Filter & Amp Sections
Filter Section
Within each Machine, the sound source is always routed through a similar Filter section, available on Tab 3.
Each Filter section then goes to an Amp section before going to the Mixer.
The Filter section is always on Tab 3 in every Machine.
This filter tab has 3 pages containing different controls. When a Tab header displays small bar icons, click its corresponding
button underneath to jump between its pages.
Main screen of Filter section
Page 1:
| Filter 1 Frequency | Filter 1 Resonance | (Filter Morph) -or- (Filter Gain) |
Filter 1 Type |
|
Controls the filter's cutoff frequency |
Controls the filter's resonance amount |
Morph between the filter types, from Low-Pass to Notch to High-pass. This parameter is only available when the filter type is set to SVF for State Variable Filter.
Set the EQ gain. This parameter is only available when the filter type is set to Bell EQ. |
Select a filter type and slope. Read the filter types reference below for details on every filter available. This parameter cannot be modulated. |
Page 2 is the same as page 1, but controls Filter 2. In order to unlock it, you need to have the Routing on page 3 on a setting other than Single.
Filter types reference:
| Off | The filter is bypassed |
| SVF | State Variable Filter model. Use Knob 3 to morph between filter types |
| K35 LP12 / HP6 | Korg 35. Inspired by the MS-20 filter. |
| TLD LP 6/12/18/24 | Transistor Ladder Filter model. Inspired by the classic Moog filter. Low-Pass with a selection of slopes from 6dB/oct to 24/dB/oct |
| TLD N 12/24 | Transistor Ladder Filter model. Inspired by the classic Moog filter. Notch filter with 12dB/oct and 24dB/oct slopes |
| TLD BP 12/24 | Transistor Ladder Filter model. Inspired by the classic Moog filter. Band-pass with 12dB/oct and 24dB/oct slopes |
| DLD LP24 | Diode Ladder Filter model. Inspired by the TB-303 filter. Low-pass with a steep 24dB/oct slope. |
| COMB +/- | Comb filter for hollowed-out sounds and wooshes effects. With positive or negative feedback (resonance) |
| FORMANT | Formant filter for vowel sounds. Morph through A-E-I-O-U with Knob 1. |
| BELL EQ | Simple 1-band equalizer to increase or decrease a selected frequency region. Knob 2 will adjust the bell width and Knob 3 will set the gain |
Page 3 allows for filter routing options:
| Routing | Balance between filter 1 and 2 | - | - |
| Select a routing configuration for the filters. Single enables only filter 1. Serial routes the output of Filter 1 to Filter 2. Para will route both filters in parallel Split will split the sound sources into the two filters, depending on the Machine select. |
Controls the volume of both filters at the output. Fully clockwise only filter 2 will be heard, and fully counter-clockwise it will be only filter 1. | - | - |
Filter routing
Single filter :
Serial filters :
Parallel filters :
Split filters :
Input reference (per machines types)
| Machine | Input 1 | Input 2 |
| Algo synth | Oscillator 1 | Oscillator 2 |
| Wavetable synth | WT Oscillator + Noise | Sub oscillator |
| Sample player | Sample | Noise |
| Crossmod | Crossmodulated output | - |
Amp Section
Within each Machine, the sound source goes through a Filter section and is then routed to the same Amp section.
Each Amp section is then routed to the Mixer.
Main screen of Amp section
The Amp section is always on Tab 4 in every Machine.
This amp tab offers settings related to amplitude and distortion:
| Drive Type | Drive | DCA1 | DCA2 |
|
Select one of the 29 distortion algorithms. Read the chart below for details regarding each one. |
Controls the amount of distortion | The first Digitally Controlled Amplifier. DCA1 and DCA2 are routed in series, with 1 usually modulated by velocity and 2 modulated by an envelope. But you can set modulations as you wish. | The second Digitally Controlled Amplifier. DCA1 and DCA2 are routed in series, with 1 usually modulated by velocity and 2 modulated by an envelope. But you can set modulations as you wish. |
Distortion algorithms reference
| Type | Description |
| Off | Bypassed |
| Soft | Applies a soft saturation effect to an audio sample by adjusting the input sample's amplitude based on a calculated drive factor, using a hyperbolic tangent function for non-linear distortion |
| Medium | Applies a medium saturation effect to an audio sample by scaling the input sample's amplitude with a drive factor and using an arctangent function to achieve a smoother, non-linear distortion. |
| Hard | Applies a hard saturation effect to an audio sample by manipulating the sample's amplitude with a drive factor and employing a combination of hyperbolic tangent and arctangent functions for a more aggressive, non-linear distortion. |
| Diode | Applies a diode-based non-linear distortion to an audio sample. It scales the input sample by a drive factor, processes it through a diode model for non-linear distortion, and then blends the processed signal with the original signal based on the drive amount, including a volume compensation factor. |
| Demon | Applies a distortion effect to an audio sample by scaling the sample with a drive factor, processing it through a sine function, and then applying a diode-like non-linearity, blending the result with the original sample based on the drive amount |
| Soft Fold | Applies a soft folding distortion to an audio sample by scaling the sample with a drive factor, processing it through a sine function to create a folding effect, and blending the result with the original sample based on the drive amount |
| Diode Fold | Applies a diode-based folding distortion to an audio sample by scaling the sample with a drive factor, processing it through a diode model and a sine function to create a folding effect, and blending the result with the original sample based on the drive amount, including volume compensation |
