implemented using the same logical block diagram as the classic
analog circuit. As such, it neglects the one-sample delay associated
with the feedback path around the four one-poles.
This extra delay alters the response, especially at high frequencies
(see reference [1] for details).
See moog_vcf_2b
below for a more accurate implementation.
Usage
_ : moog_vcf(res,fr) : _
Where:
res
: normalized amount of corner-resonance between 0 and 1 (0 is no resonance, 1 is maximum)fr
: corner-resonance frequency in Hz (less than SR/6.3 or so)
References
- https://ccrma.stanford.edu/~stilti/papers/moogvcf.pdf
- https://ccrma.stanford.edu/~jos/pasp/vegf.html
(ve.)moog_vcf_2b[n]
Moog "Voltage Controlled Filter" (VCF) as two biquads. Implementation
of the ideal Moog VCF transfer function factored into second-order
sections. As a result, it is more accurate than moog_vcf
above, but
its coefficient formulas are more complex when one or both parameters
are varied. Here, res is the fourth root of that in moog_vcf
, so, as
the sampling rate approaches infinity, moog_vcf(res,fr)
becomes equivalent
to moog_vcf_2b[n](res^4,fr)
(when res and fr are constant).
moog_vcf_2b
uses two direct-form biquads (tf2
).
moog_vcf_2bn
uses two protected normalized-ladder biquads (tf2np
).
Usage
_ : moog_vcf_2b(res,fr) : _
_ : moog_vcf_2bn(res,fr) : _
Where:
res
: normalized amount of corner-resonance between 0 and 1 (0 is min resonance, 1 is maximum)fr
: corner-resonance frequency in Hz
(ve.)moogLadder
Virtual analog model of the 4th-order Moog Ladder, which is arguably the most well-known ladder filter in analog synthesizers. Several 1st-order filters are cascaded in series. Feedback is then used, in part, to control the cut-off frequency and the resonance.
References
[Zavalishin 2012] (revision 2.1.2, February 2020):
This fix is based on Lorenzo Della Cioppa's correction to Pirkle's implementation; see this post: https://www.kvraudio.com/forum/viewtopic.php?f=33&t=571909
Usage
_ : moogLadder(normFreq,Q) : _
Where:
normFreq
: normalized frequency (0-1)Q
: quality factor between .707 (0 feedback coefficient) to 25 (feedback = 4, which is the self-oscillating threshold).
(ve.)moogHalfLadder
Virtual analog model of the 2nd-order Moog Half Ladder (simplified version of
(ve.)moogLadder
). Several 1st-order filters are cascaded in series.
Feedback is then used, in part, to control the cut-off frequency and the
resonance.
This filter was implemented in Faust by Eric Tarr during the 2019 Embedded DSP With Faust Workshop.
References
- https://www.willpirkle.com/app-notes/virtual-analog-moog-half-ladder-filter
- http://www.willpirkle.com/Downloads/AN-8MoogHalfLadderFilter.pdf
Usage
_ : moogHalfLadder(normFreq,Q) : _
Where:
normFreq
: normalized frequency (0-1)Q
: q
(ve.)diodeLadder
4th order virtual analog diode ladder filter. In addition to the individual
states used within each independent 1st-order filter, there are also additional
feedback paths found in the block diagram. These feedback paths are labeled
as connecting states. Rather than separately storing these connecting states
in the Faust implementation, they are simply implicitly calculated by
tracing back to the other states (s1
,s2
,s3
,s4
) each recursive step.
This filter was implemented in Faust by Eric Tarr during the 2019 Embedded DSP With Faust Workshop.
References
- https://www.willpirkle.com/virtual-analog-diode-ladder-filter/
- http://www.willpirkle.com/Downloads/AN-6DiodeLadderFilter.pdf
Usage
_ : diodeLadder(normFreq,Q) : _
Where:
normFreq
: normalized frequency (0-1)Q
: q
Korg 35 Filters
The following filters are virtual analog models of the Korg 35 low-pass filter and high-pass filter found in the MS-10 and MS-20 synthesizers. The virtual analog models for the LPF and HPF are different, making these filters more interesting than simply tapping different states of the same circuit.
These filters were implemented in Faust by Eric Tarr during the 2019 Embedded DSP With Faust Workshop.
Filter history:
(ve.)korg35LPF
Virtual analog models of the Korg 35 low-pass filter found in the MS-10 and MS-20 synthesizers.
Usage
_ : korg35LPF(normFreq,Q) : _
Where:
normFreq
: normalized frequency (0-1)Q
: q
(ve.)korg35HPF
Virtual analog models of the Korg 35 high-pass filter found in the MS-10 and MS-20 synthesizers.
Usage
_ : korg35HPF(normFreq,Q) : _
Where:
normFreq
: normalized frequency (0-1)Q
: q
Oberheim Filters
The following filter (4 types) is an implementation of the virtual analog model described in Section 7.2 of the Will Pirkle book, "Designing Software Synthesizer Plug-ins in C++". It is based on the block diagram in Figure 7.5.
The Oberheim filter is a state-variable filter with soft-clipping distortion within the circuit.
In many VA filters, distortion is accomplished using the "tanh" function.
For this Faust implementation, that distortion function was replaced with
the (ef.)cubicnl
function.
(ve.)oberheim
Generic multi-outputs Oberheim filter that produces the BSF, BPF, HPF and LPF outputs (see description above).
Usage
_ : oberheim(normFreq,Q) : _,_,_,_
Where:
normFreq
: normalized frequency (0-1)Q
: q
(ve.)oberheimBSF
Band-Stop Oberheim filter (see description above). Specialize the generic implementation: keep the first BSF output, the compiler will only generate the needed code.
Usage
_ : oberheimBSF(normFreq,Q) : _
Where:
normFreq
: normalized frequency (0-1)Q
: q
(ve.)oberheimBPF
Band-Pass Oberheim filter (see description above). Specialize the generic implementation: keep the second BPF output, the compiler will only generate the needed code.
Usage
_ : oberheimBPF(normFreq,Q) : _
Where:
normFreq
: normalized frequency (0-1)Q
: q
(ve.)oberheimHPF
High-Pass Oberheim filter (see description above). Specialize the generic implementation: keep the third HPF output, the compiler will only generate the needed code.
Usage
_ : oberheimHPF(normFreq,Q) : _
Where:
normFreq
: normalized frequency (0-1)Q
: q
(ve.)oberheimLPF
Low-Pass Oberheim filter (see description above). Specialize the generic implementation: keep the fourth LPF output, the compiler will only generate the needed code.
Usage
_ : oberheimLPF(normFreq,Q) : _
Where:
normFreq
: normalized frequency (0-1)Q
: q
Sallen Key Filters
The following filters were implemented based on VA models of synthesizer filters.
The modeling approach is based on a Topology Preserving Transform (TPT) to resolve the delay-free feedback loop in the corresponding analog filters.
The primary processing block used to build other filters (Moog, Korg, etc.) is based on a 1st-order Sallen-Key filter.
The filters included in this script are 1st-order LPF/HPF and 2nd-order state-variable filters capable of LPF, HPF, and BPF.
Resources:
- Vadim Zavalishin (2018) "The Art of VA Filter Design", v2.1.0
- https://www.native-instruments.com/fileadmin/ni_media/downloads/pdf/VAFilterDesign_2.1.0.pdf
- Will Pirkle (2014) "Resolving Delay-Free Loops in Recursive Filters Using
- the Modified Härmä Method", AES 137 http://www.aes.org/e-lib/browse.cfm?elib=17517
- Description and diagrams of 1st- and 2nd-order TPT filters:
- https://www.willpirkle.com/706-2/
(ve.)sallenKeyOnePole
Sallen-Key generic One Pole filter that produces the LPF and HPF outputs (see description above).
For the Faust implementation of this filter, recursion (letrec
) is used
for storing filter "states". The output (e.g. y
) is calculated by using
the input signal and the previous states of the filter.
During the current recursive step, the states of the filter (e.g. s
) for
the next step are also calculated.
Admittedly, this is not an efficient way to implement a filter because it
requires independently calculating the output and each state during each
recursive step. However, it works as a way to store and use "states"
within the constraints of Faust.
The simplest example is the 1st-order LPF (shown on the cover of Zavalishin
* 2018 and Fig 4.3 of https://www.willpirkle.com/706-2/). Here, the input
signal is split in parallel for the calculation of the output signal, y
, and
the state s
. The value of the state is only used for feedback to the next
step of recursion. It is blocked (!) from also being routed to the output.