| Dual Frequency | Applies a frequency-dependent distortion to an audio sample by splitting the sample into low and high frequency components using simple filters, applying different saturation levels to each band, and then blending the processed bands with the original sample based on the drive amount |
| Tube | Simulates tube-like distortion by scaling the input sample with a drive factor, applying a non-linear transformation to mimic the tube saturation effect, and blending the processed signal with the original sample based on the drive amount |
| Sigmoid | Applies a tube-like distortion using a sigmoid function to achieve smooth non-linear saturation, scaling the input sample with a drive factor and blending the processed signal with the original sample based on the drive amount. |
| Tape Dynamics | Applies dynamic saturation by using a high-pass pre-emphasis filter, followed by a hyperbolic tangent saturation, and then a low-pass de-emphasis filter, blending the processed signal with the original sample based on the drive amount. |
| Tape Hysteresis | Models tape hysteresis by simulating magnetic hysteresis behavior, adjusting the input sample based on coercivity and remanence factors, and applying a hyperbolic tangent function to saturate the magnetization, blending the processed signal with the original sample based on the drive amount. |
| Tape Curve | Applies a tape saturation effect by approximating a saturation curve, scaling the input sample with a drive factor, and using a non-linear transformation to mimic the characteristic response of tape saturation |
| Tape Noise | Simulates tape saturation with added noise by generating white noise, applying a hyperbolic tangent saturation to the noisy signal, and blending the processed signal with the original sample based on the drive amount, including volume compensation for higher drive levels. |
| Hard Clipping | Applies a hard clipping distortion to an audio sample by limiting the sample's amplitude to a threshold determined by the drive factor, normalizing the output, and blending the processed signal with the original sample based on the drive amount |
| Fuzz | Applies a fuzz distortion effect to an audio sample by scaling the sample with a drive factor, using an exponential function to create a non-linear distortion, and blending the processed signal with the original sample based on the drive amount |
| Chebyshev | Applies a series of Chebyshev polynomials to an audio sample, using a normalized drive factor to create a complex harmonic distortion effect, and then blends the processed signal with the original sample based on the drive amount |
| Half Rectifier | Applies a half-wave rectification effect to an audio sample by zeroing out negative values, scaling the rectified signal with a drive factor, and blending the processed signal with the original sample based on the drive amount |
| Full Rectifier | Applies a full-wave rectification effect to an audio sample by taking the absolute value of the input sample, scaling it with a drive factor, and blending the processed signal with the original sample based on the drive amount |
| Transistor | Simulates transistor-like saturation by scaling the input sample with a drive factor, applying a non-linear transformation to mimic transistor saturation characteristics, and blending the processed signal with the original sample based on the drive amount. |
| Dynamic | Applies a dynamic distortion effect to an audio sample by scaling the sample with a drive factor that is modulated by the sample's envelope, using a hyperbolic tangent function for non-linear distortion, and blending the processed signal with the original sample based on the drive amount. |
| Asymmetric | Applies an asymmetric clipping distortion to an audio sample by limiting the sample's amplitude to different positive and negative thresholds based on a scaled drive factor, normalizing the clipped signal, and blending it with the original sample based on the drive amount. |
| Feedback | Applies a feedback-based distortion effect to an audio sample by adding a feedback signal, scaled by a gain factor derived from the drive, to the input sample and then applying a hyperbolic tangent function for non-linear distortion, updating the feedback with the processed sample |
| Zero Crossing | Introduces distortion at zero crossings by adding a small spike to the audio sample whenever it crosses zero, with the spike's magnitude determined by a normalized drive factor, and updates the last sample for future comparisons. |
| Bit Reaper | Applies a bit reduction effect to an audio sample by scaling the drive factor, using it to determine a decimation factor, and then applying a bitwise reduction to the sample, followed by a non-linear saturation using a hyperbolic tangent function, blending the processed signal with the original sample based on the drive amount |
| Sample Reaper | Applies a sample rate reduction effect by holding the last sample value for a duration determined by the drive factor, updating the sample only when a counter exceeds a threshold, and blending the processed signal with the original sample based on the drive amount |
| Sample Reduction | Reduces the sample rate of an audio signal by holding the current sample value for a number of iterations determined by a drive-scaled reduction factor, effectively lowering the perceived sample rate |
| Bitwise | Applies a bitwise distortion effect to an audio sample by performing an XOR operation between the sample and a drive-scaled value, then normalizing the result and scaling it based on the drive amount |
| Ring Modulation | Applies a ring modulation effect to an audio sample by multiplying the sample with a sine wave at a frequency determined by the drive factor, updating the phase of the modulating signal to maintain continuous modulation |