A trick used for calculating the state s
is to observe that the input to
the delay block is the sum of two signal: what appears to be a feedforward
path and a feedback path. In reality, the signals being summed are identical
(signal*2) plus the value of the current state.
Usage
_ : sallenKeyOnePole(normFreq) : _,_
Where:
normFreq
: normalized frequency (0-1)
(ve.)sallenKeyOnePoleLPF
Sallen-Key One Pole lowpass filter (see description above). Specialize the generic implementation: keep the first LPF output, the compiler will only generate the needed code.
Usage
_ : sallenKeyOnePoleLPF(normFreq) : _
Where:
normFreq
: normalized frequency (0-1)
(ve.)sallenKeyOnePoleHPF
Sallen-Key One Pole Highpass filter (see description above). The dry input signal is routed in parallel to the output. The LPF'd signal is subtracted from the input so that the HPF remains. Specialize the generic implementation: keep the second HPF output, the compiler will only generate the needed code.
Usage
_ : sallenKeyOnePoleHPF(normFreq) : _
Where:
normFreq
: normalized frequency (0-1)
(ve.)sallenKey2ndOrder
Sallen-Key generic 2nd order filter that produces the LPF, BPF and HPF outputs.
This is a 2nd-order Sallen-Key state-variable filter. The idea is that by "tapping" into different points in the circuit, different filters (LPF,BPF,HPF) can be achieved. See Figure 4.6 of * https://www.willpirkle.com/706-2/
This is also a good example of the next step for generalizing the Faust
programming approach used for all these VA filters. In this case, there are
three things to calculate each recursive step (y
,s1
,s2
). For each thing, the
circuit is only calculated up to that point.
Comparing the LPF to BPF, the output signal (y
) is calculated similarly.
Except, the output of the BPF stops earlier in the circuit. Similarly, the
states (s1
and s2
) only differ in that s2
includes a couple more terms
beyond what is used for s1
.
Usage
_ : sallenKey2ndOrder(normFreq,Q) : _,_,_
Where:
normFreq
: normalized frequency (0-1)Q
: q
(ve.)sallenKey2ndOrderLPF
Sallen-Key 2nd order lowpass filter (see description above). Specialize the generic implementation: keep the first LPF output, the compiler will only generate the needed code.
Usage
_ : sallenKey2ndOrderLPF(normFreq,Q) : _
Where:
normFreq
: normalized frequency (0-1)Q
: q
(ve.)sallenKey2ndOrderBPF
Sallen-Key 2nd order bandpass filter (see description above). Specialize the generic implementation: keep the second BPF output, the compiler will only generate the needed code.
Usage
_ : sallenKey2ndOrderBPF(normFreq,Q) : _
Where:
normFreq
: normalized frequency (0-1)Q
: q
(ve.)sallenKey2ndOrderHPF
Sallen-Key 2nd order highpass filter (see description above). Specialize the generic implementation: keep the third HPF output, the compiler will only generate the needed code.
Usage
_ : sallenKey2ndOrderHPF(normFreq,Q) : _
Where:
normFreq
: normalized frequency (0-1)Q
: q
Effects
(ve.)wah4
Wah effect, 4th order.
wah4
is a standard Faust function.
Usage
_ : wah4(fr) : _
Where:
fr
: resonance frequency in Hz
Reference
(ve.)autowah
Auto-wah effect.
autowah
is a standard Faust function.
Usage
_ : autowah(level) : _
Where:
level
: amount of effect desired (0 to 1).
(ve.)crybaby
Digitized CryBaby wah pedal.
crybaby
is a standard Faust function.
Usage
_ : crybaby(wah) : _
Where:
wah
: "pedal angle" from 0 to 1
Reference
(ve.)vocoder
A very simple vocoder where the spectrum of the modulation signal
is analyzed using a filter bank.
vocoder
is a standard Faust function.
Usage
_ : vocoder(nBands,att,rel,BWRatio,source,excitation) : _
Where:
nBands
: Number of vocoder bandsatt
: Attack time in secondsrel
: Release time in secondsBWRatio
: Coefficient to adjust the bandwidth of each band (0.1 - 2)source
: Modulation signalexcitation
: Excitation/Carrier signal